From mike at jerris.com  Thu Nov  1 00:53:48 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 31 Oct 2012 17:53:48 -0400
Subject: [Freeswitch-users] sofia_reg.c:2452 Can't find user
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF231C039@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF231C039@Mail-Kilo.squay.com>
Message-ID: <EC9F18DF-0E0B-4C0A-907E-97667B626C31@jerris.com>

I suspect this user is not in your directory.  Did you try to:

define a domain called 'fsfailover.uk01.com' in your directory and add a user with the id="100" attribute
and you must configure your device to use the proper domain in it's authentication credentials.


On Oct 31, 2012, at 1:21 PM, Archana Venugopan <a.venugopan at mundio.com> wrote:

> Hi,
>  
> I am getting the below warning when I register my phone. I have registered the phone correctly but not sure from where this message is being picked up. Am i missing something? Can anyone point me out please. Many thanks
>  
> 2012-10-31 17:06:04.798150 [WARNING] sofia_reg.c:2452 Can't find user [100 at fsfailover.uk01.com]
> You must define a domain called 'fsfailover.uk01.com' in your directory and add a user with the id="100" attribute
> and you must configure your device to use the proper domain in it's authentication credentials.
>  

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From mike at jerris.com  Thu Nov  1 00:54:56 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 31 Oct 2012 17:54:56 -0400
Subject: [Freeswitch-users] FreeTDM with old a101
In-Reply-To: <C055AB75-92AA-42A3-BA4F-D384CE9B97A0@5ninesolutions.com>
References: <C055AB75-92AA-42A3-BA4F-D384CE9B97A0@5ninesolutions.com>
Message-ID: <10E66CD9-16EE-493A-9B08-040FA6EA8F57@jerris.com>

Try contacting sangoma to see what would be necessary to make this work.

Mike

On Oct 31, 2012, at 2:16 PM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:

> Hello,
> I have an old Sangoma a101 that I'd like to use with FreeSWITCH.  When I try to configure it, I receive the error "Old a101/102 card not supported in this release!!! with FreeSwitch". 
> 
> If anyone can help out with any versioning (Wanpipe, Kernel, Distro, etc.) info that would work, I'd be greatly appreciative!
> 
> It's really old I know :-)
> #wanrouter hwprobe
> 
> -------------------------------
> | Wanpipe Hardware Probe Info |
> -------------------------------
> 1 . AFT-A101c  : SLOT=3 : BUS=1 : IRQ=28 : CPU=A : PORT=PRI : V=25
> 
> Sangoma Card Count: A101-2=1 



From anthony.minessale at gmail.com  Thu Nov  1 00:55:12 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Oct 2012 16:55:12 -0500
Subject: [Freeswitch-users] SQL ERR while running freeswitch
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF231BF37@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF231BD76@Mail-Kilo.squay.com>
	<804D48104511D4468F0D60DF9D3100350AC77E8D@MAIL.millicorp.com>
	<592A9CF93E12394E8472A6CC66E66BF231BE16@Mail-Kilo.squay.com>
	<592A9CF93E12394E8472A6CC66E66BF231BECA@Mail-Kilo.squay.com>
	<592A9CF93E12394E8472A6CC66E66BF231BF37@Mail-Kilo.squay.com>
Message-ID: <CAKbxfG8q67quXEm3BQ-x1e-vFcWxQ5t9bh0kq7_OcMOLh4hxog@mail.gmail.com>

Probably some permission problem on the user, check error logs in your db.


On Wed, Oct 31, 2012 at 10:05 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> To give you further information. This is my version****
>
> ** **
>
> FreeSWITCH Version 1.2.3+git~20121004T033301Z~94664868a8 (1.2.3; git at
> commit 94664868a8 on Thu, 04 Oct 2012 03:33:01 Z)****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana
> Venugopan
> *Sent:* 31 October 2012 12:40
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SQL ERR while running freeswitch****
>
>  ** **
>
> Hi,****
>
> Database connection is executed****
>
> [root at squay-laptop-1 log]# isql -v SMEPBX smepbx smeswitch****
>
> +---------------------------------------+****
>
> | Connected!                            |****
>
> |                                       |****
>
> | sql-statement                         |****
>
> | help [tablename]                      |****
>
> | quit                                  |****
>
> |                                       |****
>
> +---------------------------------------+****
>
> SQL>****
>
> ** **
>
> But still when I run freeswitch I am getting SQL ERR errors. Can someone
> please help me out, this looks so peculiar for me. ****
>
> Many thanks.****
>
> ** **
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana
> Venugopan
> *Sent:* 31 October 2012 08:46
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SQL ERR while running freeswitch****
>
> ** **
>
> Hi,****
>
> As per the installation procedure after changing odbc.ini I ran this
> command****
>
> ** **
>
> /usr/local/freeswitch/bin/freeswitch****
>
> ** **
>
> After which I got this error.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Meade
> *Sent:* 30 October 2012 18:42
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SQL ERR while running freeswitch****
>
> ** **
>
> I know it?s simple, but did you restart Freeswitch after configuring
> odbc.ini so that FS would create the appropriate tables?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana
> Venugopan
> *Sent:* Tuesday, October 30, 2012 1:53 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] SQL ERR while running freeswitch****
>
> ** **
>
> Hi all,****
>
> ** **
>
> I have been facing issues while running freeswitch and I am facing with
> the below errors and I have installed the latest version of git. I have
> configured /etc/odbc.ini with the username and password.****
>
> ** **
>
> 012-10-29 16:53:22.108686 [DEBUG] switch_core_sqldb.c:972 SQL ERR [no such
> table: sip_registrations]****
>
> [delete from sip_registrations where (sub_host is null or contact like
> '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='
> squay-laptop-1.squay.com' and network_ip like '%' and network_port like
> '%' and sip_username like '%' and mwi_user  like '%' and mwi_host like '%'
> and orig_server_host like '%' and orig_hostname like '%']****
>
> Auto Generating Table!****
>
> 2012-10-29 16:53:22.108761 [DEBUG] switch_core_sqldb.c:979 SQL ERR [no
> such table: sip_registrations]****
>
> [drop table sip_registrations]****
>
> 2012-10-29 16:53:22.111151 [DEBUG] switch_core_sqldb.c:972 SQL ERR [no
> such table: sip_subscriptions]****
>
> [delete from sip_subscriptions where hostname='squay-laptop-1.squay.com'
> and full_to='XXX']****
>
> Auto Generating Table!****
>
> 2012-10-29 16:53:22.111246 [DEBUG] sofia.c:2413 Creating agent for internal
> ****
>
> 2012-10-29 16:53:22.111424 [DEBUG] switch_core_sqldb.c:972 SQL ERR [no
> such table: sip_subscriptions]****
>
> [delete from sip_subscriptions where hostname='squay-laptop-1.squay.com'
> and full_to='XXX']****
>
> Auto Generating Table!****
>
> 2012-10-29 16:53:22.111485 [DEBUG] switch_core_sqldb.c:979 SQL ERR [no
> such table: sip_subscriptions]****
>
> [DROP TABLE sip_subscriptions]****
>
> 2012-10-29 16:53:22.112236 [DEBUG] switch_core_sqldb.c:972 SQL ERR [no
> such table: sip_registrations]****
>
> [delete from sip_registrations where (sub_host is null or contact like
> '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='
> squay-laptop-1.squay.com' and network_ip like '%' and network_port like
> '%' and sip_username like '%' and mwi_user  like '%' and mwi_host like '%'
> and orig_server_host like '%' and orig_hostname like '%']****
>
> ** **
>
> Please do guide me on this. Many thanks.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From william at xofap.com  Thu Nov  1 06:57:31 2012
From: william at xofap.com (William Alianto)
Date: Thu, 01 Nov 2012 10:57:31 +0700
Subject: [Freeswitch-users] Nibblebill database update
Message-ID: <5091F32B.2010302@xofap.com>

Hi,

I was trying to use nibblebill as billing control of my FS. I have 
configured the odbc and the nibblebill configuration. When I tried to 
call, the query looks ok. But when I hanged up the call, it seems like 
there is something wrong with the database connection, since it's not 
updating the database at all. I got error message on CLI

2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE accounts 
SET usage=usage-0.476000 WHERE user='1001']
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 
Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL 
syntax; check the manual that corresponds to your MySQL server version 
for the right syntax to use near 'usage=usage-0.476000 WHERE 
user='1001'' at line 1
]
2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE 
accounts SET usage=usage-0.476000 WHERE user='1001']
[]
2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to log to 
database!
2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup query
[SELECT usage AS nibble_balance FROM accounts WHERE user='1001']
2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT usage 
AS nibble_balance FROM accounts WHERE user='1001']
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 
Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL 
syntax; check the manual that corresponds to your MySQL server version 
for the right syntax to use near 'usage AS nibble_balance FROM accounts 
WHERE user='1001'' at line 1
]
2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running this 
query: [SELECT usage AS nibble_balance FROM accounts WHERE user='1001']

Could anybody help me solve this issue?

Regards


From rdmitry0911 at gmail.com  Thu Nov  1 09:06:29 2012
From: rdmitry0911 at gmail.com (Dmitry R)
Date: Thu, 1 Nov 2012 10:06:29 +0400
Subject: [Freeswitch-users] Passing dtmf to an IVR behind Google
	Voiceproblem
In-Reply-To: <CAKzWOxXyas0UhGE9hywouhhJZ+NJs4UYwxEcwQMf8eJ0V-XdsQ@mail.gmail.com>
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATLu3mYWl7E69+584sdKQ4sKAAAAQAAAAvjGqVrTTe0iYHHMfw69lvgEAAAAA@gmail.com>

No, that's not my case. I can't receive dtmf while that workaround is about
sending dtmf codes
 
Dmitry

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, November 01, 2012 12:31 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Passing dtmf to an IVR behind Google
Voiceproblem


For the sake of posterity (and search engines) there's a workaround
mentioned here:
http://wiki.freeswitch.org/wiki/Dingaling#How_do_I_send_DTMF_with_Googletalk
.3F

-MC


On Wed, Oct 31, 2012 at 11:29 AM, Dmitry R <rdmitry0911 at gmail.com> wrote:


Hi, does anybody know how to pass dtmf codes to an IVR behind the Google
Voice entrance number?
A simple example code from http://wiki.freeswitch.org/wiki/Google_Voice

  <extension name="google_in" continue="true">
     <condition field="caller_id_name" expression="^(Google Voice)$" >
         <action application="log" data="Google Voice Call Incoming" />
         <action application="set" data="execute_on_answer=send_dtmf 1"/>
     </condition>
   </extension>

doesn't work for me. DTMF codes can't get thru. Everything else other then
that works fine.

Any help would be very much appreciated, Dmitry


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




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

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





-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



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From evgeniy at bestnet.kharkov.ua  Thu Nov  1 10:56:53 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Thu, 01 Nov 2012 09:56:53 +0200
Subject: [Freeswitch-users] Nibblebill database update
In-Reply-To: <5091F32B.2010302@xofap.com>
References: <5091F32B.2010302@xofap.com>
Message-ID: <50922B45.9060302@bestnet.kharkov.ua>

Hello.
Show please your nibblebill.conf.xml.

01.11.2012 05:57, William Alianto ?????:
> Hi,
>
> I was trying to use nibblebill as billing control of my FS. I have
> configured the odbc and the nibblebill configuration. When I tried to
> call, the query looks ok. But when I hanged up the call, it seems like
> there is something wrong with the database connection, since it's not
> updating the database at all. I got error message on CLI
>
> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE accounts
> SET usage=usage-0.476000 WHERE user='1001']
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
> syntax; check the manual that corresponds to your MySQL server version
> for the right syntax to use near 'usage=usage-0.476000 WHERE
> user='1001'' at line 1
> ]
> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE
> accounts SET usage=usage-0.476000 WHERE user='1001']
> []
> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to log to
> database!
> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup query
> [SELECT usage AS nibble_balance FROM accounts WHERE user='1001']
> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT usage
> AS nibble_balance FROM accounts WHERE user='1001']
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
> syntax; check the manual that corresponds to your MySQL server version
> for the right syntax to use near 'usage AS nibble_balance FROM accounts
> WHERE user='1001'' at line 1
> ]
> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running this
> query: [SELECT usage AS nibble_balance FROM accounts WHERE user='1001']
>
> Could anybody help me solve this issue?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Evgeniy Movlyan,
BestNet Ltd.


From cal.leeming at simplicitymedialtd.co.uk  Thu Nov  1 15:00:12 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 1 Nov 2012 12:00:12 +0000
Subject: [Freeswitch-users] application synchronisation between
 freeswitch and web
In-Reply-To: <CAGDHiz=WarbQftBxaok_LG5ZPLMaH_fFBnC8vU=z9dg9-qRUhA@mail.gmail.com>
References: <CAGDHiz=WarbQftBxaok_LG5ZPLMaH_fFBnC8vU=z9dg9-qRUhA@mail.gmail.com>
Message-ID: <CAHKQagFhT1qFbTqBhKA+xxgG9niefVEDYtPb5GgV7FvW6dBZJw@mail.gmail.com>

duosecurity.com is better tbh

On Wed, Oct 31, 2012 at 10:54 AM, Mumuney Abdlquadri <
abdlquadri at googlemail.com> wrote:

> Hi all,
>
> I am think of building an application like the demo here
>
> http://www.certificall.net/Try
>
> I noticed that the calls are in complete sync with the app.
>
> I am thinking to use ESL (maybe node-esl) and tunnel the events through
> socket.io.
>
> Am I thinking straight? Or does anyone have a better proposition?
>
> Regards.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bigx333 at gmail.com  Thu Nov  1 15:58:22 2012
From: bigx333 at gmail.com (Nelson Camargo)
Date: Thu, 1 Nov 2012 14:58:22 +0200
Subject: [Freeswitch-users] mod_spandsp fails to load
Message-ID: <85610303-A08D-4A53-A8AD-72D9F2AF70E5@gmail.com>

I'm having issues to get mod_spandsp to work on one of my boxes, every time I try to load I get:

2012-11-01 14:48:19.199721 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so
**/usr/local/freeswitch/mod/mod_spandsp.so: undefined symbol: jbg_enc_out**

It's happening with both HEAD master and v1.2stable, any ideas?

Thanks,
Nelson

From lxfontes at gmail.com  Thu Nov  1 20:38:18 2012
From: lxfontes at gmail.com (Lucas Fontes)
Date: Thu, 1 Nov 2012 13:38:18 -0400
Subject: [Freeswitch-users] mod_sms delivery failure
Message-ID: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>

Hi everyone,

I've been trying to determine if a message was successfully delivered to a device.
Stumbled on a post from February (subject: Testing mod_sms) http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-February/005619.html:

________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: Warren Lin <wlin121443 at yahoo.com> 
Sent: Tuesday, February 21, 2012 11:01 AM
Subject: Re: Testing mod_sms

hi you can find out how to use the lists at http://lists.freeswitch.org

try latest GIT, i put in a patch to fire the events with the heder
Failed-Delivery true
if you are listening for MESSAGE events you should catch them.
________________________________


I can't find any reference to Failed-Delivery in mod_sms.c neither anywhere else in the source tree. 
There is a field called Delivery-Failure, but seems to be used by other modules to indicate a failure on the response going out of freeswitch, not to freeswitch.
Was this overwritten / rolled back at some point ? 

thanks

--
lucas

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From rjb at contactbeep.com  Thu Nov  1 18:31:25 2012
From: rjb at contactbeep.com (Rob Boutin)
Date: Thu, 1 Nov 2012 11:31:25 -0400
Subject: [Freeswitch-users] Just starting
Message-ID: <935E2736-197D-4417-85D9-6C1B2CD503E3@contactbeep.com>

Hello,
 
I am just starting out with FreeSWITCH and I am researching it for a project at work that will hopefully end up with full implementation of FreeSWITCH and had a few short questions as I am having difficulties getting it off the ground.
 
I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system.
My tiny test network of phones includes an xlite softphone on the windows machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone.  They are being routed with a simple Linksys Etherfast router.
 
My issue is that only one phone at a time will ever be registered and able to make calls. I have four extensions configured with identical settings 300 - 303.  The peculiar thing is only the phone assigned to 301 will ever have a dial tone and be able to place calls.  It doesn't matter which device I move 301 to it is only 301 that ever works.  I was hoping to get some light shed on this, as it is likely related to some simple setting I over looked and would like to continue with my research.
 
Thanks in advance,
RJB
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From frederick at targointernet.com  Thu Nov  1 18:44:45 2012
From: frederick at targointernet.com (Frederick Pruneau)
Date: Thu, 01 Nov 2012 11:44:45 -0400
Subject: [Freeswitch-users] Parked calls cannot be retrieved
Message-ID: <509298ED.5010705@targointernet.com>

Hi all,

I am configuring a blue.box server with freeswitch. I am tweaking 
blue.box to add fifo call park. I followed instructions from this wiki 
page http://wiki.freeswitch.org/wiki/Park_%26_Retrieve .I can park the 
call but cannot retrieve it. When I dial the same extension where I 
parked the call (Usually, it is supposed to pick up the parked call), 
the call is parked in another park slot. Here is my configuration:

     <extension name="parking_slots">
         <condition field="destination_number" expression="^(18[0-2])$" 
break="on-false">
         <action inline="true" application="set" 
data="slot_count=${fifo(count $1@${domain_name})}"/>
         <action inline="true" application="set" 
data="slot_count=${slot_count:-3:2}"/>
     </condition>
     <condition field="${slot_count}" expression="^\:0$" break="always">
         <action application="unset" data="fifo_chime_list"/>
         <action application="set" data="fifo_chime_freq=0"/>
         <action application="fifo" 
data="${destination_number}@${domain_name} in undef local_stream://moh"/>
         <anti-action application="fifo" 
data="${destination_number}@${domain_name} out nowait"/>
     </condition>
     </extension>

I have found this in the log file:

EXECUTE sofia/sipinterface_1/ext at X.X.X.X 
set(slot_count=180 at X.X.X.X:0:*1*:0:0:0)

The number 1 is the parking slot number. It changes to the next 
available parking slot (In this case,  number 2) everytime I dial 180.

Am i missing something?

Thanks for your help!

Fred
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From jaykris at gmail.com  Thu Nov  1 21:54:59 2012
From: jaykris at gmail.com (JP)
Date: Thu, 1 Nov 2012 11:54:59 -0700
Subject: [Freeswitch-users] mod_http_cache with mod_xml_curl
Message-ID: <CAMRwkEB-D9vaZqVDe4qrN_e4fdt98JzbgEWfktm8GW=76QQLQQ@mail.gmail.com>

Hi,
My project has a need to use dynamically generated dial plans(stored in
LDAP). But there are instances where I will need to fetch a dialplan that
is static most of the time but may change occasionally  I was wondering if
there is a way to use mod_xml_curl, but make it go through mod_http_cache
so that I can avoid unnecessary trips to the LDAP behind a web server when
not required? If you have any experience with something like this please
share it. Your help is greatly appreciated.

-JP
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From vetali100 at gmail.com  Thu Nov  1 22:43:42 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Thu, 1 Nov 2012 12:43:42 -0700
Subject: [Freeswitch-users] Just starting
In-Reply-To: <935E2736-197D-4417-85D9-6C1B2CD503E3@contactbeep.com>
References: <935E2736-197D-4417-85D9-6C1B2CD503E3@contactbeep.com>
Message-ID: <CACc3D0VbtvPyDrR=iEiRwyLs4Fh12UbL1JrvZ0dwKb4hGJdyCw@mail.gmail.com>

If you could shutdown all devices and then capturing SIP Trace
while bringing up the device with #300 (or any # except #301) using
Wireshark, it will definitely add more info for investigation.


2012/11/1 Rob Boutin <rjb at contactbeep.com>

> Hello,****
>
> ** **
>
> I am just starting out with FreeSWITCH and I am researching it for a
> project at work that will hopefully end up with full implementation of
> FreeSWITCH and had a few short questions as I am having difficulties
> getting it off the ground.****
>
> ** **
>
> I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system.
> ****
>
> My tiny test network of phones includes an xlite softphone on the windows
> machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone.  They are being
> routed with a simple Linksys Etherfast router.****
>
> ** **
>
> My issue is that only one phone at a time will ever be registered and able
> to make calls. I have four extensions configured with identical settings
> 300 ? 303.  The peculiar thing is only the phone assigned to 301 will ever
> have a dial tone and be able to place calls.  It doesn?t matter which
> device I move 301 to it is only 301 that ever works.  I was hoping to get
> some light shed on this, as it is likely related to some simple setting I
> over looked and would like to continue with my research.****
>
> ** **
>
> Thanks in advance,****
>
> RJB****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Thu Nov  1 23:16:50 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 1 Nov 2012 20:16:50 +0000
Subject: [Freeswitch-users] mod_http_cache with mod_xml_curl
In-Reply-To: <CAMRwkEB-D9vaZqVDe4qrN_e4fdt98JzbgEWfktm8GW=76QQLQQ@mail.gmail.com>
References: <CAMRwkEB-D9vaZqVDe4qrN_e4fdt98JzbgEWfktm8GW=76QQLQQ@mail.gmail.com>
Message-ID: <CAHKQagGKuECyubTHX_UkfoSwNeqYE_xk3Y5a+aJsFm7CU_qCoQ@mail.gmail.com>

Hi JP,

This is actually mentioned in the documentation, but it wasn't very easy to
find, so I have updated the wiki page to highlight this.

Here is the updated URL;
http://wiki.freeswitch.org/wiki/Mod_xml_curl#Caching_objects

Please note, it would probably be less complex to place the caching logic
into your web application instead.

At a later stage - you could look at caching it directly within FreeSWITCH
to get a bit more performance, but you probably won't need this level of
performance to begin with.

Hope this helps.

Cal

On Thu, Nov 1, 2012 at 6:54 PM, JP <jaykris at gmail.com> wrote:

> Hi,
> My project has a need to use dynamically generated dial plans(stored in
> LDAP). But there are instances where I will need to fetch a dialplan that
> is static most of the time but may change occasionally  I was wondering if
> there is a way to use mod_xml_curl, but make it go through mod_http_cache
> so that I can avoid unnecessary trips to the LDAP behind a web server when
> not required? If you have any experience with something like this please
> share it. Your help is greatly appreciated.
>
> -JP
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ssinyagin at yahoo.com  Fri Nov  2 00:00:42 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 1 Nov 2012 14:00:42 -0700 (PDT)
Subject: [Freeswitch-users] Sofia gateway variables
Message-ID: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>

I've got the following SIP gateway which registers successfully at the ITSP:

? <gateway name="sipcall_41325133196">
??? <param name="username" value="41325133196"/>
??? <param name="password" value="XXXXXXXXXX"/>
??? <param name="extension" value="0325133196"/>
??? <param name="expire-seconds" value="125"/>
??? <param name="register" value="true"/>
??? <param name="register-transport" value="udp"/>
??? <param name="proxy" value="free2.voipgateway.org"/>
??? <param name="retry-seconds" value="30"/>
??? <param name="caller-id-in-from" value="false"/>
??? <param name="ping" value="27"/>
??? <variables>
???? <variable name="sbc_callee_ccode" data="41" direction="inbound"/>
???? <variable name="sbc_caller_ccode" data="41" direction="inbound"/>
??? </variables>
? </gateway>


Then the call from PSTN goes into the proper context, and the first condition in the context is:

?? <extension name="call_debug" continue="true">
???? <condition field="${call_debug}" expression="^true$" break="never">
?????? <action application="info"/>
?????? <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/>
???? </condition>
?? </extension>


Neither the info, nor the log show my two variables on inbound calls. 


Am I doing something wrong or is it a bug? 

I'm using git commit 94664868a8 in v1.2.stable branch



thanks,
stanislav



From jesus.rocha at overvoiplatam.com  Thu Nov  1 23:53:52 2012
From: jesus.rocha at overvoiplatam.com (Jesus Ramon Rocha Velazquez)
Date: Thu, 1 Nov 2012 14:53:52 -0600
Subject: [Freeswitch-users] Introducing me and Requesting help
Message-ID: <CAFhTFjiAaapyxqSsf+r6W0kyB7L2b_-_TOsorwFNE7C3JBND2Q@mail.gmail.com>

Hi,

I'm Just a new Member in this FS World. I love the wiki and all the
documentation that all the community has made.

I'm in this momment Trying to connect to Remote FreeSwitch But, I have This
Problem:


SERVER A -> SERVER B [OK REGISTRATION]

SERVER B -> SERVER C [OK REGISTRATION]

SERVER A -> SERVER C [OK REGISTRATION]

SERVER B -> SERVER A [ERROR]

On SERVER B:

nta.c:8164 outgoing_send() nta: resent REGISTER (35565297) to udp/
187.162.113.251:5060
tport.c:4140 tport_pend() tport_pend(0x8daa9a0): pending 0xb69d04a0 for udp/
190.158.239.105:5080 (already 1)
nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 1/2 resent, 0/2 tout,
0/0 term, 0/2 free
nta.c:1294 agent_timer() nta: timer set next to 1000 ms
nta.c:8742 _nta_outgoing_timer() nta: timer E fired, retransmit REGISTER
(35565297)
tport.c:4202 tport_release() tport_release(0x8daa9a0): 0xb6960540 by
0x8bf4b48 with (nil)
tport.c:3238 tport_tsend() tport_tsend(0x8daa9a0) tpn = udp/
187.162.113.251:5060
tport.c:4026 tport_resolve() tport_resolve addrinfo = 187.162.113.251:5060
tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x8daa9a0): not found by
name udp/187.162.113.251:5060
tport.c:3574 tport_vsend() tport_vsend(0x8daa9a0): 614 bytes of 614 to udp/
187.162.113.251:5060
tport.c:3472 tport_send_msg() tport_vsend returned 614
send 614 bytes to udp/[187.162.113.251]:5060 at 20:45:26.988826:
   ------------------------------------------------------------------------
   REGISTER sip:187.162.113.251;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 190.158.239.105:5080;rport;branch=z9hG4bK0tK51ZpF315HF
   Max-Forwards: 70
   From: <sip:contactoscolombia at 187.162.113.251>;tag=gFB9QKrcU45NQ
   To: <sip:contactoscolombia at 187.162.113.251>
   Call-ID: ec49e368-2461-11e2-a3fa-95e30a5de75a
   CSeq: 35565297 REGISTER
   Contact: <sip:gw+Contactos_1 at 190.158.239.105:5080
;transport=udp;gw=Contactos_1>
   Expires: 3600
   User-Agent: OverCall PBX 1.0.0.1
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
nta.c:8164 outgoing_send() nta: resent REGISTER (35565297) to udp/
187.162.113.251:5060
tport.c:4140 tport_pend() tport_pend(0x8daa9a0): pending 0xb6960540 for udp/
190.158.239.105:5080 (already 1)
nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 1/2 resent, 0/2 tout,
0/0 term, 0/2 free
nta.c:1294 agent_timer() nta: timer set next to 2999 ms
nta.c:8742 _nta_outgoing_timer() nta: timer E fired, retransmit REGISTER
(35565297)
tport.c:4202 tport_release() tport_release(0x8daa9a0): 0xb69d04a0 by
0xb69091b8 with (nil)
tport.c:3238 tport_tsend() tport_tsend(0x8daa9a0) tpn = udp/
187.162.113.251:5060
tport.c:4026 tport_resolve() tport_resolve addrinfo = 187.162.113.251:5060
tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x8daa9a0): not found by
name udp/187.162.113.251:5060
tport.c:3574 tport_vsend() tport_vsend(0x8daa9a0): 616 bytes of 616 to udp/
187.162.113.251:5060
tport.c:3472 tport_send_msg() tport_vsend returned 616
send 616 bytes to udp/[187.162.113.251]:5060 at 20:45:29.988221:


And Finaly a Timeour.....

ON SERVER A is kind of different:

Kecv_iovec(0xb6a0cbf0) msg 0xb55c0d58 from (udp/187.162.113.251:5060) has
620 bytes, veclen = 1
recv 620 bytes from udp/[190.158.239.105]:5080 at 20:24:00.650742:
   ------------------------------------------------------------------------
   REGISTER sip:187.162.113.251;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 190.158.239.105:5080080
;rport;branch=z9hG4bKjvHa94ayB5DcD
   Max-Forwards: 70
  From: <sip:contactoscolombia at 187.162.113.251>;tag=mj7j6X09BBpaK
  To: <sip:contactoscolombia at 187.162.113.251>
  Call-ID: ec49e368-2461-11e2-a3fa-95e30a5de75a
  CSeq: 35564655 REGISTER
  Contact: <sip:gw+Contactos_1 at 190.158.239.105:5080080
;transport=udp;gw=Contactos_1>
  Expires: 3600
  User-Agent: OverCall PBX 1.0.0.1
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
  Supported: timer, precondition, path, replaces
  Content-Length: 0
  ------------------------------------------------------------------------
tport_deliver(0xb6a0cbf0): msg 0xb55c0d58 (620 bytes) from udp/
190.158.239.105:5060/sip next=(nil)
nta: received REGISTER sip:187.162.113.251;transport=udp SIP/2.0 (CSeq
35564655)
nta: REGISTER has bad Via header
nta: timer K fired, terminate NOTIFY (35564653)
outgoing_reclaim_all((nil), (nil), 0xb671c200)
nta_outgoing_timer: 0/3 resent, 0/3 tout, 1/9 term, 1/12 free
nta: timer set next to 1 ms
nta: timer K fired, terminate NOTIFY (35564653)
outgoing_reclaim_all((nil), (nil), 0xb671c200)
nta_outgoing_timer: 0/3 resent, 0/3 tout, 1/8 term, 1/11 free
nta: timer set next to 85 ms
nta: timer E fired, retransmit NOTIFY (35564648)

I know is on the header the error the 5080080 but i don't know why is this
happening or how to fix it

NOTE: Both A & B are Freeswitch the C is a rented Itel Switch for testing.

Hope You can help me

Greetings
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From william.king at quentustech.com  Fri Nov  2 00:31:06 2012
From: william.king at quentustech.com (William King)
Date: Thu, 01 Nov 2012 14:31:06 -0700
Subject: [Freeswitch-users] mod_sms delivery failure
In-Reply-To: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>
References: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>
Message-ID: <5092EA1A.8020204@quentustech.com>

If the sms delivery failed you will see an event with the type of
SMS_MESSAGE and there will be a header 'delivery-failure' with a value
of 'true'. I'm working on a patch that will enable the event on
successful delivery.

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

On 11/01/2012 10:38 AM, Lucas Fontes wrote:
> Hi everyone,
> 
> I've been trying to determine if a message was successfully delivered to
> a device.
> Stumbled on a post from February (subject: Testing mod_sms)
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-February/005619.html:
> 
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev>>
> To: Warren Lin <wlin121443 at yahoo.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev>> 
> Sent: Tuesday, February 21, 2012 11:01 AM
> Subject: Re: Testing mod_sms
> 
> hi you can find out how to use the lists at http://lists.freeswitch.org
> 
> try latest GIT, i put in a patch to fire the events with the heder
> Failed-Delivery true
> if you are listening for MESSAGE events you should catch them.
> 
> ________________________________
> 
> 
> 
> I can't find any reference to Failed-Delivery in mod_sms.c neither
> anywhere else in the source tree. 
> There is a field called Delivery-Failure, but seems to be used by other
> modules to indicate a failure on the response going out of freeswitch,
> not to freeswitch.
> Was this overwritten / rolled back at some point ? 
> 
> thanks
> 
> --
> lucas
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From avi at avimarcus.net  Fri Nov  2 00:40:02 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 1 Nov 2012 23:40:02 +0200
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <CALNZuwkt1En=yKgmP+daaY3hU5dFKKaNaat449yNAh6V7PVFjw@mail.gmail.com>

Several silly questions to help start debugging:

1) Did you kill the gateway and restart it since adding in this variable?
reloadxml doesn't update active gateways.
2) Are you sure the call is actually coming in through the gateway and not
directly to the sip address?
3) Is the debug extension actually executing? Are you
sure ${call_debug}==true -- that lines runs in your logs?

-Avi


On Thu, Nov 1, 2012 at 11:00 PM, Stanislav Sinyagin <ssinyagin at yahoo.com>wrote:

> I've got the following SIP gateway which registers successfully at the
> ITSP:
>
>   <gateway name="sipcall_41325133196">
>     <param name="username" value="41325133196"/>
>     <param name="password" value="XXXXXXXXXX"/>
>     <param name="extension" value="0325133196"/>
>     <param name="expire-seconds" value="125"/>
>     <param name="register" value="true"/>
>     <param name="register-transport" value="udp"/>
>     <param name="proxy" value="free2.voipgateway.org"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="caller-id-in-from" value="false"/>
>     <param name="ping" value="27"/>
>     <variables>
>      <variable name="sbc_callee_ccode" data="41" direction="inbound"/>
>      <variable name="sbc_caller_ccode" data="41" direction="inbound"/>
>     </variables>
>   </gateway>
>
>
> Then the call from PSTN goes into the proper context, and the first
> condition in the context is:
>
>    <extension name="call_debug" continue="true">
>      <condition field="${call_debug}" expression="^true$" break="never">
>        <action application="info"/>
>        <action application="log" data="INFO Country code:
> ${sbc_callee_ccode}"/>
>      </condition>
>    </extension>
>
>
> Neither the info, nor the log show my two variables on inbound calls.
>
>
> Am I doing something wrong or is it a bug?
>
> I'm using git commit 94664868a8 in v1.2.stable branch
>
>
>
> thanks,
> stanislav
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Chad.Engler at patlive.com  Fri Nov  2 00:46:29 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Thu, 1 Nov 2012 17:46:29 -0400
Subject: [Freeswitch-users] mod_sms delivery failure
In-Reply-To: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>
References: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606B937CB@mail2.patlive.local>

I don't think SIP MESSAGEs have a response/ack. I know that our provider
doesn't send anything except the 200 OK when we send a message, and that
doesn't mean it actually made it to the device. As far as I know there
is no way to know if and when the device gets the actual message without
the device responding with another message back through the via chain.

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Lucas Fontes
Sent: Thursday, November 01, 2012 1:38 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_sms delivery failure

 

Hi everyone,

 

I've been trying to determine if a message was successfully delivered to
a device.

Stumbled on a post from February (subject: Testing mod_sms)
http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-February/00561
9.html:

 

________________________________
From: Anthony Minessale <anthony.minessale at gmail.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> >
To: Warren Lin <wlin121443 at yahoo.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> > 
Sent: Tuesday, February 21, 2012 11:01 AM
Subject: Re: Testing mod_sms
 
hi you can find out how to use the lists at http://lists.freeswitch.org
 
try latest GIT, i put in a patch to fire the events with the heder
Failed-Delivery true
if you are listening for MESSAGE events you should catch them.
________________________________

 

 

I can't find any reference to Failed-Delivery in mod_sms.c neither
anywhere else in the source tree. 

There is a field called Delivery-Failure, but seems to be used by other
modules to indicate a failure on the response going out of freeswitch,
not to freeswitch.

Was this overwritten / rolled back at some point ? 

 

thanks

 

--

lucas

 

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From paul at cupis.co.uk  Fri Nov  2 01:15:07 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Thu, 01 Nov 2012 22:15:07 +0000
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <5092F46B.5060709@cupis.co.uk>

On 01/11/12 21:00, Stanislav Sinyagin wrote:
> Then the call from PSTN goes into the proper context,

Are you sure? Can you get a log of such a call and put it on pastebin so 
we can see what is happening, please?


Regards,


From ssinyagin at yahoo.com  Fri Nov  2 01:31:03 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 1 Nov 2012 15:31:03 -0700 (PDT)
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <1351809063.48212.YahooMailNeo@web39305.mail.mud.yahoo.com>

[the mailing list is very slow lately, but I've seen some replies in the list archive].

freeswitch daemon was freshly started after XML editing. Also I'm sure the call 
comes into this gateway, as I get 
variable_sip_gateway: [sipcall_41325133196]
in the info output.

The full log for the call is here:
https://gist.github.com/3997100

the call does not end anywhere, as there are no matching conditions 
yet that would end in a transfer or a bridge, so the caller gets 
the busy signal. But anyway I believe info should display the inbound variables.









----- Original Message -----
> From: Stanislav Sinyagin <ssinyagin at yahoo.com>
> To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Thursday, November 1, 2012 10:00 PM
> Subject: Sofia gateway variables
> 
> I've got the following SIP gateway which registers successfully at the ITSP:
> 
> ? <gateway name="sipcall_41325133196">
> ??? <param name="username" value="41325133196"/>
> ??? <param name="password" value="XXXXXXXXXX"/>
> ??? <param name="extension" value="0325133196"/>
> ??? <param name="expire-seconds" value="125"/>
> ??? <param name="register" value="true"/>
> ??? <param name="register-transport" value="udp"/>
> ??? <param name="proxy" 
> value="free2.voipgateway.org"/>
> ??? <param name="retry-seconds" value="30"/>
> ??? <param name="caller-id-in-from" value="false"/>
> ??? <param name="ping" value="27"/>
> ??? <variables>
> ???? <variable name="sbc_callee_ccode" data="41" 
> direction="inbound"/>
> ???? <variable name="sbc_caller_ccode" data="41" 
> direction="inbound"/>
> ??? </variables>
> ? </gateway>
> 
> 
> Then the call from PSTN goes into the proper context, and the first condition in 
> the context is:
> 
> ?? <extension name="call_debug" continue="true">
> ???? <condition field="${call_debug}" expression="^true$" 
> break="never">
> ?????? <action application="info"/>
> ?????? <action application="log" data="INFO Country code: 
> ${sbc_callee_ccode}"/>
> ???? </condition>
> ?? </extension>
> 
> 
> Neither the info, nor the log show my two variables on inbound calls. 
> 
> 
> Am I doing something wrong or is it a bug? 
> 
> I'm using git commit 94664868a8 in v1.2.stable branch
> 
> 
> 
> thanks,
> stanislav
> 


From ssinyagin at yahoo.com  Fri Nov  2 01:45:28 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 1 Nov 2012 15:45:28 -0700 (PDT)
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <1351809063.48212.YahooMailNeo@web39305.mail.mud.yahoo.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
	<1351809063.48212.YahooMailNeo@web39305.mail.mud.yahoo.com>
Message-ID: <1351809928.71573.YahooMailNeo@web39302.mail.mud.yahoo.com>

actually here's the whole FreeSWITCH config, except for the SIP gateway 
with the real password:

https://github.com/xlab1/sipfe_freeswitch_common

It's pretty short currently, and I re-define some global variables 
with X-PRE-PROCESS statements, so there is a potentially problem with that, 
such as corrupted memory. Particularly, I define the IPv4 addresses in 
vars.xml, and set them again with the proper values in site_vars.xml
I don't know if the pre-processor is designed to handle this, but it 
works so far...







----- Original Message -----
> From: Stanislav Sinyagin <ssinyagin at yahoo.com>
> To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Thursday, November 1, 2012 11:31 PM
> Subject: Re: Sofia gateway variables
> 
> [the mailing list is very slow lately, but I've seen some replies in the 
> list archive].
> 
> freeswitch daemon was freshly started after XML editing. Also I'm sure the 
> call 
> comes into this gateway, as I get 
> variable_sip_gateway: [sipcall_41325133196]
> in the info output.
> 
> The full log for the call is here:
> https://gist.github.com/3997100
> 
> the call does not end anywhere, as there are no matching conditions 
> yet that would end in a transfer or a bridge, so the caller gets 
> the busy signal. But anyway I believe info should display the inbound variables.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> ----- Original Message -----
>>  From: Stanislav Sinyagin <ssinyagin at yahoo.com>
>>  To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>  Cc: 
>>  Sent: Thursday, November 1, 2012 10:00 PM
>>  Subject: Sofia gateway variables
>> 
>>  I've got the following SIP gateway which registers successfully at the 
> ITSP:
>> 
>>  ? <gateway name="sipcall_41325133196">
>>  ??? <param name="username" value="41325133196"/>
>>  ??? <param name="password" value="XXXXXXXXXX"/>
>>  ??? <param name="extension" value="0325133196"/>
>>  ??? <param name="expire-seconds" value="125"/>
>>  ??? <param name="register" value="true"/>
>>  ??? <param name="register-transport" 
> value="udp"/>
>>  ??? <param name="proxy" 
>>  value="free2.voipgateway.org"/>
>>  ??? <param name="retry-seconds" value="30"/>
>>  ??? <param name="caller-id-in-from" 
> value="false"/>
>>  ??? <param name="ping" value="27"/>
>>  ??? <variables>
>>  ???? <variable name="sbc_callee_ccode" data="41" 
>>  direction="inbound"/>
>>  ???? <variable name="sbc_caller_ccode" data="41" 
>>  direction="inbound"/>
>>  ??? </variables>
>>  ? </gateway>
>> 
>> 
>>  Then the call from PSTN goes into the proper context, and the first 
> condition in 
>>  the context is:
>> 
>>  ?? <extension name="call_debug" continue="true">
>>  ???? <condition field="${call_debug}" 
> expression="^true$" 
>>  break="never">
>>  ?????? <action application="info"/>
>>  ?????? <action application="log" data="INFO Country code: 
> 
>>  ${sbc_callee_ccode}"/>
>>  ???? </condition>
>>  ?? </extension>
>> 
>> 
>>  Neither the info, nor the log show my two variables on inbound calls. 
>> 
>> 
>>  Am I doing something wrong or is it a bug? 
>> 
>>  I'm using git commit 94664868a8 in v1.2.stable branch
>> 
>> 
>> 
>>  thanks,
>>  stanislav
>> 
> 


From lxfontes at gmail.com  Fri Nov  2 02:10:36 2012
From: lxfontes at gmail.com (Lucas Fontes)
Date: Thu, 1 Nov 2012 19:10:36 -0400
Subject: [Freeswitch-users] mod_sms delivery failure
In-Reply-To: <5092EA1A.8020204@quentustech.com>
References: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>
	<5092EA1A.8020204@quentustech.com>
Message-ID: <980ECC15-B873-42EC-AEE0-1D23149E778C@gmail.com>

Thanks William,

 I've tried with 1.3.0+git~20121016T230329Z~c51aebf621  and did not see that event even after all timers expired.

Is this event generated by mod_sms or gsmopen ?
Reason I ask is because my use case is around SIP messages only and we want to track delivery to devices.
Ex:
client connected via eventsocket -> sendevent SMS::SEND_MESSAGE 
Freeswitch consuming SMS::SEND_MESSAGE -> dispatch SIP message to client device
client returning anything other than 200 OK -> fire MESSAGE event tagged as failure

If that's the same understanding, then I will dig further here as I'm probably not catching such event (although pretty sure it is subscribed to all events right now).

cheers
--
lucas



On 2012-11-01, at 5:31 PM, William King <william.king at quentustech.com> wrote:

> If the sms delivery failed you will see an event with the type of
> SMS_MESSAGE and there will be a header 'delivery-failure' with a value
> of 'true'. I'm working on a patch that will enable the event on
> successful delivery.
> 
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
> 
> On 11/01/2012 10:38 AM, Lucas Fontes wrote:
>> Hi everyone,
>> 
>> I've been trying to determine if a message was successfully delivered to
>> a device.
>> Stumbled on a post from February (subject: Testing mod_sms)
>> http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-February/005619.html:
>> 
>> ________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev>>
>> To: Warren Lin <wlin121443 at yahoo.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev>> 
>> Sent: Tuesday, February 21, 2012 11:01 AM
>> Subject: Re: Testing mod_sms
>> 
>> hi you can find out how to use the lists at http://lists.freeswitch.org
>> 
>> try latest GIT, i put in a patch to fire the events with the heder
>> Failed-Delivery true
>> if you are listening for MESSAGE events you should catch them.
>> 
>> ________________________________
>> 
>> 
>> 
>> I can't find any reference to Failed-Delivery in mod_sms.c neither
>> anywhere else in the source tree. 
>> There is a field called Delivery-Failure, but seems to be used by other
>> modules to indicate a failure on the response going out of freeswitch,
>> not to freeswitch.
>> Was this overwritten / rolled back at some point ? 
>> 
>> thanks
>> 
>> --
>> lucas
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lxfontes at gmail.com  Fri Nov  2 02:28:57 2012
From: lxfontes at gmail.com (Lucas Fontes)
Date: Thu, 1 Nov 2012 19:28:57 -0400
Subject: [Freeswitch-users] mod_sms delivery failure
In-Reply-To: <BD87D3B1554F614190EB6E54CE95DDB606B937CB@mail2.patlive.local>
References: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>
	<BD87D3B1554F614190EB6E54CE95DDB606B937CB@mail2.patlive.local>
Message-ID: <38522381-38CB-497F-AA19-FE0759A8CE68@gmail.com>


Hi Chad,
thanks for that awesome ESL binding for node.js !

Section 7 of rfc3428

> A UAS that receives a MESSAGE request processes it following the
>    rules of SIP [1].
> 
>    A UAS receiving a MESSAGE request SHOULD respond with a final
>    response immediately.  Note, however, that the UAS is not obliged to
>    display the message to the user either before or after responding
>    with a 200 OK.  That is, a 200 OK response does not necessarily mean
>    the user has read the message.
> 
>    A 2xx response to a MESSAGE request MUST NOT contain a body.  A UAS
>    MUST NOT insert a Contact header field into a 2xx response.
> 
>    A UAS which is, in fact, a message relay, storing the message and
>    forwarding it later on, or forwarding it into a non-SIP domain,
>    SHOULD return a 202 (Accepted) [5] response indicating that the
>    message was accepted, but end to end delivery has not been
>    guaranteed.
> 
>    A 4xx or 5xx response indicates that the message was not delivered
>    successfully.  A 6xx response means it was delivered successfully,
>    but refused.


At Fongo (fongo.com) we have a few proxies for store and forward, also for SMS integration. We use 200, 402, 404 and 488 to control delivery and payments.
So in our case, the final device always returns 200 (as it is the final), but a proxy in between might generate a 4xx.
we have an opensips cluster in front of freeswitch and steer MESSAGE methods away from freeswitch to handle those scenarios. I'm  looking into removing this logic from opensips and move it to freeswitch to keep it consistent with dialplan and rating engine. 

cheers
--
lucas



On 2012-11-01, at 5:46 PM, Chad Engler <Chad.Engler at patlive.com> wrote:

> I don?t think SIP MESSAGEs have a response/ack. I know that our provider doesn?t send anything except the 200 OK when we send a message, and that doesn?t mean it actually made it to the device. As far as I know there is no way to know if and when the device gets the actual message without the device responding with another message back through the via chain.
>  
> -Chad
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lucas Fontes
> Sent: Thursday, November 01, 2012 1:38 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] mod_sms delivery failure
>  
> Hi everyone,
>  
> I've been trying to determine if a message was successfully delivered to a device.
> Stumbled on a post from February (subject: Testing mod_sms)http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-February/005619.html:
>  
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: Warren Lin <wlin121443 at yahoo.com> 
> Sent: Tuesday, February 21, 2012 11:01 AM
> Subject: Re: Testing mod_sms
>  
> hi you can find out how to use the lists at http://lists.freeswitch.org
>  
> try latest GIT, i put in a patch to fire the events with the heder
> Failed-Delivery true
> if you are listening for MESSAGE events you should catch them.
> ________________________________
>  
>  
> I can't find any reference to Failed-Delivery in mod_sms.c neither anywhere else in the source tree. 
> There is a field called Delivery-Failure, but seems to be used by other modules to indicate a failure on the response going out of freeswitch, not to freeswitch.
> Was this overwritten / rolled back at some point ? 
>  
> thanks
>  
> --
> lucas
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From 8f27e956 at gmail.com  Fri Nov  2 03:47:31 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Thu, 1 Nov 2012 20:47:31 -0400
Subject: [Freeswitch-users] mod_sms delivery failure
In-Reply-To: <BD87D3B1554F614190EB6E54CE95DDB606B937CB@mail2.patlive.local>
References: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com>
	<BD87D3B1554F614190EB6E54CE95DDB606B937CB@mail2.patlive.local>
Message-ID: <8804583403483732041@unknownmsgid>

This technique has worked on 2G gsm nets.  3G and 4G, it's more carrier
selective.  For a given message TEXT (not the address#), prefix the msg
string with one of either,

111
-or-
*noti#

E.g.  111Hello World -or- *noti#Hello World
(Fido in Canada supports/ed 111, T-Mobile supports/ed *noti#)

On 2012-11-01, at 19:49, Chad Engler <Chad.Engler at patlive.com> wrote:

I don?t think SIP MESSAGEs have a response/ack. I know that our provider
doesn?t send anything except the 200 OK when we send a message, and that
doesn?t mean it actually made it to the device. As far as I know there is
no way to know if and when the device gets the actual message without the
device responding with another message back through the via chain.



-Chad



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lucas Fontes
*Sent:* Thursday, November 01, 2012 1:38 PM
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* [Freeswitch-users] mod_sms delivery failure



Hi everyone,



I've been trying to determine if a message was successfully delivered to a
device.

Stumbled on a post from February (subject: Testing mod_sms)
http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-February/005619.html:



________________________________

From: Anthony Minessale <anthony.minessale at gmail.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev>>

To: Warren Lin <wlin121443 at yahoo.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev>>

Sent: Tuesday, February 21, 2012 11:01 AM

Subject: Re: Testing mod_sms



hi you can find out how to use the lists at http://lists.freeswitch.org



try latest GIT, i put in a patch to fire the events with the heder

Failed-Delivery true

if you are listening for MESSAGE events you should catch them.

________________________________





I can't find any reference to Failed-Delivery in mod_sms.c neither anywhere
else in the source tree.

There is a field called Delivery-Failure, but seems to be used by other
modules to indicate a failure on the response going out of freeswitch, not
to freeswitch.

Was this overwritten / rolled back at some point ?



thanks



--

lucas



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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From william at xofap.com  Fri Nov  2 06:54:06 2012
From: william at xofap.com (William Alianto)
Date: Fri, 02 Nov 2012 10:54:06 +0700
Subject: [Freeswitch-users] Nibblebill database update
In-Reply-To: <50922B45.9060302@bestnet.kharkov.ua>
References: <5091F32B.2010302@xofap.com> <50922B45.9060302@bestnet.kharkov.ua>
Message-ID: <509343DE.6070905@xofap.com>

Hi,

This is my nibblebill.conf.xml

<configuration name="nibblebill.conf" description="Nibble Billing">
<settings>
<!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with 
these options -->

<!-- Information for connecting to your database -->
<param name="db_username" value="root"/>
<param name="db_password" value="xxxxxxxx"/>
<param name="db_dsn" value="billing"/>

<!-- The database table where your CASH column is located -->
<param name="db_table" value="accounts"/>

<!-- The column name where we store the value of the account -->
<param name="db_column_cash" value="usage"/>

<!-- The column name for the unique ID identifying the account -->
<param name="db_column_account" value="user"/>

<!-- Custom SQL for loading current balance - overrides column names
channel vars are interpreted.
field nibble_balance is used for balance info

<param name="custom_sql_lookup" value="SELECT cash AS nibble_balance 
FROM accounts WHERE account_code='${nibble_account}'"/>
-->

<!-- Custom SQL for loading current balance - overrides column names
channel vars are interpreted.
nibble_increment is the amount to update

<param name="custom_sql_save" value="UPDATE accounts SET 
cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
-->


<!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e. 
bill only at end of call) -->
<param name="global_heartbeat" value="1"/>

<!-- By default, warn a caller when their balance is at $5.00. You can 
set this to a negative number. -->
<!--<param name="lowbal_amt" value="5"/>-->
<!--<param name="lowbal_action" value="play ding"/>-->

<!-- By default, terminate a caller when their balance hits $0.00. You 
can set this to a negative number. -->
<!--<param name="nobal_amt" value="0"/>
<param name="nobal_action" value="hangup"/>-->

<!-- If a call goes beyond a certain dollar amount, flag or terminate it -->
<!--<param name="percall_max_amt" value="100"/>
<param name="percall_action" value="hangup"/>-->

</settings>
</configuration>


On 11/01/2012 02:56 PM, Evgeniy Movlyan wrote:
> Hello.
> Show please your nibblebill.conf.xml.
>
> 01.11.2012 05:57, William Alianto ?????:
>> Hi,
>>
>> I was trying to use nibblebill as billing control of my FS. I have
>> configured the odbc and the nibblebill configuration. When I tried to
>> call, the query looks ok. But when I hanged up the call, it seems like
>> there is something wrong with the database connection, since it's not
>> updating the database at all. I got error message on CLI
>>
>> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE accounts
>> SET usage=age-0.476000 WHERE user='1001']
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
>> syntax; check the manual that corresponds to your MySQL server version
>> for the right syntax to use near 'usage=age-0.476000 WHERE
>> user=001'' at line 1
>> ]
>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE
>> accounts SET usage=age-0.476000 WHERE user='1001']
>> []
>> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to log to
>> database!
>> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup 
>> query
>> [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT usage
>> AS nibble_balance FROM accounts WHERE user=001']
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
>> syntax; check the manual that corresponds to your MySQL server version
>> for the right syntax to use near 'usage AS nibble_balance FROM accounts
>> WHERE user=001'' at line 1
>> ]
>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running this
>> query: [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
>>
>> Could anybody help me solve this issue?
>>
>> Regards
>>
>> _________________________________________________________________________ 
>>
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From succer110 at tiscali.it  Fri Nov  2 07:30:13 2012
From: succer110 at tiscali.it (succer110 at tiscali.it)
Date: Fri, 2 Nov 2012 05:30:13 +0100 (GMT+01:00)
Subject: [Freeswitch-users] freeswitch talk
Message-ID: <25807156.201351830613489.JavaMail.defaultUser@defaultHost>

Hi! I'm a freeswitch user, and i have a little experience in developing freeswitch modules.Within a month, I'm goint to do a talk in a conference, focused on freeswitch. Specifically, i was thinking about this topics:
-difference between FS and other pbx in the market-Where is the best place for FS in a VoIP architecture? (registrar, media encoding, application server, ivr, etc..)-how to scale
If you have some *in depth* documentation about this topics i will be very glad.I've found a lot of blog post about comparison, but i didn't found a specific analysis of code and benchmarks between various app.(in this days i'll watch a just discovered source of information: ClueCon channel on youtube :D )
Also, if you want to suggest some new topics to talk about, it would be very cool for me to study something new, if it's a good way for helping the community :)thank you!







Invita i tuoi amici e Tiscali ti premia! Il consiglio di un amico vale pi? di uno spot in TV. Per ogni nuovo abbonato 30 ? di premio per te e per lui! Un amico al mese e parli e navighi sempre gratis: http://freelosophy.tiscali.it/

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From belsarepankaj at gmail.com  Fri Nov  2 06:16:16 2012
From: belsarepankaj at gmail.com (Pankaj Belsare)
Date: Fri, 2 Nov 2012 08:46:16 +0530
Subject: [Freeswitch-users] survey after the call.
Message-ID: <CAMHZFZ5VGUQpjemV+rZe7MG=G5gM74F5tGH0N6SDymxcXiUQfA@mail.gmail.com>

Hi,
i am trying out a dialplan to take survey and append the database with the
responses and the call is hangup by the "agent" , can any body help with
suggestion/example

Pankaj
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From sms at icefire.qza.net.au  Fri Nov  2 07:16:39 2012
From: sms at icefire.qza.net.au (sms at icefire.qza.net.au)
Date: Fri, 2 Nov 2012 15:16:39 +1100
Subject: [Freeswitch-users] Remote pickup of on-hold extensions
Message-ID: <bc4d4e4f822530c463361a4f72774f6a.squirrel@icefire.qza.net.au>

Hi all,

I have a scenario that I'm thinking of implementing, but not sure if it's
possible, so here's the details:

Extensions 200 to 250 are regular internal sip phones, contained in a call
group (Ext 1000)

Extensions 300 to 305 are SIP-GSM and SIP-PSTN gateways with SIP client
firmware, namely those cheap GOIP GSM units on ebay, plus an SPA3000.
These are to be configured with the logon credentials of the extension and
behave as a typical SIP client.

The dialplan will route incoming and outgoing calls via these gateways as
if they were trunks. They will restricted privileges to prevent toll
fraud. The phones will either be Yealink T28p or Grandstream GXP2124, it
depends if GS can respond with a fix for their horrible AGC or not. The
gateways will be subscribed to the phone's DSS keys as BLF's.

Now, all this so far is reasonably straight forward. The next part is the
tricky part (you might've guessed where this is going....)

Let's say a call comes in on 300, routes to the call group and is picked
up by 1000, who then puts the call on hold. The BLF key for 300 shows it
as busy/on hold. 1000 then calls 1001 to take the call. 1001 then presses
BLF 300, which causes the call to be transferred to them and automatically
answered.

So is there any way in freeswitch to replicate this behavior? I know it's
possible to do SAA, but this gets tricky if I want to have time based call
groups, or to share incoming lines between branches. The method above
allows the lines to be present on all phones, while still configuring call
groups in the usual way and more or less replicating old key system
behavior.

Cheers,
Francis



From evgeniy at bestnet.kharkov.ua  Fri Nov  2 10:22:12 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Fri, 02 Nov 2012 09:22:12 +0200
Subject: [Freeswitch-users] Nibblebill database update
In-Reply-To: <509343DE.6070905@xofap.com>
References: <5091F32B.2010302@xofap.com> <50922B45.9060302@bestnet.kharkov.ua>
	<509343DE.6070905@xofap.com>
Message-ID: <509374A4.1090203@bestnet.kharkov.ua>

Seems you need to uncomment custom SQL-queries and rewrite it to 
according to your needs.

02.11.2012 05:54, William Alianto ?????:
> Hi,
>
> This is my nibblebill.conf.xml
>
> <configuration name="nibblebill.conf" description="Nibble Billing">
> <settings>
> <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
> these options -->
>
> <!-- Information for connecting to your database -->
> <param name="db_username" value="root"/>
> <param name="db_password" value="xxxxxxxx"/>
> <param name="db_dsn" value="billing"/>
>
> <!-- The database table where your CASH column is located -->
> <param name="db_table" value="accounts"/>
>
> <!-- The column name where we store the value of the account -->
> <param name="db_column_cash" value="usage"/>
>
> <!-- The column name for the unique ID identifying the account -->
> <param name="db_column_account" value="user"/>
>
> <!-- Custom SQL for loading current balance - overrides column names
> channel vars are interpreted.
> field nibble_balance is used for balance info
>
> <param name="custom_sql_lookup" value="SELECT cash AS nibble_balance
> FROM accounts WHERE account_code='${nibble_account}'"/>
> -->
>
> <!-- Custom SQL for loading current balance - overrides column names
> channel vars are interpreted.
> nibble_increment is the amount to update
>
> <param name="custom_sql_save" value="UPDATE accounts SET
> cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
> -->
>
>
> <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
> bill only at end of call) -->
> <param name="global_heartbeat" value="1"/>
>
> <!-- By default, warn a caller when their balance is at $5.00. You can
> set this to a negative number. -->
> <!--<param name="lowbal_amt" value="5"/>-->
> <!--<param name="lowbal_action" value="play ding"/>-->
>
> <!-- By default, terminate a caller when their balance hits $0.00. You
> can set this to a negative number. -->
> <!--<param name="nobal_amt" value="0"/>
> <param name="nobal_action" value="hangup"/>-->
>
> <!-- If a call goes beyond a certain dollar amount, flag or terminate it
> -->
> <!--<param name="percall_max_amt" value="100"/>
> <param name="percall_action" value="hangup"/>-->
>
> </settings>
> </configuration>
>
>
> On 11/01/2012 02:56 PM, Evgeniy Movlyan wrote:
>> Hello.
>> Show please your nibblebill.conf.xml.
>>
>> 01.11.2012 05:57, William Alianto ?????:
>>> Hi,
>>>
>>> I was trying to use nibblebill as billing control of my FS. I have
>>> configured the odbc and the nibblebill configuration. When I tried to
>>> call, the query looks ok. But when I hanged up the call, it seems like
>>> there is something wrong with the database connection, since it's not
>>> updating the database at all. I got error message on CLI
>>>
>>> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE accounts
>>> SET usage=age-0.476000 WHERE user='1001']
>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
>>> syntax; check the manual that corresponds to your MySQL server version
>>> for the right syntax to use near 'usage=age-0.476000 WHERE
>>> user=001'' at line 1
>>> ]
>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE
>>> accounts SET usage=age-0.476000 WHERE user='1001']
>>> []
>>> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to log to
>>> database!
>>> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup
>>> query
>>> [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT usage
>>> AS nibble_balance FROM accounts WHERE user=001']
>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
>>> syntax; check the manual that corresponds to your MySQL server version
>>> for the right syntax to use near 'usage AS nibble_balance FROM accounts
>>> WHERE user=001'' at line 1
>>> ]
>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running this
>>> query: [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
>>>
>>> Could anybody help me solve this issue?
>>>
>>> Regards
>>>
>>> _________________________________________________________________________
>>>
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>

-- 
Evgeniy Movlyan,
BestNet Ltd.


From william at xofap.com  Fri Nov  2 10:42:41 2012
From: william at xofap.com (William Alianto)
Date: Fri, 02 Nov 2012 14:42:41 +0700
Subject: [Freeswitch-users] Nibblebill database update
In-Reply-To: <509374A4.1090203@bestnet.kharkov.ua>
References: <5091F32B.2010302@xofap.com> <50922B45.9060302@bestnet.kharkov.ua>
	<509343DE.6070905@xofap.com> <509374A4.1090203@bestnet.kharkov.ua>
Message-ID: <50937971.4090809@xofap.com>

Thanks for pointing that out. I think I missed that part when I edited 
the config.

On 11/02/2012 02:22 PM, Evgeniy Movlyan wrote:
> Seems you need to uncomment custom SQL-queries and rewrite it to 
> according to your needs.
>
> 02.11.2012 05:54, William Alianto ?????:
>> Hi,
>>
>> This is my nibblebill.conf.xml
>>
>> <configuration name="nibblebill.conf" description="Nibble Billing">
>> <settings>
>> <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
>> these options -->
>>
>> <!-- Information for connecting to your database -->
>> <param name="db_username" value="root"/>
>> <param name="db_password" value="xxxxxxxx"/>
>> <param name="db_dsn" value="billing"/>
>>
>> <!-- The database table where your CASH column is located -->
>> <param name="db_table" value="accounts"/>
>>
>> <!-- The column name where we store the value of the account -->
>> <param name="db_column_cash" value="usage"/>
>>
>> <!-- The column name for the unique ID identifying the account -->
>> <param name="db_column_account" value="user"/>
>>
>> <!-- Custom SQL for loading current balance - overrides column names
>> channel vars are interpreted.
>> field nibble_balance is used for balance info
>>
>> <param name="custom_sql_lookup" value="SELECT cash AS nibble_balance
>> FROM accounts WHERE account_code='${nibble_account}'"/>
>> -->
>>
>> <!-- Custom SQL for loading current balance - overrides column names
>> channel vars are interpreted.
>> nibble_increment is the amount to update
>>
>> <param name="custom_sql_save" value="UPDATE accounts SET
>> cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
>> -->
>>
>>
>> <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
>> bill only at end of call) -->
>> <param name="global_heartbeat" value="1"/>
>>
>> <!-- By default, warn a caller when their balance is at $5.00. You can
>> set this to a negative number. -->
>> <!--<param name="lowbal_amt" value="5"/>-->
>> <!--<param name="lowbal_action" value="play ding"/>-->
>>
>> <!-- By default, terminate a caller when their balance hits $0.00. You
>> can set this to a negative number. -->
>> <!--<param name="nobal_amt" value="0"/>
>> <param name="nobal_action" value="hangup"/>-->
>>
>> <!-- If a call goes beyond a certain dollar amount, flag or terminate it
>> -->
>> <!--<param name="percall_max_amt" value="100"/>
>> <param name="percall_action" value="hangup"/>-->
>>
>> </settings>
>> </configuration>
>>
>>
>> On 11/01/2012 02:56 PM, Evgeniy Movlyan wrote:
>>> Hello.
>>> Show please your nibblebill.conf.xml.
>>>
>>> 01.11.2012 05:57, William Alianto ?????:
>>>> Hi,
>>>>
>>>> I was trying to use nibblebill as billing control of my FS. I have
>>>> configured the odbc and the nibblebill configuration. When I tried to
>>>> call, the query looks ok. But when I hanged up the call, it seems like
>>>> there is something wrong with the database connection, since it's not
>>>> updating the database at all. I got error message on CLI
>>>>
>>>> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE 
>>>> accounts
>>>> SET usage=age-0.476000 WHERE user='1001']
>>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
>>>> syntax; check the manual that corresponds to your MySQL server version
>>>> for the right syntax to use near 'usage=age-0.476000 WHERE
>>>> user=001'' at line 1
>>>> ]
>>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE
>>>> accounts SET usage=age-0.476000 WHERE user='1001']
>>>> []
>>>> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to 
>>>> log to
>>>> database!
>>>> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup
>>>> query
>>>> [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
>>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT 
>>>> usage
>>>> AS nibble_balance FROM accounts WHERE user=001']
>>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
>>>> syntax; check the manual that corresponds to your MySQL server version
>>>> for the right syntax to use near 'usage AS nibble_balance FROM 
>>>> accounts
>>>> WHERE user=001'' at line 1
>>>> ]
>>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running 
>>>> this
>>>> query: [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
>>>>
>>>> Could anybody help me solve this issue?
>>>>
>>>> Regards
>>>>
>>>> _________________________________________________________________________ 
>>>>
>>>>
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>>>
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>



From tomasz.szuster at gmail.com  Fri Nov  2 11:44:14 2012
From: tomasz.szuster at gmail.com (Tomasz Szuster)
Date: Fri, 2 Nov 2012 09:44:14 +0100
Subject: [Freeswitch-users] survey after the call.
In-Reply-To: <CAMHZFZ5VGUQpjemV+rZe7MG=G5gM74F5tGH0N6SDymxcXiUQfA@mail.gmail.com>
References: <CAMHZFZ5VGUQpjemV+rZe7MG=G5gM74F5tGH0N6SDymxcXiUQfA@mail.gmail.com>
Message-ID: <CAMAX9-2iVGqrM9tpBpE2WK-6eUHeXq6FK9wvtHFvHFtrc-XnJg@mail.gmail.com>

Hi Pankaj.

There is newfies dialer which meet your requirements.

Regards Tom.
On 2 Nov 2012 07:20, "Pankaj Belsare" <belsarepankaj at gmail.com> wrote:

> Hi,
> i am trying out a dialplan to take survey and append the database with the
> responses and the call is hangup by the "agent" , can any body help with
> suggestion/example
>
> Pankaj
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tahir at ictinnovations.com  Fri Nov  2 13:01:21 2012
From: tahir at ictinnovations.com (Tahir Almas)
Date: Fri, 2 Nov 2012 15:01:21 +0500
Subject: [Freeswitch-users] survey after the call.
In-Reply-To: <CAMAX9-2iVGqrM9tpBpE2WK-6eUHeXq6FK9wvtHFvHFtrc-XnJg@mail.gmail.com>
References: <CAMHZFZ5VGUQpjemV+rZe7MG=G5gM74F5tGH0N6SDymxcXiUQfA@mail.gmail.com>
	<CAMAX9-2iVGqrM9tpBpE2WK-6eUHeXq6FK9wvtHFvHFtrc-XnJg@mail.gmail.com>
Message-ID: <CAOL0t+JE404Bi0oFiGBjfpWyq3mhiE3V+-BKQx4bYGK7v3QFtw@mail.gmail.com>

If you like to customize open source code , check  ICTDialer
http://www.ictdialer.org and  ICTDialer IVR Designer
http://forum.ictdialer.org/viewtopic.php?f=7&t=2355,  It may  help you

and If you like a ready made solution, check
http://www.ictbroadcast.comwhich has survey campaign support with
detail reporting as well

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT

****************************************************************************************************************
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On Fri, Nov 2, 2012 at 1:44 PM, Tomasz Szuster <tomasz.szuster at gmail.com>wrote:

> Hi Pankaj.
>
> There is newfies dialer which meet your requirements.
>
> Regards Tom.
> On 2 Nov 2012 07:20, "Pankaj Belsare" <belsarepankaj at gmail.com> wrote:
>
>> Hi,
>> i am trying out a dialplan to take survey and append the database with
>> the responses and the call is hangup by the "agent" , can any body help
>> with suggestion/example
>>
>> Pankaj
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Fri Nov  2 13:25:28 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 02 Nov 2012 06:25:28 -0400
Subject: [Freeswitch-users] Nibblebill database update
In-Reply-To: <50937971.4090809@xofap.com>
References: <5091F32B.2010302@xofap.com> <509374A4.1090203@bestnet.kharkov.ua>
	<50937971.4090809@xofap.com>
Message-ID: <8724844.crljyV7uo8@sos>

 What is the structure of "accounts" table?

On Friday 02 November 2012 14:42:41 William Alianto wrote:
> Thanks for pointing that out. I think I missed that part when I edited
> the config.
> 
> On 11/02/2012 02:22 PM, Evgeniy Movlyan wrote:
> > Seems you need to uncomment custom SQL-queries and rewrite it to
> > according to your needs.
> > 
> > 02.11.2012 05:54, William Alianto ?????:
> >> Hi,
> >> 
> >> This is my nibblebill.conf.xml
> >> 
> >> <configuration name="nibblebill.conf" description="Nibble Billing">
> >> <settings>
> >> <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
> >> these options -->
> >> 
> >> <!-- Information for connecting to your database -->
> >> <param name="db_username" value="root"/>
> >> <param name="db_password" value="xxxxxxxx"/>
> >> <param name="db_dsn" value="billing"/>
> >> 
> >> <!-- The database table where your CASH column is located -->
> >> <param name="db_table" value="accounts"/>
> >> 
> >> <!-- The column name where we store the value of the account -->
> >> <param name="db_column_cash" value="usage"/>
> >> 
> >> <!-- The column name for the unique ID identifying the account -->
> >> <param name="db_column_account" value="user"/>
> >> 
> >> <!-- Custom SQL for loading current balance - overrides column names
> >> channel vars are interpreted.
> >> field nibble_balance is used for balance info
> >> 
> >> <param name="custom_sql_lookup" value="SELECT cash AS nibble_balance
> >> FROM accounts WHERE account_code='${nibble_account}'"/>
> >> -->
> >> 
> >> <!-- Custom SQL for loading current balance - overrides column names
> >> channel vars are interpreted.
> >> nibble_increment is the amount to update
> >> 
> >> <param name="custom_sql_save" value="UPDATE accounts SET
> >> cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
> >> -->
> >> 
> >> 
> >> <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
> >> bill only at end of call) -->
> >> <param name="global_heartbeat" value="1"/>
> >> 
> >> <!-- By default, warn a caller when their balance is at $5.00. You can
> >> set this to a negative number. -->
> >> <!--<param name="lowbal_amt" value="5"/>-->
> >> <!--<param name="lowbal_action" value="play ding"/>-->
> >> 
> >> <!-- By default, terminate a caller when their balance hits $0.00. You
> >> can set this to a negative number. -->
> >> <!--<param name="nobal_amt" value="0"/>
> >> <param name="nobal_action" value="hangup"/>-->
> >> 
> >> <!-- If a call goes beyond a certain dollar amount, flag or terminate it
> >> -->
> >> <!--<param name="percall_max_amt" value="100"/>
> >> <param name="percall_action" value="hangup"/>-->
> >> 
> >> </settings>
> >> </configuration>
> >> 
> >> On 11/01/2012 02:56 PM, Evgeniy Movlyan wrote:
> >>> Hello.
> >>> Show please your nibblebill.conf.xml.
> >>> 
> >>> 01.11.2012 05:57, William Alianto ?????:
> >>>> Hi,
> >>>> 
> >>>> I was trying to use nibblebill as billing control of my FS. I have
> >>>> configured the odbc and the nibblebill configuration. When I tried to
> >>>> call, the query looks ok. But when I hanged up the call, it seems like
> >>>> there is something wrong with the database connection, since it's not
> >>>> updating the database at all. I got error message on CLI
> >>>> 
> >>>> 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE
> >>>> accounts
> >>>> SET usage=age-0.476000 WHERE user='1001']
> >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
> >>>> syntax; check the manual that corresponds to your MySQL server version
> >>>> for the right syntax to use near 'usage=age-0.476000 WHERE
> >>>> user=001'' at line 1
> >>>> ]
> >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE
> >>>> accounts SET usage=age-0.476000 WHERE user='1001']
> >>>> []
> >>>> 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to
> >>>> log to
> >>>> database!
> >>>> 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup
> >>>> query
> >>>> [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
> >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT
> >>>> usage
> >>>> AS nibble_balance FROM accounts WHERE user=001']
> >>>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> >>>> Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL
> >>>> syntax; check the manual that corresponds to your MySQL server version
> >>>> for the right syntax to use near 'usage AS nibble_balance FROM
> >>>> accounts
> >>>> WHERE user=001'' at line 1
> >>>> ]
> >>>> 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running
> >>>> this
> >>>> query: [SELECT usage AS nibble_balance FROM accounts WHERE user=001']
> >>>> 
> >>>> Could anybody help me solve this issue?
> >>>> 
> >>>> Regards
> >>>> 
> >>>> _______________________________________________________________________
> >>>> __
> >>>> 
> >>>> 
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>> 
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>> s
> >>>> 
> >>>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vbvbrj at gmail.com  Fri Nov  2 15:35:51 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 02 Nov 2012 14:35:51 +0200
Subject: [Freeswitch-users] mod_callcenter agents didn't change to Receiving
Message-ID: <5093BE27.7000804@gmail.com>

Hello.

Got a problem when using mod_callcenter. Agents where in list at state 
'Available - Waiting'. They where tiered to a queue. Suddenly clients 
were queued but no agent were changed to Receiving. Mod_callcenter just 
kept clients in queue and didn't offer any agent to them. There were an 
agent for test which were logged out. When this agent logged in, it 
started to receive clients. The other agents remained as Waiting. When 
this test agent were logged out, again clients just sit in queue and 
agents with status Waiting were not tried to bridge with clients. Even a 
complete restart of FS didn't help. I had to delete these agents from 
agent's list completely with "callcenter_config agent del" and add them 
again.

What happened, why I had to completely remove agents from agent's list 
and not use agent log out?

In log didn't appear anything. Just info that to clients the position is 
announced. No info about trying any agent.

-- 
Mimiko desu.


From rjb at contactbeep.com  Fri Nov  2 15:54:19 2012
From: rjb at contactbeep.com (Rob Boutin)
Date: Fri, 2 Nov 2012 08:54:19 -0400
Subject: [Freeswitch-users] Just starting
In-Reply-To: <CACc3D0VbtvPyDrR=iEiRwyLs4Fh12UbL1JrvZ0dwKb4hGJdyCw@mail.gmail.com>
Message-ID: <800283A1-7157-411F-9751-532C76B318F4@contactbeep.com>

Thank you, 
 
Using this I was able to realize the embarrassingly simple solution.
My XML editor at some point during my setup lost permission to change files in the FreeSWITCH folder and wasn't throwing errors for some reason. 
 
 
Thanks again,
RJB
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vitalie Colosov
Sent: Thursday, November 01, 2012 3:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Just starting
 
If you could shutdown all devices and then capturing SIP Trace while bringing up the device with #300 (or any # except #301) using Wireshark, it will definitely add more info for investigation.
 
2012/11/1 Rob Boutin <rjb at contactbeep.com>
Hello,
 
I am just starting out with FreeSWITCH and I am researching it for a project at work that will hopefully end up with full implementation of FreeSWITCH and had a few short questions as I am having difficulties getting it off the ground.
 
I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system.
My tiny test network of phones includes an xlite softphone on the windows machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone.  They are being routed with a simple Linksys Etherfast router.
 
My issue is that only one phone at a time will ever be registered and able to make calls. I have four extensions configured with identical settings 300 - 303.  The peculiar thing is only the phone assigned to 301 will ever have a dial tone and be able to place calls.  It doesn't matter which device I move 301 to it is only 301 that ever works.  I was hoping to get some light shed on this, as it is likely related to some simple setting I over looked and would like to continue with my research.
 
Thanks in advance,
RJB

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
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From mkovalenko at cybervisiontech.com  Fri Nov  2 17:13:53 2012
From: mkovalenko at cybervisiontech.com (Max Kovalenko)
Date: Fri, 2 Nov 2012 10:13:53 -0400 (EDT)
Subject: [Freeswitch-users] media proxy
In-Reply-To: <29307400.127.1351848959125.JavaMail.master@VoiceJuggler>
Message-ID: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>

Hello,

There are two modes of media stream to be proxied or bypassed. 

1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc.
2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT traversal purposes. No media stream parsing is supported
3. Media bypass - RTP streams are bypassing FreeSWITCH at all.

There are also two channel parameters affecting above modes: bypass_media and proxy_media.

- Are these parameters independent? Meaning the combination: bypass_media=true AND proxy_media=true is possible. What effect will be?

- Does it means that if I want to turn on "light proxy" I would always need to set bypass_media=false AND proxy_media=true?

- How to set "hard proxy" (full media proxy) per channel before to bridge legs?

Waiting for your replay ASAP. Thank you in advance.

Best Regards.

Max Kovalenko 
Team Leader 
VoIP & UC Team 
Managed Services Dept. 
CyberVision Inc. 
------------------------------------------------- 
tel. +1 (201) 585-9809 ext. 215 
Email: mkovalenko at cybervisiontech.com
Skype: mkovalenko_cv, panzer_meister
WWW: www.cybervisiontech.com





From rjb at contactbeep.com  Fri Nov  2 17:48:52 2012
From: rjb at contactbeep.com (Rob Boutin)
Date: Fri, 2 Nov 2012 10:48:52 -0400
Subject: [Freeswitch-users] Just starting
In-Reply-To: <800283A1-7157-411F-9751-532C76B318F4@contactbeep.com>
Message-ID: <52E7F8D4-D9AA-403D-A982-D9D3316F3E59@contactbeep.com>

Alright so my phones have ring tones and can place calls to the predefined extensions (1000 - 1020) but upon trying to call one another it always says "temporarily unavailable" there is no difference in the actual extension files between the working ones and the new ones I defined but the server doesn't seem able to connect the calls from my new extensions to my other new extensions for some reason. 
 
Any ideas on this would be very helpful, and I feel like whatever this is should be the last issue I face.
 
 
Thanks again,
RJB 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Boutin
Sent: Friday, November 02, 2012 8:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Just starting
 
Thank you, 
 
Using this I was able to realize the embarrassingly simple solution.
My XML editor at some point during my setup lost permission to change files in the FreeSWITCH folder and wasn't throwing errors for some reason. 
 
 
Thanks again,
RJB
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vitalie Colosov
Sent: Thursday, November 01, 2012 3:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Just starting
 
If you could shutdown all devices and then capturing SIP Trace while bringing up the device with #300 (or any # except #301) using Wireshark, it will definitely add more info for investigation.
 
2012/11/1 Rob Boutin <rjb at contactbeep.com>
Hello,
 
I am just starting out with FreeSWITCH and I am researching it for a project at work that will hopefully end up with full implementation of FreeSWITCH and had a few short questions as I am having difficulties getting it off the ground.
 
I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system.
My tiny test network of phones includes an xlite softphone on the windows machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone.  They are being routed with a simple Linksys Etherfast router.
 
My issue is that only one phone at a time will ever be registered and able to make calls. I have four extensions configured with identical settings 300 - 303.  The peculiar thing is only the phone assigned to 301 will ever have a dial tone and be able to place calls.  It doesn't matter which device I move 301 to it is only 301 that ever works.  I was hoping to get some light shed on this, as it is likely related to some simple setting I over looked and would like to continue with my research.
 
Thanks in advance,
RJB

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
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From krice at freeswitch.org  Fri Nov  2 18:12:11 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 02 Nov 2012 10:12:11 -0500
Subject: [Freeswitch-users] media proxy
In-Reply-To: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>
Message-ID: <CCB94CFB.5EDC5%krice@freeswitch.org>

Default, proxy and bybass media are exclusive of each other, you can not
combine them on a single call...

Proxy media is just that proxy the media

Bypass media is just that, bypass freeswitch and send the media direct
between the end point...

Setting this options is fairly well documented on the wiki...



On 11/2/12 9:13 AM, "Max Kovalenko" <mkovalenko at cybervisiontech.com> wrote:

> Hello,
> 
> There are two modes of media stream to be proxied or bypassed.
> 
> 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc.
> 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> traversal purposes. No media stream parsing is supported
> 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> 
> There are also two channel parameters affecting above modes: bypass_media and
> proxy_media.
> 
> - Are these parameters independent? Meaning the combination: bypass_media=true
> AND proxy_media=true is possible. What effect will be?
> 
> - Does it means that if I want to turn on "light proxy" I would always need to
> set bypass_media=false AND proxy_media=true?
> 
> - How to set "hard proxy" (full media proxy) per channel before to bridge
> legs?
> 
> Waiting for your replay ASAP. Thank you in advance.
> 
> Best Regards.
> 
> Max Kovalenko 
> Team Leader 
> VoIP & UC Team 
> Managed Services Dept.
> CyberVision Inc. 
> -------------------------------------------------
> tel. +1 (201) 585-9809 ext. 215
> Email: mkovalenko at cybervisiontech.com
> Skype: mkovalenko_cv, panzer_meister
> WWW: www.cybervisiontech.com
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From mike at jerris.com  Fri Nov  2 18:39:27 2012
From: mike at jerris.com (Michael Jerris)
Date: Fri, 2 Nov 2012 11:39:27 -0400
Subject: [Freeswitch-users] media proxy
In-Reply-To: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>
References: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>
Message-ID: <2818E18E-E28D-4F3A-B535-690CFDBFE23B@jerris.com>

There should be no reason to ever use proxy mode (2).

default mode that you call "hard proxy" requires no settings to set it, its the default.

On Nov 2, 2012, at 10:13 AM, Max Kovalenko <mkovalenko at cybervisiontech.com> wrote:

> Hello,
> 
> There are two modes of media stream to be proxied or bypassed. 
> 
> 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc.
> 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT traversal purposes. No media stream parsing is supported
> 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> 
> There are also two channel parameters affecting above modes: bypass_media and proxy_media.
> 
> - Are these parameters independent? Meaning the combination: bypass_media=true AND proxy_media=true is possible. What effect will be?
> 
> - Does it means that if I want to turn on "light proxy" I would always need to set bypass_media=false AND proxy_media=true?
> 
> - How to set "hard proxy" (full media proxy) per channel before to bridge legs?
> 
> Waiting for your replay ASAP. Thank you in advance.



From rdmitry0911 at gmail.com  Fri Nov  2 18:28:25 2012
From: rdmitry0911 at gmail.com (Dmitry R)
Date: Fri, 2 Nov 2012 19:28:25 +0400
Subject: [Freeswitch-users] Passing dtmf to an IVR behind Google Voice
	problem
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATLu3mYWl7E69+584sdKQ4sKAAAAQAAAA1JHoHyZlVEC7jGSvxqpKoAEAAAAA@gmail.com>

I've just tested the same configuration under asterisk 11 with
chan_motif/xmpp. Everything works great. DTMF codes get thru to IVR behind a
Google Voice entrance. This means, that either FS dingaling driver doesn't
work correctly or there is a mess in dtmf signaling between dingaling and
sofia. 
Is there any way to fix this?
 
Dmitry

  _____  

From: Dmitry R [mailto:rdmitry0911 at gmail.com] 
Sent: Thursday, November 01, 2012 10:06 AM
To: 'FreeSWITCH Users Help'
Subject: RE: [Freeswitch-users] Passing dtmf to an IVR behind Google
Voiceproblem


No, that's not my case. I can't receive dtmf while that workaround is about
sending dtmf codes
 
Dmitry

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, November 01, 2012 12:31 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Passing dtmf to an IVR behind Google
Voiceproblem


For the sake of posterity (and search engines) there's a workaround
mentioned here:
http://wiki.freeswitch.org/wiki/Dingaling#How_do_I_send_DTMF_with_Googletalk
.3F

-MC


On Wed, Oct 31, 2012 at 11:29 AM, Dmitry R <rdmitry0911 at gmail.com> wrote:


Hi, does anybody know how to pass dtmf codes to an IVR behind the Google
Voice entrance number?
A simple example code from http://wiki.freeswitch.org/wiki/Google_Voice

  <extension name="google_in" continue="true">
     <condition field="caller_id_name" expression="^(Google Voice)$" >
         <action application="log" data="Google Voice Call Incoming" />
         <action application="set" data="execute_on_answer=send_dtmf 1"/>
     </condition>
   </extension>

doesn't work for me. DTMF codes can't get thru. Everything else other then
that works fine.

Any help would be very much appreciated, Dmitry


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




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

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





-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



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From kris at kriskinc.com  Fri Nov  2 19:06:00 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 2 Nov 2012 12:06:00 -0400
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <CAKDfjgczaZ0kkeJ-ZZR7xG9_+Z1MriOM_x0UwoapP3f7V0n+rQ@mail.gmail.com>

Where are you setting call_debug=true?  Are you doing it before a
transfer?  Maybe you need to use inline?

http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

On Thu, Nov 1, 2012 at 5:00 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> I've got the following SIP gateway which registers successfully at the ITSP:
>
>   <gateway name="sipcall_41325133196">
>     <param name="username" value="41325133196"/>
>     <param name="password" value="XXXXXXXXXX"/>
>     <param name="extension" value="0325133196"/>
>     <param name="expire-seconds" value="125"/>
>     <param name="register" value="true"/>
>     <param name="register-transport" value="udp"/>
>     <param name="proxy" value="free2.voipgateway.org"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="caller-id-in-from" value="false"/>
>     <param name="ping" value="27"/>
>     <variables>
>      <variable name="sbc_callee_ccode" data="41" direction="inbound"/>
>      <variable name="sbc_caller_ccode" data="41" direction="inbound"/>
>     </variables>
>   </gateway>
>
>
> Then the call from PSTN goes into the proper context, and the first condition in the context is:
>
>    <extension name="call_debug" continue="true">
>      <condition field="${call_debug}" expression="^true$" break="never">
>        <action application="info"/>
>        <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/>
>      </condition>
>    </extension>
>
>
> Neither the info, nor the log show my two variables on inbound calls.
>
>
> Am I doing something wrong or is it a bug?
>
> I'm using git commit 94664868a8 in v1.2.stable branch
>
>
>
> thanks,
> stanislav
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From sos at sokhapkin.dyndns.org  Fri Nov  2 19:32:06 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 02 Nov 2012 12:32:06 -0400
Subject: [Freeswitch-users] media proxy
In-Reply-To: <2818E18E-E28D-4F3A-B535-690CFDBFE23B@jerris.com>
References: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>
	<2818E18E-E28D-4F3A-B535-690CFDBFE23B@jerris.com>
Message-ID: <1945566.6VRVmflkY8@sos>

 Will proxy mode 2 work with codecs not supported by freeswitch?

On Friday 02 November 2012 11:39:27 Michael Jerris wrote:
> There should be no reason to ever use proxy mode (2).
> 
> default mode that you call "hard proxy" requires no settings to set it, its
> the default.
> On Nov 2, 2012, at 10:13 AM, Max Kovalenko <mkovalenko at cybervisiontech.com> 
wrote:
> > Hello,
> > 
> > There are two modes of media stream to be proxied or bypassed.
> > 
> > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc.
> > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > traversal purposes. No media stream parsing is supported 3. Media bypass
> > - RTP streams are bypassing FreeSWITCH at all.
> > 
> > There are also two channel parameters affecting above modes: bypass_media
> > and proxy_media.
> > 
> > - Are these parameters independent? Meaning the combination:
> > bypass_media=true AND proxy_media=true is possible. What effect will be?
> > 
> > - Does it means that if I want to turn on "light proxy" I would always
> > need to set bypass_media=false AND proxy_media=true?
> > 
> > - How to set "hard proxy" (full media proxy) per channel before to bridge
> > legs?
> > 
> > Waiting for your replay ASAP. Thank you in advance.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From andrew at cassidywebservices.co.uk  Fri Nov  2 19:56:07 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 2 Nov 2012 16:56:07 +0000
Subject: [Freeswitch-users] Remote pickup of on-hold extensions
In-Reply-To: <bc4d4e4f822530c463361a4f72774f6a.squirrel@icefire.qza.net.au>
References: <bc4d4e4f822530c463361a4f72774f6a.squirrel@icefire.qza.net.au>
Message-ID: <CAMbW9oKH0j_oRuMjaDOGb6A=eBMf79XZMezSBeggWTdmYJDrDg@mail.gmail.com>

Personally I'd just use the Blind Transfer features on your handsets. 1000
picks up the call, presses transfer on the phone, then dials 1001 and hangs
up. 1001 rings and is connected to the transferred call. Sure you'e not got
the BLF, but the idea works fine. You could use call parking, but that has
more steps to it.

On 2 November 2012 04:16, <sms at icefire.qza.net.au> wrote:

> Hi all,
>
> I have a scenario that I'm thinking of implementing, but not sure if it's
> possible, so here's the details:
>
> Extensions 200 to 250 are regular internal sip phones, contained in a call
> group (Ext 1000)
>
> Extensions 300 to 305 are SIP-GSM and SIP-PSTN gateways with SIP client
> firmware, namely those cheap GOIP GSM units on ebay, plus an SPA3000.
> These are to be configured with the logon credentials of the extension and
> behave as a typical SIP client.
>
> The dialplan will route incoming and outgoing calls via these gateways as
> if they were trunks. They will restricted privileges to prevent toll
> fraud. The phones will either be Yealink T28p or Grandstream GXP2124, it
> depends if GS can respond with a fix for their horrible AGC or not. The
> gateways will be subscribed to the phone's DSS keys as BLF's.
>
> Now, all this so far is reasonably straight forward. The next part is the
> tricky part (you might've guessed where this is going....)
>
> Let's say a call comes in on 300, routes to the call group and is picked
> up by 1000, who then puts the call on hold. The BLF key for 300 shows it
> as busy/on hold. 1000 then calls 1001 to take the call. 1001 then presses
> BLF 300, which causes the call to be transferred to them and automatically
> answered.
>
> So is there any way in freeswitch to replicate this behavior? I know it's
> possible to do SAA, but this gets tricky if I want to have time based call
> groups, or to share incoming lines between branches. The method above
> allows the lines to be present on all phones, while still configuring call
> groups in the usual way and more or less replicating old key system
> behavior.
>
> Cheers,
> Francis
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From anthony.minessale at gmail.com  Fri Nov  2 20:00:57 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Nov 2012 12:00:57 -0500
Subject: [Freeswitch-users] Just starting
In-Reply-To: <52E7F8D4-D9AA-403D-A982-D9D3316F3E59@contactbeep.com>
References: <800283A1-7157-411F-9751-532C76B318F4@contactbeep.com>
	<52E7F8D4-D9AA-403D-A982-D9D3316F3E59@contactbeep.com>
Message-ID: <CAKbxfG-Qmt1COKz38op9xo1_=C8P7CMGM6Lsh5njRjcbH8B_YA@mail.gmail.com>

look in dialplan/default.xml at Local_Extension

<condition field="destination_number" expression="^(10[01][0-9])$">

Change that regex to also cover your new extensions.

You may find some use on the wiki section on xml dialplan or possibly one
of our tech books.



On Fri, Nov 2, 2012 at 9:48 AM, Rob Boutin <rjb at contactbeep.com> wrote:

> Alright so my phones have ring tones and can place calls to the predefined
> extensions (1000 ? 1020) but upon trying to call one another it always says
> ?temporarily unavailable? there is no difference in the actual extension
> files between the working ones and the new ones I defined but the server
> doesn?t seem able to connect the calls from my new extensions to my other
> new extensions for some reason. ****
>
> ** **
>
> Any ideas on this would be very helpful, and I feel like whatever this is
> should be the last issue I face.****
>
> ** **
>
> ** **
>
> Thanks again,****
>
> RJB ****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rob Boutin
> *Sent:* Friday, November 02, 2012 8:54 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Just starting****
>
> ** **
>
> Thank you, ****
>
> ** **
>
> Using this I was able to realize the embarrassingly simple solution.****
>
> My XML editor at some point during my setup lost permission to change
> files in the FreeSWITCH folder and wasn?t throwing errors for some reason.
> ****
>
> ** **
>
> ** **
>
> Thanks again,****
>
> RJB****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *On Behalf Of *Vitalie Colosov
> *Sent:* Thursday, November 01, 2012 3:44 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Just starting****
>
> ** **
>
> If you could shutdown all devices and then capturing SIP Trace
> while bringing up the device with #300 (or any # except #301) using
> Wireshark, it will definitely add more info for investigation.****
>
> ** **
>
> 2012/11/1 Rob Boutin <rjb at contactbeep.com>****
>
> Hello,****
>
>  ****
>
> I am just starting out with FreeSWITCH and I am researching it for a
> project at work that will hopefully end up with full implementation of
> FreeSWITCH and had a few short questions as I am having difficulties
> getting it off the ground.****
>
>  ****
>
> I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system.
> ****
>
> My tiny test network of phones includes an xlite softphone on the windows
> machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone.  They are being
> routed with a simple Linksys Etherfast router.****
>
>  ****
>
> My issue is that only one phone at a time will ever be registered and able
> to make calls. I have four extensions configured with identical settings
> 300 ? 303.  The peculiar thing is only the phone assigned to 301 will ever
> have a dial tone and be able to place calls.  It doesn?t matter which
> device I move 301 to it is only 301 that ever works.  I was hoping to get
> some light shed on this, as it is likely related to some simple setting I
> over looked and would like to continue with my research.****
>
>  ****
>
> Thanks in advance,****
>
> RJB****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From vetali100 at gmail.com  Fri Nov  2 20:20:32 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Fri, 2 Nov 2012 10:20:32 -0700
Subject: [Freeswitch-users] Just starting
In-Reply-To: <52E7F8D4-D9AA-403D-A982-D9D3316F3E59@contactbeep.com>
References: <800283A1-7157-411F-9751-532C76B318F4@contactbeep.com>
	<52E7F8D4-D9AA-403D-A982-D9D3316F3E59@contactbeep.com>
Message-ID: <CACc3D0Ufrxc+sty0paAS43mnd0aynXYwNyiqFBkU0e9KDhFk7Q@mail.gmail.com>

It should be another simple issue. Check your default.xml dialplan. It has
entries for 1000-1020 numbers, but you must similarly add for your new #s.



2012/11/2 Rob Boutin <rjb at contactbeep.com>

> Alright so my phones have ring tones and can place calls to the predefined
> extensions (1000 ? 1020) but upon trying to call one another it always says
> ?temporarily unavailable? there is no difference in the actual extension
> files between the working ones and the new ones I defined but the server
> doesn?t seem able to connect the calls from my new extensions to my other
> new extensions for some reason. ****
>
> ** **
>
> Any ideas on this would be very helpful, and I feel like whatever this is
> should be the last issue I face.****
>
> ** **
>
> ** **
>
> Thanks again,****
>
> RJB ****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rob Boutin
> *Sent:* Friday, November 02, 2012 8:54 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Just starting****
>
> ** **
>
> Thank you, ****
>
> ** **
>
> Using this I was able to realize the embarrassingly simple solution.****
>
> My XML editor at some point during my setup lost permission to change
> files in the FreeSWITCH folder and wasn?t throwing errors for some reason.
> ****
>
> ** **
>
> ** **
>
> Thanks again,****
>
> RJB****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *On Behalf Of *Vitalie Colosov
> *Sent:* Thursday, November 01, 2012 3:44 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Just starting****
>
> ** **
>
> If you could shutdown all devices and then capturing SIP Trace
> while bringing up the device with #300 (or any # except #301) using
> Wireshark, it will definitely add more info for investigation.****
>
> ** **
>
> 2012/11/1 Rob Boutin <rjb at contactbeep.com>****
>
> Hello,****
>
>  ****
>
> I am just starting out with FreeSWITCH and I am researching it for a
> project at work that will hopefully end up with full implementation of
> FreeSWITCH and had a few short questions as I am having difficulties
> getting it off the ground.****
>
>  ****
>
> I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system.
> ****
>
> My tiny test network of phones includes an xlite softphone on the windows
> machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone.  They are being
> routed with a simple Linksys Etherfast router.****
>
>  ****
>
> My issue is that only one phone at a time will ever be registered and able
> to make calls. I have four extensions configured with identical settings
> 300 ? 303.  The peculiar thing is only the phone assigned to 301 will ever
> have a dial tone and be able to place calls.  It doesn?t matter which
> device I move 301 to it is only 301 that ever works.  I was hoping to get
> some light shed on this, as it is likely related to some simple setting I
> over looked and would like to continue with my research.****
>
>  ****
>
> Thanks in advance,****
>
> RJB****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Fri Nov  2 20:25:39 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 02 Nov 2012 12:25:39 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
Message-ID: <CCB96C43.5EDD5%krice@freeswitch.org>

Hey Guys,

There?s some new Database Goodness in the core of FreeSWITCH that can lead
to some unexpected things for you guys updating existing installations using
ODBC.

We now have Native PostgreSQL support in the core, and along with this comes
some changes to the various ?odbc-dsn? settings around the tree.

If you are using the format ?dsn:username:password? you wont be affected,
however if you are just specifying a DSN as ?dsn? you will need to listen up

The settings for this field have changed.

pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
options='-c client_min_messages=NOTICE for postgresql (the stuff after
pgsql:// is a standard libpq connect string for you programmer types)
odbc://dns:username:password  for ODBC   ( dsn:: should also work or
dns:username: )
sqlite://filename   for sqlite different SQLite Databases

I think we still need to doc this up good, but its there and its coming
strong...

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

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From david at styleflare.com  Fri Nov  2 22:05:53 2012
From: david at styleflare.com (David J)
Date: Fri, 2 Nov 2012 15:05:53 -0400
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCB96C43.5EDD5%krice@freeswitch.org>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
Message-ID: <CAPoPLj8pFu+vijaVzWZMytvMzD+YzBrFneCpLyESK+FebFmVNg@mail.gmail.com>

WOW this is awesome news

Thx
On Nov 2, 2012 3:02 PM, "Ken Rice" <krice at freeswitch.org> wrote:

>  Hey Guys,
>
> There?s some new Database Goodness in the core of FreeSWITCH that can lead
> to some unexpected things for you guys updating existing installations
> using ODBC.
>
> We now have Native PostgreSQL support in the core, and along with this
> comes some changes to the various ?odbc-dsn? settings around the tree.
>
> If you are using the format ?dsn:username:password? you wont be affected,
> however if you are just specifying a DSN as ?dsn? you will need to listen up
>
> The settings for this field have changed.
>
> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> options='-c client_min_messages=NOTICE for postgresql (the stuff after
> pgsql:// is a standard libpq connect string for you programmer types)
> odbc://dns:username:password  for ODBC   ( dsn:: should also work or
> dns:username: )
> sqlite://filename   for sqlite different SQLite Databases
>
> I think we still need to doc this up good, but its there and its coming
> strong...
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gabe at gundy.org  Fri Nov  2 22:10:40 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 2 Nov 2012 13:10:40 -0600
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCB96C43.5EDD5%krice@freeswitch.org>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
Message-ID: <CAMwZ-tJiuyA1CmzwTHGpTxr8cy_n-9rjgTkGqYu=DxeHDRz5mA@mail.gmail.com>

On Fri, Nov 2, 2012 at 11:25 AM, Ken Rice <krice at freeswitch.org> wrote:
> We now have Native PostgreSQL support in the core, and along with this comes
> some changes to the various ?odbc-dsn? settings around the tree.

Yey for PortgreSQL!!!


Gabe


From krice at freeswitch.org  Fri Nov  2 22:14:42 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 02 Nov 2012 14:14:42 -0500
Subject: [Freeswitch-users] I think its FreeForAll Time
Message-ID: <CCB985D2.5EE23%krice@freeswitch.org>

Sure we shoot for 4PM Eastern....

But lets get it started now....

sip:888 at conference.freeswitch.org as usual... Get in here...

Today?s Topic: TACOS! (or something)


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

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From ssinyagin at yahoo.com  Fri Nov  2 23:14:23 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Fri, 2 Nov 2012 13:14:23 -0700 (PDT)
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <CAKDfjgczaZ0kkeJ-ZZR7xG9_+Z1MriOM_x0UwoapP3f7V0n+rQ@mail.gmail.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
	<CAKDfjgczaZ0kkeJ-ZZR7xG9_+Z1MriOM_x0UwoapP3f7V0n+rQ@mail.gmail.com>
Message-ID: <1351887263.12126.YahooMailNeo@web39305.mail.mud.yahoo.com>

it is set in the usual place, in vars.xml as a 
global variable. Besides, I do get the full Info 
output, but those gateway variables are missing:
https://gist.github.com/3997100





----- Original Message -----
> From: Kristian Kielhofner <kris at kriskinc.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Friday, November 2, 2012 5:06 PM
> Subject: Re: [Freeswitch-users] Sofia gateway variables
> 
> Where are you setting call_debug=true?? Are you doing it before a
> transfer?? Maybe you need to use inline?
> 
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
> 
> On Thu, Nov 1, 2012 at 5:00 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> 
> wrote:
>>  I've got the following SIP gateway which registers successfully at the 
> ITSP:
>> 
>> ?  <gateway name="sipcall_41325133196">
>> ? ?  <param name="username" value="41325133196"/>
>> ? ?  <param name="password" value="XXXXXXXXXX"/>
>> ? ?  <param name="extension" value="0325133196"/>
>> ? ?  <param name="expire-seconds" value="125"/>
>> ? ?  <param name="register" value="true"/>
>> ? ?  <param name="register-transport" 
> value="udp"/>
>> ? ?  <param name="proxy" 
> value="free2.voipgateway.org"/>
>> ? ?  <param name="retry-seconds" value="30"/>
>> ? ?  <param name="caller-id-in-from" 
> value="false"/>
>> ? ?  <param name="ping" value="27"/>
>> ? ?  <variables>
>> ? ? ? <variable name="sbc_callee_ccode" data="41" 
> direction="inbound"/>
>> ? ? ? <variable name="sbc_caller_ccode" data="41" 
> direction="inbound"/>
>> ? ?  </variables>
>> ?  </gateway>
>> 
>> 
>>  Then the call from PSTN goes into the proper context, and the first 
> condition in the context is:
>> 
>> ? ? <extension name="call_debug" continue="true">
>> ? ? ? <condition field="${call_debug}" 
> expression="^true$" break="never">
>> ? ? ? ? <action application="info"/>
>> ? ? ? ? <action application="log" data="INFO Country code: 
> ${sbc_callee_ccode}"/>
>> ? ? ? </condition>
>> ? ? </extension>
>> 
>> 
>>  Neither the info, nor the log show my two variables on inbound calls.
>> 
>> 
>>  Am I doing something wrong or is it a bug?
>> 
>>  I'm using git commit 94664868a8 in v1.2.stable branch
>> 
>> 
>> 
>>  thanks,
>>  stanislav
>> 
>> 
>>  _________________________________________________________________________
>>  Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org
>>  http://www.freeswitchsolutions.com
>> 
>>  
>>  
>> 
>>  Official FreeSWITCH Sites
>>  http://www.freeswitch.org
>>  http://wiki.freeswitch.org
>>  http://www.cluecon.com
>> 
>>  FreeSWITCH-users mailing list
>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
> 
> 
> 
> -- 
> Kristian Kielhofner
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From 8f27e956 at gmail.com  Fri Nov  2 23:34:32 2012
From: 8f27e956 at gmail.com (Scott)
Date: Fri, 2 Nov 2012 16:34:32 -0400
Subject: [Freeswitch-users] sqlite3 and regex
Message-ID: <CA+Y7Lf6B0LUjqOpq2M2NFtpCCfhirg7JO4KCo_dGu+i2VP35ng@mail.gmail.com>

"LIKE" notwithstanding, sqlite3 does not have a built-in true regex
function; it does allow for a a c-language hook to one.  Given fs extensive
use of the regex engine and of sqlite3, we're wondering if the hook is
already written and rolled.  If so, can the rest of us hook it to our
sqlite3 uses (e.g. from dial plan lua sqlite3).

With thanks,
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From anthony.minessale at gmail.com  Fri Nov  2 23:40:40 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Nov 2012 15:40:40 -0500
Subject: [Freeswitch-users] freeswitch talk
In-Reply-To: <25807156.201351830613489.JavaMail.defaultUser@defaultHost>
References: <25807156.201351830613489.JavaMail.defaultUser@defaultHost>
Message-ID: <CAKbxfG8NFk2ov-hRHi==hwrrsbh5UFy+hZEzN3bjwyaEGYgpoQ@mail.gmail.com>

The docs are mostly about using FS but the more you learn about what you
can do on the outside the easier it becomes to learn about the inside but
the best documentation we have on making modules are the modules themselves
which are designed to showcase capability of the core.  I recommend you
hang out on IRC to get started.

irc.freenode.net #freeswitch



On Thu, Nov 1, 2012 at 11:30 PM, succer110 at tiscali.it
<succer110 at tiscali.it>wrote:

> Hi!
> I'm a freeswitch user, and i have a little experience in developing
> freeswitch modules.
> Within a month, I'm goint to do a talk in a conference, focused on
> freeswitch.
> Specifically, i was thinking about this topics:
>
> -difference between FS and other pbx in the market
> -Where is the best place for FS in a VoIP architecture? (registrar, media
> encoding, application server, ivr, etc..)
> -how to scale
>
> If you have some *in depth* documentation about this topics i will be very
> glad.
> I've found a lot of blog post about comparison, but i didn't found a
> specific analysis of code and benchmarks between various app.
> (in this days i'll watch a just discovered source of information: ClueCon
> channel on youtube :D )
>
> Also, if you want to suggest some new topics to talk about, it would be
> very cool for me to study something new,
> if it's a good way for helping the community :)
> thank you!
>
>
>
>
>
>
>
>
> Invita i tuoi amici e Tiscali ti premia! Il consiglio di un amico vale pi?
> di uno spot in TV. Per ogni nuovo abbonato 30 ? di premio per te e per lui!
> Un amico al mese e parli e navighi sempre gratis:
> http://freelosophy.tiscali.it/
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From yiftah at choochee.com  Fri Nov  2 23:50:23 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 2 Nov 2012 13:50:23 -0700
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAPoPLj8pFu+vijaVzWZMytvMzD+YzBrFneCpLyESK+FebFmVNg@mail.gmail.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<CAPoPLj8pFu+vijaVzWZMytvMzD+YzBrFneCpLyESK+FebFmVNg@mail.gmail.com>
Message-ID: <CAFVUaV8Uz=Bj70JUED_s4WJGDsEnU1UkxYWS4tzKH0FxeHsgAQ@mail.gmail.com>

+1 for this adding db for core FreeSWITCH would be a good news for us as
well


On Fri, Nov 2, 2012 at 12:05 PM, David J <david at styleflare.com> wrote:

> WOW this is awesome news
>
> Thx
> On Nov 2, 2012 3:02 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>
>>  Hey Guys,
>>
>> There?s some new Database Goodness in the core of FreeSWITCH that can
>> lead to some unexpected things for you guys updating existing installations
>> using ODBC.
>>
>> We now have Native PostgreSQL support in the core, and along with this
>> comes some changes to the various ?odbc-dsn? settings around the tree.
>>
>> If you are using the format ?dsn:username:password? you wont be affected,
>> however if you are just specifying a DSN as ?dsn? you will need to listen up
>>
>> The settings for this field have changed.
>>
>> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
>> options='-c client_min_messages=NOTICE for postgresql (the stuff after
>> pgsql:// is a standard libpq connect string for you programmer types)
>> odbc://dns:username:password  for ODBC   ( dsn:: should also work or
>> dns:username: )
>> sqlite://filename   for sqlite different SQLite Databases
>>
>> I think we still need to doc this up good, but its there and its coming
>> strong...
>>
>> --
>> Ken
>> *http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> *irc.freenode.net #freeswitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vipkilla at gmail.com  Fri Nov  2 23:54:08 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 2 Nov 2012 16:54:08 -0400
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAPoPLj8pFu+vijaVzWZMytvMzD+YzBrFneCpLyESK+FebFmVNg@mail.gmail.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<CAPoPLj8pFu+vijaVzWZMytvMzD+YzBrFneCpLyESK+FebFmVNg@mail.gmail.com>
Message-ID: <CAC-LwPOy7zsk-kdA72DiD0i7f=6seWtWNjs1R1jaZTvsABBsow@mail.gmail.com>

Can we do the same for MySQL?
:)


From ash at url.net.au  Sat Nov  3 01:06:52 2012
From: ash at url.net.au (Ashley Breeden)
Date: Sat, 3 Nov 2012 09:06:52 +1100
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCB96C43.5EDD5%krice@freeswitch.org>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
Message-ID: <F1691313-1D90-48BF-B051-3548A64AC297@url.net.au>

Hi Ken,

Excellent news for Postgres and FreeSWITCH. 


I'll ask this as I am sure there are a lot of people wondering, are there plans to support MySQL natively as well?


A.



On 03/11/2012, at 4:25 AM, Ken Rice wrote:

> Hey Guys,
> 
> There?s some new Database Goodness in the core of FreeSWITCH that can lead to some unexpected things for you guys updating existing installations using ODBC.
> 
> We now have Native PostgreSQL support in the core, and along with this comes some changes to the various ?odbc-dsn? settings around the tree.
> 
> If you are using the format ?dsn:username:password? you wont be affected, however if you are just specifying a DSN as ?dsn? you will need to listen up
> 
> The settings for this field have changed.
> 
> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE for postgresql (the stuff after pgsql:// is a standard libpq connect string for you programmer types)
> odbc://dns:username:password  for ODBC   ( dsn:: should also work or dns:username: )
> sqlite://filename   for sqlite different SQLite Databases
> 
> I think we still need to doc this up good, but its there and its coming strong...
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sat Nov  3 01:08:05 2012
From: mike at jerris.com (Michael Jerris)
Date: Fri, 2 Nov 2012 18:08:05 -0400
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAC-LwPOy7zsk-kdA72DiD0i7f=6seWtWNjs1R1jaZTvsABBsow@mail.gmail.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<CAPoPLj8pFu+vijaVzWZMytvMzD+YzBrFneCpLyESK+FebFmVNg@mail.gmail.com>
	<CAC-LwPOy7zsk-kdA72DiD0i7f=6seWtWNjs1R1jaZTvsABBsow@mail.gmail.com>
Message-ID: <D876AEF7-EF2E-4C1F-BC4F-0EE48CDD3AEB@jerris.com>

No.

On Nov 2, 2012, at 4:54 PM, Vik Killa <vipkilla at gmail.com> wrote:

> Can we do the same for MySQL?
> :)
> 



From anthony.minessale at gmail.com  Sat Nov  3 01:48:14 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Nov 2012 17:48:14 -0500
Subject: [Freeswitch-users] sqlite3 and regex
In-Reply-To: <CA+Y7Lf6B0LUjqOpq2M2NFtpCCfhirg7JO4KCo_dGu+i2VP35ng@mail.gmail.com>
References: <CA+Y7Lf6B0LUjqOpq2M2NFtpCCfhirg7JO4KCo_dGu+i2VP35ng@mail.gmail.com>
Message-ID: <CAKbxfG8v-uUp33khup9NPi69U4zmgG-8=-q97ChMN_KxF0WG8A@mail.gmail.com>

No, That is not implemented.
On Nov 2, 2012 5:27 PM, "Scott" <8f27e956 at gmail.com> wrote:

> "LIKE" notwithstanding, sqlite3 does not have a built-in true regex
> function; it does allow for a a c-language hook to one.  Given fs extensive
> use of the regex engine and of sqlite3, we're wondering if the hook is
> already written and rolled.  If so, can the rest of us hook it to our
> sqlite3 uses (e.g. from dial plan lua sqlite3).
>
> With thanks,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Sat Nov  3 02:03:51 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 02 Nov 2012 18:03:51 -0500
Subject: [Freeswitch-users] sqlite3 and regex
In-Reply-To: <CA+Y7Lf6B0LUjqOpq2M2NFtpCCfhirg7JO4KCo_dGu+i2VP35ng@mail.gmail.com>
Message-ID: <CCB9BB87.5EE38%krice@freeswitch.org>

No and this wont happen anytime soon... The SQL interfaces for FreeSwitch
are kept generic as we support more then just sqlite from common code and if
we did it for sqlite we would have to make sure its implemented equally well
for postgresql and mysql and mssql and any other database someone might want
to use via ODBC

K


On 11/2/12 3:34 PM, "Scott" <8f27e956 at gmail.com> wrote:

> "LIKE" notwithstanding, sqlite3 does not have a built-in true regex function;
> it does allow for a a c-language hook to one.? Given fs extensive use of the
> regex engine and of sqlite3, we're wondering if the hook is already written
> and rolled.? If so, can the rest of us hook it to our sqlite3 uses (e.g. from
> dial plan lua sqlite3).
> 
> With thanks,
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From krice at freeswitch.org  Sat Nov  3 02:04:24 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 02 Nov 2012 18:04:24 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <D876AEF7-EF2E-4C1F-BC4F-0EE48CDD3AEB@jerris.com>
Message-ID: <CCB9BBA8.5EE39%krice@freeswitch.org>

What Jerris said....


On 11/2/12 5:08 PM, "Michael Jerris" <mike at jerris.com> wrote:

> No.
> 
> On Nov 2, 2012, at 4:54 PM, Vik Killa <vipkilla at gmail.com> wrote:
> 
>> Can we do the same for MySQL?
>> :)
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From yiftah at choochee.com  Sat Nov  3 02:12:45 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 2 Nov 2012 16:12:45 -0700
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAC-LwPOy7zsk-kdA72DiD0i7f=6seWtWNjs1R1jaZTvsABBsow@mail.gmail.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<CAPoPLj8pFu+vijaVzWZMytvMzD+YzBrFneCpLyESK+FebFmVNg@mail.gmail.com>
	<CAC-LwPOy7zsk-kdA72DiD0i7f=6seWtWNjs1R1jaZTvsABBsow@mail.gmail.com>
Message-ID: <CAFVUaV-SywxGC0eN_Niy9Y9ug2KgGRaykC3OruPHecz=9epQPw@mail.gmail.com>

+1000000 to this

On Fri, Nov 2, 2012 at 1:54 PM, Vik Killa <vipkilla at gmail.com> wrote:

> Can we do the same for MySQL?
> :)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From basit.engg at gmail.com  Sat Nov  3 04:56:51 2012
From: basit.engg at gmail.com (Abdul Basit)
Date: Sat, 3 Nov 2012 06:56:51 +0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <F1691313-1D90-48BF-B051-3548A64AC297@url.net.au>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<F1691313-1D90-48BF-B051-3548A64AC297@url.net.au>
Message-ID: <CAGQG5MZ+Vmop-ksXF8X3nwLkq3xbQoF0OZEaSdPuVKdFGft4vQ@mail.gmail.com>

+1 for mysql support


On Sat, Nov 3, 2012 at 3:06 AM, Ashley Breeden <ash at url.net.au> wrote:

> Hi Ken,
>
> Excellent news for Postgres and FreeSWITCH.
>
>
> I'll ask this as I am sure there are a lot of people wondering, are there
> plans to support MySQL natively as well?
>
>
> A.
>
>
>
> On 03/11/2012, at 4:25 AM, Ken Rice wrote:
>
>  Hey Guys,
>
> There?s some new Database Goodness in the core of FreeSWITCH that can lead
> to some unexpected things for you guys updating existing installations
> using ODBC.
>
> We now have Native PostgreSQL support in the core, and along with this
> comes some changes to the various ?odbc-dsn? settings around the tree.
>
> If you are using the format ?dsn:username:password? you wont be affected,
> however if you are just specifying a DSN as ?dsn? you will need to listen up
>
> The settings for this field have changed.
>
> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> options='-c client_min_messages=NOTICE for postgresql (the stuff after
> pgsql:// is a standard libpq connect string for you programmer types)
> odbc://dns:username:password  for ODBC   ( dsn:: should also work or
> dns:username: )
> sqlite://filename   for sqlite different SQLite Databases
>
> I think we still need to doc this up good, but its there and its coming
> strong...
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,

Abdul Basit | P: +92 32 1416 4196 | O: +92 30 0841 1445 | UK: +447937421194
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From lists at kavun.ch  Sat Nov  3 08:06:23 2012
From: lists at kavun.ch (Emrah)
Date: Sat, 3 Nov 2012 01:06:23 -0400
Subject: [Freeswitch-users] A-leg hangup cause is blank
Message-ID: <51EC5321-C965-4553-9DE4-F84B1FC460DD@kavun.ch>

Hi all,

To be simple, I am trying to play around with hangup causes and see what I get in my environment. 
I must have a pretty shy set up because FS never discloses any hangup cause? It's continuously blank.
This is what I have before my bridge action:
<action application="set" data="api_hangup_hook=system /root/hangup_cause.sh ${hangup_cause} ${hangup_cause_q850}" />

My shell script just stores the hangup_cause I get in a file.

Ideally, I am trying to achieve the following:

1. User a calls user b; user a hangs up before user b picks up; I get a hangup_cause that is relevant and I can notify user b of a missed call.
2. User a calls user b; call goes to Voicemail; user a hangs up without recording a message; I get a hangup_cause that is relevant and I can notify user b of a missed call.

And so on and so forth.

With Asterisk, I would get channel statuses like NOANSWER, CHANUNAVAIL, CANCEL? and could act upon them. 
How do I get my FS to talk to me and give me some hangup_cause for my a-leg?

Thanks a bunch for all your help,
Emrah

From curriegrad2004 at gmail.com  Sat Nov  3 08:55:28 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 2 Nov 2012 22:55:28 -0700
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAGQG5MZ+Vmop-ksXF8X3nwLkq3xbQoF0OZEaSdPuVKdFGft4vQ@mail.gmail.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<F1691313-1D90-48BF-B051-3548A64AC297@url.net.au>
	<CAGQG5MZ+Vmop-ksXF8X3nwLkq3xbQoF0OZEaSdPuVKdFGft4vQ@mail.gmail.com>
Message-ID: <CAMWiMy8bpBkRMe-s2uJEz=KNOPQQMU_1XVeS=6+agzhNdTnUuQ@mail.gmail.com>

Quite impossible. The reason is because of the license that the MySQL
libs are licensed under. iirc, they are GPL licensed and they are not
compatible with the MPL that FreeSWITCH uses.

On Fri, Nov 2, 2012 at 6:56 PM, Abdul Basit <basit.engg at gmail.com> wrote:
> +1 for mysql support
>
>
> On Sat, Nov 3, 2012 at 3:06 AM, Ashley Breeden <ash at url.net.au> wrote:
>>
>> Hi Ken,
>>
>> Excellent news for Postgres and FreeSWITCH.
>>
>>
>> I'll ask this as I am sure there are a lot of people wondering, are there
>> plans to support MySQL natively as well?
>>
>>
>> A.
>>
>>
>>
>> On 03/11/2012, at 4:25 AM, Ken Rice wrote:
>>
>> Hey Guys,
>>
>> There?s some new Database Goodness in the core of FreeSWITCH that can lead
>> to some unexpected things for you guys updating existing installations using
>> ODBC.
>>
>> We now have Native PostgreSQL support in the core, and along with this
>> comes some changes to the various ?odbc-dsn? settings around the tree.
>>
>> If you are using the format ?dsn:username:password? you wont be affected,
>> however if you are just specifying a DSN as ?dsn? you will need to listen up
>>
>> The settings for this field have changed.
>>
>> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
>> options='-c client_min_messages=NOTICE for postgresql (the stuff after
>> pgsql:// is a standard libpq connect string for you programmer types)
>> odbc://dns:username:password  for ODBC   ( dsn:: should also work or
>> dns:username: )
>> sqlite://filename   for sqlite different SQLite Databases
>>
>> I think we still need to doc this up good, but its there and its coming
>> strong...
>>
>> --
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards,
>
> Abdul Basit | P: +92 32 1416 4196 | O: +92 30 0841 1445 | UK: +447937421194
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gabe at gundy.org  Sat Nov  3 09:40:23 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 3 Nov 2012 00:40:23 -0600
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAMWiMy8bpBkRMe-s2uJEz=KNOPQQMU_1XVeS=6+agzhNdTnUuQ@mail.gmail.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<F1691313-1D90-48BF-B051-3548A64AC297@url.net.au>
	<CAGQG5MZ+Vmop-ksXF8X3nwLkq3xbQoF0OZEaSdPuVKdFGft4vQ@mail.gmail.com>
	<CAMWiMy8bpBkRMe-s2uJEz=KNOPQQMU_1XVeS=6+agzhNdTnUuQ@mail.gmail.com>
Message-ID: <CAMwZ-t+9jFFtmPZfyTvFLdTJRGgjJEcAeZnzQ5a7bmBBXEmj4Q@mail.gmail.com>

On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> Quite impossible. The reason is because of the license that the MySQL
> libs are licensed under. iirc, they are GPL licensed and they are not
> compatible with the MPL that FreeSWITCH uses.

I've got PostgreSQL, so I've got all I need :) But, it seems like this
legal issue is easily resolved:

http://www.mysql.com/about/legal/licensing/foss-exception/

Anyway, I'm not a lawyer or even a very thorough reader ;)


Best,
Gabe


From bigx333 at gmail.com  Sat Nov  3 09:42:17 2012
From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado)
Date: Sat, 3 Nov 2012 08:42:17 +0200
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAMWiMy8bpBkRMe-s2uJEz=KNOPQQMU_1XVeS=6+agzhNdTnUuQ@mail.gmail.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<F1691313-1D90-48BF-B051-3548A64AC297@url.net.au>
	<CAGQG5MZ+Vmop-ksXF8X3nwLkq3xbQoF0OZEaSdPuVKdFGft4vQ@mail.gmail.com>
	<CAMWiMy8bpBkRMe-s2uJEz=KNOPQQMU_1XVeS=6+agzhNdTnUuQ@mail.gmail.com>
Message-ID: <CAPt4n69xOrpB515cS3XvzFwmCVVqgjOb9DghXkAv7o+GUkLEgg@mail.gmail.com>

No one in his right mind should be using MySQL anyway :p
On Nov 3, 2012 8:31 AM, "curriegrad2004" <curriegrad2004 at gmail.com> wrote:

> Quite impossible. The reason is because of the license that the MySQL
> libs are licensed under. iirc, they are GPL licensed and they are not
> compatible with the MPL that FreeSWITCH uses.
>
> On Fri, Nov 2, 2012 at 6:56 PM, Abdul Basit <basit.engg at gmail.com> wrote:
> > +1 for mysql support
> >
> >
> > On Sat, Nov 3, 2012 at 3:06 AM, Ashley Breeden <ash at url.net.au> wrote:
> >>
> >> Hi Ken,
> >>
> >> Excellent news for Postgres and FreeSWITCH.
> >>
> >>
> >> I'll ask this as I am sure there are a lot of people wondering, are
> there
> >> plans to support MySQL natively as well?
> >>
> >>
> >> A.
> >>
> >>
> >>
> >> On 03/11/2012, at 4:25 AM, Ken Rice wrote:
> >>
> >> Hey Guys,
> >>
> >> There?s some new Database Goodness in the core of FreeSWITCH that can
> lead
> >> to some unexpected things for you guys updating existing installations
> using
> >> ODBC.
> >>
> >> We now have Native PostgreSQL support in the core, and along with this
> >> comes some changes to the various ?odbc-dsn? settings around the tree.
> >>
> >> If you are using the format ?dsn:username:password? you wont be
> affected,
> >> however if you are just specifying a DSN as ?dsn? you will need to
> listen up
> >>
> >> The settings for this field have changed.
> >>
> >> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> >> options='-c client_min_messages=NOTICE for postgresql (the stuff after
> >> pgsql:// is a standard libpq connect string for you programmer types)
> >> odbc://dns:username:password  for ODBC   ( dsn:: should also work or
> >> dns:username: )
> >> sqlite://filename   for sqlite different SQLite Databases
> >>
> >> I think we still need to doc this up good, but its there and its coming
> >> strong...
> >>
> >> --
> >> Ken
> >> http://www.FreeSWITCH.org
> >> http://www.ClueCon.com
> >> http://www.OSTAG.org
> >> irc.freenode.net #freeswitch
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Regards,
> >
> > Abdul Basit | P: +92 32 1416 4196 | O: +92 30 0841 1445 | UK:
> +447937421194
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anton.jugatsu at gmail.com  Sat Nov  3 14:59:01 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sat, 3 Nov 2012 15:59:01 +0400
Subject: [Freeswitch-users] media proxy
In-Reply-To: <1945566.6VRVmflkY8@sos>
References: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>
	<2818E18E-E28D-4F3A-B535-690CFDBFE23B@jerris.com>
	<1945566.6VRVmflkY8@sos>
Message-ID: <CANBhyPBYrbUuDfcA1Dg=Z9nePZwfcSf043qQdDUBgD=sxPTpNw@mail.gmail.com>

Yes, you can pass-thru g729.

2012/11/2 Sergey Okhapkin <sos at sokhapkin.dyndns.org>

>  Will proxy mode 2 work with codecs not supported by freeswitch?
>
> On Friday 02 November 2012 11:39:27 Michael Jerris wrote:
> > There should be no reason to ever use proxy mode (2).
> >
> > default mode that you call "hard proxy" requires no settings to set it,
> its
> > the default.
> > On Nov 2, 2012, at 10:13 AM, Max Kovalenko <
> mkovalenko at cybervisiontech.com>
> wrote:
> > > Hello,
> > >
> > > There are two modes of media stream to be proxied or bypassed.
> > >
> > > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
> etc.
> > > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > > traversal purposes. No media stream parsing is supported 3. Media
> bypass
> > > - RTP streams are bypassing FreeSWITCH at all.
> > >
> > > There are also two channel parameters affecting above modes:
> bypass_media
> > > and proxy_media.
> > >
> > > - Are these parameters independent? Meaning the combination:
> > > bypass_media=true AND proxy_media=true is possible. What effect will
> be?
> > >
> > > - Does it means that if I want to turn on "light proxy" I would always
> > > need to set bypass_media=false AND proxy_media=true?
> > >
> > > - How to set "hard proxy" (full media proxy) per channel before to
> bridge
> > > legs?
> > >
> > > Waiting for your replay ASAP. Thank you in advance.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Sat Nov  3 16:55:35 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 03 Nov 2012 09:55:35 -0400
Subject: [Freeswitch-users] media proxy
In-Reply-To: <CANBhyPBYrbUuDfcA1Dg=Z9nePZwfcSf043qQdDUBgD=sxPTpNw@mail.gmail.com>
References: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>
	<1945566.6VRVmflkY8@sos>
	<CANBhyPBYrbUuDfcA1Dg=Z9nePZwfcSf043qQdDUBgD=sxPTpNw@mail.gmail.com>
Message-ID: <2003146.NZnh1akVdp@sos>

 So, there there is a reason to use proxy mode (2) sometime.

On Saturday 03 November 2012 15:59:01 Anton Kvashenkin wrote:
> Yes, you can pass-thru g729.
> 
> 2012/11/2 Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> 
> >  Will proxy mode 2 work with codecs not supported by freeswitch?
> > 
> > On Friday 02 November 2012 11:39:27 Michael Jerris wrote:
> > > There should be no reason to ever use proxy mode (2).
> > > 
> > > default mode that you call "hard proxy" requires no settings to set it,
> > 
> > its
> > 
> > > the default.
> > > On Nov 2, 2012, at 10:13 AM, Max Kovalenko <
> > 
> > mkovalenko at cybervisiontech.com>
> > 
> > wrote:
> > > > Hello,
> > > > 
> > > > There are two modes of media stream to be proxied or bypassed.
> > > > 
> > > > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
> > 
> > etc.
> > 
> > > > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > > > traversal purposes. No media stream parsing is supported 3. Media
> > 
> > bypass
> > 
> > > > - RTP streams are bypassing FreeSWITCH at all.
> > 
> > > > There are also two channel parameters affecting above modes:
> > bypass_media
> > 
> > > > and proxy_media.
> > > > 
> > > > - Are these parameters independent? Meaning the combination:
> > > > bypass_media=true AND proxy_media=true is possible. What effect will
> > 
> > be?
> > 
> > > > - Does it means that if I want to turn on "light proxy" I would always
> > > > need to set bypass_media=false AND proxy_media=true?
> > > > 
> > > > - How to set "hard proxy" (full media proxy) per channel before to
> > 
> > bridge
> > 
> > > > legs?
> > > > 
> > > > Waiting for your replay ASAP. Thank you in advance.
> > > 
> > > ________________________________________________________________________
> > > _
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > > 
> > > 
> > > 
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From krice at freeswitch.org  Sat Nov  3 20:32:20 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 03 Nov 2012 12:32:20 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAMwZ-t+9jFFtmPZfyTvFLdTJRGgjJEcAeZnzQ5a7bmBBXEmj4Q@mail.gmail.com>
Message-ID: <CCBABF54.5EF0B%krice@freeswitch.org>

While this may solve the issue... This does not address a number of other
issues

A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...

B) There is that whole threadsafe vs non-threadsafe mysql client lib issue,
so we would have to come up with a way in the build system to detect which
we have, and never use the non-thread safe one... (if we don't do this, the
code would never be stable as we could never know for sure which one
actually got detected and linked against)





On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org> wrote:

> On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
> <curriegrad2004 at gmail.com> wrote:
>> Quite impossible. The reason is because of the license that the MySQL
>> libs are licensed under. iirc, they are GPL licensed and they are not
>> compatible with the MPL that FreeSWITCH uses.
> 
> I've got PostgreSQL, so I've got all I need :) But, it seems like this
> legal issue is easily resolved:
> 
> http://www.mysql.com/about/legal/licensing/foss-exception/
> 
> Anyway, I'm not a lawyer or even a very thorough reader ;)
> 
> 
> Best,
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From anthony.minessale at gmail.com  Sat Nov  3 21:36:29 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Nov 2012 13:36:29 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCBABF54.5EF0B%krice@freeswitch.org>
References: <CAMwZ-t+9jFFtmPZfyTvFLdTJRGgjJEcAeZnzQ5a7bmBBXEmj4Q@mail.gmail.com>
	<CCBABF54.5EF0B%krice@freeswitch.org>
Message-ID: <CAKbxfG-eCoU+-JfcAu=GMTQDJLbifEKb6_QiLYLqMRHoVCBROg@mail.gmail.com>

Plus as a friend of mine used to say: "It's not MY sql its YOUR sql!"

It is a fact that we have an incredible amount of bugs come up that end up
being mysql+odbc memory errors and thread safety issues.
I would be concerned with more blame falling on us of we had the code
deeper into our core.

I would not condone adding any more db support unless we we architected it
to use db modules so we could still properly blame it on mysql when it was
not working.




On Sat, Nov 3, 2012 at 12:32 PM, Ken Rice <krice at freeswitch.org> wrote:

> While this may solve the issue... This does not address a number of other
> issues
>
> A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...
>
> B) There is that whole threadsafe vs non-threadsafe mysql client lib issue,
> so we would have to come up with a way in the build system to detect which
> we have, and never use the non-thread safe one... (if we don't do this, the
> code would never be stable as we could never know for sure which one
> actually got detected and linked against)
>
>
>
>
>
> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>
> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
> > <curriegrad2004 at gmail.com> wrote:
> >> Quite impossible. The reason is because of the license that the MySQL
> >> libs are licensed under. iirc, they are GPL licensed and they are not
> >> compatible with the MPL that FreeSWITCH uses.
> >
> > I've got PostgreSQL, so I've got all I need :) But, it seems like this
> > legal issue is easily resolved:
> >
> > http://www.mysql.com/about/legal/licensing/foss-exception/
> >
> > Anyway, I'm not a lawyer or even a very thorough reader ;)
> >
> >
> > Best,
> > Gabe
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From lazyvirus at gmx.com  Sat Nov  3 22:46:37 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sat, 3 Nov 2012 20:46:37 +0100
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAKbxfG-eCoU+-JfcAu=GMTQDJLbifEKb6_QiLYLqMRHoVCBROg@mail.gmail.com>
References: <CAMwZ-t+9jFFtmPZfyTvFLdTJRGgjJEcAeZnzQ5a7bmBBXEmj4Q@mail.gmail.com>
	<CCBABF54.5EF0B%krice@freeswitch.org>
	<CAKbxfG-eCoU+-JfcAu=GMTQDJLbifEKb6_QiLYLqMRHoVCBROg@mail.gmail.com>
Message-ID: <20121103204637.7916aacc@anubis.defcon1>

On Sat, 3 Nov 2012 13:36:29 -0500
Anthony Minessale <anthony.minessale at gmail.com> wrote:

> 
> I would not condone adding any more db support unless we we architected it
> to use db modules so we could still properly blame it on mysql when it was
> not working.

Support access, it's even easier to blame :)

-- 
Peter : nonsense! look by the window, kids riding bikes outside...
tom : they don't have a computer or what?


From darcy at Vex.Net  Sat Nov  3 23:11:33 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Sat, 3 Nov 2012 16:11:33 -0400
Subject: [Freeswitch-users] Trying to wrap my head around variables.
Message-ID: <20121103161133.19bf89e8@dilbert>

Here is what I have tried so far:

http://pastebin.freeswitch.org/20166

The idea is to replace %BRIDGE_DATA% with ${MyBridgeData}.  When I do
that the call goes straight to voicemail.  The log line is supposed to
show what MyBridgeData is but all it displays is the literal
"${MyBridgeData}".  Can someone point me in the right direction please.

By the way, here is what this convoluted mess is supposed to do.  I
give each of my clients two numbers, a real DID and an internal number
which is +1 999 99x-xxxx where "xxxxx" is the user's client code.  When
a call comes in for them I send the call to their main account.  If
they have registered the internal number then it will simultaneously
ring that number after 5 seconds.  If not and they have registered
their cell number in their control panel then I use that for the
secondary one instead.

Cheers.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From curriegrad2004 at gmail.com  Sat Nov  3 23:25:34 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 3 Nov 2012 13:25:34 -0700
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAKbxfG-eCoU+-JfcAu=GMTQDJLbifEKb6_QiLYLqMRHoVCBROg@mail.gmail.com>
References: <CAMwZ-t+9jFFtmPZfyTvFLdTJRGgjJEcAeZnzQ5a7bmBBXEmj4Q@mail.gmail.com>
	<CCBABF54.5EF0B%krice@freeswitch.org>
	<CAKbxfG-eCoU+-JfcAu=GMTQDJLbifEKb6_QiLYLqMRHoVCBROg@mail.gmail.com>
Message-ID: <CAMWiMy95x7GAiisCG2f3M0pBQ=EmVjyf39D=sB5sxXdgKOOK2g@mail.gmail.com>

ODBC's there for this reason. Why invent more wheels when we already
have that wheel sitting around?

On Sat, Nov 3, 2012 at 11:36 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Plus as a friend of mine used to say: "It's not MY sql its YOUR sql!"
>
> It is a fact that we have an incredible amount of bugs come up that end up
> being mysql+odbc memory errors and thread safety issues.
> I would be concerned with more blame falling on us of we had the code deeper
> into our core.
>
> I would not condone adding any more db support unless we we architected it
> to use db modules so we could still properly blame it on mysql when it was
> not working.
>
>
>
>
> On Sat, Nov 3, 2012 at 12:32 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> While this may solve the issue... This does not address a number of other
>> issues
>>
>> A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...
>>
>> B) There is that whole threadsafe vs non-threadsafe mysql client lib
>> issue,
>> so we would have to come up with a way in the build system to detect which
>> we have, and never use the non-thread safe one... (if we don't do this,
>> the
>> code would never be stable as we could never know for sure which one
>> actually got detected and linked against)
>>
>>
>>
>>
>>
>> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>>
>> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
>> > <curriegrad2004 at gmail.com> wrote:
>> >> Quite impossible. The reason is because of the license that the MySQL
>> >> libs are licensed under. iirc, they are GPL licensed and they are not
>> >> compatible with the MPL that FreeSWITCH uses.
>> >
>> > I've got PostgreSQL, so I've got all I need :) But, it seems like this
>> > legal issue is easily resolved:
>> >
>> > http://www.mysql.com/about/legal/licensing/foss-exception/
>> >
>> > Anyway, I'm not a lawyer or even a very thorough reader ;)
>> >
>> >
>> > Best,
>> > Gabe
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ssinyagin at yahoo.com  Sun Nov  4 00:32:46 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sat, 3 Nov 2012 14:32:46 -0700 (PDT)
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <1351978366.55811.YahooMailNeo@web39302.mail.mud.yahoo.com>

Problem is resolved:
http://jira.freeswitch.org/browse/FS-4791

Wiki documented a wrong syntax (data="xx" instead of value="xx"), and I 
updated the wiki pages where I could find it.





----- Original Message -----
> From: Stanislav Sinyagin <ssinyagin at yahoo.com>
> To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Thursday, November 1, 2012 10:00 PM
> Subject: [Freeswitch-users] Sofia gateway variables
> 
> I've got the following SIP gateway which registers successfully at the ITSP:
> 
> ? <gateway name="sipcall_41325133196">
> ??? <param name="username" value="41325133196"/>
> ??? <param name="password" value="XXXXXXXXXX"/>
> ??? <param name="extension" value="0325133196"/>
> ??? <param name="expire-seconds" value="125"/>
> ??? <param name="register" value="true"/>
> ??? <param name="register-transport" value="udp"/>
> ??? <param name="proxy" 
> value="free2.voipgateway.org"/>
> ??? <param name="retry-seconds" value="30"/>
> ??? <param name="caller-id-in-from" value="false"/>
> ??? <param name="ping" value="27"/>
> ??? <variables>
> ???? <variable name="sbc_callee_ccode" data="41" 
> direction="inbound"/>
> ???? <variable name="sbc_caller_ccode" data="41" 
> direction="inbound"/>
> ??? </variables>
> ? </gateway>
> 
> 
> Then the call from PSTN goes into the proper context, and the first condition in 
> the context is:
> 
> ?? <extension name="call_debug" continue="true">
> ???? <condition field="${call_debug}" expression="^true$" 
> break="never">
> ?????? <action application="info"/>
> ?????? <action application="log" data="INFO Country code: 
> ${sbc_callee_ccode}"/>
> ???? </condition>
> ?? </extension>
> 
> 
> Neither the info, nor the log show my two variables on inbound calls. 
> 
> 
> Am I doing something wrong or is it a bug? 
> 
> I'm using git commit 94664868a8 in v1.2.stable branch
> 
> 
> 
> thanks,
> stanislav
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From avi at avimarcus.net  Sun Nov  4 00:45:40 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 3 Nov 2012 23:45:40 +0200
Subject: [Freeswitch-users] Incoming Calls, use From instead of PID?
Message-ID: <CALNZuw=kEWvmhAYdJ_j8_QyKXmpYu=UO3S41HStFCKhu8ye8og@mail.gmail.com>

Hi - I'm getting calls from a carrier with the proper e.164 FROM but their
PID is coming in inconsistently. I was told to ignore the PID.

Apparently, FS is using the PID for the profile Caller-Caller-ID-Number
variable.

Are there any toggles for how to manage *inbound *caller ID?
This is currently coming in via ACL to public context... can I even set the
sip_cid for that?

Suggestions?

Thanks,

-Avi Marcus
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From sms at icefire.qza.net.au  Sat Nov  3 13:46:07 2012
From: sms at icefire.qza.net.au (sms at icefire.qza.net.au)
Date: Sat, 3 Nov 2012 21:46:07 +1100
Subject: [Freeswitch-users] Remote pickup of on-hold extensions
In-Reply-To: <CAMbW9oKH0j_oRuMjaDOGb6A=eBMf79XZMezSBeggWTdmYJDrDg@mail.gmail.com>
References: <bc4d4e4f822530c463361a4f72774f6a.squirrel@icefire.qza.net.au>
	<CAMbW9oKH0j_oRuMjaDOGb6A=eBMf79XZMezSBeggWTdmYJDrDg@mail.gmail.com>
Message-ID: <e62dfe57ead648fc2e67de37fc38726f.squirrel@icefire.qza.net.au>

I've thought about those options... the obstacle is mostly the users, who
aren't the most trainable people in the world, and they're used to the
present key system which allows them to have pretty colored buttons that
they only press once. I dread the attempt to explain the concept of
transferring a call, and the question will inevitably come back: "Why
can't we have line keys like the last system?" Calls will likely get lost
or dropped, and of course, the system gets blamed ;-)

The other option, again if possible, is to program a soft key as a
one-touch "Park" key that announces the park extension number, which could
also be assigned to a BLF, then parks the call. Then we have one touch
park, one touch retrieve.

I have to sell this to them in such a way that it doesn't come across as a
feature downgrade, or we may end up stuck with the present system until it
dies and no inter-branch connectivity.

I suppose it's the same old predicament... things would be so simple if
management just said yes :)



> Personally I'd just use the Blind Transfer features on your handsets. 1000
> picks up the call, presses transfer on the phone, then dials 1001 and
> hangs
> up. 1001 rings and is connected to the transferred call. Sure you'e not
> got
> the BLF, but the idea works fine. You could use call parking, but that has
> more steps to it.
>
> On 2 November 2012 04:16, <sms at icefire.qza.net.au> wrote:
>
>> Hi all,
>>
>> I have a scenario that I'm thinking of implementing, but not sure if
>> it's
>> possible, so here's the details:
>>
>> Extensions 200 to 250 are regular internal sip phones, contained in a
>> call
>> group (Ext 1000)
>>
>> Extensions 300 to 305 are SIP-GSM and SIP-PSTN gateways with SIP client
>> firmware, namely those cheap GOIP GSM units on ebay, plus an SPA3000.
>> These are to be configured with the logon credentials of the extension
>> and
>> behave as a typical SIP client.
>>
>> The dialplan will route incoming and outgoing calls via these gateways
>> as
>> if they were trunks. They will restricted privileges to prevent toll
>> fraud. The phones will either be Yealink T28p or Grandstream GXP2124, it
>> depends if GS can respond with a fix for their horrible AGC or not. The
>> gateways will be subscribed to the phone's DSS keys as BLF's.
>>
>> Now, all this so far is reasonably straight forward. The next part is
>> the
>> tricky part (you might've guessed where this is going....)
>>
>> Let's say a call comes in on 300, routes to the call group and is picked
>> up by 1000, who then puts the call on hold. The BLF key for 300 shows it
>> as busy/on hold. 1000 then calls 1001 to take the call. 1001 then
>> presses
>> BLF 300, which causes the call to be transferred to them and
>> automatically
>> answered.
>>
>> So is there any way in freeswitch to replicate this behavior? I know
>> it's
>> possible to do SAA, but this gets tricky if I want to have time based
>> call
>> groups, or to share incoming lines between branches. The method above
>> allows the lines to be present on all phones, while still configuring
>> call
>> groups in the usual way and more or less replicating old key system
>> behavior.
>>
>> Cheers,
>> Francis
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960
> *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From anthony.minessale at gmail.com  Sun Nov  4 01:05:51 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Nov 2012 16:05:51 -0600
Subject: [Freeswitch-users] Trying to wrap my head around variables.
In-Reply-To: <20121103161133.19bf89e8@dilbert>
References: <20121103161133.19bf89e8@dilbert>
Message-ID: <CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>

Have you read the wiki article on the xml dialplan or the chapter on ot in
our book?

The thing that tends to be the most difficult to grok is that when the
module is parsing the xml, its not doing the things right as it sees them,
rather it collects the matching stuff and distills it down to a linear set
of instructions.   So if you set variables in one action then expect it to
exist, you will be disappointed because it has not actually executed the
set command, it only adds it to the list of work.   However, there is a way
to get what you want by adding inline="true" to the action when you set
variables so they actually are set immediately and won't count in the list
of work.  This only works for set and a small list of apps.......
On Nov 3, 2012 4:50 PM, "D'Arcy J.M. Cain" <darcy at vex.net> wrote:

> Here is what I have tried so far:
>
> http://pastebin.freeswitch.org/20166
>
> The idea is to replace %BRIDGE_DATA% with ${MyBridgeData}.  When I do
> that the call goes straight to voicemail.  The log line is supposed to
> show what MyBridgeData is but all it displays is the literal
> "${MyBridgeData}".  Can someone point me in the right direction please.
>
> By the way, here is what this convoluted mess is supposed to do.  I
> give each of my clients two numbers, a real DID and an internal number
> which is +1 999 99x-xxxx where "xxxxx" is the user's client code.  When
> a call comes in for them I send the call to their main account.  If
> they have registered the internal number then it will simultaneously
> ring that number after 5 seconds.  If not and they have registered
> their cell number in their control panel then I use that for the
> secondary one instead.
>
> Cheers.
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.Net
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Sun Nov  4 01:07:11 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 4 Nov 2012 00:07:11 +0200
Subject: [Freeswitch-users] Sofia gateway variables
In-Reply-To: <1351978366.55811.YahooMailNeo@web39302.mail.mud.yahoo.com>
References: <1351803642.57390.YahooMailNeo@web39302.mail.mud.yahoo.com>
	<1351978366.55811.YahooMailNeo@web39302.mail.mud.yahoo.com>
Message-ID: <CALNZuwnidSEijB67mbnvL28O3oC5HTe73KzNcwbh3EJUw-S66g@mail.gmail.com>

Eek, looks like I doc'd that. I can't find where I had copied it from, I
must have simply copied it from the dialplan, which uses "data" instead of
"value".

Sorry for the trouble and thanks for tracking it down!
-Avi

On Sat, Nov 3, 2012 at 11:32 PM, Stanislav Sinyagin <ssinyagin at yahoo.com>wrote:

> Problem is resolved:
> http://jira.freeswitch.org/browse/FS-4791
>
> Wiki documented a wrong syntax (data="xx" instead of value="xx"), and I
> updated the wiki pages where I could find it.
>
>
>
>
>
> ----- Original Message -----
> > From: Stanislav Sinyagin <ssinyagin at yahoo.com>
> > To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Cc:
> > Sent: Thursday, November 1, 2012 10:00 PM
> > Subject: [Freeswitch-users] Sofia gateway variables
> >
> > I've got the following SIP gateway which registers successfully at the
> ITSP:
> >
> >   <gateway name="sipcall_41325133196">
> >     <param name="username" value="41325133196"/>
> >     <param name="password" value="XXXXXXXXXX"/>
> >     <param name="extension" value="0325133196"/>
> >     <param name="expire-seconds" value="125"/>
> >     <param name="register" value="true"/>
> >     <param name="register-transport" value="udp"/>
> >     <param name="proxy"
> > value="free2.voipgateway.org"/>
> >     <param name="retry-seconds" value="30"/>
> >     <param name="caller-id-in-from" value="false"/>
> >     <param name="ping" value="27"/>
> >     <variables>
> >      <variable name="sbc_callee_ccode" data="41"
> > direction="inbound"/>
> >      <variable name="sbc_caller_ccode" data="41"
> > direction="inbound"/>
> >     </variables>
> >   </gateway>
> >
> >
> > Then the call from PSTN goes into the proper context, and the first
> condition in
> > the context is:
> >
> >    <extension name="call_debug" continue="true">
> >      <condition field="${call_debug}" expression="^true$"
> > break="never">
> >        <action application="info"/>
> >        <action application="log" data="INFO Country code:
> > ${sbc_callee_ccode}"/>
> >      </condition>
> >    </extension>
> >
> >
> > Neither the info, nor the log show my two variables on inbound calls.
> >
> >
> > Am I doing something wrong or is it a bug?
> >
> > I'm using git commit 94664868a8 in v1.2.stable branch
> >
> >
> >
> > thanks,
> > stanislav
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ksims.ml at gmail.com  Sun Nov  4 04:42:58 2012
From: ksims.ml at gmail.com (KPS Maillinglist)
Date: Sat, 3 Nov 2012 20:42:58 -0500
Subject: [Freeswitch-users] ODBC with mod_voicemail and storage-dir
Message-ID: <CAOjvyPkhnDPAfKTV8wP+Dt_AiaLird5FsuH4UYLXsT+qrp33XA@mail.gmail.com>

I cant seem to get mod_voicemail to store vm information in the database.
It also is not using the storage-dir param to place the vm files. Please
help.


<profile name="DEMO1_CDI_TEST1">
<settings>
<param name="callback-context" value="DEMO1_CDI_TEST1"/>
<param name="callback-dialplan" value="XML"/>
<param name="choose-greeting-key" value="2"/>
<param name="config-menu-key" value="5"/>
<param name="delete-file-key" value="7"/>
<param name="digit-timeout" value="10000"/>
<param name="email-key" value="4"/>
<param name="ff-key" value="6"/>
<param name="file-extension" value="mp3"/>
<param name="listen-file-key" value="1"/>
<param name="main-menu-key" value="*"/>
<param name="max-login-attempts" value="3"/>
<param name="max-record-length" value="300"/>
<param name="odbc-dsn" value="fs_regsrv_vm:fs_cs:staysynced"/>
<param name="operator-extension" value="operator XML DEMO1_CDI_TEST1"/>
<param name="operator-key" value="9"/>
<param name="pause-key" value="0"/>
<param name="play-new-messages-key" value="1"/>
<param name="play-saved-messages-key" value="2"/>
<param name="record-file-key" value="3"/>
<param name="record-greeting-key" value="1"/>
<param name="record-name-key" value="3"/>
<param name="record-silence-hits" value="2"/>
<param name="record-silence-threshold" value="200"/>
<param name="restart-key" value="1"/>
<param name="rew-key" value="4"/>
<param name="save-file-key" value="9"/>
<param name="storage-dir" value="/fs/cust/vm/DEMO1_CDI_TEST1"/>
<param name="terminator-key" value="#"/>
<param name="tone-spec" value="%(1000, 0, 640)"/>
<param name="undelete-file-key" value="8"/>
<param name="web-template-file" value="web-vm.tpl"/>
<email>
<param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
<param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
<param name="notify-template-file" value="notify-voicemail.tpl"/>
<param name="template-file" value="voicemail.tpl"/>
</email>
</settings>
</profile>
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From gabe at gundy.org  Sun Nov  4 05:58:10 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 3 Nov 2012 20:58:10 -0600
Subject: [Freeswitch-users] http://wiki.freeswitch.org/wiki/Mod_xml_curl
 updated and big thanks to Cal
In-Reply-To: <CCB6B8C4.5ECE2%krice@freeswitch.org>
References: <CCB6B8C4.5ECE2%krice@freeswitch.org>
Message-ID: <CAMwZ-tJPvQjP3AWM6snQ87FvCQ8WZ-1tERRWxMwKvX9MC8_1CA@mail.gmail.com>

On Wed, Oct 31, 2012 at 10:15 AM, Ken Rice <krice at freeswitch.org> wrote:
> Cal had a problem with the documentation as many of us do, he got on the
> mailing list and figured his issues out, then took the time to totally
> re-write the mod_xml_curl page on the wiki.

Solid work, Cal. Thanks!

Gabe


From darcy at Vex.Net  Sun Nov  4 05:58:43 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Sat, 3 Nov 2012 22:58:43 -0400
Subject: [Freeswitch-users] Trying to wrap my head around variables.
In-Reply-To: <CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>
References: <20121103161133.19bf89e8@dilbert>
	<CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>
Message-ID: <20121103225843.7a69aa2c@dilbert>

On Sat, 3 Nov 2012 16:05:51 -0600
Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Have you read the wiki article on the xml dialplan or the chapter on
> ot in our book?

I have pored over the wiki quite a bit.  In fact, some of that code
comes from the examples there.  I suppose I could have misread it.  Is
there something in particular in my dialplan that I am doing wrong?

> The thing that tends to be the most difficult to grok is that when the
> module is parsing the xml, its not doing the things right as it sees
> them, rather it collects the matching stuff and distills it down to a
> linear set of instructions.   So if you set variables in one action
> then expect it to exist, you will be disappointed because it has not
> actually executed the set command, it only adds it to the list of

That's exactly what I hope, that it executes the set actions at run
time.  Specifically, the sofia_contact(99999%UID%@${domain_name}) has
to happen during the call setup since the state of registration can
change all the time.

> work.   However, there is a way to get what you want by adding
> inline="true" to the action when you set variables so they actually
> are set immediately and won't count in the list of work.  This only
> works for set and a small list of apps.

Definitely not what I want in this case.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From yiftah at choochee.com  Sun Nov  4 06:59:48 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sat, 3 Nov 2012 20:59:48 -0700
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAMWiMy95x7GAiisCG2f3M0pBQ=EmVjyf39D=sB5sxXdgKOOK2g@mail.gmail.com>
References: <CAMwZ-t+9jFFtmPZfyTvFLdTJRGgjJEcAeZnzQ5a7bmBBXEmj4Q@mail.gmail.com>
	<CCBABF54.5EF0B%krice@freeswitch.org>
	<CAKbxfG-eCoU+-JfcAu=GMTQDJLbifEKb6_QiLYLqMRHoVCBROg@mail.gmail.com>
	<CAMWiMy95x7GAiisCG2f3M0pBQ=EmVjyf39D=sB5sxXdgKOOK2g@mail.gmail.com>
Message-ID: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>

It is just about how fast is ODBC compare to native mysql
http://www.devtoolshed.com/content/performance-benchmarks-odbc-vs-oracle-mysql-sql-server-net-providers
Also for what ken wrote we used the non thread safe version and protected
the call to the database with mutex


On Sat, Nov 3, 2012 at 1:25 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> ODBC's there for this reason. Why invent more wheels when we already
> have that wheel sitting around?
>
> On Sat, Nov 3, 2012 at 11:36 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Plus as a friend of mine used to say: "It's not MY sql its YOUR sql!"
> >
> > It is a fact that we have an incredible amount of bugs come up that end
> up
> > being mysql+odbc memory errors and thread safety issues.
> > I would be concerned with more blame falling on us of we had the code
> deeper
> > into our core.
> >
> > I would not condone adding any more db support unless we we architected
> it
> > to use db modules so we could still properly blame it on mysql when it
> was
> > not working.
> >
> >
> >
> >
> > On Sat, Nov 3, 2012 at 12:32 PM, Ken Rice <krice at freeswitch.org> wrote:
> >>
> >> While this may solve the issue... This does not address a number of
> other
> >> issues
> >>
> >> A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...
> >>
> >> B) There is that whole threadsafe vs non-threadsafe mysql client lib
> >> issue,
> >> so we would have to come up with a way in the build system to detect
> which
> >> we have, and never use the non-thread safe one... (if we don't do this,
> >> the
> >> code would never be stable as we could never know for sure which one
> >> actually got detected and linked against)
> >>
> >>
> >>
> >>
> >>
> >> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
> >>
> >> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
> >> > <curriegrad2004 at gmail.com> wrote:
> >> >> Quite impossible. The reason is because of the license that the MySQL
> >> >> libs are licensed under. iirc, they are GPL licensed and they are not
> >> >> compatible with the MPL that FreeSWITCH uses.
> >> >
> >> > I've got PostgreSQL, so I've got all I need :) But, it seems like this
> >> > legal issue is easily resolved:
> >> >
> >> > http://www.mysql.com/about/legal/licensing/foss-exception/
> >> >
> >> > Anyway, I'm not a lawyer or even a very thorough reader ;)
> >> >
> >> >
> >> > Best,
> >> > Gabe
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >> --
> >> Ken
> >> http://www.FreeSWITCH.org
> >> http://www.ClueCon.com
> >> http://www.OSTAG.org
> >> irc.freenode.net #freeswitch
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darcy at Vex.Net  Sun Nov  4 07:20:02 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Sun, 4 Nov 2012 00:20:02 -0400
Subject: [Freeswitch-users] Trying to wrap my head around variables.
In-Reply-To: <20121103225843.7a69aa2c@dilbert>
References: <20121103161133.19bf89e8@dilbert>
	<CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>
	<20121103225843.7a69aa2c@dilbert>
Message-ID: <20121104002002.1bf512c4@dilbert>

On Sat, 3 Nov 2012 22:58:43 -0400
"D'Arcy J.M. Cain" <darcy at Vex.Net> wrote:
> > work.   However, there is a way to get what you want by adding
> > inline="true" to the action when you set variables so they actually
> > are set immediately and won't count in the list of work.  This only
> > works for set and a small list of apps.
> 
> Definitely not what I want in this case.

I think I misunderstood.  I read "immediately" as being at reloadxml
time.  I see that it means during the hunting phase.  I modified my
dialplan but it still isn't quite right.  See the updated pastebin:

http://pastebin.freeswitch.org/20168

See the log extract at the end.  It shows cellPhone being set to
something but when it tests it it is blank.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From avi at avimarcus.net  Sun Nov  4 09:15:57 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 4 Nov 2012 08:15:57 +0200
Subject: [Freeswitch-users] ODBC with mod_voicemail and storage-dir
In-Reply-To: <CAOjvyPkhnDPAfKTV8wP+Dt_AiaLird5FsuH4UYLXsT+qrp33XA@mail.gmail.com>
References: <CAOjvyPkhnDPAfKTV8wP+Dt_AiaLird5FsuH4UYLXsT+qrp33XA@mail.gmail.com>
Message-ID: <CALNZuwk+yAGbt7ebMa3yGnpXM6kFsQDfpQo0SMxVkeeWpnZAXA@mail.gmail.com>

Did you reload mod_voicemail since saving these changes? reloadxml usually
isn't enough to reload for module-configuration.
-Avi

On Sun, Nov 4, 2012 at 3:42 AM, KPS Maillinglist <ksims.ml at gmail.com> wrote:

> I cant seem to get mod_voicemail to store vm information in the database.
> It also is not using the storage-dir param to place the vm files. Please
> help.
>
>
> <profile name="DEMO1_CDI_TEST1">
> <settings>
> <param name="callback-context" value="DEMO1_CDI_TEST1"/>
> <param name="callback-dialplan" value="XML"/>
> <param name="choose-greeting-key" value="2"/>
> <param name="config-menu-key" value="5"/>
> <param name="delete-file-key" value="7"/>
> <param name="digit-timeout" value="10000"/>
> <param name="email-key" value="4"/>
> <param name="ff-key" value="6"/>
> <param name="file-extension" value="mp3"/>
> <param name="listen-file-key" value="1"/>
> <param name="main-menu-key" value="*"/>
> <param name="max-login-attempts" value="3"/>
> <param name="max-record-length" value="300"/>
> <param name="odbc-dsn" value="fs_regsrv_vm:fs_cs:staysynced"/>
> <param name="operator-extension" value="operator XML DEMO1_CDI_TEST1"/>
> <param name="operator-key" value="9"/>
> <param name="pause-key" value="0"/>
> <param name="play-new-messages-key" value="1"/>
> <param name="play-saved-messages-key" value="2"/>
> <param name="record-file-key" value="3"/>
> <param name="record-greeting-key" value="1"/>
> <param name="record-name-key" value="3"/>
> <param name="record-silence-hits" value="2"/>
> <param name="record-silence-threshold" value="200"/>
> <param name="restart-key" value="1"/>
> <param name="rew-key" value="4"/>
> <param name="save-file-key" value="9"/>
> <param name="storage-dir" value="/fs/cust/vm/DEMO1_CDI_TEST1"/>
> <param name="terminator-key" value="#"/>
> <param name="tone-spec" value="%(1000, 0, 640)"/>
> <param name="undelete-file-key" value="8"/>
> <param name="web-template-file" value="web-vm.tpl"/>
> <email>
> <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
> <param name="email-from" value="${voicemail_account}@${voicemail_domain}"
> />
> <param name="notify-template-file" value="notify-voicemail.tpl"/>
> <param name="template-file" value="voicemail.tpl"/>
> </email>
> </settings>
> </profile>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david at styleflare.com  Sun Nov  4 14:32:27 2012
From: david at styleflare.com (David | StyleFlare)
Date: Sun, 04 Nov 2012 06:32:27 -0500
Subject: [Freeswitch-users] PGSQL - Support in Core. - Error During configure
Message-ID: <5096524B.6010604@styleflare.com>

I am trying to build latest freeswitch from master.

I have the --enable-postgres... flag = true when I run config

I get this error message.

checking whether to include odbc... no
checking for pg_config... /usr/local/pgsql/bin/pg_config
checking for PostgreSQL libraries... checking for PQgetvalue in -lpq... no
configure: error: no usable libpq; please install PostgreSQL devel 
package or equivalent


IN MY "config.log"

I see

## ----------- ##
## confdefs.h. ##
## ----------- ##

#define PACKAGE_NAME "freeswitch"
#define PACKAGE_TARNAME "freeswitch"
#define PACKAGE_VERSION "1.3.0"
#define PACKAGE_STRING "freeswitch 1.3.0"
#define PACKAGE_BUGREPORT "BUG-REPORT-ADDRESS"
#define PACKAGE "libfreeswitch"
#define VERSION "0.1"
#define SWITCH_MOD_DIR "/usr/local/freeswitch/mod"
#define SWITCH_RUN_DIR "/usr/local/freeswitch/run"
#define SWITCH_LOG_DIR "/usr/local/freeswitch/log"
#define SWITCH_DB_DIR "/usr/local/freeswitch/db"
#define SWITCH_HTDOCS_DIR "/usr/local/freeswitch/htdocs"
#define SWITCH_SOUNDS_DIR "/usr/local/freeswitch/sounds"
#define SWITCH_GRAMMAR_DIR "/usr/local/freeswitch/grammar"
#define SWITCH_SCRIPT_DIR "/usr/local/freeswitch/scripts"
#define SWITCH_RECORDINGS_DIR "/usr/local/freeswitch/recordings"
#define SWITCH_CONF_DIR "/usr/local/freeswitch/conf"
#define STDC_HEADERS 1
#define HAVE_SYS_TYPES_H 1
#define HAVE_SYS_STAT_H 1
#define HAVE_STDLIB_H 1
#define HAVE_STRING_H 1
#define HAVE_MEMORY_H 1
#define HAVE_STRINGS_H 1
#define HAVE_INTTYPES_H 1
#define HAVE_STDINT_H 1
#define HAVE_UNISTD_H 1
#define HAVE_DLFCN_H 1
#define LT_OBJDIR ".libs/"
#define DEBUG /**/
#define SIZEOF_LONG 8
#define SWITCH_HAVE_PGSQL 1
#define POSTGRESQL_VERSION "9.2.1"
#define POSTGRESQL_MAJOR_VERSION 9
#define POSTGRESQL_MINOR_VERSION 2
#define POSTGRESQL_PATCH_VERSION 1

configure: exit 1

Here is the output of my pg_config

BINDIR = /usr/local/pgsql/bin
DOCDIR = /usr/local/pgsql/share/doc
HTMLDIR = /usr/local/pgsql/share/doc
INCLUDEDIR = /usr/local/pgsql/include
PKGINCLUDEDIR = /usr/local/pgsql/include
INCLUDEDIR-SERVER = /usr/local/pgsql/include/server
LIBDIR = /usr/local/pgsql/lib
PKGLIBDIR = /usr/local/pgsql/lib
LOCALEDIR = /usr/local/pgsql/share/locale
MANDIR = /usr/local/pgsql/share/man
SHAREDIR = /usr/local/pgsql/share
SYSCONFDIR = /usr/local/pgsql/etc
PGXS = /usr/local/pgsql/lib/pgxs/src/makefiles/pgxs.mk
CONFIGURE =
CC = gcc
CPPFLAGS = -D_GNU_SOURCE
CFLAGS = -O2 -Wall -Wmissing-prototypes -Wpointer-arith 
-Wdeclaration-after-statement -Wendif-labels -Wmissing-format-attribute 
-Wformat-security -fno-strict-aliasing -fwrapv
CFLAGS_SL = -fpic
LDFLAGS = -Wl,--as-needed 
-Wl,-rpath,'/usr/local/pgsql/lib',--enable-new-dtags
LDFLAGS_EX =
LDFLAGS_SL =
LIBS = -lpgport -lz -lreadline -lcrypt -ldl -lm
VERSION = PostgreSQL 9.2.1


Any ideas where I am going wrong?

Thanks.




From darcy at Vex.Net  Sun Nov  4 15:18:20 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Sun, 4 Nov 2012 07:18:20 -0500
Subject: [Freeswitch-users] PGSQL - Support in Core. - Error During
 configure
In-Reply-To: <5096524B.6010604@styleflare.com>
References: <5096524B.6010604@styleflare.com>
Message-ID: <20121104071820.4b7ee6eb@dilbert>

On Sun, 04 Nov 2012 06:32:27 -0500
David | StyleFlare <david at styleflare.com> wrote:
> I get this error message.
> 
> checking whether to include odbc... no
> checking for pg_config... /usr/local/pgsql/bin/pg_config
> checking for PostgreSQL libraries... checking for PQgetvalue in
> -lpq... no configure: error: no usable libpq; please install
> PostgreSQL devel package or equivalent
> ...
> Any ideas where I am going wrong?

You don't say whether PostgreSQL is installed or not.  Is it?  This
message implies that it is not.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From anton.jugatsu at gmail.com  Sun Nov  4 17:07:57 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sun, 4 Nov 2012 18:07:57 +0400
Subject: [Freeswitch-users] PGSQL - Support in Core. - Error During
	configure
In-Reply-To: <20121104071820.4b7ee6eb@dilbert>
References: <5096524B.6010604@styleflare.com> <20121104071820.4b7ee6eb@dilbert>
Message-ID: <CANBhyPA_52az5yjMusUfERLK_Lad+q7S56OWP5xtTC_CFhCVqA@mail.gmail.com>

aptitude search ^postgres~ndev
i   postgresql-server-dev-9.1
    - development files for PostgreSQL 9.1 server-side programming
p   postgresql-server-dev-all
    - extension build tool for multiple PostgreSQL versions

You should install -dev files.

2012/11/4 D'Arcy J.M. Cain <darcy at vex.net>

> On Sun, 04 Nov 2012 06:32:27 -0500
> David | StyleFlare <david at styleflare.com> wrote:
> > I get this error message.
> >
> > checking whether to include odbc... no
> > checking for pg_config... /usr/local/pgsql/bin/pg_config
> > checking for PostgreSQL libraries... checking for PQgetvalue in
> > -lpq... no configure: error: no usable libpq; please install
> > PostgreSQL devel package or equivalent
> > ...
> > Any ideas where I am going wrong?
>
> You don't say whether PostgreSQL is installed or not.  Is it?  This
> message implies that it is not.
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.Net
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ksims.ml at gmail.com  Sun Nov  4 18:26:25 2012
From: ksims.ml at gmail.com (KPS Maillinglist)
Date: Sun, 4 Nov 2012 09:26:25 -0600
Subject: [Freeswitch-users] ODBC with mod_voicemail and storage-dir
In-Reply-To: <CALNZuwk+yAGbt7ebMa3yGnpXM6kFsQDfpQo0SMxVkeeWpnZAXA@mail.gmail.com>
References: <CAOjvyPkhnDPAfKTV8wP+Dt_AiaLird5FsuH4UYLXsT+qrp33XA@mail.gmail.com>
	<CALNZuwk+yAGbt7ebMa3yGnpXM6kFsQDfpQo0SMxVkeeWpnZAXA@mail.gmail.com>
Message-ID: <CAOjvyPk1vvA0Tjx4cz=j7NocoWzBU1JsVWZW7pR+jWJCh4MP-w@mail.gmail.com>

Yes I have reloaded the voicemail module as well as did a full shutdown and
startup of FS .



>
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From brian.wiese.freeswitch at gmail.com  Sun Nov  4 18:33:26 2012
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Sun, 4 Nov 2012 09:33:26 -0600
Subject: [Freeswitch-users] Trying to wrap my head around variables.
In-Reply-To: <20121104002002.1bf512c4@dilbert>
References: <20121103161133.19bf89e8@dilbert>
	<CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>
	<20121103225843.7a69aa2c@dilbert> <20121104002002.1bf512c4@dilbert>
Message-ID: <CADEQS5wtho+WLWGFRGrs=nHxnugB2eP5_xQW13SMtGYwvQHxkg@mail.gmail.com>

What's happening is that you're trying to use cellPhone as a built-in
variable when instead it is a channel variable that you've set (read
the whole Variables section of this Wiki page:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Variables).

Try "<condition field="${cellPhone}" expression......."

~Brian

On Sat, Nov 3, 2012 at 11:20 PM, D'Arcy J.M. Cain <darcy at vex.net> wrote:
> On Sat, 3 Nov 2012 22:58:43 -0400
> "D'Arcy J.M. Cain" <darcy at Vex.Net> wrote:
>> > work.   However, there is a way to get what you want by adding
>> > inline="true" to the action when you set variables so they actually
>> > are set immediately and won't count in the list of work.  This only
>> > works for set and a small list of apps.
>>
>> Definitely not what I want in this case.
>
> I think I misunderstood.  I read "immediately" as being at reloadxml
> time.  I see that it means during the hunting phase.  I modified my
> dialplan but it still isn't quite right.  See the updated pastebin:
>
> http://pastebin.freeswitch.org/20168
>
> See the log extract at the end.  It shows cellPhone being set to
> something but when it tests it it is blank.
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.Net
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From krice at freeswitch.org  Sun Nov  4 18:07:45 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 04 Nov 2012 10:07:45 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
Message-ID: <CCBBEEF1.5EF42%krice@freeswitch.org>

Oh yeah, lets let 1 thread run at a time access the database... Mysql is the
only one that shows this issue... So why should we cripple the DB interfaces
for just mysql



On 11/3/12 10:59 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:

> Also for what ken wrote we used the non thread safe version and protected the
> call to the database with mutex
> ?
> 
> On Sat, Nov 3, 2012 at 1:25 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>> ODBC's there for this reason. Why invent more wheels when we already
>> have that wheel sitting around?
>> 
>> On Sat, Nov 3, 2012 at 11:36 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> > Plus as a friend of mine used to say: "It's not MY sql its YOUR sql!"
>>> >
>>> > It is a fact that we have an incredible amount of bugs come up that end up
>>> > being mysql+odbc memory errors and thread safety issues.
>>> > I would be concerned with more blame falling on us of we had the code
>>> deeper
>>> > into our core.
>>> >
>>> > I would not condone adding any more db support unless we we architected it
>>> > to use db modules so we could still properly blame it on mysql when it was
>>> > not working.
>>> >
>>> >
>>> >
>>> >
>>> > On Sat, Nov 3, 2012 at 12:32 PM, Ken Rice <krice at freeswitch.org> wrote:
>>>> >>
>>>> >> While this may solve the issue... This does not address a number of
>>>> other
>>>> >> issues
>>>> >>
>>>> >> A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...
>>>> >>
>>>> >> B) There is that whole threadsafe vs non-threadsafe mysql client lib
>>>> >> issue,
>>>> >> so we would have to come up with a way in the build system to detect
>>>> which
>>>> >> we have, and never use the non-thread safe one... (if we don't do this,
>>>> >> the
>>>> >> code would never be stable as we could never know for sure which one
>>>> >> actually got detected and linked against)
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>>>> >>
>>>>> >> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
>>>>> >> > <curriegrad2004 at gmail.com> wrote:
>>>>>> >> >> Quite impossible. The reason is because of the license that the
MySQL
>>>>>> >> >> libs are licensed under. iirc, they are GPL licensed and they are
not
>>>>>> >> >> compatible with the MPL that FreeSWITCH uses.
>>>>> >> >
>>>>> >> > I've got PostgreSQL, so I've got all I need :) But, it seems like
this
>>>>> >> > legal issue is easily resolved:
>>>>> >> >
>>>>> >> > http://www.mysql.com/about/legal/licensing/foss-exception/
>>>>> >> >
>>>>> >> > Anyway, I'm not a lawyer or even a very thorough reader ;)
>>>>> >> >
>>>>> >> >
>>>>> >> > Best,
>>>>> >> > Gabe
>>>>> >> >
>>>>> >> >
>>>>> >> > 
>>>>> _________________________________________________________________________
>>>>> >> > Professional FreeSWITCH Consulting Services:
>>>>> >> > consulting at freeswitch.org
>>>>> >> > http://www.freeswitchsolutions.com
>>>>> >> >
>>>>> >> > 
>>>>> >> > 
>>>>> >> >
>>>>> >> > Official FreeSWITCH Sites
>>>>> >> > http://www.freeswitch.org
>>>>> >> > http://wiki.freeswitch.org
>>>>> >> > http://www.cluecon.com
>>>>> >> >
>>>>> >> > FreeSWITCH-users mailing list
>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> > 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> > http://www.freeswitch.org
>>>> >>
>>>> >> --
>>>> >> Ken
>>>> >> http://www.FreeSWITCH.org
>>>> >> http://www.ClueCon.com
>>>> >> http://www.OSTAG.org
>>>> >> irc.freenode.net <http://irc.freenode.net>  #freeswitch
>>>> >>
>>>> >>
>>>> >>
>>>> >> 
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> 
>>>> >> 
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://wiki.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>> > IRC: irc.freenode.net <http://irc.freenode.net>  #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> <mailto:sip%3A888 at conference.freeswitch.org>
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> > pstn:+19193869900 <tel:%2B19193869900>
>>> >
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From krice at freeswitch.org  Sun Nov  4 18:16:30 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 04 Nov 2012 10:16:30 -0500
Subject: [Freeswitch-users] PGSQL - Support in Core. - Error During
 configure
In-Reply-To: <CANBhyPA_52az5yjMusUfERLK_Lad+q7S56OWP5xtTC_CFhCVqA@mail.gmail.com>
Message-ID: <CCBBF0FE.5EF44%krice@freeswitch.org>

That?s the server side stuff...

Do you have libpq-dev installed?


On 11/4/12 9:07 AM, "Anton Kvashenkin" <anton.jugatsu at gmail.com> wrote:

> 
> aptitude search ^postgres~ndev
> i ? postgresql-server-dev-9.1 ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
> ? - development files for PostgreSQL 9.1 server-side programming
> p ? postgresql-server-dev-all ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
> ? - extension build tool for multiple PostgreSQL versions
> 
> You should install -dev files.
> 
> 2012/11/4 D'Arcy J.M. Cain <darcy at vex.net>
>> On Sun, 04 Nov 2012 06:32:27 -0500
>> David | StyleFlare <david at styleflare.com> wrote:
>>> > I get this error message.
>>> >
>>> > checking whether to include odbc... no
>>> > checking for pg_config... /usr/local/pgsql/bin/pg_config
>>> > checking for PostgreSQL libraries... checking for PQgetvalue in
>>> > -lpq... no configure: error: no usable libpq; please install
>>> > PostgreSQL devel package or equivalent
>>> > ...
>>> > Any ideas where I am going wrong?
>> 
>> You don't say whether PostgreSQL is installed or not. ?Is it? ?This
>> message implies that it is not.
>> 
>> --
>> D'Arcy J.M. Cain
>> System Administrator, Vex.Net
>> http://www.Vex.Net/ IM:darcy at Vex.Net
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From darcy at Vex.Net  Sun Nov  4 19:31:07 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Sun, 4 Nov 2012 11:31:07 -0500
Subject: [Freeswitch-users] Trying to wrap my head around variables.
In-Reply-To: <CADEQS5wtho+WLWGFRGrs=nHxnugB2eP5_xQW13SMtGYwvQHxkg@mail.gmail.com>
References: <20121103161133.19bf89e8@dilbert>
	<CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>
	<20121103225843.7a69aa2c@dilbert> <20121104002002.1bf512c4@dilbert>
	<CADEQS5wtho+WLWGFRGrs=nHxnugB2eP5_xQW13SMtGYwvQHxkg@mail.gmail.com>
Message-ID: <20121104113107.421f76b6@dilbert>

On Sun, 4 Nov 2012 09:33:26 -0600
Brian Wiese <brian.wiese.freeswitch at gmail.com> wrote:
> What's happening is that you're trying to use cellPhone as a built-in
> variable when instead it is a channel variable that you've set (read
> the whole Variables section of this Wiki page:
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Variables).
> 
> Try "<condition field="${cellPhone}" expression......."

The light goes on.  Thanks.  I have added a specific example of this in
the wiki just to make it a little clearer.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From darcy at Vex.Net  Sun Nov  4 19:34:33 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Sun, 4 Nov 2012 11:34:33 -0500
Subject: [Freeswitch-users] Trying to wrap my head around variables.
In-Reply-To: <CADEQS5wtho+WLWGFRGrs=nHxnugB2eP5_xQW13SMtGYwvQHxkg@mail.gmail.com>
References: <20121103161133.19bf89e8@dilbert>
	<CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>
	<20121103225843.7a69aa2c@dilbert> <20121104002002.1bf512c4@dilbert>
	<CADEQS5wtho+WLWGFRGrs=nHxnugB2eP5_xQW13SMtGYwvQHxkg@mail.gmail.com>
Message-ID: <20121104113433.4a796d90@dilbert>

On Sun, 4 Nov 2012 09:33:26 -0600
Brian Wiese <brian.wiese.freeswitch at gmail.com> wrote:
> What's happening is that you're trying to use cellPhone as a built-in

By the way, I have managed to do what I want using a Python module.
It's much cleaner and more flexible anyway.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From babak.freeswitch at gmail.com  Sun Nov  4 20:59:59 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Sun, 4 Nov 2012 21:29:59 +0330
Subject: [Freeswitch-users] duplicate fax pages
Message-ID: <CAEVfcaYyvO-tz145EY=QmYFpOCO4U4p-1UFy8nNGT6m2bh2-Tg@mail.gmail.com>

Hi
I'm using mod_spandsp to send fax to different targets. when destination is
another freeswitch or elastix everything works fine. but when sending faxes
to old fax machines if tiff file contains only 1 page, sending will
succeed. but if tiff contains more than 1 page FS keeps sending first page
multiple times till I issue hupall and it stops sending(I should mention
that sometimes it sends first page 3 times second page 3 times and keeps
eating  remote fax machine paper roll).
I'm using this command:
originate {fax_ident=$faxidnt,fax_header=$faxhdr,origination_caller_id_number=$callerid,tx_owner_id=$userid}sofia/gateway/fax_outbound/999
&txfax('path')
in php to send faxes. I've set global t38_passthru to true and in
spandsp.conf:
<fax-settings>
<param name="use-ecm" value="true"/>
<param name="verbose" value="true"/>
<param name="disable-v17 " value="false"/>
<param name="ident" value="test"/>
<param name="header" value="test"/>

<param name="spool-dir" value="/tmp"/>
<param name="file-prefix" value="faxrx"/>
    </fax-settings>
and the logs for spandsp verbose mode:
http://pastebin.freeswitch.org/20169
and the version:
FreeSWITCH Version 1.3.0+git~20120908T211235Z~36cee285b0 (1.3.0; git at
commit 36cee285b0 on Sat, 08 Sep 2012 21:12:35 Z)
thanx
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From anthony.minessale at gmail.com  Sun Nov  4 21:11:38 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 4 Nov 2012 12:11:38 -0600
Subject: [Freeswitch-users] Trying to wrap my head around variables.
In-Reply-To: <CADEQS5wtho+WLWGFRGrs=nHxnugB2eP5_xQW13SMtGYwvQHxkg@mail.gmail.com>
References: <20121103161133.19bf89e8@dilbert>
	<CAKbxfG_iBhXSjYALwYOuU0dk+PC5D_cZaw-eom070_oUZxCu6w@mail.gmail.com>
	<20121103225843.7a69aa2c@dilbert> <20121104002002.1bf512c4@dilbert>
	<CADEQS5wtho+WLWGFRGrs=nHxnugB2eP5_xQW13SMtGYwvQHxkg@mail.gmail.com>
Message-ID: <CAKbxfG_t_GGQ5Y3bfSb-Gtf86+Hn8CHQ-1T6-+TKFmkOo9f5mQ@mail.gmail.com>

Make one extension to set vars then call transfer to another one that does
things based on the vars.



On Sun, Nov 4, 2012 at 9:33 AM, Brian Wiese <
brian.wiese.freeswitch at gmail.com> wrote:

> What's happening is that you're trying to use cellPhone as a built-in
> variable when instead it is a channel variable that you've set (read
> the whole Variables section of this Wiki page:
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Variables).
>
> Try "<condition field="${cellPhone}" expression......."
>
> ~Brian
>
> On Sat, Nov 3, 2012 at 11:20 PM, D'Arcy J.M. Cain <darcy at vex.net> wrote:
> > On Sat, 3 Nov 2012 22:58:43 -0400
> > "D'Arcy J.M. Cain" <darcy at Vex.Net> wrote:
> >> > work.   However, there is a way to get what you want by adding
> >> > inline="true" to the action when you set variables so they actually
> >> > are set immediately and won't count in the list of work.  This only
> >> > works for set and a small list of apps.
> >>
> >> Definitely not what I want in this case.
> >
> > I think I misunderstood.  I read "immediately" as being at reloadxml
> > time.  I see that it means during the hunting phase.  I modified my
> > dialplan but it still isn't quite right.  See the updated pastebin:
> >
> > http://pastebin.freeswitch.org/20168
> >
> > See the log extract at the end.  It shows cellPhone being set to
> > something but when it tests it it is blank.
> >
> > --
> > D'Arcy J.M. Cain
> > System Administrator, Vex.Net
> > http://www.Vex.Net/ IM:darcy at Vex.Net
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From mkovalenko at cybervisiontech.com  Sun Nov  4 22:57:27 2012
From: mkovalenko at cybervisiontech.com (Max Kovalenko)
Date: Sun, 4 Nov 2012 14:57:27 -0500 (EST)
Subject: [Freeswitch-users] media proxy
In-Reply-To: <30168183.121352058620772.JavaMail.root@inter.cybervisiontech.com>
Message-ID: <31733028.141352059047097.JavaMail.root@inter.cybervisiontech.com>

Unfortunately, it isn't well documented, or else I would not ask about  :))))))

Those parameters perhaps are exclusive of each other, by mechanics they affects to is almost the same. Or at least, I would call them two sides of ame thing.

Theoretically, if you consider the situation of bypass_media=true, that means media flow bypass around Freeswitch, it also automatically annihilates settings of proxy_media variable. It doesn't matter, even you would set proxy_media=true for same channel, it should not provide you "light proxy" of media flow, because your settign of bypass_proxy comprises Freeswitch don't proxy media.
Thus I guess in order to set "light" or "hard" proxy the variable bypass_media must be set to FALSE first, for same channel of course.

My task requires either bypass media or light/hard proxy to be set for a channel depending of some conditions of a session. In the other words, I need to set up one of three modes per each channel.

Perhaps I undestand something not so right - I will appreciate if you will correct me with the following:

1. Default (full media proxy) : bypass_media=false ; proxy_media=false

2. Transparent proxy (mine "light proxy"): bypass_media=false; proxy_media=true

3. Bypass media: bupass_media=true  (it's doesn't matter what currect value of proxy_media is)

Thanks

Max Kovalenko 
Team Leader 
VoIP & UC Team 
Managed Services Dept. 
CyberVision Inc. 
------------------------------------------------- 
tel. +1 (201) 585-9809 ext. 215 
Email: mkovalenko at cybervisiontech.com
Skype: mkovalenko_cv, panzer_meister
WWW: www.cybervisiontech.com



----- ???????? ????????? -----
??: "Ken Rice" <krice at freeswitch.org>
????: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????, ????????, ???????
????: Re: [Freeswitch-users] media proxy

Default, proxy and bybass media are exclusive of each other, you can not
combine them on a single call...

Proxy media is just that proxy the media

Bypass media is just that, bypass freeswitch and send the media direct
between the end point...

Setting this options is fairly well documented on the wiki...



On 11/2/12 9:13 AM, "Max Kovalenko" <mkovalenko at cybervisiontech.com> wrote:

> Hello,
> 
> There are two modes of media stream to be proxied or bypassed.
> 
> 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc.
> 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> traversal purposes. No media stream parsing is supported
> 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> 
> There are also two channel parameters affecting above modes: bypass_media and
> proxy_media.
> 
> - Are these parameters independent? Meaning the combination: bypass_media=true
> AND proxy_media=true is possible. What effect will be?
> 
> - Does it means that if I want to turn on "light proxy" I would always need to
> set bypass_media=false AND proxy_media=true?
> 
> - How to set "hard proxy" (full media proxy) per channel before to bridge
> legs?
> 
> Waiting for your replay ASAP. Thank you in advance.
> 
> Best Regards.
> 
> Max Kovalenko 
> Team Leader 
> VoIP & UC Team 
> Managed Services Dept.
> CyberVision Inc. 
> -------------------------------------------------
> tel. +1 (201) 585-9809 ext. 215
> Email: mkovalenko at cybervisiontech.com
> Skype: mkovalenko_cv, panzer_meister
> WWW: www.cybervisiontech.com
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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



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




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

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


From ben at langfeld.co.uk  Sun Nov  4 23:46:00 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sun, 4 Nov 2012 12:46:00 -0800
Subject: [Freeswitch-users] media proxy
In-Reply-To: <31733028.141352059047097.JavaMail.root@inter.cybervisiontech.com>
References: <30168183.121352058620772.JavaMail.root@inter.cybervisiontech.com>
	<31733028.141352059047097.JavaMail.root@inter.cybervisiontech.com>
Message-ID: <CAAyX+KGSjJZ8ttMQ9WRFORUBzOoo=DXo2RiYsKKaqOJX4Qs-9g@mail.gmail.com>

Can you point to other software that provides this distinction in media
proxying?

Regards,
Ben Langfeld


On 4 November 2012 11:57, Max Kovalenko <mkovalenko at cybervisiontech.com>wrote:

> Unfortunately, it isn't well documented, or else I would not ask about
>  :))))))
>
> Those parameters perhaps are exclusive of each other, by mechanics they
> affects to is almost the same. Or at least, I would call them two sides of
> ame thing.
>
> Theoretically, if you consider the situation of bypass_media=true, that
> means media flow bypass around Freeswitch, it also automatically
> annihilates settings of proxy_media variable. It doesn't matter, even you
> would set proxy_media=true for same channel, it should not provide you
> "light proxy" of media flow, because your settign of bypass_proxy comprises
> Freeswitch don't proxy media.
> Thus I guess in order to set "light" or "hard" proxy the variable
> bypass_media must be set to FALSE first, for same channel of course.
>
> My task requires either bypass media or light/hard proxy to be set for a
> channel depending of some conditions of a session. In the other words, I
> need to set up one of three modes per each channel.
>
> Perhaps I undestand something not so right - I will appreciate if you will
> correct me with the following:
>
> 1. Default (full media proxy) : bypass_media=false ; proxy_media=false
>
> 2. Transparent proxy (mine "light proxy"): bypass_media=false;
> proxy_media=true
>
> 3. Bypass media: bupass_media=true  (it's doesn't matter what currect
> value of proxy_media is)
>
> Thanks
>
> Max Kovalenko
> Team Leader
> VoIP & UC Team
> Managed Services Dept.
> CyberVision Inc.
> -------------------------------------------------
> tel. +1 (201) 585-9809 ext. 215
> Email: mkovalenko at cybervisiontech.com
> Skype: mkovalenko_cv, panzer_meister
> WWW: www.cybervisiontech.com
>
>
>
> ----- ???????? ????????? -----
> ??: "Ken Rice" <krice at freeswitch.org>
> ????: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> ????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????,
> ????????, ???????
> ????: Re: [Freeswitch-users] media proxy
>
> Default, proxy and bybass media are exclusive of each other, you can not
> combine them on a single call...
>
> Proxy media is just that proxy the media
>
> Bypass media is just that, bypass freeswitch and send the media direct
> between the end point...
>
> Setting this options is fairly well documented on the wiki...
>
>
>
> On 11/2/12 9:13 AM, "Max Kovalenko" <mkovalenko at cybervisiontech.com>
> wrote:
>
> > Hello,
> >
> > There are two modes of media stream to be proxied or bypassed.
> >
> > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
> etc.
> > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > traversal purposes. No media stream parsing is supported
> > 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> >
> > There are also two channel parameters affecting above modes:
> bypass_media and
> > proxy_media.
> >
> > - Are these parameters independent? Meaning the combination:
> bypass_media=true
> > AND proxy_media=true is possible. What effect will be?
> >
> > - Does it means that if I want to turn on "light proxy" I would always
> need to
> > set bypass_media=false AND proxy_media=true?
> >
> > - How to set "hard proxy" (full media proxy) per channel before to bridge
> > legs?
> >
> > Waiting for your replay ASAP. Thank you in advance.
> >
> > Best Regards.
> >
> > Max Kovalenko
> > Team Leader
> > VoIP & UC Team
> > Managed Services Dept.
> > CyberVision Inc.
> > -------------------------------------------------
> > tel. +1 (201) 585-9809 ext. 215
> > Email: mkovalenko at cybervisiontech.com
> > Skype: mkovalenko_cv, panzer_meister
> > WWW: www.cybervisiontech.com
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yiftah at choochee.com  Mon Nov  5 05:37:35 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sun, 4 Nov 2012 18:37:35 -0800
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCBBEEF1.5EF42%krice@freeswitch.org>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
Message-ID: <CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>

I do not know Postgres (and maybe it is a good chance to try it) but
probably there is a penalty with Postgres as well
It is just that OpenSIPs goes very well with FreeSWITCH (at least for us)
and OpenSIPs use mysql as its primary db
Therefore using one DB for both makes a lot of sense for us
BTW, I did not get an answer for posting the Germany number that we added
for the conference, do you want it or should I return it to our poll?

Thanks,
Yiftach.


On Sun, Nov 4, 2012 at 7:07 AM, Ken Rice <krice at freeswitch.org> wrote:

>  Oh yeah, lets let 1 thread run at a time access the database... Mysql is
> the only one that shows this issue... So why should we cripple the DB
> interfaces for just mysql
>
>
>
> On 11/3/12 10:59 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>
> Also for what ken wrote we used the non thread safe version and protected
> the call to the database with mutex
>
>
> On Sat, Nov 3, 2012 at 1:25 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>
> ODBC's there for this reason. Why invent more wheels when we already
> have that wheel sitting around?
>
> On Sat, Nov 3, 2012 at 11:36 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Plus as a friend of mine used to say: "It's not MY sql its YOUR sql!"
> >
> > It is a fact that we have an incredible amount of bugs come up that end
> up
> > being mysql+odbc memory errors and thread safety issues.
> > I would be concerned with more blame falling on us of we had the code
> deeper
> > into our core.
> >
> > I would not condone adding any more db support unless we we architected
> it
> > to use db modules so we could still properly blame it on mysql when it
> was
> > not working.
> >
> >
> >
> >
> > On Sat, Nov 3, 2012 at 12:32 PM, Ken Rice <krice at freeswitch.org> wrote:
> >>
> >> While this may solve the issue... This does not address a number of
> other
> >> issues
> >>
> >> A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...
> >>
> >> B) There is that whole threadsafe vs non-threadsafe mysql client lib
> >> issue,
> >> so we would have to come up with a way in the build system to detect
> which
> >> we have, and never use the non-thread safe one... (if we don't do this,
> >> the
> >> code would never be stable as we could never know for sure which one
> >> actually got detected and linked against)
> >>
> >>
> >>
> >>
> >>
> >> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
> >>
> >> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
> >> > <curriegrad2004 at gmail.com> wrote:
> >> >> Quite impossible. The reason is because of the license that the MySQL
> >> >> libs are licensed under. iirc, they are GPL licensed and they are not
> >> >> compatible with the MPL that FreeSWITCH uses.
> >> >
> >> > I've got PostgreSQL, so I've got all I need :) But, it seems like this
> >> > legal issue is easily resolved:
> >> >
> >> > http://www.mysql.com/about/legal/licensing/foss-exception/
> >> >
> >> > Anyway, I'm not a lawyer or even a very thorough reader ;)
> >> >
> >> >
> >> > Best,
> >> > Gabe
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >> --
> >> Ken
> >> http://www.FreeSWITCH.org
> >> http://www.ClueCon.com
> >> http://www.OSTAG.org
> >> irc.freenode.net <http://irc.freenode.net>  #freeswitch
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > 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<MSN%3Aanthony_minessale at hotmail.com>>
>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <
> mailto:PAYPAL%3Aanthony.minessale at gmail.com<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<sip%3A888 at conference.freeswitch.org>>
>
> > googletalk:conf+888 at conference.freeswitch.org <
> mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>>
>
> > pstn:+19193869900 <tel:%2B19193869900>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yiftah at choochee.com  Mon Nov  5 05:38:46 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sun, 4 Nov 2012 18:38:46 -0800
Subject: [Freeswitch-users] http://wiki.freeswitch.org/wiki/Mod_xml_curl
 updated and big thanks to Cal
In-Reply-To: <CAMwZ-tJPvQjP3AWM6snQ87FvCQ8WZ-1tERRWxMwKvX9MC8_1CA@mail.gmail.com>
References: <CCB6B8C4.5ECE2%krice@freeswitch.org>
	<CAMwZ-tJPvQjP3AWM6snQ87FvCQ8WZ-1tERRWxMwKvX9MC8_1CA@mail.gmail.com>
Message-ID: <CAFVUaV-48AF+-ZD2GA9n43e8TB1ZyotOVJfYpEjCHyCVe6sSsA@mail.gmail.com>

+1 to that also helped me


On Sat, Nov 3, 2012 at 7:58 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Wed, Oct 31, 2012 at 10:15 AM, Ken Rice <krice at freeswitch.org> wrote:
> > Cal had a problem with the documentation as many of us do, he got on the
> > mailing list and figured his issues out, then took the time to totally
> > re-write the mod_xml_curl page on the wiki.
>
> Solid work, Cal. Thanks!
>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Mon Nov  5 05:21:12 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 04 Nov 2012 21:21:12 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
Message-ID: <CCBC8CC8.5EF65%krice@freeswitch.org>

You are more then welcome to keep the number pointed to the Conference
server... Don?t expect a lot of action on things like that during the
weekend... Most of us that work for the project do so full time and do take
time for the family...

As far as OpenSIPs, they support PostgreSQL via the db_postgres engine...



On 11/4/12 9:37 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:

> I do not know Postgres (and maybe it is a good chance to try it) but probably
> there is a penalty with Postgres as well
> It is just that OpenSIPs goes very well with FreeSWITCH (at least for us) and
> OpenSIPs use mysql as its primary db
> Therefore using one DB for both makes a lot of sense for us
> BTW, I did not get an answer for posting the Germany number that we added for
> the conference, do you want it or should I return it to our poll?
> 
> Thanks,
> Yiftach.
> 
> 
> On Sun, Nov 4, 2012 at 7:07 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Oh yeah, lets let 1 thread run at a time access the database... Mysql is the
>> only one that shows this issue... So why should we cripple the DB interfaces
>> for just mysql
>> 
>> 
>> 
>> On 11/3/12 10:59 PM, "Yiftach Golan" <yiftah at choochee.com
>> <http://yiftah at choochee.com> > wrote:
>> 
>>> Also for what ken wrote we used the non thread safe version and protected
>>> the call to the database with mutex
>>> ?
>>> 
>>> On Sat, Nov 3, 2012 at 1:25 PM, curriegrad2004 <curriegrad2004 at gmail.com
>>> <http://curriegrad2004 at gmail.com> > wrote:
>>>> ODBC's there for this reason. Why invent more wheels when we already
>>>> have that wheel sitting around?
>>>> 
>>>> On Sat, Nov 3, 2012 at 11:36 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com <http://anthony.minessale at gmail.com> > wrote:
>>>>> > Plus as a friend of mine used to say: "It's not MY sql its YOUR sql!"
>>>>> >
>>>>> > It is a fact that we have an incredible amount of bugs come up that end
>>>>> up
>>>>> > being mysql+odbc memory errors and thread safety issues.
>>>>> > I would be concerned with more blame falling on us of we had the code
>>>>> deeper
>>>>> > into our core.
>>>>> >
>>>>> > I would not condone adding any more db support unless we we architected
>>>>> it
>>>>> > to use db modules so we could still properly blame it on mysql when it
>>>>> was
>>>>> > not working.
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Sat, Nov 3, 2012 at 12:32 PM, Ken Rice <krice at freeswitch.org
>>>>> <http://krice at freeswitch.org> > wrote:
>>>>>> >>
>>>>>> >> While this may solve the issue... This does not address a number of
>>>>>> other
>>>>>> >> issues
>>>>>> >>
>>>>>> >> A) 90+% of the primary FreeSWITCH developers use PostgreSQL
>>>>>> primarily...
>>>>>> >>
>>>>>> >> B) There is that whole threadsafe vs non-threadsafe mysql client lib
>>>>>> >> issue,
>>>>>> >> so we would have to come up with a way in the build system to detect
>>>>>> which
>>>>>> >> we have, and never use the non-thread safe one... (if we don't do
>>>>>> this,
>>>>>> >> the
>>>>>> >> code would never be stable as we could never know for sure which one
>>>>>> >> actually got detected and linked against)
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org
>>>>>> <http://gabe at gundy.org> > wrote:
>>>>>> >>
>>>>>>> >> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
>>>>>>> >> > <curriegrad2004 at gmail.com <http://curriegrad2004 at gmail.com> >
>>>>>>> wrote:
>>>>>>>> >> >> Quite impossible. The reason is because of the license that the
MySQL
>>>>>>>> >> >> libs are licensed under. iirc, they are GPL licensed and they are
not
>>>>>>>> >> >> compatible with the MPL that FreeSWITCH uses.
>>>>>>> >> >
>>>>>>> >> > I've got PostgreSQL, so I've got all I need :) But, it seems like
this
>>>>>>> >> > legal issue is easily resolved:
>>>>>>> >> >
>>>>>>> >> > http://www.mysql.com/about/legal/licensing/foss-exception/
>>>>>>> >> >
>>>>>>> >> > Anyway, I'm not a lawyer or even a very thorough reader ;)
>>>>>>> >> >
>>>>>>> >> >
>>>>>>> >> > Best,
>>>>>>> >> > Gabe
>>>>>>> >> >
>>>>>>> >> >
>>>>>>> >> > 
>>>>>>> 
_________________________________________________________________________
>>>>>>> >> > Professional FreeSWITCH Consulting Services:
>>>>>>> >> > consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>>>>> >> > http://www.freeswitchsolutions.com
>>>>>>> >> >
>>>>>>> >> > 
>>>>>>> >> > 
>>>>>>> >> >
>>>>>>> >> > Official FreeSWITCH Sites
>>>>>>> >> > http://www.freeswitch.org
>>>>>>> >> > http://wiki.freeswitch.org
>>>>>>> >> > http://www.cluecon.com
>>>>>>> >> >
>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> > 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >> > http://www.freeswitch.org
>>>>>> >>
>>>>>> >> --
>>>>>> >> Ken
>>>>>> >> http://www.FreeSWITCH.org
>>>>>> >> http://www.ClueCon.com
>>>>>> >> http://www.OSTAG.org
>>>>>> >> irc.freenode.net <http://irc.freenode.net>  <http://irc.freenode.net>
>>>>>> ?#freeswitch
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >> 
>>>>>> _________________________________________________________________________
>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>> >> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>>>> >> http://www.freeswitchsolutions.com
>>>>>> >>
>>>>>> >> 
>>>>>> >> 
>>>>>> >>
>>>>>> >> Official FreeSWITCH Sites
>>>>>> >> http://www.freeswitch.org
>>>>>> >> http://wiki.freeswitch.org
>>>>>> >> http://www.cluecon.com
>>>>>> >>
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> <http://FreeSWITCH-users at 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 <http://anthony_minessale at hotmail.com>
>>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> <http://anthony.minessale at gmail.com>
>>>>> <mailto:PAYPAL%3Aanthony.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 <http://888 at conference.freeswitch.org>
>>>>> <mailto:sip%3A888 at conference.freeswitch.org>
>>>>> > googletalk:conf+888 at conference.freeswitch.org
>>>>> <http://conf+888 at conference.freeswitch.org>
>>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> > pstn:+19193869900 <tel:%2B19193869900>  <tel:%2B19193869900>
>>>>> >
>>>>> > 
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > 
>>>>> > 
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://wiki.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org

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

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From vbvbrj at gmail.com  Mon Nov  5 09:28:11 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 05 Nov 2012 08:28:11 +0200
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
Message-ID: <50975C7B.3090807@gmail.com>

I was wondering why developers prefers PostgreSQL other the MySQL? 
Despite the "thread" issue are the PostgreSQL more preferable in 
everyday working? I want to integrate a DB for FS, web, and other 
projects. I already use MySQL, but more and more I see that PostgreSQL 
is more preferable.

I've seen PostgreSQL is used by ArcGIS Server. I didn't like the DB 
Management interface. In MySQL it is more intuitive.

Reading the internet I didn't find a comprehensive conclusion on this 
PostgreSQL vs MySQL. Also, does PostgreSQL work in fault-tolerance, 
replication?

Thanks.

-- 
Mimiko desu.


From gabe at gundy.org  Mon Nov  5 09:39:30 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sun, 4 Nov 2012 23:39:30 -0700
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
	<CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
Message-ID: <CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>

On Sun, Nov 4, 2012 at 7:37 PM, Yiftach Golan <yiftah at choochee.com> wrote:
> OpenSIPs goes very well with FreeSWITCH (at least for us) and OpenSIPs use
> mysql as its primary db

OpenSIPS works great with PostgreSQL.


Gabe


From itamar at ispbrasil.com.br  Mon Nov  5 10:08:49 2012
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Mon, 5 Nov 2012 05:08:49 -0200
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <50975C7B.3090807@gmail.com>
References: <50975C7B.3090807@gmail.com>
Message-ID: <CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>

On Mon, Nov 5, 2012 at 4:28 AM, Vbvbrj <vbvbrj at gmail.com> wrote:

> I was wondering why developers prefers PostgreSQL other the MySQL?
> Despite the "thread" issue are the PostgreSQL more preferable in
> everyday working? I want to integrate a DB for FS, web, and other
> projects. I already use MySQL, but more and more I see that PostgreSQL
> is more preferable.
>
> I've seen PostgreSQL is used by ArcGIS Server. I didn't like the DB
> Management interface. In MySQL it is more intuitive.
>
> Reading the internet I didn't find a comprehensive conclusion on this
> PostgreSQL vs MySQL. Also, does PostgreSQL work in fault-tolerance,
> replication?
>
> Thanks.
>
> --
> Mimiko desu.
>
>
>
postgresql is the true free and opensource datase

mysql is owned by oracle, his future is not clear.


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

Itamar Reis Peixoto
http://www.quebarato.com.br/perfil/itamarjp
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From b2m at a-cti.com  Mon Nov  5 11:37:29 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Mon, 5 Nov 2012 14:07:29 +0530
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
Message-ID: <CAP2fbbedYYU9k67wUDwq5BZKExZOg6+TqjN7MEX8R+69VGzrAA@mail.gmail.com>

+100 for PostgreSQL.

Thanks,
Bala

On Mon, Nov 5, 2012 at 12:38 PM, Itamar Reis Peixoto <
itamar at ispbrasil.com.br> wrote:

>
>
>
> On Mon, Nov 5, 2012 at 4:28 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
>
>> I was wondering why developers prefers PostgreSQL other the MySQL?
>> Despite the "thread" issue are the PostgreSQL more preferable in
>> everyday working? I want to integrate a DB for FS, web, and other
>> projects. I already use MySQL, but more and more I see that PostgreSQL
>> is more preferable.
>>
>> I've seen PostgreSQL is used by ArcGIS Server. I didn't like the DB
>> Management interface. In MySQL it is more intuitive.
>>
>> Reading the internet I didn't find a comprehensive conclusion on this
>> PostgreSQL vs MySQL. Also, does PostgreSQL work in fault-tolerance,
>> replication?
>>
>> Thanks.
>>
>> --
>> Mimiko desu.
>>
>>
>>
> postgresql is the true free and opensource datase
>
> mysql is owned by oracle, his future is not clear.
>
>
> --
> ------------
>
> Itamar Reis Peixoto
> http://www.quebarato.com.br/perfil/itamarjp
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eagle.antonio at gmail.com  Mon Nov  5 12:14:59 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 5 Nov 2012 09:14:59 +0000
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
Message-ID: <CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>

The issue is not that simple , i consider PostgreSQL more "enterprise" is
just my point of view.
One thing i love about postgres is the Transactions  + an extremely good
engine , PostgreSQL is not made as easy as mysql you don't get fancy
"phpmyadmin" but it does the work extremely well even in some corner cases.

You can find many pages like this one
http://www.wikivs.com/wiki/MySQL_vs_PostgreSQL , anyway you can always test
but from my experience , running FS clusters for three years PostgreSQL
never let me down even when the RAID card failed , the internal triggers
"locked down" the database , no rows were lost , the transaction engine
rolled back current transactions and it saved my day :).

The lesson is , test first deploy later , PostegreSQL is for me , more
feature reach and more stable but it requires more skills , mysql is more
easy ofc less features and less stable ( this isn't true for all the cases).

Its up to you use something that you feel comfortable so when problems rise
you can actually solve problems this means you understand the database and
how it works.

P.S one point positive for mysql OPENSIPS tend to use mysql way more :)

There is also a more "oracle" version of PostgreSQL ,
http://www.enterprisedb.com/

Regards
Antonio



2012/11/5 Itamar Reis Peixoto <itamar at ispbrasil.com.br>

>
>
>
> On Mon, Nov 5, 2012 at 4:28 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
>
>> I was wondering why developers prefers PostgreSQL other the MySQL?
>> Despite the "thread" issue are the PostgreSQL more preferable in
>> everyday working? I want to integrate a DB for FS, web, and other
>> projects. I already use MySQL, but more and more I see that PostgreSQL
>> is more preferable.
>>
>> I've seen PostgreSQL is used by ArcGIS Server. I didn't like the DB
>> Management interface. In MySQL it is more intuitive.
>>
>> Reading the internet I didn't find a comprehensive conclusion on this
>> PostgreSQL vs MySQL. Also, does PostgreSQL work in fault-tolerance,
>> replication?
>>
>> Thanks.
>>
>> --
>> Mimiko desu.
>>
>>
>>
> postgresql is the true free and opensource datase
>
> mysql is owned by oracle, his future is not clear.
>
>
> --
> ------------
>
> Itamar Reis Peixoto
> http://www.quebarato.com.br/perfil/itamarjp
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vbvbrj at gmail.com  Mon Nov  5 12:17:36 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 05 Nov 2012 11:17:36 +0200
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
Message-ID: <50978430.9050308@gmail.com>

On 05.11.2012 09:08, Itamar Reis Peixoto wrote:
>
> postgresql is the true free and opensource datase
>
> mysql is owned by oracle, his future is not clear.

Actually, this is not an argument. Oracle buyied MySQL several years 
ago. Developers of MySQL supposed if MySQL will close, they will cerate 
another project instead and will continue MySQL'ed code and direction as 
open source.

-- 
Mimiko desu.


From vallimamod.abdullah at imtelecom.fr  Mon Nov  5 13:46:07 2012
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Mon, 5 Nov 2012 11:46:07 +0100
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
Message-ID: <3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>

Hi,

On Nov 5, 2012, at 10:14 AM, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

>  PostgreSQL is not made as easy as mysql you don't get fancy "phpmyadmin" but it does the work extremely well even in some corner cases.

You may be interested by adminer (http://www.adminer.org): it's a very light, one php file "phpmyadmin" for mysql, postgres, etc.

Cheers,
Vallimamod
.




From evgeniy at bestnet.kharkov.ua  Mon Nov  5 14:37:07 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Mon, 05 Nov 2012 13:37:07 +0200
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
Message-ID: <5097A4E3.9080202@bestnet.kharkov.ua>

I'm using phppgamin 
(http://phppgadmin.sourceforge.net/doku.php?id=start). Description from 
official site:

-Administer multiple servers
-Support for PostgreSQL 7.4.x, 8.0.x, 8.1.x, 8.2.x, 8.3.x, 8.4.x, 9.0.x
-Manage all aspects of:
-Users & groups
-Databases
-Schemas
-Tables, indexes, constraints, triggers, rules & privileges
-Views, sequences & functions
-Advanced objects
-Reports
-Easy data manipulation:
-Browse tables, views & reports
-Execute arbitrary SQL
-Select, insert, update and delete
-Dump table data in a variety of formats: SQL, COPY, XML, XHTML, CSV, 
-Tabbed, pg_dump
-Import SQL scripts, COPY data, XML, CSV and Tabbed
-Supports the Slony master-slave replication engine
-Excellent language support:
   Available in 27 languages
-No encoding conflicts. Edit Russian data using a Japanese interface!
-Easy to install and configure

05.11.2012 11:14, Antonio Teixeira ?????:
> The issue is not that simple , i consider PostgreSQL more "enterprise" is
> just my point of view.
> One thing i love about postgres is the Transactions  + an extremely good
> engine , PostgreSQL is not made as easy as mysql you don't get fancy
> "phpmyadmin" but it does the work extremely well even in some corner cases.
>
> You can find many pages like this one
> http://www.wikivs.com/wiki/MySQL_vs_PostgreSQL , anyway you can always test
> but from my experience , running FS clusters for three years PostgreSQL
> never let me down even when the RAID card failed , the internal triggers
> "locked down" the database , no rows were lost , the transaction engine
> rolled back current transactions and it saved my day :).
>
> The lesson is , test first deploy later , PostegreSQL is for me , more
> feature reach and more stable but it requires more skills , mysql is more
> easy ofc less features and less stable ( this isn't true for all the cases).
>
> Its up to you use something that you feel comfortable so when problems rise
> you can actually solve problems this means you understand the database and
> how it works.
>
> P.S one point positive for mysql OPENSIPS tend to use mysql way more :)
>
> There is also a more "oracle" version of PostgreSQL ,
> http://www.enterprisedb.com/
>
> Regards
> Antonio
>
>
>
> 2012/11/5 Itamar Reis Peixoto<itamar at ispbrasil.com.br>
>
>>
>>
>>
>> On Mon, Nov 5, 2012 at 4:28 AM, Vbvbrj<vbvbrj at gmail.com>  wrote:
>>
>>> I was wondering why developers prefers PostgreSQL other the MySQL?
>>> Despite the "thread" issue are the PostgreSQL more preferable in
>>> everyday working? I want to integrate a DB for FS, web, and other
>>> projects. I already use MySQL, but more and more I see that PostgreSQL
>>> is more preferable.
>>>
>>> I've seen PostgreSQL is used by ArcGIS Server. I didn't like the DB
>>> Management interface. In MySQL it is more intuitive.
>>>
>>> Reading the internet I didn't find a comprehensive conclusion on this
>>> PostgreSQL vs MySQL. Also, does PostgreSQL work in fault-tolerance,
>>> replication?
>>>
>>> Thanks.
>>>
>>> --
>>> Mimiko desu.
>>>
>>>
>>>
>> postgresql is the true free and opensource datase
>>
>> mysql is owned by oracle, his future is not clear.
>>
>>
>> --
>> ------------
>>
>> Itamar Reis Peixoto
>> http://www.quebarato.com.br/perfil/itamarjp
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From avi at avimarcus.net  Mon Nov  5 15:08:38 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 5 Nov 2012 14:08:38 +0200
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
Message-ID: <CALNZuwkLBZM2e5AWV4-mXn4y_tvoF6WTO0RcMJD8S9Cw8Mo=Yw@mail.gmail.com>

On Mon, Nov 5, 2012 at 12:46 PM, Vallimamod ABDULLAH <
vallimamod.abdullah at imtelecom.fr> wrote:

> You may be interested by adminer (http://www.adminer.org): it's a very
> light, one php file "phpmyadmin" for mysql, postgres, etc.
>

+1 to adminer. It's integrated into FusionPBX and via PHP drivers, allows
editing of mysql, postgres, sqlite, ms sql and oracle. It seems to do
nearly everything phpmyadmin does and is a clean interface and quite fast.

-Avi
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From lazyvirus at gmx.com  Mon Nov  5 15:35:49 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Mon, 5 Nov 2012 13:35:49 +0100
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
Message-ID: <20121105133549.11b900ff@anubis.defcon1>

On Mon, 5 Nov 2012 09:14:59 +0000
Antonio Teixeira <eagle.antonio at gmail.com> wrote:

> The issue is not that simple , i consider PostgreSQL more "enterprise" is
> just my point of view.

Hmm, I see the question from another point of view: how can people use
mysql for _any_ enterprise work.

> One thing i love about postgres is the Transactions  + an extremely good
> engine , PostgreSQL is not made as easy as mysql you don't get fancy
> "phpmyadmin" but it does the work extremely well even in some corner cases.

That isn't true: as a web application you have phppgadmin
(http://phppgadmin.sourceforge.net/doku.php?id=start), as a
standalone program you have pgadmin3 (http://www.pgadmin.org/).

> 
> There is also a more "oracle" version of PostgreSQL ,
> http://www.enterprisedb.com/

In fact, Pg is seen as the open-source oracle; enterprisedb brings the
advantage of long term bug fixing & support for its chosen Pg versions :)

Jiff
-- 
<cell> apt sex
<apt> sex is probably updatedb; locate; talk; date; cd; unzip; strip; look;
      touch; finger; head; mount; fsck; more; yes; yes; umount; make clean;
      sleep
<RST2003> is there a female equivalent of that command line?
<lazarus> RST2003: sleep :)
		-- in #debian


From eagle.antonio at gmail.com  Mon Nov  5 15:42:34 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 5 Nov 2012 12:42:34 +0000
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
Message-ID: <CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>

My second No No Besides Mysql ... PHP :) , but seems interesting !.
Thanks


2012/11/5 Vallimamod ABDULLAH <vallimamod.abdullah at imtelecom.fr>

> Hi,
>
> On Nov 5, 2012, at 10:14 AM, Antonio Teixeira <eagle.antonio at gmail.com>
> wrote:
>
> >  PostgreSQL is not made as easy as mysql you don't get fancy
> "phpmyadmin" but it does the work extremely well even in some corner cases.
>
> You may be interested by adminer (http://www.adminer.org): it's a very
> light, one php file "phpmyadmin" for mysql, postgres, etc.
>
> Cheers,
> Vallimamod
> .
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sparklezou at 163.com  Mon Nov  5 11:54:02 2012
From: sparklezou at 163.com (sparklezou)
Date: Mon, 5 Nov 2012 16:54:02 +0800
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
Message-ID: <2f17f98c.442d.13acfc76aa3.Coremail.sparklezou@163.com>

Hi Sir/Madam,

I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept

And also implement it on FS.

Here I want to know, does FS could implement such features? I know such features are working on some digital phone system.

1. "KAKA" & "GAGA" are in the same "Dial Groups".
2. When someone inside/outside call "KAKA", there will be visible sentens on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name), "12345678 call KAKA" (outside display the number). 

Is it posible to implement?

Thanks in advance!

2012-11-05



sparklezou
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From sparklezou at 163.com  Mon Nov  5 11:59:36 2012
From: sparklezou at 163.com (sparklezou)
Date: Mon, 5 Nov 2012 16:59:36 +0800
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
In-Reply-To: <2f17f98c.442d.13acfc76aa3.Coremail.sparklezou@163.com>
References: <2f17f98c.442d.13acfc76aa3.Coremail.sparklezou@163.com>
Message-ID: <75a548fa.7db4.13acfcc850d.Coremail.sparklezou@163.com>




????sparklezou
?????2012-11-05 16:54
???About "Dial Groups" or "Callgroup intercept"
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>,"freeswitch-users"<freeswitch-users at lists.freeswitch.org>
???

Hi Sir/Madam,

I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept

And also implement it on FS.

Here I want to know, does FS could implement such features? I know such features are working on some digital phone system.

1. "KAKA" & "GAGA" are in the same "Dial Groups".
2. When someone inside/outside call "KAKA", there will be visible sentens on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name), "12345678 call KAKA" (outside display the number). 

Is it posible to implement?

Thanks in advance!

2012-11-05



sparklezou
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From mike at jerris.com  Mon Nov  5 16:26:33 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Nov 2012 08:26:33 -0500
Subject: [Freeswitch-users] media proxy
In-Reply-To: <2003146.NZnh1akVdp@sos>
References: <23050389.219.1351865910300.JavaMail.master@VoiceJuggler>
	<1945566.6VRVmflkY8@sos>
	<CANBhyPBYrbUuDfcA1Dg=Z9nePZwfcSf043qQdDUBgD=sxPTpNw@mail.gmail.com>
	<2003146.NZnh1akVdp@sos>
Message-ID: <320FAB79-A5F6-4FD3-A1AA-411CD022C798@jerris.com>

The correct solution for this is to add passthrough codecs for those that are not supported.

On Nov 3, 2012, at 9:55 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> So, there there is a reason to use proxy mode (2) sometime.
> 
> On Saturday 03 November 2012 15:59:01 Anton Kvashenkin wrote:
>> Yes, you can pass-thru g729.
>> 
>> 2012/11/2 Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>> 
>>> Will proxy mode 2 work with codecs not supported by freeswitch?
>>> 
>>> On Friday 02 November 2012 11:39:27 Michael Jerris wrote:
>>>> There should be no reason to ever use proxy mode (2).
>>>> 
>>>> default mode that you call "hard proxy" requires no settings to set it,
>>> 
>>> its
>>> 
>>>> the default.
>>>> On Nov 2, 2012, at 10:13 AM, Max Kovalenko <
>>> 
>>> mkovalenko at cybervisiontech.com>
>>> 
>>> wrote:
>>>>> Hello,
>>>>> 
>>>>> There are two modes of media stream to be proxied or bypassed.
>>>>> 
>>>>> 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
>>> 
>>> etc.
>>> 
>>>>> 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
>>>>> traversal purposes. No media stream parsing is supported 3. Media
>>> 
>>> bypass
>>> 
>>>>> - RTP streams are bypassing FreeSWITCH at all.
>>> 
>>>>> There are also two channel parameters affecting above modes:
>>> bypass_media
>>> 
>>>>> and proxy_media.
>>>>> 
>>>>> - Are these parameters independent? Meaning the combination:
>>>>> bypass_media=true AND proxy_media=true is possible. What effect will
>>> 
>>> be?
>>> 
>>>>> - Does it means that if I want to turn on "light proxy" I would always
>>>>> need to set bypass_media=false AND proxy_media=true?
>>>>> 
>>>>> - How to set "hard proxy" (full media proxy) per channel before to
>>> 
>>> bridge
>>> 
>>>>> legs?
>>>>> 
>>>>> Waiting for your replay ASAP. Thank you in advance.
>>>> 



From mike at jerris.com  Mon Nov  5 16:29:02 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Nov 2012 08:29:02 -0500
Subject: [Freeswitch-users] Incoming Calls, use From instead of PID?
In-Reply-To: <CALNZuw=kEWvmhAYdJ_j8_QyKXmpYu=UO3S41HStFCKhu8ye8og@mail.gmail.com>
References: <CALNZuw=kEWvmhAYdJ_j8_QyKXmpYu=UO3S41HStFCKhu8ye8og@mail.gmail.com>
Message-ID: <7CA92F6E-54F3-4AFA-87ED-95562A75AD80@jerris.com>

I am not sure why they would tell you to ignore PID, that seem broken to me.  However, in this case, you could resolve it in a couple ways.

1) you could just look at the sip_from_user variable
2) you could set a param on the gateway for caller id in from (this may be less sucessful based on if we can match this inbound call to the gateway)

Mike

On Nov 3, 2012, at 5:45 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Hi - I'm getting calls from a carrier with the proper e.164 FROM but their PID is coming in inconsistently. I was told to ignore the PID.
> 
> Apparently, FS is using the PID for the profile Caller-Caller-ID-Number variable.
> 
> Are there any toggles for how to manage inbound caller ID?
> This is currently coming in via ACL to public context... can I even set the sip_cid for that?
> 
> Suggestions?
> 
> Thanks,
> 
> -Avi Marcus

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From mike at jerris.com  Mon Nov  5 16:31:07 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Nov 2012 08:31:07 -0500
Subject: [Freeswitch-users] ODBC with mod_voicemail and storage-dir
In-Reply-To: <CAOjvyPkhnDPAfKTV8wP+Dt_AiaLird5FsuH4UYLXsT+qrp33XA@mail.gmail.com>
References: <CAOjvyPkhnDPAfKTV8wP+Dt_AiaLird5FsuH4UYLXsT+qrp33XA@mail.gmail.com>
Message-ID: <98C09FCA-749A-4F71-BBFC-4ACEB0A1DA75@jerris.com>

How are you calling voicemail?  Are you specifying to use the profile listed below?

On Nov 3, 2012, at 9:42 PM, KPS Maillinglist <ksims.ml at gmail.com> wrote:

> I cant seem to get mod_voicemail to store vm information in the database. It also is not using the storage-dir param to place the vm files. Please help. 
> 
> 
> <profile name="DEMO1_CDI_TEST1">
> <settings>
> <param name="callback-context" value="DEMO1_CDI_TEST1"/>
> <param name="callback-dialplan" value="XML"/>
> <param name="choose-greeting-key" value="2"/>
> <param name="config-menu-key" value="5"/>
> <param name="delete-file-key" value="7"/>
> <param name="digit-timeout" value="10000"/>
> <param name="email-key" value="4"/>
> <param name="ff-key" value="6"/>
> <param name="file-extension" value="mp3"/>
> <param name="listen-file-key" value="1"/>
> <param name="main-menu-key" value="*"/>
> <param name="max-login-attempts" value="3"/>
> <param name="max-record-length" value="300"/>
> <param name="odbc-dsn" value="fs_regsrv_vm:fs_cs:staysynced"/>
> <param name="operator-extension" value="operator XML DEMO1_CDI_TEST1"/>
> <param name="operator-key" value="9"/>
> <param name="pause-key" value="0"/>
> <param name="play-new-messages-key" value="1"/>
> <param name="play-saved-messages-key" value="2"/>
> <param name="record-file-key" value="3"/>
> <param name="record-greeting-key" value="1"/>
> <param name="record-name-key" value="3"/>
> <param name="record-silence-hits" value="2"/>
> <param name="record-silence-threshold" value="200"/>
> <param name="restart-key" value="1"/>
> <param name="rew-key" value="4"/>
> <param name="save-file-key" value="9"/>
> <param name="storage-dir" value="/fs/cust/vm/DEMO1_CDI_TEST1"/>
> <param name="terminator-key" value="#"/>
> <param name="tone-spec" value="%(1000, 0, 640)"/>
> <param name="undelete-file-key" value="8"/>
> <param name="web-template-file" value="web-vm.tpl"/>
> <email>
> <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
> <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
> <param name="notify-template-file" value="notify-voicemail.tpl"/>
> <param name="template-file" value="voicemail.tpl"/>
> </email>
> </settings>
> </profile>
> _________________________________________________________________________

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From sos at sokhapkin.dyndns.org  Mon Nov  5 16:32:54 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 05 Nov 2012 08:32:54 -0500
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
	<CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>
Message-ID: <3261345.GBi1nAZ3EQ@sos>

 Does Postgres support circular multi-master replication like mysql does? Are 
there some ways to run geographically distributed DB cluster?

On Monday 05 November 2012 12:42:34 Antonio Teixeira wrote:
> My second No No Besides Mysql ... PHP :) , but seems interesting !.
> Thanks
> 
> 
> 2012/11/5 Vallimamod ABDULLAH <vallimamod.abdullah at imtelecom.fr>
> 
> > Hi,
> > 
> > On Nov 5, 2012, at 10:14 AM, Antonio Teixeira <eagle.antonio at gmail.com>
> > 
> > wrote:
> > >  PostgreSQL is not made as easy as mysql you don't get fancy
> > 
> > "phpmyadmin" but it does the work extremely well even in some corner
> > cases.
> > 
> > You may be interested by adminer (http://www.adminer.org): it's a very
> > light, one php file "phpmyadmin" for mysql, postgres, etc.
> > 
> > Cheers,
> > Vallimamod
> > .
> > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From vbvbrj at gmail.com  Mon Nov  5 16:43:39 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 05 Nov 2012 15:43:39 +0200
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
Message-ID: <5097C28B.2070101@gmail.com>

On 05.11.2012 11:14, Antonio Teixeira wrote:
> The issue is not that simple , i consider PostgreSQL more "enterprise"
> is just my point of view.

What means more "enterprise" from your point of view?

> One thing i love about postgres is the Transactions  + an extremely good
> engine , PostgreSQL is not made as easy as mysql you don't get fancy
> "phpmyadmin" but it does the work extremely well even in some corner cases.

I don't use phpmyadmin, I use the MySQL GUI Tools distributed by the 
developers. PostreSQL also has its GUI, but its not intuitive. Ok. This 
is a like only. Eventually anyone can get used with administering the DB 
with some tools.

I am about the work progress, replication, fault-tolerance building, 
transactions is, foreign keys. Innodb in MySQL also is transactional and 
has foreign key with updates and deletes handling.

Does PostgreSQL has some stability improvements, db speed access, 
sequrity issues better then MySQL?

-- 
Mimiko desu.


From avi at avimarcus.net  Mon Nov  5 17:07:14 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 5 Nov 2012 16:07:14 +0200
Subject: [Freeswitch-users] Incoming Calls, use From instead of PID?
In-Reply-To: <7CA92F6E-54F3-4AFA-87ED-95562A75AD80@jerris.com>
References: <CALNZuw=kEWvmhAYdJ_j8_QyKXmpYu=UO3S41HStFCKhu8ye8og@mail.gmail.com>
	<7CA92F6E-54F3-4AFA-87ED-95562A75AD80@jerris.com>
Message-ID: <CALNZuwnuRBMWWJ8iSfJ7s49Fc4fA4PQFhDF-9N=7OyN5mPE0LQ@mail.gmail.com>

Thanks Mike.
Yep, it seems broken, but they don't seem to think it's their issue.

1) This would mean that I have to handle this carriers and others
carrier separately. I do a bunch of things to process the inbound caller ID
so that's not really ideal..
2) I think gateway caller-id-in-from is for outbound, not inbound. Perhaps
cid_type=none but right now it's just an ACL for inbound so I'd have to
create a user for each incoming IP address and set the var there. Not
ideal...

-Avi


On Mon, Nov 5, 2012 at 3:29 PM, Michael Jerris <mike at jerris.com> wrote:

> I am not sure why they would tell you to ignore PID, that seem broken to
> me.  However, in this case, you could resolve it in a couple ways.
>
> 1) you could just look at the sip_from_user variable
> 2) you could set a param on the gateway for caller id in from (this may be
> less sucessful based on if we can match this inbound call to the gateway)
>
> Mike
>
> On Nov 3, 2012, at 5:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> Hi - I'm getting calls from a carrier with the proper e.164 FROM but their
> PID is coming in inconsistently. I was told to ignore the PID.
>
> Apparently, FS is using the PID for the profile Caller-Caller-ID-Number
> variable.
>
> Are there any toggles for how to manage *inbound *caller ID?
> This is currently coming in via ACL to public context... can I even set
> the sip_cid for that?
>
> Suggestions?
>
> Thanks,
>
> -Avi Marcus
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Chad.Engler at patlive.com  Mon Nov  5 17:07:40 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Mon, 5 Nov 2012 09:07:40 -0500
Subject: [Freeswitch-users] mod_sms delivery failure
In-Reply-To: <38522381-38CB-497F-AA19-FE0759A8CE68@gmail.com>
References: <C0F15DFC-F3F0-49F1-B90D-74F9C0DD1576@gmail.com><BD87D3B1554F614190EB6E54CE95DDB606B937CB@mail2.patlive.local>
	<38522381-38CB-497F-AA19-FE0759A8CE68@gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606B93883@mail2.patlive.local>

No problem! Thanks for using it! I've already got a couple of good pull
requests that have been merged in, and a couple other bugs fixed as
well. I want to do a little more cleanup, but npm should be seeing a new
version this week; so be ready for that!

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Lucas Fontes
Sent: Thursday, November 01, 2012 7:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_sms delivery failure

 

 

Hi Chad,

thanks for that awesome ESL binding for node.js !

 

Section 7 of rfc3428

 

	A UAS that receives a MESSAGE request processes it following the
	   rules of SIP [1].
	 
	   A UAS receiving a MESSAGE request SHOULD respond with a final
	   response immediately.  Note, however, that the UAS is not
obliged to
	   display the message to the user either before or after
responding
	   with a 200 OK.  That is, a 200 OK response does not
necessarily mean
	   the user has read the message.
	 
	   A 2xx response to a MESSAGE request MUST NOT contain a body.
A UAS
	   MUST NOT insert a Contact header field into a 2xx response.
	 
	   A UAS which is, in fact, a message relay, storing the message
and
	   forwarding it later on, or forwarding it into a non-SIP
domain,
	   SHOULD return a 202 (Accepted) [5] response indicating that
the
	   message was accepted, but end to end delivery has not been
	   guaranteed.
	 
	   A 4xx or 5xx response indicates that the message was not
delivered
	   successfully.  A 6xx response means it was delivered
successfully,
	   but refused.

 

 

At Fongo (fongo.com) we have a few proxies for store and forward, also
for SMS integration. We use 200, 402, 404 and 488 to control delivery
and payments.

So in our case, the final device always returns 200 (as it is the
final), but a proxy in between might generate a 4xx.

we have an opensips cluster in front of freeswitch and steer MESSAGE
methods away from freeswitch to handle those scenarios. I'm  looking
into removing this logic from opensips and move it to freeswitch to keep
it consistent with dialplan and rating engine. 

 

cheers

--

lucas

 

 

On 2012-11-01, at 5:46 PM, Chad Engler <Chad.Engler at patlive.com> wrote:





I don't think SIP MESSAGEs have a response/ack. I know that our provider
doesn't send anything except the 200 OK when we send a message, and that
doesn't mean it actually made it to the device. As far as I know there
is no way to know if and when the device gets the actual message without
the device responding with another message back through the via chain.

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Lucas Fontes
Sent: Thursday, November 01, 2012 1:38 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_sms delivery failure

 

Hi everyone,

 

I've been trying to determine if a message was successfully delivered to
a device.

Stumbled on a post from February (subject: Testing
mod_sms)http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-Februa
ry/005619.html:
<http://lists.freeswitch.org/pipermail/freeswitch-dev/2012-February/0056
19.html:> 

 

________________________________
From: Anthony Minessale <anthony.minessale at gmail.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> >
To: Warren Lin <wlin121443 at yahoo.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> > 
Sent: Tuesday, February 21, 2012 11:01 AM
Subject: Re: Testing mod_sms
 
hi you can find out how to use the lists at http://lists.freeswitch.org
<http://lists.freeswitch.org> 
 
try latest GIT, i put in a patch to fire the events with the heder
Failed-Delivery true
if you are listening for MESSAGE events you should catch them.
________________________________

 

 

I can't find any reference to Failed-Delivery in mod_sms.c neither
anywhere else in the source tree. 

There is a field called Delivery-Failure, but seems to be used by other
modules to indicate a failure on the response going out of freeswitch,
not to freeswitch.

Was this overwritten / rolled back at some point ? 

 

thanks

 

--

lucas

 

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


 <> 

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

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

 

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From mike at jerris.com  Mon Nov  5 17:07:36 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Nov 2012 09:07:36 -0500
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <20121105133549.11b900ff@anubis.defcon1>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<20121105133549.11b900ff@anubis.defcon1>
Message-ID: <475D1248-9FAD-48D0-9F92-7B6B684EBAF6@jerris.com>

Freeswitch creates all the tables and indexes for you.  There is no need for a GUI admin as there is not really anything to admin to set up freeswitch.  The only place this might not be true is if you are trying to setup replication, where mysql still has the edge in functionality.  I clearly am biased away from YourSQL, but in the end, I see no good reason to add it to FreeSWITCH. If someone feels strongly the other way, we would entertain patches to change this, under the guidelines set out earlier by Tony (and a few clarifications/additions).

1) We would need proof of some advantage over ODBC.  People often state that ODBC is slower, but I have yet to see evidence this is actually true to an extent that warrants the additional trouble.
2) A modular interface would need to be added if we add any more specific db interfaces.
3) Proper configure (and possibly runtime) check will need to be added to ensure we are using proper mysql libraries checking for transaction handling and thread safety

Postgres was added to the core in order to address specific limitations in the current interface with postgres ODBC related to how it makes blocking calls to the database, not for performance reasons, and was done so by our team due to our own needs.  If someone feels strongly this needs to be added, has the skillset required to make these patches, and is willing to put in the time to do so, please talk to me off list and I am happy to discuss requirements in more detail.

Mike

On Nov 5, 2012, at 7:35 AM, Bzzz <lazyvirus at gmx.com> wrote:

> On Mon, 5 Nov 2012 09:14:59 +0000
> Antonio Teixeira <eagle.antonio at gmail.com> wrote:
> 
>> The issue is not that simple , i consider PostgreSQL more "enterprise" is
>> just my point of view.
> 
> Hmm, I see the question from another point of view: how can people use
> mysql for _any_ enterprise work.
> 
>> One thing i love about postgres is the Transactions  + an extremely good
>> engine , PostgreSQL is not made as easy as mysql you don't get fancy
>> "phpmyadmin" but it does the work extremely well even in some corner cases.
> 
> That isn't true: as a web application you have phppgadmin
> (http://phppgadmin.sourceforge.net/doku.php?id=start), as a
> standalone program you have pgadmin3 (http://www.pgadmin.org/).
> 
>> 
>> There is also a more "oracle" version of PostgreSQL ,
>> http://www.enterprisedb.com/
> 
> In fact, Pg is seen as the open-source oracle; enterprisedb brings the
> advantage of long term bug fixing & support for its chosen Pg versions :)
> 



From Chad.Engler at patlive.com  Mon Nov  5 17:25:19 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Mon, 5 Nov 2012 09:25:19 -0500
Subject: [Freeswitch-users] application synchronisation between
	freeswitchand web
In-Reply-To: <CAGDHiz=WarbQftBxaok_LG5ZPLMaH_fFBnC8vU=z9dg9-qRUhA@mail.gmail.com>
References: <CAGDHiz=WarbQftBxaok_LG5ZPLMaH_fFBnC8vU=z9dg9-qRUhA@mail.gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606B9388A@mail2.patlive.local>

That is just what the channels
<https://github.com/englercj/node-esl/tree/master/examples/channels>
example of node-esl is doing. It basically dumps the changes in a
channel to the client to update a list that the client maintains (the
server also buffers the changes, but that is because the example
demonstrates 3 different methods of displaying the data).

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Mumuney Abdlquadri
Sent: Wednesday, October 31, 2012 6:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] application synchronisation between
freeswitchand web

 

Hi all,

 

I am think of building an application like the demo here 

 

http://www.certificall.net/Try

 

I noticed that the calls are in complete sync with the app.

 

I am thinking to use ESL (maybe node-esl) and tunnel the events through
socket.io.

 

Am I thinking straight? Or does anyone have a better proposition?

 

Regards.

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From itamar at ispbrasil.com.br  Mon Nov  5 17:35:37 2012
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Mon, 5 Nov 2012 12:35:37 -0200
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <3261345.GBi1nAZ3EQ@sos>
References: <50975C7B.3090807@gmail.com>
	<3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
	<CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>
	<3261345.GBi1nAZ3EQ@sos>
Message-ID: <CACyzUwLgJuxkE5wFDHttDPns3Daxdj4So1bZqBfq0LJyy6DnvQ@mail.gmail.com>

On Mon, Nov 5, 2012 at 11:32 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

>  Does Postgres support circular multi-master replication like mysql does?
> Are
> there some ways to run geographically distributed DB cluster?
>
> yes, bucardo.org



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

Itamar Reis Peixoto
http://www.quebarato.com.br/perfil/itamarjp
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From cal.leeming at simplicitymedialtd.co.uk  Mon Nov  5 17:37:18 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 5 Nov 2012 14:37:18 +0000
Subject: [Freeswitch-users] http://wiki.freeswitch.org/wiki/Mod_xml_curl
 updated and big thanks to Cal
In-Reply-To: <CAFVUaV-48AF+-ZD2GA9n43e8TB1ZyotOVJfYpEjCHyCVe6sSsA@mail.gmail.com>
References: <CCB6B8C4.5ECE2%krice@freeswitch.org>
	<CAMwZ-tJPvQjP3AWM6snQ87FvCQ8WZ-1tERRWxMwKvX9MC8_1CA@mail.gmail.com>
	<CAFVUaV-48AF+-ZD2GA9n43e8TB1ZyotOVJfYpEjCHyCVe6sSsA@mail.gmail.com>
Message-ID: <CAHKQagF_-u8YP-MG0rSNMWEVSSiuPc_mstPh7WxFeJ625x432Q@mail.gmail.com>

Glad it's helped!

I'll be updating it over the next few weeks, as there is still a lot of
information missing.

Cal

On Mon, Nov 5, 2012 at 2:38 AM, Yiftach Golan <yiftah at choochee.com> wrote:

> +1 to that also helped me
>
>
> On Sat, Nov 3, 2012 at 7:58 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Wed, Oct 31, 2012 at 10:15 AM, Ken Rice <krice at freeswitch.org> wrote:
>> > Cal had a problem with the documentation as many of us do, he got on the
>> > mailing list and figured his issues out, then took the time to totally
>> > re-write the mod_xml_curl page on the wiki.
>>
>> Solid work, Cal. Thanks!
>>
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Nov  5 17:50:02 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Nov 2012 08:50:02 -0600
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
	<CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
	<CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>
Message-ID: <CAKbxfG9OYOR9cfjSBmnow76d-0C8d6NgqVzuUPv_3CpWB_4mPQ@mail.gmail.com>

As I already said.  I would not add any new db interface unless
we re-factored it into pluggable modules.  I am not planning on working on
this so this any time soon so this is really a m00t discussion.  I would
prefer to focus attention on things we actually have the time and energy
for that are on our roadmap.



On Mon, Nov 5, 2012 at 12:39 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Nov 4, 2012 at 7:37 PM, Yiftach Golan <yiftah at choochee.com> wrote:
> > OpenSIPs goes very well with FreeSWITCH (at least for us) and OpenSIPs
> use
> > mysql as its primary db
>
> OpenSIPS works great with PostgreSQL.
>
>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From ocset at the800group.com  Mon Nov  5 17:55:38 2012
From: ocset at the800group.com (ocset)
Date: Mon, 05 Nov 2012 22:55:38 +0800
Subject: [Freeswitch-users] Play music while transferring new incoming call
Message-ID: <5097D36A.7010803@the800group.com>

Hi

I am trying to to configure FS to play music while transferring a new 
incoming call. Like when you phone a business and you get the "your call 
in important to use, please wait while we transfer..." message.

I found this suggestion on the web but it does not work for me. I have 
tested the two <action> command separately to make sure they work but 
when they are put together, the call is never answered and no music is 
played.


<include>
  <extension name="ext_8888">
         <condition field="destination_number" expression="^(8888)$">
                 <action application="playback" data="music/test.wav"/>
                 <action application="bridge" 
data="{ignore_early_media}sofia/internal/1005 at 192.168.0.150"/>
         </condition>
  </extension>
</include>

ps. I cannot find the answer in either of the two FS books. Would have 
expected this to be there, at least in the cookbook? All I can find is 
examples of playing music and then hanging up. Happy to be corrected if 
I missed it:-)

All help greatly appreciated
Thanks
O.


From anthony.minessale at gmail.com  Mon Nov  5 18:01:18 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Nov 2012 09:01:18 -0600
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
	<CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>
Message-ID: <CAKbxfG-HCVKksr63KPuwiNq+RxZkGWrAT0nBT3_+2Qiz3PYjXg@mail.gmail.com>

Its simpler than that.

I have used both db in my time as a developer and I have been mad at both
for one reason or another.
It all boils down to Mysql is not as stable and has several thread safety
and memory leak issues when its used in a multi-threaded app like
FreeSWITCH.  In fact that preference is irrelevant because technically the
default database in FS is sqlite because the design goal of the project is
for the software to run anywhere in a minimal capacity and be able to grow
and scale with the user's needs.  So anyone can use any database they want.
 However, when we get a bug report that shows Mysql crashing, that is not
something we can do anything about.  Since we do have better luck with
postgres, we allowed the inclusion of native api but that was a 3rd party
contribution.

So bottom line is we don't care what other people do, we just do what we
like doing and share it.





On Mon, Nov 5, 2012 at 6:42 AM, Antonio Teixeira <eagle.antonio at gmail.com>wrote:

> My second No No Besides Mysql ... PHP :) , but seems interesting !.
> Thanks
>
>
>
> 2012/11/5 Vallimamod ABDULLAH <vallimamod.abdullah at imtelecom.fr>
>
>> Hi,
>>
>> On Nov 5, 2012, at 10:14 AM, Antonio Teixeira <eagle.antonio at gmail.com>
>> wrote:
>>
>> >  PostgreSQL is not made as easy as mysql you don't get fancy
>> "phpmyadmin" but it does the work extremely well even in some corner cases.
>>
>> You may be interested by adminer (http://www.adminer.org): it's a very
>> light, one php file "phpmyadmin" for mysql, postgres, etc.
>>
>> Cheers,
>> Vallimamod
>> .
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From avi at avimarcus.net  Mon Nov  5 18:25:08 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 5 Nov 2012 17:25:08 +0200
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAKbxfG9OYOR9cfjSBmnow76d-0C8d6NgqVzuUPv_3CpWB_4mPQ@mail.gmail.com>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
	<CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
	<CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>
	<CAKbxfG9OYOR9cfjSBmnow76d-0C8d6NgqVzuUPv_3CpWB_4mPQ@mail.gmail.com>
Message-ID: <CALNZuwmuiufpioV-ktnSDky0YrtZwwi1g_MT5mCws+z5f48GvA@mail.gmail.com>

Is there a public roadmap somewhere -- for your coding, and for the project
as a whole (e.g. packaging, docs, books, etc) and who is doing each one?

-Avi
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From avi at avimarcus.net  Mon Nov  5 18:30:32 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 5 Nov 2012 17:30:32 +0200
Subject: [Freeswitch-users] Play music while transferring new incoming
	call
In-Reply-To: <5097D36A.7010803@the800group.com>
References: <5097D36A.7010803@the800group.com>
Message-ID: <CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>

You probably want ringback and/or transfer ringback. That's for playing
something while the call is waiting to be answered.

See:
http://wiki.freeswitch.org/wiki/Time_of_Day_Routing#Example_for_office_open_09:00-16:00
there's
something set there, using:


       <action application="set" data="ringback=local_stream://moh"/>
       <action application="set" data="transfer_ringback=local_stream://moh"/>

You're line isn't quite right -- "{ignore_early_media*=true*}", you need to
set the variable data.

Playback is blocking -- it plays and then moves on. Transfer_ringback sets
what to do when the transfer occurs.

I hope that helps and you can clean up the wiki when you've got it all
figured out.

-Avi



On Mon, Nov 5, 2012 at 4:55 PM, ocset <ocset at the800group.com> wrote:

> Hi
>
> I am trying to to configure FS to play music while transferring a new
> incoming call. Like when you phone a business and you get the "your call
> in important to use, please wait while we transfer..." message.
>
> I found this suggestion on the web but it does not work for me. I have
> tested the two <action> command separately to make sure they work but
> when they are put together, the call is never answered and no music is
> played.
>
>
> <include>
>   <extension name="ext_8888">
>          <condition field="destination_number" expression="^(8888)$">
>                  <action application="playback" data="music/test.wav"/>
>                  <action application="bridge"
> data="{ignore_early_media}sofia/internal/1005 at 192.168.0.150"/>
>          </condition>
>   </extension>
> </include>
>
> ps. I cannot find the answer in either of the two FS books. Would have
> expected this to be there, at least in the cookbook? All I can find is
> examples of playing music and then hanging up. Happy to be corrected if
> I missed it:-)
>
> All help greatly appreciated
> Thanks
> O.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From manieq at wp.eu  Mon Nov  5 18:33:38 2012
From: manieq at wp.eu (Mariusz Czulada)
Date: Mon, 05 Nov 2012 16:33:38 +0100
Subject: [Freeswitch-users] How to add SIP Route URI params?
Message-ID: <5097dc52ea28a7.83268813@wp.pl>

Hello,

I faced a problem for which I cannot find a solution yet. 

[Environment: freeswitch 1.2.0-rc2, RHEL5.7, FreeSWITCH server attached to ALU IMS platform as an MRF]

I have to make an outgoing call. I was told by an IMS vendor, that I should use "Route:" header with ";orig" attribute in it. I try to execute:

 originate
  {origination_caller_id_number=+48399555555,ignore_early_media=true,originate_timeout=15}
  sofia/service/+48223301008 at neofon.tp.pl:5090;user=phone;
  fs_path=sip:+48223301008 at icsf-stdn.imsgroup0-000.isc01.ims.tp.pl:5090 &echo
[I splitted this long line]

and I get: 

   INVITE sip:+48223301008 at neofon.tp.pl:5090;user=phone SIP/2.0
   Via: SIP/2.0/UDP 10.79.21.107;rport;branch=z9hG4bK3yaUK4vQN0mSQ
   Route: <sip:+48223301008 at icsf-stdn.imsgroup0-000.isc01.ims.tp.pl:5090>
   [...]

ALU request that Route URI must also contain ";lr;orig" attributes so next I tried:

  {origination_caller_id_number=+48399555555,ignore_early_media=true,originate_timeout=15}
  sofia/service/+48223301008 at neofon.tp.pl:5090;user=phone;
  fs_path=sip:+48223301008 at icsf-stdn.imsgroup0-000.isc01.ims.tp.pl:5090;lr;orig &echo

but it didn't change the header. What I try to get (and requested to) is:

   Route: <sip:+48223301008 at icsf-stdn.imsgroup0-000.isc01.ims.tp.pl:5090;lr;orig>

Can anybody help me with this?

regards,

Mariusz




From darcy at Vex.Net  Mon Nov  5 18:34:30 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Mon, 5 Nov 2012 10:34:30 -0500
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <5097C28B.2070101@gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<5097C28B.2070101@gmail.com>
Message-ID: <20121105103430.408659fd@dilbert>

On Mon, 05 Nov 2012 15:43:39 +0200
Vbvbrj <vbvbrj at gmail.com> wrote:
> I don't use phpmyadmin, I use the MySQL GUI Tools distributed by the 
> developers. PostreSQL also has its GUI, but its not intuitive. Ok.
> This is a like only. Eventually anyone can get used with
> administering the DB with some tools.

I don't understand this obsession with GUI tools for working on
databases.  If we are talking about enterprise level databases I assume
we mean enterprise level deployment as well.  That means sitting down
and designing a database from requirements.  The database should be
created in a text file that can be dropped into a versioning system.
Don't say that binary project files can be saved because you can't
diff binary files in any useful way.

And don't get me started on the dog's breakfast that is PHP.  I won't
even install it on my central database server.

> I am about the work progress, replication, fault-tolerance building, 
> transactions is, foreign keys. Innodb in MySQL also is transactional
> and has foreign key with updates and deletes handling.

Of course PostgreSQL has all that.  Who do you think MySQL is trying to
catch up to.

> 
> Does PostgreSQL has some stability improvements, db speed access, 
> sequrity issues better then MySQL?

Absolutely.  I wouldn't trust my critical data to any other open source
database and very few commercial ones.

Run a busy database application in both PostgreSQL and MySQL and pull
the plug on the server once in a while.  Want to take bets on which one
corrupts your data and requires a restore from backup?

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From andrew at cassidywebservices.co.uk  Mon Nov  5 18:57:43 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Mon, 5 Nov 2012 15:57:43 +0000
Subject: [Freeswitch-users] Remote pickup of on-hold extensions
In-Reply-To: <e62dfe57ead648fc2e67de37fc38726f.squirrel@icefire.qza.net.au>
References: <bc4d4e4f822530c463361a4f72774f6a.squirrel@icefire.qza.net.au>
	<CAMbW9oKH0j_oRuMjaDOGb6A=eBMf79XZMezSBeggWTdmYJDrDg@mail.gmail.com>
	<e62dfe57ead648fc2e67de37fc38726f.squirrel@icefire.qza.net.au>
Message-ID: <CAMbW9o+3NsdJJwUN4B6oiG5BrLTy=7ieq7u1HQed=WzBF+Rv6g@mail.gmail.com>

You can one-touch park using the valet_park application, and you set up
your BLF keys to monitor/park/unpark calls on the parking extesions.

For example, for a 10 slot (700 to 709) lot you'd have:

<extension name="parkinglot">
 <condition field="destination_number" expression="^(park+)?(70\d)$">
  <action application="valet_park" data="lot $1" />
 </condition>
</extension>

Then set you BLF keys to monitor/dial park+700 to park+709 and if your
handsets support it properly and are configured correctly, just pressing
one of the free line keys should park the call in that slot, and illuminate
the LED.

You can also dial 700-709 directly, if you wanted to.

Is that a bit closer?

On 3 November 2012 10:46, <sms at icefire.qza.net.au> wrote:

> I've thought about those options... the obstacle is mostly the users, who
> aren't the most trainable people in the world, and they're used to the
> present key system which allows them to have pretty colored buttons that
> they only press once. I dread the attempt to explain the concept of
> transferring a call, and the question will inevitably come back: "Why
> can't we have line keys like the last system?" Calls will likely get lost
> or dropped, and of course, the system gets blamed ;-)
>
> The other option, again if possible, is to program a soft key as a
> one-touch "Park" key that announces the park extension number, which could
> also be assigned to a BLF, then parks the call. Then we have one touch
> park, one touch retrieve.
>
> I have to sell this to them in such a way that it doesn't come across as a
> feature downgrade, or we may end up stuck with the present system until it
> dies and no inter-branch connectivity.
>
> I suppose it's the same old predicament... things would be so simple if
> management just said yes :)
>
>
>
> > Personally I'd just use the Blind Transfer features on your handsets.
> 1000
> > picks up the call, presses transfer on the phone, then dials 1001 and
> > hangs
> > up. 1001 rings and is connected to the transferred call. Sure you'e not
> > got
> > the BLF, but the idea works fine. You could use call parking, but that
> has
> > more steps to it.
> >
> > On 2 November 2012 04:16, <sms at icefire.qza.net.au> wrote:
> >
> >> Hi all,
> >>
> >> I have a scenario that I'm thinking of implementing, but not sure if
> >> it's
> >> possible, so here's the details:
> >>
> >> Extensions 200 to 250 are regular internal sip phones, contained in a
> >> call
> >> group (Ext 1000)
> >>
> >> Extensions 300 to 305 are SIP-GSM and SIP-PSTN gateways with SIP client
> >> firmware, namely those cheap GOIP GSM units on ebay, plus an SPA3000.
> >> These are to be configured with the logon credentials of the extension
> >> and
> >> behave as a typical SIP client.
> >>
> >> The dialplan will route incoming and outgoing calls via these gateways
> >> as
> >> if they were trunks. They will restricted privileges to prevent toll
> >> fraud. The phones will either be Yealink T28p or Grandstream GXP2124, it
> >> depends if GS can respond with a fix for their horrible AGC or not. The
> >> gateways will be subscribed to the phone's DSS keys as BLF's.
> >>
> >> Now, all this so far is reasonably straight forward. The next part is
> >> the
> >> tricky part (you might've guessed where this is going....)
> >>
> >> Let's say a call comes in on 300, routes to the call group and is picked
> >> up by 1000, who then puts the call on hold. The BLF key for 300 shows it
> >> as busy/on hold. 1000 then calls 1001 to take the call. 1001 then
> >> presses
> >> BLF 300, which causes the call to be transferred to them and
> >> automatically
> >> answered.
> >>
> >> So is there any way in freeswitch to replicate this behavior? I know
> >> it's
> >> possible to do SAA, but this gets tricky if I want to have time based
> >> call
> >> groups, or to share incoming lines between branches. The method above
> >> allows the lines to be present on all phones, while still configuring
> >> call
> >> groups in the usual way and more or less replicating old key system
> >> behavior.
> >>
> >> Cheers,
> >> Francis
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > *Andrew Cassidy BSc (Hons) MBCS SSCA*
> > Managing Director
> >
> >
> > *T <info at cassidywebservices.co.uk> *03300 100 960
> > *F<info at cassidywebservices.co.uk>
> >  *03300 100 961
> > *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> > *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From krice at freeswitch.org  Mon Nov  5 17:57:46 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 05 Nov 2012 09:57:46 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CALNZuwmuiufpioV-ktnSDky0YrtZwwi1g_MT5mCws+z5f48GvA@mail.gmail.com>
Message-ID: <CCBD3E1A.5EFC7%krice@freeswitch.org>

That?s something I thing we need to work on...


On 11/5/12 10:25 AM, "Avi Marcus" <avi at avimarcus.net> wrote:

> Is there a public roadmap somewhere -- for your coding, and for the project as
> a whole (e.g. packaging, docs, books, etc) and who is doing each one?
> 
> -Avi
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From lazyvirus at gmx.com  Mon Nov  5 19:12:07 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Mon, 5 Nov 2012 17:12:07 +0100
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <3261345.GBi1nAZ3EQ@sos>
References: <50975C7B.3090807@gmail.com>
	<3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
	<CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>
	<3261345.GBi1nAZ3EQ@sos>
Message-ID: <20121105171207.3f4f469f@anubis.defcon1>

On Mon, 05 Nov 2012 08:32:54 -0500
Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

>  Does Postgres support circular multi-master replication like mysql does?

You have several solutions, depending whether you want sync (a bit of
weakness her) or async: PGCluster, Bucardo, Postgres-R, RubyRep...
(http://wiki.postgresql.org/wiki/Replication,_Clustering,_and_Connection_Pooling)

EnterpriseDB is also presenting a solution (xDB), but it is in a beta state
though.

AFAIK, the most achieved sync project is Postgres-XC
(See http://postgres-xc.sourceforge.net/ &
http://en.wikipedia.org/wiki/Multi-master_replication#PostgreSQL) 
and Rubirep for async, (http://www.rubyrep.org/) - open-source speaking of
course, 'cos there are other (proprietary) solutions.

> Are 
> there some ways to run geographically distributed DB cluster?

As far as you can partition and sync replicate, you can do that with
Pg.

-- 
If it takes a bloodbath, lets get it over with.  No more appeasement.
		-- Ronald Reagan


From ksims.ml at gmail.com  Mon Nov  5 19:16:56 2012
From: ksims.ml at gmail.com (KPS Maillinglist)
Date: Mon, 5 Nov 2012 10:16:56 -0600
Subject: [Freeswitch-users] ODBC with mod_voicemail and storage-dir
In-Reply-To: <98C09FCA-749A-4F71-BBFC-4ACEB0A1DA75@jerris.com>
References: <CAOjvyPkhnDPAfKTV8wP+Dt_AiaLird5FsuH4UYLXsT+qrp33XA@mail.gmail.com>
	<98C09FCA-749A-4F71-BBFC-4ACEB0A1DA75@jerris.com>
Message-ID: <CAOjvyPmu+PfD4=bmP6NFd5dg5Fr4M-mVRySFdPcuahg-g3b9fA@mail.gmail.com>

yes i am. I am calling the dialplan as follows

to send a caller to voicemail
<condition field="destination_number" expression="^EXT-(\d{4})$">
   <action application="bridge" data="user/$1 at demo1.cleardialin.com"/>
   <action application="answer"/>
   <action application="sleep" data="1000"/>
   <action application="hash"
data="insert/${domain_name}-retrieve_vm/global/${uuid}"/>
   <action application="bridge"
data="loopback/app=voicemail:DEMO1_CDI_TEST1 demo1.cleardialin.com $1"/>
</condition>

this is to retrieve voicemail
<condition field="destination_number"
expression="^vmain$|^\*7777$|^\*98$|^voicemail$">
<action application="voicemail" data="check DEMO1_CDI_TEST1
demo1.cleardialin.com"/>
</condition>



On Mon, Nov 5, 2012 at 7:31 AM, Michael Jerris <mike at jerris.com> wrote:

> How are you calling voicemail?  Are you specifying to use the profile
> listed below?
>
> On Nov 3, 2012, at 9:42 PM, KPS Maillinglist <ksims.ml at gmail.com> wrote:
>
> I cant seem to get mod_voicemail to store vm information in the database.
> It also is not using the storage-dir param to place the vm files. Please
> help.
>
>
> <profile name="DEMO1_CDI_TEST1">
> <settings>
> <param name="callback-context" value="DEMO1_CDI_TEST1"/>
> <param name="callback-dialplan" value="XML"/>
> <param name="choose-greeting-key" value="2"/>
> <param name="config-menu-key" value="5"/>
> <param name="delete-file-key" value="7"/>
> <param name="digit-timeout" value="10000"/>
> <param name="email-key" value="4"/>
> <param name="ff-key" value="6"/>
> <param name="file-extension" value="mp3"/>
> <param name="listen-file-key" value="1"/>
> <param name="main-menu-key" value="*"/>
> <param name="max-login-attempts" value="3"/>
> <param name="max-record-length" value="300"/>
> <param name="odbc-dsn" value="fs_regsrv_vm:fs_cs:staysynced"/>
> <param name="operator-extension" value="operator XML DEMO1_CDI_TEST1"/>
> <param name="operator-key" value="9"/>
> <param name="pause-key" value="0"/>
> <param name="play-new-messages-key" value="1"/>
> <param name="play-saved-messages-key" value="2"/>
> <param name="record-file-key" value="3"/>
> <param name="record-greeting-key" value="1"/>
> <param name="record-name-key" value="3"/>
> <param name="record-silence-hits" value="2"/>
> <param name="record-silence-threshold" value="200"/>
> <param name="restart-key" value="1"/>
> <param name="rew-key" value="4"/>
> <param name="save-file-key" value="9"/>
> <param name="storage-dir" value="/fs/cust/vm/DEMO1_CDI_TEST1"/>
> <param name="terminator-key" value="#"/>
> <param name="tone-spec" value="%(1000, 0, 640)"/>
> <param name="undelete-file-key" value="8"/>
> <param name="web-template-file" value="web-vm.tpl"/>
> <email>
> <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
> <param name="email-from" value="${voicemail_account}@${voicemail_domain}"
> />
> <param name="notify-template-file" value="notify-voicemail.tpl"/>
> <param name="template-file" value="voicemail.tpl"/>
> </email>
> </settings>
> </profile>
> _________________________________________________________________________
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eagle.antonio at gmail.com  Mon Nov  5 19:52:33 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 5 Nov 2012 16:52:33 +0000
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <475D1248-9FAD-48D0-9F92-7B6B684EBAF6@jerris.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<20121105133549.11b900ff@anubis.defcon1>
	<475D1248-9FAD-48D0-9F92-7B6B684EBAF6@jerris.com>
Message-ID: <CAF+3j-stnp9oY17K9aS2Xqv1p1W07OpH4HSS0N38VznoL1AQ9A@mail.gmail.com>

>From my point of view this is a not a discussing that my db is bigger than
yours ( sorry better ...).



2012/11/5 Michael Jerris <mike at jerris.com>

> Freeswitch creates all the tables and indexes for you.  There is no need
> for a GUI admin as there is not really anything to admin to set up
> freeswitch.  The only place this might not be true is if you are trying to
> setup replication, where mysql still has the edge in functionality.  I
> clearly am biased away from YourSQL, but in the end, I see no good reason
> to add it to FreeSWITCH. If someone feels strongly the other way, we would
> entertain patches to change this, under the guidelines set out earlier by
> Tony (and a few clarifications/additions).
>
> 1) We would need proof of some advantage over ODBC.  People often state
> that ODBC is slower, but I have yet to see evidence this is actually true
> to an extent that warrants the additional trouble.
> 2) A modular interface would need to be added if we add any more specific
> db interfaces.
> 3) Proper configure (and possibly runtime) check will need to be added to
> ensure we are using proper mysql libraries checking for transaction
> handling and thread safety
>
> Postgres was added to the core in order to address specific limitations in
> the current interface with postgres ODBC related to how it makes blocking
> calls to the database, not for performance reasons, and was done so by our
> team due to our own needs.  If someone feels strongly this needs to be
> added, has the skillset required to make these patches, and is willing to
> put in the time to do so, please talk to me off list and I am happy to
> discuss requirements in more detail.
>
> Mike
>
> On Nov 5, 2012, at 7:35 AM, Bzzz <lazyvirus at gmx.com> wrote:
>
> > On Mon, 5 Nov 2012 09:14:59 +0000
> > Antonio Teixeira <eagle.antonio at gmail.com> wrote:
> >
> >> The issue is not that simple , i consider PostgreSQL more "enterprise"
> is
> >> just my point of view.
> >
> > Hmm, I see the question from another point of view: how can people use
> > mysql for _any_ enterprise work.
> >
> >> One thing i love about postgres is the Transactions  + an extremely good
> >> engine , PostgreSQL is not made as easy as mysql you don't get fancy
> >> "phpmyadmin" but it does the work extremely well even in some corner
> cases.
> >
> > That isn't true: as a web application you have phppgadmin
> > (http://phppgadmin.sourceforge.net/doku.php?id=start), as a
> > standalone program you have pgadmin3 (http://www.pgadmin.org/).
> >
> >>
> >> There is also a more "oracle" version of PostgreSQL ,
> >> http://www.enterprisedb.com/
> >
> > In fact, Pg is seen as the open-source oracle; enterprisedb brings the
> > advantage of long term bug fixing & support for its chosen Pg versions :)
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eagle.antonio at gmail.com  Mon Nov  5 20:01:05 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 5 Nov 2012 17:01:05 +0000
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-stnp9oY17K9aS2Xqv1p1W07OpH4HSS0N38VznoL1AQ9A@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<20121105133549.11b900ff@anubis.defcon1>
	<475D1248-9FAD-48D0-9F92-7B6B684EBAF6@jerris.com>
	<CAF+3j-stnp9oY17K9aS2Xqv1p1W07OpH4HSS0N38VznoL1AQ9A@mail.gmail.com>
Message-ID: <CAF+3j-vtL3yXHwHHzjMNbx6C2QU_rabk7F56J0U47h6YaC_-SQ@mail.gmail.com>

>From my point of view this is a not a discussing that my db is bigger than
yours ( sorry better ...).
No one can ever say that a database is more secure , more stable and X , Y
, Z technology is advancing far faster than we can support/understand it.

Postgresql as by far outperformed mysql , from my work experience i have
saw raid car failure with the db holding up , for me i will always prefer a
DB  that has one unique engine geared towards stability , etc mysql with
its internal engines appears to be a hammer that will also work as a brush.

So for me a DB that i haven't touched for always 3 years , inserting close
to 1M rows per day ( cdr's) with a plain and simple vaccumm ( reindex ).





2012/11/5 Antonio Teixeira <eagle.antonio at gmail.com>

> From my point of view this is a not a discussing that my db is bigger than
> yours ( sorry better ...).
>
>
>
>
> 2012/11/5 Michael Jerris <mike at jerris.com>
>
>> Freeswitch creates all the tables and indexes for you.  There is no need
>> for a GUI admin as there is not really anything to admin to set up
>> freeswitch.  The only place this might not be true is if you are trying to
>> setup replication, where mysql still has the edge in functionality.  I
>> clearly am biased away from YourSQL, but in the end, I see no good reason
>> to add it to FreeSWITCH. If someone feels strongly the other way, we would
>> entertain patches to change this, under the guidelines set out earlier by
>> Tony (and a few clarifications/additions).
>>
>> 1) We would need proof of some advantage over ODBC.  People often state
>> that ODBC is slower, but I have yet to see evidence this is actually true
>> to an extent that warrants the additional trouble.
>> 2) A modular interface would need to be added if we add any more specific
>> db interfaces.
>> 3) Proper configure (and possibly runtime) check will need to be added to
>> ensure we are using proper mysql libraries checking for transaction
>> handling and thread safety
>>
>> Postgres was added to the core in order to address specific limitations
>> in the current interface with postgres ODBC related to how it makes
>> blocking calls to the database, not for performance reasons, and was done
>> so by our team due to our own needs.  If someone feels strongly this needs
>> to be added, has the skillset required to make these patches, and is
>> willing to put in the time to do so, please talk to me off list and I am
>> happy to discuss requirements in more detail.
>>
>> Mike
>>
>> On Nov 5, 2012, at 7:35 AM, Bzzz <lazyvirus at gmx.com> wrote:
>>
>> > On Mon, 5 Nov 2012 09:14:59 +0000
>> > Antonio Teixeira <eagle.antonio at gmail.com> wrote:
>> >
>> >> The issue is not that simple , i consider PostgreSQL more "enterprise"
>> is
>> >> just my point of view.
>> >
>> > Hmm, I see the question from another point of view: how can people use
>> > mysql for _any_ enterprise work.
>> >
>> >> One thing i love about postgres is the Transactions  + an extremely
>> good
>> >> engine , PostgreSQL is not made as easy as mysql you don't get fancy
>> >> "phpmyadmin" but it does the work extremely well even in some corner
>> cases.
>> >
>> > That isn't true: as a web application you have phppgadmin
>> > (http://phppgadmin.sourceforge.net/doku.php?id=start), as a
>> > standalone program you have pgadmin3 (http://www.pgadmin.org/).
>> >
>> >>
>> >> There is also a more "oracle" version of PostgreSQL ,
>> >> http://www.enterprisedb.com/
>> >
>> > In fact, Pg is seen as the open-source oracle; enterprisedb brings the
>> > advantage of long term bug fixing & support for its chosen Pg versions
>> :)
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From slickwilly2000 at gmx.de  Mon Nov  5 20:22:38 2012
From: slickwilly2000 at gmx.de (=?iso-8859-1?Q?=22Alex_M=FCller=22?=)
Date: Mon, 05 Nov 2012 18:22:38 +0100
Subject: [Freeswitch-users] Pickup
Message-ID: <20121105172238.137730@gmx.net>

Hi,

today, I implemented the pickup feature with the "intercept-application". So I am able to pick up an internal call that isn't answered yet. That works perfect!

The only problem is that I do not see the counterpart at the phone that picked up the call.


An example to illustrate this situation: Alice calls Bob, so Bob's phone is ringing. Charlie picks up the call, so now is bridged to Alice. Charlie does not see that he is connected to Alice (name/number).

Can anybody help how to get this information from the remaining leg after intercept?



Thanks in advance!


(Just for your information: I already know how so set these information (effective_callee_id_name/effective_callee_id_number) but do not know how to get this from the bleg)


From fdelawarde at wirelessmundi.com  Mon Nov  5 20:33:05 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 05 Nov 2012 18:33:05 +0100
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAKbxfG-HCVKksr63KPuwiNq+RxZkGWrAT0nBT3_+2Qiz3PYjXg@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<3E768A43-8D37-45F1-8DCE-5AD4C71E15A9@imtelecom.fr>
	<CAF+3j-sY78NnANw9j0d_gHvxp1ds+SDoP6y5+xUYsv_uOjSozQ@mail.gmail.com>
	<CAKbxfG-HCVKksr63KPuwiNq+RxZkGWrAT0nBT3_+2Qiz3PYjXg@mail.gmail.com>
Message-ID: <1352136785.6373.42.camel@luna.madrid.commsmundi.com>

Plus, when you say it outloud, PostgreSQL sounds way better than MySQL
(which sounds a bit childish).



On Mon, 2012-11-05 at 09:01 -0600, Anthony Minessale wrote:
> Its simpler than that. 
> 
> 
> I have used both db in my time as a developer and I have been mad at
> both for one reason or another.
> It all boils down to Mysql is not as stable and has several
> thread safety and memory leak issues when its used in a multi-threaded
> app like FreeSWITCH.  In fact that preference is irrelevant because
> technically the default database in FS is sqlite because the design
> goal of the project is for the software to run anywhere in a minimal
> capacity and be able to grow and scale with the user's needs.  So
> anyone can use any database they want.  However, when we get a bug
> report that shows Mysql crashing, that is not something we can do
> anything about.  Since we do have better luck with postgres, we
> allowed the inclusion of native api but that was a 3rd party
> contribution.
> 
> 
> So bottom line is we don't care what other people do, we just do what
> we like doing and share it.
> 
> 
> 
> 
>   
> 
> 
> On Mon, Nov 5, 2012 at 6:42 AM, Antonio Teixeira
> <eagle.antonio at gmail.com> wrote:
>         My second No No Besides Mysql ... PHP :) , but seems
>         interesting !.
>         Thanks 
>         
>         
>         
>         2012/11/5 Vallimamod ABDULLAH
>         <vallimamod.abdullah at imtelecom.fr>
>                 Hi,
>                 
>                 On Nov 5, 2012, at 10:14 AM, Antonio Teixeira
>                 <eagle.antonio at gmail.com> wrote:
>                 
>                 >  PostgreSQL is not made as easy as mysql you don't
>                 get fancy "phpmyadmin" but it does the work extremely
>                 well even in some corner cases.
>                 
>                 You may be interested by adminer
>                 (http://www.adminer.org): it's a very light, one php
>                 file "phpmyadmin" for mysql, postgres, etc.
>                 
>                 Cheers,
>                 Vallimamod 
>                 .
>                 
>                 
>                 
>                 _________________________________________________________________________
>                 Professional FreeSWITCH Consulting Services:
>                 consulting at freeswitch.org
>                 http://www.freeswitchsolutions.com
>                 
>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>                 Server
>                 
>                 
>                 Official FreeSWITCH Sites
>                 http://www.freeswitch.org
>                 http://wiki.freeswitch.org
>                 http://www.cluecon.com
>                 
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>                 
>         
>         
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Mon Nov  5 20:42:56 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Nov 2012 11:42:56 -0600
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CALNZuwmuiufpioV-ktnSDky0YrtZwwi1g_MT5mCws+z5f48GvA@mail.gmail.com>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
	<CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
	<CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>
	<CAKbxfG9OYOR9cfjSBmnow76d-0C8d6NgqVzuUPv_3CpWB_4mPQ@mail.gmail.com>
	<CALNZuwmuiufpioV-ktnSDky0YrtZwwi1g_MT5mCws+z5f48GvA@mail.gmail.com>
Message-ID: <CAKbxfG-VDXx8wbwN4EW8K4nABEWmcdiBhb+hUK6-bp-b=avauA@mail.gmail.com>

No but I can start one ....


ROADMAP

1) Get someone to make a roadmap and organize it for us.




On Mon, Nov 5, 2012 at 9:25 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Is there a public roadmap somewhere -- for your coding, and for the
> project as a whole (e.g. packaging, docs, books, etc) and who is doing each
> one?
>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From yiftah at choochee.com  Mon Nov  5 21:05:51 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Mon, 5 Nov 2012 10:05:51 -0800
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCBC8CC8.5EF65%krice@freeswitch.org>
References: <CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
	<CCBC8CC8.5EF65%krice@freeswitch.org>
Message-ID: <CAFVUaV9xk=m70Se1H13ap3=TPOXG+h7x+ZW8DjBf45tedjf2Yw@mail.gmail.com>

OK thanks

On Sun, Nov 4, 2012 at 6:21 PM, Ken Rice <krice at freeswitch.org> wrote:

>  You are more then welcome to keep the number pointed to the Conference
> server... Don?t expect a lot of action on things like that during the
> weekend... Most of us that work for the project do so full time and do take
> time for the family...
>
> As far as OpenSIPs, they support PostgreSQL via the db_postgres engine...
>
>
>
> On 11/4/12 9:37 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>
> I do not know Postgres (and maybe it is a good chance to try it) but
> probably there is a penalty with Postgres as well
> It is just that OpenSIPs goes very well with FreeSWITCH (at least for us)
> and OpenSIPs use mysql as its primary db
> Therefore using one DB for both makes a lot of sense for us
> BTW, I did not get an answer for posting the Germany number that we added
> for the conference, do you want it or should I return it to our poll?
>
> Thanks,
> Yiftach.
>
>
> On Sun, Nov 4, 2012 at 7:07 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> Oh yeah, lets let 1 thread run at a time access the database... Mysql is
> the only one that shows this issue... So why should we cripple the DB
> interfaces for just mysql
>
>
>
> On 11/3/12 10:59 PM, "Yiftach Golan" <yiftah at choochee.com <
> http://yiftah at choochee.com> > wrote:
>
> Also for what ken wrote we used the non thread safe version and protected
> the call to the database with mutex
>
>
> On Sat, Nov 3, 2012 at 1:25 PM, curriegrad2004 <curriegrad2004 at gmail.com <
> http://curriegrad2004 at gmail.com> > wrote:
>
> ODBC's there for this reason. Why invent more wheels when we already
> have that wheel sitting around?
>
> On Sat, Nov 3, 2012 at 11:36 AM, Anthony Minessale
> <anthony.minessale at gmail.com <http://anthony.minessale at gmail.com> > wrote:
> > Plus as a friend of mine used to say: "It's not MY sql its YOUR sql!"
> >
> > It is a fact that we have an incredible amount of bugs come up that end
> up
> > being mysql+odbc memory errors and thread safety issues.
> > I would be concerned with more blame falling on us of we had the code
> deeper
> > into our core.
> >
> > I would not condone adding any more db support unless we we architected
> it
> > to use db modules so we could still properly blame it on mysql when it
> was
> > not working.
> >
> >
> >
> >
> > On Sat, Nov 3, 2012 at 12:32 PM, Ken Rice <krice at freeswitch.org <
> http://krice at freeswitch.org> > wrote:
> >>
> >> While this may solve the issue... This does not address a number of
> other
> >> issues
> >>
> >> A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...
> >>
> >> B) There is that whole threadsafe vs non-threadsafe mysql client lib
> >> issue,
> >> so we would have to come up with a way in the build system to detect
> which
> >> we have, and never use the non-thread safe one... (if we don't do this,
> >> the
> >> code would never be stable as we could never know for sure which one
> >> actually got detected and linked against)
> >>
> >>
> >>
> >>
> >>
> >> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org <
> http://gabe at gundy.org> > wrote:
> >>
> >> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
> >> > <curriegrad2004 at gmail.com <http://curriegrad2004 at gmail.com> > wrote:
> >> >> Quite impossible. The reason is because of the license that the MySQL
> >> >> libs are licensed under. iirc, they are GPL licensed and they are not
> >> >> compatible with the MPL that FreeSWITCH uses.
> >> >
> >> > I've got PostgreSQL, so I've got all I need :) But, it seems like this
> >> > legal issue is easily resolved:
> >> >
> >> > http://www.mysql.com/about/legal/licensing/foss-exception/
> >> >
> >> > Anyway, I'm not a lawyer or even a very thorough reader ;)
> >> >
> >> >
> >> > Best,
> >> > Gabe
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org <http://consulting at freeswitch.org>
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >> --
> >> Ken
> >> http://www.FreeSWITCH.org
> >> http://www.ClueCon.com
> >> http://www.OSTAG.org
> >> irc.freenode.net <http://irc.freenode.net>  <http://irc.freenode.net>
>  #freeswitch
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org <http://consulting at freeswitch.org>
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at 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 <http://anthony_minessale at hotmail.com>
>  <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>>
>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <
> http://anthony.minessale at gmail.com>  <
> mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%3Aanthony.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 <http://888 at conference.freeswitch.org>
>  <mailto:sip%3A888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>>
>
> > googletalk:conf+888 at conference.freeswitch.org <
> http://conf+888 at conference.freeswitch.org>  <
> mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>>
>
> > pstn:+19193869900 <tel:%2B19193869900>  <tel:%2B19193869900>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <http://consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vbvbrj at gmail.com  Mon Nov  5 21:20:33 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 05 Nov 2012 20:20:33 +0200
Subject: [Freeswitch-users] Remote pickup of on-hold extensions
In-Reply-To: <CAMbW9o+3NsdJJwUN4B6oiG5BrLTy=7ieq7u1HQed=WzBF+Rv6g@mail.gmail.com>
References: <bc4d4e4f822530c463361a4f72774f6a.squirrel@icefire.qza.net.au>
	<CAMbW9oKH0j_oRuMjaDOGb6A=eBMf79XZMezSBeggWTdmYJDrDg@mail.gmail.com>
	<e62dfe57ead648fc2e67de37fc38726f.squirrel@icefire.qza.net.au>
	<CAMbW9o+3NsdJJwUN4B6oiG5BrLTy=7ieq7u1HQed=WzBF+Rv6g@mail.gmail.com>
Message-ID: <50980371.2060002@gmail.com>

On 05.11.2012 17:57, Andrew Cassidy wrote:
> You can one-touch park using the valet_park application, and you set up
> your BLF keys to monitor/park/unpark calls on the parking extesions.
>
> For example, for a 10 slot (700 to 709) lot you'd have:
>
> <extension name="parkinglot">
>   <condition field="destination_number" expression="^(park+)?(70\d)$">
>    <action application="valet_park" data="lot $1" />
>   </condition>
> </extension>
>
> Then set you BLF keys to monitor/dial park+700 to park+709 and if your
> handsets support it properly and are configured correctly, just pressing
> one of the free line keys should park the call in that slot, and
> illuminate the LED.
>
> You can also dial 700-709 directly, if you wanted to.
>
> Is that a bit closer?

Recently I've thought and read about this. The callcenter which I 
implement will have some other users which want (as on old system 
phones) to see on the phone a LED blinking when other line is ringing. 
But I understood that LED call presence is not fully implemented in FS. 
Is it so? Or it depends on the phone?

-- 
Mimiko desu.


From oseslija at gmail.com  Mon Nov  5 21:24:58 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Nov 2012 19:24:58 +0100
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
In-Reply-To: <2f17f98c.442d.13acfc76aa3.Coremail.sparklezou@163.com>
References: <2f17f98c.442d.13acfc76aa3.Coremail.sparklezou@163.com>
Message-ID: <CAM21H-PU=gk1x+yD34cOcxmV+=PuLHNHJZ__bHiAuO8H16z4XA@mail.gmail.com>

That feature is phone dependent. Afaik it's working on snom, but not on
Linksys/Cisco.
FS already sets everything needed.

On Mon, Nov 5, 2012 at 9:54 AM, sparklezou <sparklezou at 163.com> wrote:

> **
> **
> Hi Sir/Madam,
>
> I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept
>
> And also implement it on FS.
>
> Here I want to know, does FS could implement such features? I know such
> features are working on some digital phone system.
>
> 1. "KAKA" & "GAGA" are in the same "Dial Groups".
> 2. When someone inside/outside call "KAKA", there will be visible sentens
> on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name),
> "12345678 call KAKA" (outside display the number).
>
> Is it posible to implement?
>
> Thanks in advance!
>
> 2012-11-05
> ------------------------------
>  sparklezou
> **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From darcy at Vex.Net  Mon Nov  5 21:37:21 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Mon, 5 Nov 2012 13:37:21 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CAKbxfG-VDXx8wbwN4EW8K4nABEWmcdiBhb+hUK6-bp-b=avauA@mail.gmail.com>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
	<CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
	<CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>
	<CAKbxfG9OYOR9cfjSBmnow76d-0C8d6NgqVzuUPv_3CpWB_4mPQ@mail.gmail.com>
	<CALNZuwmuiufpioV-ktnSDky0YrtZwwi1g_MT5mCws+z5f48GvA@mail.gmail.com>
	<CAKbxfG-VDXx8wbwN4EW8K4nABEWmcdiBhb+hUK6-bp-b=avauA@mail.gmail.com>
Message-ID: <20121105133721.01da8d23@dilbert>

On Mon, 5 Nov 2012 11:42:56 -0600
Anthony Minessale <anthony.minessale at gmail.com> wrote:
> ROADMAP
> 
> 1) Get someone to make a roadmap and organize it for us.

http://jira.freeswitch.org/browse/FS#selectedTab=com.atlassian.jira.plugin.system.project%3Aroadmap-panel

That's 1a.  Now 1b is to organize it.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From anthony.minessale at gmail.com  Mon Nov  5 22:10:02 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Nov 2012 13:10:02 -0600
Subject: [Freeswitch-users] [OffTopic] Why PostgreSQL better then MySQL?
In-Reply-To: <CAF+3j-stnp9oY17K9aS2Xqv1p1W07OpH4HSS0N38VznoL1AQ9A@mail.gmail.com>
References: <50975C7B.3090807@gmail.com>
	<CACyzUwK-cSydAKDQFtAnaTMyAKEfc-4Mygj698SHT0R7A=N+SQ@mail.gmail.com>
	<CAF+3j-s5Hq=ypmvps-gji_zZApXtEupVCEi5_Ls81mbMHhwvNQ@mail.gmail.com>
	<20121105133549.11b900ff@anubis.defcon1>
	<475D1248-9FAD-48D0-9F92-7B6B684EBAF6@jerris.com>
	<CAF+3j-stnp9oY17K9aS2Xqv1p1W07OpH4HSS0N38VznoL1AQ9A@mail.gmail.com>
Message-ID: <CAKbxfG_GC2-vdMh_1TX=oVe-NVEQdtw--OF95-bBFk0EjJdyHA@mail.gmail.com>

Like I mentioned... There is enough typing here to have fixed several wiki
pages.  So everyone should work on that or discuss something that actually
matters with the same conviction and we would make even more progress....



On Mon, Nov 5, 2012 at 10:52 AM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> >From my point of view this is a not a discussing that my db is bigger
> than yours ( sorry better ...).
>
>
>
>
> 2012/11/5 Michael Jerris <mike at jerris.com>
>
>> Freeswitch creates all the tables and indexes for you.  There is no need
>> for a GUI admin as there is not really anything to admin to set up
>> freeswitch.  The only place this might not be true is if you are trying to
>> setup replication, where mysql still has the edge in functionality.  I
>> clearly am biased away from YourSQL, but in the end, I see no good reason
>> to add it to FreeSWITCH. If someone feels strongly the other way, we would
>> entertain patches to change this, under the guidelines set out earlier by
>> Tony (and a few clarifications/additions).
>>
>> 1) We would need proof of some advantage over ODBC.  People often state
>> that ODBC is slower, but I have yet to see evidence this is actually true
>> to an extent that warrants the additional trouble.
>> 2) A modular interface would need to be added if we add any more specific
>> db interfaces.
>> 3) Proper configure (and possibly runtime) check will need to be added to
>> ensure we are using proper mysql libraries checking for transaction
>> handling and thread safety
>>
>> Postgres was added to the core in order to address specific limitations
>> in the current interface with postgres ODBC related to how it makes
>> blocking calls to the database, not for performance reasons, and was done
>> so by our team due to our own needs.  If someone feels strongly this needs
>> to be added, has the skillset required to make these patches, and is
>> willing to put in the time to do so, please talk to me off list and I am
>> happy to discuss requirements in more detail.
>>
>> Mike
>>
>> On Nov 5, 2012, at 7:35 AM, Bzzz <lazyvirus at gmx.com> wrote:
>>
>> > On Mon, 5 Nov 2012 09:14:59 +0000
>> > Antonio Teixeira <eagle.antonio at gmail.com> wrote:
>> >
>> >> The issue is not that simple , i consider PostgreSQL more "enterprise"
>> is
>> >> just my point of view.
>> >
>> > Hmm, I see the question from another point of view: how can people use
>> > mysql for _any_ enterprise work.
>> >
>> >> One thing i love about postgres is the Transactions  + an extremely
>> good
>> >> engine , PostgreSQL is not made as easy as mysql you don't get fancy
>> >> "phpmyadmin" but it does the work extremely well even in some corner
>> cases.
>> >
>> > That isn't true: as a web application you have phppgadmin
>> > (http://phppgadmin.sourceforge.net/doku.php?id=start), as a
>> > standalone program you have pgadmin3 (http://www.pgadmin.org/).
>> >
>> >>
>> >> There is also a more "oracle" version of PostgreSQL ,
>> >> http://www.enterprisedb.com/
>> >
>> > In fact, Pg is seen as the open-source oracle; enterprisedb brings the
>> > advantage of long term bug fixing & support for its chosen Pg versions
>> :)
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
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pstn:+19193869900
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From oseslija at gmail.com  Mon Nov  5 22:37:05 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Nov 2012 20:37:05 +0100
Subject: [Freeswitch-users] Pickup
In-Reply-To: <20121105172238.137730@gmx.net>
References: <20121105172238.137730@gmx.net>
Message-ID: <CAM21H-NP7oUyw_pqUggrSdC8y8CDjXTZ0wpVWp-DqbTOYy7XuA@mail.gmail.com>

What phones do you have?



On Mon, Nov 5, 2012 at 6:22 PM, "Alex M?ller" <slickwilly2000 at gmx.de> wrote:

> Hi,
>
> today, I implemented the pickup feature with the "intercept-application".
> So I am able to pick up an internal call that isn't answered yet. That
> works perfect!
>
> The only problem is that I do not see the counterpart at the phone that
> picked up the call.
>
>
> An example to illustrate this situation: Alice calls Bob, so Bob's phone
> is ringing. Charlie picks up the call, so now is bridged to Alice. Charlie
> does not see that he is connected to Alice (name/number).
>
> Can anybody help how to get this information from the remaining leg after
> intercept?
>
>
>
> Thanks in advance!
>
>
> (Just for your information: I already know how so set these information
> (effective_callee_id_name/effective_callee_id_number) but do not know how
> to get this from the bleg)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at kavun.ch  Mon Nov  5 23:01:23 2012
From: lists at kavun.ch (Emrah)
Date: Mon, 5 Nov 2012 15:01:23 -0500
Subject: [Freeswitch-users] A-leg hangup cause is blank
In-Reply-To: <51EC5321-C965-4553-9DE4-F84B1FC460DD@kavun.ch>
References: <51EC5321-C965-4553-9DE4-F84B1FC460DD@kavun.ch>
Message-ID: <6B9182CC-772E-4B56-80BB-A9A2DDC32425@kavun.ch>

Hi guys,

It makes sense that hangup_cause is blank because at the time the variable is set, the channel hasn't been hanged-up? 

What can be done to have this set post-hangup?

Thanks a bunch and all the best,
Emrah

On Nov 3, 2012, at 1:06 AM, Emrah <lists at kavun.ch> wrote:

> Hi all,
> 
> To be simple, I am trying to play around with hangup causes and see what I get in my environment. 
> I must have a pretty shy set up because FS never discloses any hangup cause? It's continuously blank.
> This is what I have before my bridge action:
> <action application="set" data="api_hangup_hook=system /root/hangup_cause.sh ${hangup_cause} ${hangup_cause_q850}" />
> 
> My shell script just stores the hangup_cause I get in a file.
> 
> Ideally, I am trying to achieve the following:
> 
> 1. User a calls user b; user a hangs up before user b picks up; I get a hangup_cause that is relevant and I can notify user b of a missed call.
> 2. User a calls user b; call goes to Voicemail; user a hangs up without recording a message; I get a hangup_cause that is relevant and I can notify user b of a missed call.
> 
> And so on and so forth.
> 
> With Asterisk, I would get channel statuses like NOANSWER, CHANUNAVAIL, CANCEL? and could act upon them. 
> How do I get my FS to talk to me and give me some hangup_cause for my a-leg?
> 
> Thanks a bunch for all your help,
> Emrah



From vbvbrj at gmail.com  Mon Nov  5 23:32:54 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 05 Nov 2012 22:32:54 +0200
Subject: [Freeswitch-users] Strange problem with phones.
Message-ID: <50982276.50202@gmail.com>

As I don't know what this problem is related to: FS or Phone, I will 
describe here.

Testing mod_callcenter for real response to calls. There are about 7 
VoIP phones from D-Link (DPH-150S/E/F1 and DPH-150S/F2A). Agents log in 
and out when they need. All is working but suddenly one phone 
(DPH-150S/E/F1) started to reject calls as it was from log (see log at 
the end). There is a lot of "entering state [terminated][480]" and 
"[CS_CONSUME_MEDIA] [NO_USER_RESPONSE]". The phone does not ring, but 
instead its missing call value on display is rising quickly, so is the 
missing call count in the callcenter's DB for that agent. The other 
phones where receiving calls normal.

I looked on the management page of that phone and it was in state 
"Registered" to the FS. I tried to change the password to some erradic 
to unregister and set password back to register, and the problem 
persisted. I then reset phone to its factory defaults and restored 
settings from a backup file for this particular phone and account. And 
it get back to receive calls normally.

The same happened later with other phone (DPH-150S/F2A). In matter of 
seconds, the missing calls value raised to hundreds. So I asked to 
disconnect that phone from power outlet and connect another free phone. 
The logging of "[CS_CONSUME_MEDIA] [NO_USER_RESPONSE]" stopped for that 
phone.

So I don't know, this is a bug in FS or something with D-Link phone. 
This behavior must happen only when the agent presses "Reject" button on 
the phone, but its like the phone presses immediately this button for 
itself.

Did someone encountered such problem with such phone or other phones?

Log:
2012-11-05 20:38:41.208702 [DEBUG] mod_callcenter.c:1049 Updated Agent 
624 at pbx01.domain.md set state = Receiving
2012-11-05 20:38:41.208702 [DEBUG] switch_ivr_originate.c:2013 Parsing 
global variables
2012-11-05 20:38:41.208702 [DEBUG] switch_ivr_originate.c:2433 Parsing 
session specific variables
2012-11-05 20:38:41.208702 [DEBUG] switch_event.c:1570 Parsing variable 
[call_timeout]=[10]
2012-11-05 20:38:41.208702 [DEBUG] switch_ivr_originate.c:2013 Parsing 
global variables
2012-11-05 20:38:41.208702 [DEBUG] switch_event.c:1570 Parsing variable 
[sip_invite_domain]=[pbx01.domain.md]
2012-11-05 20:38:41.208702 [DEBUG] switch_event.c:1570 Parsing variable 
[presence_id]=[624 at pbx01.domain.md]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [NOTICE] 
switch_channel.c:951 New Channel 
sofia/internal/sip:624 at 132.101.16.24:5060 
[ebbcda11-e628-4fb4-a8c5-653b9c35f54e]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
mod_sofia.c:4880 (sofia/internal/sip:624 at 132.101.16.24:5060) State 
Change CS_NEW -> CS_INIT
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_session.c:490 sofia/internal/sip:624 at 132.101.16.24:5060 set 
UUID=ebbcda11-e628-4fb4-a8c5-653b9c35f54e
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_INIT
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:437 
(sofia/internal/sip:624 at 132.101.16.24:5060) State INIT
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
mod_sofia.c:86 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA INIT
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
sofia_glue.c:2637 Local SDP:
ebbcda11-e628-4fb4-a8c5-653b9c35f54e v=0
ebbcda11-e628-4fb4-a8c5-653b9c35f54e o=FreeSWITCH 1352122361 1352122362 
IN IP4 132.101.0.58
ebbcda11-e628-4fb4-a8c5-653b9c35f54e s=FreeSWITCH
ebbcda11-e628-4fb4-a8c5-653b9c35f54e c=IN IP4 132.101.0.58
ebbcda11-e628-4fb4-a8c5-653b9c35f54e t=0 0
ebbcda11-e628-4fb4-a8c5-653b9c35f54e m=audio 18360 RTP/AVP 98 9 0 8 3 101 13
ebbcda11-e628-4fb4-a8c5-653b9c35f54e a=rtpmap:98 SPEEX/32000
ebbcda11-e628-4fb4-a8c5-653b9c35f54e a=rtpmap:101 telephone-event/8000
ebbcda11-e628-4fb4-a8c5-653b9c35f54e a=fmtp:101 0-16
ebbcda11-e628-4fb4-a8c5-653b9c35f54e a=ptime:20
ebbcda11-e628-4fb4-a8c5-653b9c35f54e a=sendrecv
ebbcda11-e628-4fb4-a8c5-653b9c35f54e
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
mod_sofia.c:126 (sofia/internal/sip:624 at 132.101.16.24:5060) State Change 
CS_INIT -> CS_ROUTING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:437 
(sofia/internal/sip:624 at 132.101.16.24:5060) State INIT going to sleep
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_ROUTING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_channel.c:1964 (sofia/internal/sip:624 at 132.101.16.24:5060) 
Callstate Change DOWN -> RINGING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:446 
(sofia/internal/sip:624 at 132.101.16.24:5060) State ROUTING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
mod_sofia.c:149 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA ROUTING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_ivr_originate.c:67 (sofia/internal/sip:624 at 132.101.16.24:5060) 
State Change CS_ROUTING -> CS_CONSUME_MEDIA
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:446 
(sofia/internal/sip:624 at 132.101.16.24:5060) State ROUTING going to sleep
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change 
CS_CONSUME_MEDIA
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:465 
(sofia/internal/sip:624 at 132.101.16.24:5060) State CONSUME_MEDIA
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_state_machine.c:465 
(sofia/internal/sip:624 at 132.101.16.24:5060) State CONSUME_MEDIA going to 
sleep
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.208702 [DEBUG] 
sofia.c:6308 Channel sofia/internal/sip:624 at 132.101.16.24:5060 entering 
state [calling][0]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
sofia.c:6308 Channel sofia/internal/sip:624 at 132.101.16.24:5060 entering 
state [terminated][480]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
switch_channel.c:2950 (sofia/internal/sip:624 at 132.101.16.24:5060) 
Callstate Change RINGING -> HANGUP
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [NOTICE] 
sofia.c:7108 Hangup sofia/internal/sip:624 at 132.101.16.24:5060 
[CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
switch_channel.c:2973 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [KILL]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_HANGUP
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
switch_core_state_machine.c:638 
(sofia/internal/sip:624 at 132.101.16.24:5060) State HANGUP
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.248617 [DEBUG] 
mod_sofia.c:483 Channel sofia/internal/sip:624 at 132.101.16.24:5060 
hanging up, cause: NO_USER_RESPONSE
2012-11-05 20:38:41.248617 [DEBUG] switch_ivr_originate.c:3516 Originate 
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
2012-11-05 20:38:41.248617 [NOTICE] switch_ivr_originate.c:2599 Cannot 
create outgoing channel of type [user] cause: [NO_USER_RESPONSE]
2012-11-05 20:38:41.248617 [DEBUG] switch_ivr_originate.c:3516 Originate 
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.248617 [DEBUG] 
mod_callcenter.c:1728 Agent 624 at pbx01.domain.md Origination Canceled : 
NO_USER_RESPONSE
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:48 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard HANGUP, cause: NO_USER_RESPONSE
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:638 
(sofia/internal/sip:624 at 132.101.16.24:5060) State HANGUP going to sleep
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:429 
(sofia/internal/sip:624 at 132.101.16.24:5060) State Change CS_HANGUP -> 
CS_REPORTING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change 
CS_REPORTING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:720 
(sofia/internal/sip:624 at 132.101.16.24:5060) State REPORTING
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:92 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard REPORTING, cause: NO_USER_RESPONSE
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:720 
(sofia/internal/sip:624 at 132.101.16.24:5060) State REPORTING going to sleep
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:423 
(sofia/internal/sip:624 at 132.101.16.24:5060) State Change CS_REPORTING -> 
CS_DESTROY
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_session.c:1415 Session 5700 
(sofia/internal/sip:624 at 132.101.16.24:5060) Locked, Waiting on external 
entities
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [NOTICE] 
switch_core_session.c:1433 Session 5700 
(sofia/internal/sip:624 at 132.101.16.24:5060) Ended
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [NOTICE] 
switch_core_session.c:1437 Close Channel 
sofia/internal/sip:624 at 132.101.16.24:5060 [CS_DESTROY]
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:527 
(sofia/internal/sip:624 at 132.101.16.24:5060) Callstate Change HANGUP -> DOWN
2012-11-05 20:38:41.258623 [DEBUG] mod_callcenter.c:1049 Updated Agent 
624 at pbx01.domain.md set state = Waiting
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:530 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_DESTROY
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:540 
(sofia/internal/sip:624 at 132.101.16.24:5060) State DESTROY
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
mod_sofia.c:376 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA DESTROY
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:99 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard DESTROY
ebbcda11-e628-4fb4-a8c5-653b9c35f54e 2012-11-05 20:38:41.258623 [DEBUG] 
switch_core_state_machine.c:540 
(sofia/internal/sip:624 at 132.101.16.24:5060) State DESTROY going to sleep
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.298620 [DEBUG] 
switch_core_session.c:1042 Send signal 
sofia/orange/022256999 at 192.168.0.5:5061 [BREAK]
2012-11-05 20:38:41.308625 [DEBUG] mod_callcenter.c:1049 Updated Agent 
624 at pbx01.domain.md set state = Receiving
2012-11-05 20:38:41.308625 [DEBUG] switch_ivr_originate.c:2013 Parsing 
global variables
2012-11-05 20:38:41.308625 [DEBUG] switch_ivr_originate.c:2433 Parsing 
session specific variables
2012-11-05 20:38:41.308625 [DEBUG] switch_event.c:1570 Parsing variable 
[call_timeout]=[10]
2012-11-05 20:38:41.308625 [DEBUG] switch_ivr_originate.c:2013 Parsing 
global variables
2012-11-05 20:38:41.308625 [DEBUG] switch_event.c:1570 Parsing variable 
[sip_invite_domain]=[pbx01.domain.md]
2012-11-05 20:38:41.308625 [DEBUG] switch_event.c:1570 Parsing variable 
[presence_id]=[624 at pbx01.domain.md]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [NOTICE] 
switch_channel.c:951 New Channel 
sofia/internal/sip:624 at 132.101.16.24:5060 
[b6eb501f-10c0-4138-ac95-51bc5abc572f]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
mod_sofia.c:4880 (sofia/internal/sip:624 at 132.101.16.24:5060) State 
Change CS_NEW -> CS_INIT
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_session.c:490 sofia/internal/sip:624 at 132.101.16.24:5060 set 
UUID=b6eb501f-10c0-4138-ac95-51bc5abc572f
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_INIT
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:437 
(sofia/internal/sip:624 at 132.101.16.24:5060) State INIT
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
mod_sofia.c:86 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA INIT
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
sofia_glue.c:2637 Local SDP:
b6eb501f-10c0-4138-ac95-51bc5abc572f v=0
b6eb501f-10c0-4138-ac95-51bc5abc572f o=FreeSWITCH 1352117743 1352117744 
IN IP4 132.101.0.58
b6eb501f-10c0-4138-ac95-51bc5abc572f s=FreeSWITCH
b6eb501f-10c0-4138-ac95-51bc5abc572f c=IN IP4 132.101.0.58
b6eb501f-10c0-4138-ac95-51bc5abc572f t=0 0
b6eb501f-10c0-4138-ac95-51bc5abc572f m=audio 22978 RTP/AVP 98 9 0 8 3 101 13
b6eb501f-10c0-4138-ac95-51bc5abc572f a=rtpmap:98 SPEEX/32000
b6eb501f-10c0-4138-ac95-51bc5abc572f a=rtpmap:101 telephone-event/8000
b6eb501f-10c0-4138-ac95-51bc5abc572f a=fmtp:101 0-16
b6eb501f-10c0-4138-ac95-51bc5abc572f a=ptime:20
b6eb501f-10c0-4138-ac95-51bc5abc572f a=sendrecv
b6eb501f-10c0-4138-ac95-51bc5abc572f
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
mod_sofia.c:126 (sofia/internal/sip:624 at 132.101.16.24:5060) State Change 
CS_INIT -> CS_ROUTING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:437 
(sofia/internal/sip:624 at 132.101.16.24:5060) State INIT going to sleep
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_ROUTING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_channel.c:1964 (sofia/internal/sip:624 at 132.101.16.24:5060) 
Callstate Change DOWN -> RINGING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:446 
(sofia/internal/sip:624 at 132.101.16.24:5060) State ROUTING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
mod_sofia.c:149 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA ROUTING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_ivr_originate.c:67 (sofia/internal/sip:624 at 132.101.16.24:5060) 
State Change CS_ROUTING -> CS_CONSUME_MEDIA
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:446 
(sofia/internal/sip:624 at 132.101.16.24:5060) State ROUTING going to sleep
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change 
CS_CONSUME_MEDIA
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:465 
(sofia/internal/sip:624 at 132.101.16.24:5060) State CONSUME_MEDIA
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_state_machine.c:465 
(sofia/internal/sip:624 at 132.101.16.24:5060) State CONSUME_MEDIA going to 
sleep
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.308625 [DEBUG] 
sofia.c:6308 Channel sofia/internal/sip:624 at 132.101.16.24:5060 entering 
state [calling][0]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.348625 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
sofia.c:6308 Channel sofia/internal/sip:624 at 132.101.16.24:5060 entering 
state [terminated][480]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_channel.c:2950 (sofia/internal/sip:624 at 132.101.16.24:5060) 
Callstate Change RINGING -> HANGUP
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [NOTICE] 
sofia.c:7108 Hangup sofia/internal/sip:624 at 132.101.16.24:5060 
[CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_channel.c:2973 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [KILL]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_HANGUP
2012-11-05 20:38:41.358620 [DEBUG] switch_ivr_originate.c:3516 Originate 
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
2012-11-05 20:38:41.358620 [NOTICE] switch_ivr_originate.c:2599 Cannot 
create outgoing channel of type [user] cause: [NO_USER_RESPONSE]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:638 
(sofia/internal/sip:624 at 132.101.16.24:5060) State HANGUP
2012-11-05 20:38:41.358620 [DEBUG] switch_ivr_originate.c:3516 Originate 
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
mod_sofia.c:483 Channel sofia/internal/sip:624 at 132.101.16.24:5060 
hanging up, cause: NO_USER_RESPONSE
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.358620 [DEBUG] 
mod_callcenter.c:1728 Agent 624 at pbx01.domain.md Origination Canceled : 
NO_USER_RESPONSE
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:48 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard HANGUP, cause: NO_USER_RESPONSE
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:638 
(sofia/internal/sip:624 at 132.101.16.24:5060) State HANGUP going to sleep
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:429 
(sofia/internal/sip:624 at 132.101.16.24:5060) State Change CS_HANGUP -> 
CS_REPORTING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change 
CS_REPORTING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:720 
(sofia/internal/sip:624 at 132.101.16.24:5060) State REPORTING
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:92 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard REPORTING, cause: NO_USER_RESPONSE
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:720 
(sofia/internal/sip:624 at 132.101.16.24:5060) State REPORTING going to sleep
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:423 
(sofia/internal/sip:624 at 132.101.16.24:5060) State Change CS_REPORTING -> 
CS_DESTROY
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_session.c:1415 Session 5701 
(sofia/internal/sip:624 at 132.101.16.24:5060) Locked, Waiting on external 
entities
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [NOTICE] 
switch_core_session.c:1433 Session 5701 
(sofia/internal/sip:624 at 132.101.16.24:5060) Ended
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [NOTICE] 
switch_core_session.c:1437 Close Channel 
sofia/internal/sip:624 at 132.101.16.24:5060 [CS_DESTROY]
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:527 
(sofia/internal/sip:624 at 132.101.16.24:5060) Callstate Change HANGUP -> DOWN
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:530 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_DESTROY
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:540 
(sofia/internal/sip:624 at 132.101.16.24:5060) State DESTROY
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
mod_sofia.c:376 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA DESTROY
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:99 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard DESTROY
b6eb501f-10c0-4138-ac95-51bc5abc572f 2012-11-05 20:38:41.358620 [DEBUG] 
switch_core_state_machine.c:540 
(sofia/internal/sip:624 at 132.101.16.24:5060) State DESTROY going to sleep
2012-11-05 20:38:41.358620 [DEBUG] mod_callcenter.c:1049 Updated Agent 
624 at pbx01.domain.md set state = Waiting
2012-11-05 20:38:41.408622 [DEBUG] mod_callcenter.c:1049 Updated Agent 
624 at pbx01.domain.md set state = Receiving
2012-11-05 20:38:41.408622 [DEBUG] switch_ivr_originate.c:2013 Parsing 
global variables
2012-11-05 20:38:41.408622 [DEBUG] switch_ivr_originate.c:2433 Parsing 
session specific variables
2012-11-05 20:38:41.408622 [DEBUG] switch_event.c:1570 Parsing variable 
[call_timeout]=[10]
2012-11-05 20:38:41.408622 [DEBUG] switch_ivr_originate.c:2013 Parsing 
global variables
2012-11-05 20:38:41.408622 [DEBUG] switch_event.c:1570 Parsing variable 
[sip_invite_domain]=[pbx01.domain.md]
2012-11-05 20:38:41.408622 [DEBUG] switch_event.c:1570 Parsing variable 
[presence_id]=[624 at pbx01.domain.md]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [NOTICE] 
switch_channel.c:951 New Channel 
sofia/internal/sip:624 at 132.101.16.24:5060 
[b00f1fc8-c252-48d2-9065-8124d137d16c]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
mod_sofia.c:4880 (sofia/internal/sip:624 at 132.101.16.24:5060) State 
Change CS_NEW -> CS_INIT
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_session.c:490 sofia/internal/sip:624 at 132.101.16.24:5060 set 
UUID=b00f1fc8-c252-48d2-9065-8124d137d16c
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_INIT
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:437 
(sofia/internal/sip:624 at 132.101.16.24:5060) State INIT
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
mod_sofia.c:86 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA INIT
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
sofia_glue.c:2637 Local SDP:
b00f1fc8-c252-48d2-9065-8124d137d16c v=0
b00f1fc8-c252-48d2-9065-8124d137d16c o=FreeSWITCH 1352123977 1352123978 
IN IP4 132.101.0.58
b00f1fc8-c252-48d2-9065-8124d137d16c s=FreeSWITCH
b00f1fc8-c252-48d2-9065-8124d137d16c c=IN IP4 132.101.0.58
b00f1fc8-c252-48d2-9065-8124d137d16c t=0 0
b00f1fc8-c252-48d2-9065-8124d137d16c m=audio 16744 RTP/AVP 98 9 0 8 3 101 13
b00f1fc8-c252-48d2-9065-8124d137d16c a=rtpmap:98 SPEEX/32000
b00f1fc8-c252-48d2-9065-8124d137d16c a=rtpmap:101 telephone-event/8000
b00f1fc8-c252-48d2-9065-8124d137d16c a=fmtp:101 0-16
b00f1fc8-c252-48d2-9065-8124d137d16c a=ptime:20
b00f1fc8-c252-48d2-9065-8124d137d16c a=sendrecv
b00f1fc8-c252-48d2-9065-8124d137d16c
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
mod_sofia.c:126 (sofia/internal/sip:624 at 132.101.16.24:5060) State Change 
CS_INIT -> CS_ROUTING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:437 
(sofia/internal/sip:624 at 132.101.16.24:5060) State INIT going to sleep
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_ROUTING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_channel.c:1964 (sofia/internal/sip:624 at 132.101.16.24:5060) 
Callstate Change DOWN -> RINGING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:446 
(sofia/internal/sip:624 at 132.101.16.24:5060) State ROUTING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
mod_sofia.c:149 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA ROUTING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_ivr_originate.c:67 (sofia/internal/sip:624 at 132.101.16.24:5060) 
State Change CS_ROUTING -> CS_CONSUME_MEDIA
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:446 
(sofia/internal/sip:624 at 132.101.16.24:5060) State ROUTING going to sleep
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change 
CS_CONSUME_MEDIA
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:465 
(sofia/internal/sip:624 at 132.101.16.24:5060) State CONSUME_MEDIA
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
switch_core_state_machine.c:465 
(sofia/internal/sip:624 at 132.101.16.24:5060) State CONSUME_MEDIA going to 
sleep
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.408622 [DEBUG] 
sofia.c:6308 Channel sofia/internal/sip:624 at 132.101.16.24:5060 entering 
state [calling][0]
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.418617 [DEBUG] 
switch_ivr.c:605 sofia/orange/022256999 at 192.168.0.5:5061 Command Execute 
phrase(queue-position,1)
d7a61a90-f498-41be-9017-0c821157e1e1 EXECUTE 
sofia/orange/022256999 at 192.168.0.5:5061 phrase(queue-position,1)
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.418617 [DEBUG] 
mod_dptools.c:2507 Execute queue-position(1) lang
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.418617 [DEBUG] 
switch_ivr_play_say.c:67 No language specified - Using [ro]
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.428626 [DEBUG] 
switch_ivr_play_say.c:244 Handle play-file:[ro/queue-youarenext.wav] (ro:ro)
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.428626 [DEBUG] 
switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_session.c:905 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
sofia.c:6308 Channel sofia/internal/sip:624 at 132.101.16.24:5060 entering 
state [terminated][480]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_channel.c:2950 (sofia/internal/sip:624 at 132.101.16.24:5060) 
Callstate Change RINGING -> HANGUP
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [NOTICE] 
sofia.c:7108 Hangup sofia/internal/sip:624 at 132.101.16.24:5060 
[CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_channel.c:2973 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [KILL]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_HANGUP
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:638 
(sofia/internal/sip:624 at 132.101.16.24:5060) State HANGUP
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
mod_sofia.c:483 Channel sofia/internal/sip:624 at 132.101.16.24:5060 
hanging up, cause: NO_USER_RESPONSE
2012-11-05 20:38:41.458617 [DEBUG] switch_ivr_originate.c:3516 Originate 
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
2012-11-05 20:38:41.458617 [NOTICE] switch_ivr_originate.c:2599 Cannot 
create outgoing channel of type [user] cause: [NO_USER_RESPONSE]
2012-11-05 20:38:41.458617 [DEBUG] switch_ivr_originate.c:3516 Originate 
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
d7a61a90-f498-41be-9017-0c821157e1e1 2012-11-05 20:38:41.458617 [DEBUG] 
mod_callcenter.c:1728 Agent 624 at pbx01.domain.md Origination Canceled : 
NO_USER_RESPONSE
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:48 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard HANGUP, cause: NO_USER_RESPONSE
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:638 
(sofia/internal/sip:624 at 132.101.16.24:5060) State HANGUP going to sleep
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:429 
(sofia/internal/sip:624 at 132.101.16.24:5060) State Change CS_HANGUP -> 
CS_REPORTING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:398 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change 
CS_REPORTING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:720 
(sofia/internal/sip:624 at 132.101.16.24:5060) State REPORTING
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:92 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard REPORTING, cause: NO_USER_RESPONSE
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:720 
(sofia/internal/sip:624 at 132.101.16.24:5060) State REPORTING going to sleep
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:423 
(sofia/internal/sip:624 at 132.101.16.24:5060) State Change CS_REPORTING -> 
CS_DESTROY
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_session.c:1210 Send signal 
sofia/internal/sip:624 at 132.101.16.24:5060 [BREAK]
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_session.c:1415 Session 5702 
(sofia/internal/sip:624 at 132.101.16.24:5060) Locked, Waiting on external 
entities
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [NOTICE] 
switch_core_session.c:1433 Session 5702 
(sofia/internal/sip:624 at 132.101.16.24:5060) Ended
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [NOTICE] 
switch_core_session.c:1437 Close Channel 
sofia/internal/sip:624 at 132.101.16.24:5060 [CS_DESTROY]
2012-11-05 20:38:41.458617 [DEBUG] mod_callcenter.c:1049 Updated Agent 
624 at pbx01.domain.md set state = Waiting
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:527 
(sofia/internal/sip:624 at 132.101.16.24:5060) Callstate Change HANGUP -> DOWN
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:530 
(sofia/internal/sip:624 at 132.101.16.24:5060) Running State Change CS_DESTROY
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:540 
(sofia/internal/sip:624 at 132.101.16.24:5060) State DESTROY
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
mod_sofia.c:376 sofia/internal/sip:624 at 132.101.16.24:5060 SOFIA DESTROY
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:99 sofia/internal/sip:624 at 132.101.16.24:5060 
Standard DESTROY
b00f1fc8-c252-48d2-9065-8124d137d16c 2012-11-05 20:38:41.458617 [DEBUG] 
switch_core_state_machine.c:540 
(sofia/internal/sip:624 at 132.101.16.24:5060) State DESTROY going to sleep

-- 
Mimiko desu.


From spencer at 5ninesolutions.com  Tue Nov  6 00:32:44 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 5 Nov 2012 13:32:44 -0800
Subject: [Freeswitch-users] SIP to TDM t38 gateway
Message-ID: <5BA9AADC-BB63-4FF5-B1A6-1DEC1BE3E931@5ninesolutions.com>

Hello,
I'm trying to use Freeswitch as a SIP to TDM gateway.  I'd like to use t38_gateway to detect fax tones and send a ReINVITE to t38.

Have a very minimal config with one profile that simply relays to FreeTDM

My dialplan is:
<include>
  <context name="sip">

    <extension name="dial-trunk">
      <condition field="destination_number" expression="^(.*)$">
        <action application="set" data="fax_enable_t38=true"/>
        <action application="set" data="fax_enable_t38_request=true"/>
        <action application="set" data="execute_on_answer=t38_gateway self"/>
        <action application="bridge" data="freetdm/1/a/$1"/>
      </condition>
    </extension>

  </context>
</include>

The problem is a media bug is created on the channel but almost immediately destroyed so fax tones are never detected. 

See:
http://pastebin.freeswitch.org/20162

Thanks for any assistance,
Spencer





From lists at telefaks.de  Tue Nov  6 01:09:01 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 05 Nov 2012 23:09:01 +0100
Subject: [Freeswitch-users] Pass presence through Freeswitch B2BUA
Message-ID: <509838FD.5010205@telefaks.de>

Hello,

I am wondering, if anybody had found a way to pass through any presence
information through one Freeswitch to another.

What is the scenario I am thinking of?

  * I have a central Freeswitch server
  * at a remote location we have a minimum Freeswitch server which
    handles local registrations of local phones. This remote Freeswitch
    acts as a B2BUA via VPN to the central server (in order to eliminate
    NAT and Network issues and encrypt traffic to the remote site)
  * With the remote Freeswitch we can easily handle presence (BLF) at
    the remote site, as all phones are local.
  * In this B2BUA scenario, for every remote phone the remote freeswitch
    is registering the same phone number to the central Freeswitch,
    which handles some other phones, which are connected to the central
    Freeswitch directly.
  * What we want to achieve is, that a phone behind the remote
    Freeswitch can do BLF with another phone which is directly connected
    to the central Freeswitch

[Central Freeswitch] <=== registers 200 ===== [remote Freeswitch] <===
registers 200 =====[phone 200]
                     |<=== registers 201 =====[phone 201]

So in terms of this short drawing the phones 200 and 201 need to share
presence.
So the main question is: How can I pass the presence information of 201
from the central Freeswitch through the remote Freeswitch  to phone 200
and vice versa?

Is there a way t do that and how could one do that?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de


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From lpradovera at mojolingo.com  Tue Nov  6 01:13:50 2012
From: lpradovera at mojolingo.com (Luca Pradovera)
Date: Mon, 5 Nov 2012 23:13:50 +0100
Subject: [Freeswitch-users] NAT behavior
Message-ID: <079800ED-3488-4CB5-8FD8-0D286509FEE2@mojolingo.com>

Hello,
without going in too much detail, we have a working setup with Asterisk over a pretty complex network topology.
The setting that makes it work is nat=force_rport,comedia, and we have been trying to replicate that behavior on FreeSWITCH.
We basically want t force all RTP traffic to go through a set IP, no matter what SDP says or wants to do.

Thanks,

Luca
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From sparklezou at 163.com  Tue Nov  6 04:32:53 2012
From: sparklezou at 163.com (sparklezou)
Date: Tue, 6 Nov 2012 09:32:53 +0800
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
In-Reply-To: <CAM21H-PU=gk1x+yD34cOcxmV+=PuLHNHJZ__bHiAuO8H16z4XA@mail.gmail.com>
References: <CAM21H-PU=gk1x+yD34cOcxmV+=PuLHNHJZ__bHiAuO8H16z4XA@mail.gmail.com>
Message-ID: <711bb1b6.b10.13ad35a26bc.Coremail.sparklezou@163.com>

Hi Ognjen,

Could you please provide more info, which kind of phone support such feature?

Or it's tested on which kind of phones?

Thanks!

2012-11-06



sparklezou



????Ognjen Seslija
?????2012-11-06 02:24
???Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

That feature is phone dependent. Afaik it's working on snom, but not on Linksys/Cisco.
FS already sets everything needed.


On Mon, Nov 5, 2012 at 9:54 AM, sparklezou <sparklezou at 163.com> wrote:

Hi Sir/Madam,

I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept

And also implement it on FS.

Here I want to know, does FS could implement such features? I know such features are working on some digital phone system.

1. "KAKA" & "GAGA" are in the same "Dial Groups".
2. When someone inside/outside call "KAKA", there will be visible sentens on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name), "12345678 call KAKA" (outside display the number). 

Is it posible to implement?

Thanks in advance!

2012-11-05



sparklezou

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From sms at icefire.qza.net.au  Tue Nov  6 03:46:37 2012
From: sms at icefire.qza.net.au (sms at icefire.qza.net.au)
Date: Tue, 6 Nov 2012 11:46:37 +1100
Subject: [Freeswitch-users] Remote pickup of on-hold extensions
In-Reply-To: <CAMbW9o+3NsdJJwUN4B6oiG5BrLTy=7ieq7u1HQed=WzBF+Rv6g@mail.gmail.com>
References: <bc4d4e4f822530c463361a4f72774f6a.squirrel@icefire.qza.net.au>
	<CAMbW9oKH0j_oRuMjaDOGb6A=eBMf79XZMezSBeggWTdmYJDrDg@mail.gmail.com>
	<e62dfe57ead648fc2e67de37fc38726f.squirrel@icefire.qza.net.au>
	<CAMbW9o+3NsdJJwUN4B6oiG5BrLTy=7ieq7u1HQed=WzBF+Rv6g@mail.gmail.com>
Message-ID: <37f918cfdb8f9523f94611b33f841a0f.squirrel@icefire.qza.net.au>

Certainly is! Thanks, I'll give this a go. I'm going to have a shot at
configuring GSM_OPEN later this week so hopefully I'll have a success to
report back.

I've purchased the two freeswitch books and I'm slowly inching through
them. I'll try reproducing my first scenario when I've digested enough
information. The primary difference I see between key system behavior and
SAA/BLA/Park is that in a KS, the line always has the same address (line
1, line 2 etc) regardless of call state or ring assignments. The latter
configurations overcome the difficulty of managing large numbers of
calls/channels, at the expense of the former, this makes sense for large
systems. I'm liking freeswitch so far, it seems a whole lot more
configurable than Asterisk.


> You can one-touch park using the valet_park application, and you set up
> your BLF keys to monitor/park/unpark calls on the parking extesions.
>
> For example, for a 10 slot (700 to 709) lot you'd have:
>
> <extension name="parkinglot">
>  <condition field="destination_number" expression="^(park+)?(70\d)$">
>   <action application="valet_park" data="lot $1" />
>  </condition>
> </extension>
>
> Then set you BLF keys to monitor/dial park+700 to park+709 and if your
> handsets support it properly and are configured correctly, just pressing
> one of the free line keys should park the call in that slot, and
> illuminate
> the LED.
>
> You can also dial 700-709 directly, if you wanted to.
>
> Is that a bit closer?
>
> On 3 November 2012 10:46, <sms at icefire.qza.net.au> wrote:
>
>> I've thought about those options... the obstacle is mostly the users,
>> who
>> aren't the most trainable people in the world, and they're used to the
>> present key system which allows them to have pretty colored buttons that
>> they only press once. I dread the attempt to explain the concept of
>> transferring a call, and the question will inevitably come back: "Why
>> can't we have line keys like the last system?" Calls will likely get
>> lost
>> or dropped, and of course, the system gets blamed ;-)
>>
>> The other option, again if possible, is to program a soft key as a
>> one-touch "Park" key that announces the park extension number, which
>> could
>> also be assigned to a BLF, then parks the call. Then we have one touch
>> park, one touch retrieve.
>>
>> I have to sell this to them in such a way that it doesn't come across as
>> a
>> feature downgrade, or we may end up stuck with the present system until
>> it
>> dies and no inter-branch connectivity.
>>
>> I suppose it's the same old predicament... things would be so simple if
>> management just said yes :)
>>
>>
>>
>> > Personally I'd just use the Blind Transfer features on your handsets.
>> 1000
>> > picks up the call, presses transfer on the phone, then dials 1001 and
>> > hangs
>> > up. 1001 rings and is connected to the transferred call. Sure you'e
>> not
>> > got
>> > the BLF, but the idea works fine. You could use call parking, but that
>> has
>> > more steps to it.
>> >
>> > On 2 November 2012 04:16, <sms at icefire.qza.net.au> wrote:
>> >
>> >> Hi all,
>> >>
>> >> I have a scenario that I'm thinking of implementing, but not sure if
>> >> it's
>> >> possible, so here's the details:
>> >>
>> >> Extensions 200 to 250 are regular internal sip phones, contained in a
>> >> call
>> >> group (Ext 1000)
>> >>
>> >> Extensions 300 to 305 are SIP-GSM and SIP-PSTN gateways with SIP
>> client
>> >> firmware, namely those cheap GOIP GSM units on ebay, plus an SPA3000.
>> >> These are to be configured with the logon credentials of the
>> extension
>> >> and
>> >> behave as a typical SIP client.
>> >>
>> >> The dialplan will route incoming and outgoing calls via these
>> gateways
>> >> as
>> >> if they were trunks. They will restricted privileges to prevent toll
>> >> fraud. The phones will either be Yealink T28p or Grandstream GXP2124,
>> it
>> >> depends if GS can respond with a fix for their horrible AGC or not.
>> The
>> >> gateways will be subscribed to the phone's DSS keys as BLF's.
>> >>
>> >> Now, all this so far is reasonably straight forward. The next part is
>> >> the
>> >> tricky part (you might've guessed where this is going....)
>> >>
>> >> Let's say a call comes in on 300, routes to the call group and is
>> picked
>> >> up by 1000, who then puts the call on hold. The BLF key for 300 shows
>> it
>> >> as busy/on hold. 1000 then calls 1001 to take the call. 1001 then
>> >> presses
>> >> BLF 300, which causes the call to be transferred to them and
>> >> automatically
>> >> answered.
>> >>
>> >> So is there any way in freeswitch to replicate this behavior? I know
>> >> it's
>> >> possible to do SAA, but this gets tricky if I want to have time based
>> >> call
>> >> groups, or to share incoming lines between branches. The method above
>> >> allows the lines to be present on all phones, while still configuring
>> >> call
>> >> groups in the usual way and more or less replicating old key system
>> >> behavior.
>> >>
>> >> Cheers,
>> >> Francis
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> > Managing Director
>> >
>> >
>> > *T <info at cassidywebservices.co.uk> *03300 100 960
>> > *F<info at cassidywebservices.co.uk>
>> >  *03300 100 961
>> > *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> > *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960
> *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From sparklezou at 163.com  Tue Nov  6 05:00:06 2012
From: sparklezou at 163.com (sparklezou)
Date: Tue, 6 Nov 2012 10:00:06 +0800
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
In-Reply-To: <711bb1b6.b10.13ad35a26bc.Coremail.sparklezou@163.com>
References: <CAM21H-PU=gk1x+yD34cOcxmV+=PuLHNHJZ__bHiAuO8H16z4XA@mail.gmail.com><711bb1b6.b10.13ad35a26bc.Coremail.sparklezou@163.com>
Message-ID: <4fad929.1ca9.13ad373167c.Coremail.sparklezou@163.com>

Hi Ognjen,

There is a "Notify" message to inform all of the users in the Group? Correct?

How implment at FS side?

2012-11-06



sparklezou



????sparklezou
?????2012-11-06 09:33
???Re: Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>,"Ognjen Seslija"<oseslija at gmail.com>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Ognjen,

Could you please provide more info, which kind of phone support such feature?

Or it's tested on which kind of phones?

Thanks!

2012-11-06



sparklezou



????Ognjen Seslija
?????2012-11-06 02:24
???Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

That feature is phone dependent. Afaik it's working on snom, but not on Linksys/Cisco. 
FS already sets everything needed.


On Mon, Nov 5, 2012 at 9:54 AM, sparklezou <sparklezou at 163.com> wrote:

Hi Sir/Madam,

I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept

And also implement it on FS.

Here I want to know, does FS could implement such features? I know such features are working on some digital phone system.

1. "KAKA" & "GAGA" are in the same "Dial Groups".
2. When someone inside/outside call "KAKA", there will be visible sentens on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name), "12345678 call KAKA" (outside display the number). 

Is it posible to implement?

Thanks in advance!

2012-11-05



sparklezou

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From ocset at the800group.com  Tue Nov  6 06:30:56 2012
From: ocset at the800group.com (ocset)
Date: Tue, 06 Nov 2012 11:30:56 +0800
Subject: [Freeswitch-users] Play music while transferring new incoming
 call
In-Reply-To: <CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>
References: <5097D36A.7010803@the800group.com>
	<CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>
Message-ID: <50988470.1040708@the800group.com>

Thanks Avi

I have tried the following two dial plans but the phone just rings - no 
music.

<extension name="ext_8888">
         <condition field="destination_number" expression="^(8888)$">
                 <action application="set" 
data="transfer_ringback=local_stream://moh"/>
                 <action application="set" 
data="ringback=local_stream://moh"/>
                 <action application="transfer" data="1005 XML default"/>
         </condition>
  </extension>

<extension name="ext_8888">
         <condition field="destination_number" expression="^(8888)$">
                 <action application="set" 
data="transfer_ringback=local_stream://moh"/>
                 <action application="set" 
data="ringback=local_stream://moh"/>
                 <action application="bridge" 
data="sofia/internal/1005 at 192.168.0.150"/>
         </condition>
  </extension>

What am I missing? Should there be an "answer" directive in there 
somewhere also?

Thanks
O.

On 05/11/12 23:30, Avi Marcus wrote:
> You probably want ringback and/or transfer ringback. That's for 
> playing something while the call is waiting to be answered.
>
> See: 
> http://wiki.freeswitch.org/wiki/Time_of_Day_Routing#Example_for_office_open_09:00-16:00 there's 
> something set there, using:
>         <action application="set" data="ringback=local_stream://moh"/>
>         <action application="set" data="transfer_ringback=local_stream://moh"/>
> You're line isn't quite right -- "{ignore_early_media_=true_}", you 
> need to set the variable data.
>
> Playback is blocking -- it plays and then moves on. Transfer_ringback 
> sets what to do when the transfer occurs.
>
> I hope that helps and you can clean up the wiki when you've got it all 
> figured out.
>
> -Avi
>
>
>
> On Mon, Nov 5, 2012 at 4:55 PM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     I am trying to to configure FS to play music while transferring a new
>     incoming call. Like when you phone a business and you get the
>     "your call
>     in important to use, please wait while we transfer..." message.
>
>     I found this suggestion on the web but it does not work for me. I have
>     tested the two <action> command separately to make sure they work but
>     when they are put together, the call is never answered and no music is
>     played.
>
>
>     <include>
>       <extension name="ext_8888">
>              <condition field="destination_number" expression="^(8888)$">
>                      <action application="playback"
>     data="music/test.wav"/>
>                      <action application="bridge"
>     data="{ignore_early_media}sofia/internal/1005 at 192.168.0.150
>     <mailto:1005 at 192.168.0.150>"/>
>              </condition>
>       </extension>
>     </include>
>
>     ps. I cannot find the answer in either of the two FS books. Would have
>     expected this to be there, at least in the cookbook? All I can find is
>     examples of playing music and then hanging up. Happy to be
>     corrected if
>     I missed it:-)
>
>     All help greatly appreciated
>     Thanks
>     O.
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gabe at gundy.org  Tue Nov  6 09:11:43 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 5 Nov 2012 23:11:43 -0700
Subject: [Freeswitch-users] Play music while transferring new incoming
	call
In-Reply-To: <50988470.1040708@the800group.com>
References: <5097D36A.7010803@the800group.com>
	<CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>
	<50988470.1040708@the800group.com>
Message-ID: <CAMwZ-tLL+JaW-RPnzAXw1huJ4DSP6_tWmHMMwVX9yVxxMnQu4w@mail.gmail.com>

On Mon, Nov 5, 2012 at 8:30 PM, ocset <ocset at the800group.com> wrote:
> <action application="set" data="ringback=local_stream://moh"/>

Is your "local_stream://moh" known to work in other cases? Can you set
it up as MOH and try it without involving ring/transfer back?

Gabe


From gabe at gundy.org  Tue Nov  6 09:21:58 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 5 Nov 2012 23:21:58 -0700
Subject: [Freeswitch-users] How to handle SIP REFER message
In-Reply-To: <CAFvodxXhamdDWG=UdiUK3tb5Mj=pL=CRNAkAPTjf8EdQ9ON3qA@mail.gmail.com>
References: <CAFvodxXhamdDWG=UdiUK3tb5Mj=pL=CRNAkAPTjf8EdQ9ON3qA@mail.gmail.com>
Message-ID: <CAMwZ-tJYmEw19KnzyMwbiY5BepciVVdf16RT96GAaePbMr5FuQ@mail.gmail.com>

On Fri, Oct 26, 2012 at 6:36 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:
> I am trying to handle SIP REFER mesage from the dialplan but i
> could not ,actually our intention is when we get the SIP REFER message
> based on the refered-to header we need to bridge the call.I am not
> getting any idea to achieve this flow.

I'm not sure what you're trying to do, but you might be looking for
something that can should be done from a SIP proxy.

Anyway, the dialplan is only consulted when the call hits the routing
stage. If you have a call setup and you want FreeSWITCH to send the
refer, you can use this:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect


If you you're the one getting the REFER, FreeSWITCH should already do
the work of setting up the new channel.

Again, I don't know for sure what you're looking to do, but I hope that helps.


Best,
Gabe


From gabe at gundy.org  Tue Nov  6 09:23:31 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 5 Nov 2012 23:23:31 -0700
Subject: [Freeswitch-users] How to handle SIP REFER message
In-Reply-To: <CAMwZ-tJYmEw19KnzyMwbiY5BepciVVdf16RT96GAaePbMr5FuQ@mail.gmail.com>
References: <CAFvodxXhamdDWG=UdiUK3tb5Mj=pL=CRNAkAPTjf8EdQ9ON3qA@mail.gmail.com>
	<CAMwZ-tJYmEw19KnzyMwbiY5BepciVVdf16RT96GAaePbMr5FuQ@mail.gmail.com>
Message-ID: <CAMwZ-tJQgaX1mCnzYnKmmMjhOokFdZLYMOGCKRhdGbGpJ9B5ig@mail.gmail.com>

On Mon, Nov 5, 2012 at 11:21 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> you might be looking for something that can should be done from a SIP proxy.

Sorry, that should be:

"something that can/should be done from a SIP proxy"


Gabe


From gabe at gundy.org  Tue Nov  6 09:33:17 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 5 Nov 2012 23:33:17 -0700
Subject: [Freeswitch-users] A-leg hangup cause is blank
In-Reply-To: <6B9182CC-772E-4B56-80BB-A9A2DDC32425@kavun.ch>
References: <51EC5321-C965-4553-9DE4-F84B1FC460DD@kavun.ch>
	<6B9182CC-772E-4B56-80BB-A9A2DDC32425@kavun.ch>
Message-ID: <CAMwZ-tLJRodQ8Jbrm6uSrsM4OsKu7bWFAmYhqyWfFqDzMxB3AQ@mail.gmail.com>

On Mon, Nov 5, 2012 at 1:01 PM, Emrah <lists at kavun.ch> wrote:
> It makes sense that hangup_cause is blank because at the time the variable is set, the channel hasn't been hanged-up?

Depending on what you're doing, you may have to look to the
CS_REPORTING state. This is where you get your CDRs written out to
disk, HTTP, DB or whatever.

http://wiki.freeswitch.org/wiki/Channel_States

Is listening on the event socket an option? How about watching for
CDRs to be written to disk?


Good luck,
Gabe


From gabe at gundy.org  Tue Nov  6 09:35:27 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 5 Nov 2012 23:35:27 -0700
Subject: [Freeswitch-users] duplicate fax pages
In-Reply-To: <CAEVfcaYyvO-tz145EY=QmYFpOCO4U4p-1UFy8nNGT6m2bh2-Tg@mail.gmail.com>
References: <CAEVfcaYyvO-tz145EY=QmYFpOCO4U4p-1UFy8nNGT6m2bh2-Tg@mail.gmail.com>
Message-ID: <CAMwZ-tK4wfK0wEVhkvq=ou-29ogygyE6-DARBss0CTyn=etvtg@mail.gmail.com>

On Sun, Nov 4, 2012 at 10:59 AM, Babak Yakhchali
<babak.freeswitch at gmail.com> wrote:
> I'm using mod_spandsp to send fax to different targets. when destination is
> another freeswitch or elastix everything works fine. but when sending faxes
> to old fax machines if tiff file contains only 1 page, sending will succeed.
> but if tiff contains more than 1 page FS keeps sending first page multiple
> times till I issue hupall and it stops sending(I should mention that
> sometimes it sends first page 3 times second page 3 times and keeps eating
> remote fax machine paper roll).

Any updates on this?


Gabe


From gabe at gundy.org  Tue Nov  6 09:41:02 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 5 Nov 2012 23:41:02 -0700
Subject: [Freeswitch-users] Introducing me and Requesting help
In-Reply-To: <CAFhTFjiAaapyxqSsf+r6W0kyB7L2b_-_TOsorwFNE7C3JBND2Q@mail.gmail.com>
References: <CAFhTFjiAaapyxqSsf+r6W0kyB7L2b_-_TOsorwFNE7C3JBND2Q@mail.gmail.com>
Message-ID: <CAMwZ-tLw-pHqLM0y1OBwBudAHN9jUESRRQsQgMPL2cKPPAL5uQ@mail.gmail.com>

On Thu, Nov 1, 2012 at 2:53 PM, Jesus Ramon Rocha Velazquez
<jesus.rocha at overvoiplatam.com> wrote:
> I'm Just a new Member in this FS World. I love the wiki and all the
> documentation that all the community has made.

First off, welcome to the community. Hope you find what you need here!



> I'm in this momment Trying to connect to Remote FreeSwitch But, I have This
> Problem:
>
>
> SERVER A -> SERVER B [OK REGISTRATION]
>
> SERVER B -> SERVER C [OK REGISTRATION]
>
> SERVER A -> SERVER C [OK REGISTRATION]
>
> SERVER B -> SERVER A [ERROR]

Have you made any progress with this? BTW, I'd consider forgoing
REGISTER and just whitelist the IP addresses of the servers. That's a
more common approach.

Anyway, it you haven't figured this out, update us and we'll see what we can do.


Best,
Gabe


From gabe at gundy.org  Tue Nov  6 09:42:38 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 5 Nov 2012 23:42:38 -0700
Subject: [Freeswitch-users] libtermcap,
	libcurses or libncurses are required!
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AC777B5@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AC777B5@MAIL.millicorp.com>
Message-ID: <CAMwZ-tJehq+4nSvN5EnMdw_ZQB7Lcx31+vpQHJVx3Xjats9S5Q@mail.gmail.com>

On Tue, Oct 30, 2012 at 12:03 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:
> Any ideas?   I?ve done this install many many times on normal CentOS 5.8
> servers.

Have you updated your sources and tried again?

If you're still having issues, it might be a build bug. If that's the
case, report on Jira.


Best,
Gabe


From peter.olsson at visionutveckling.se  Tue Nov  6 09:54:55 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 6 Nov 2012 06:54:55 +0000
Subject: [Freeswitch-users] Play music while transferring new
	incoming	call
Message-ID: <1FFF97C269757C458224B7C895F35F151B5545@cantor.std.visionutv.se>

You will probably need "ignore_early_media=true" on the outgoing leg as well, or that media will be passed on instead of the music.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gabriel Gunderson
Skickat: den 6 november 2012 07:12
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Play music while transferring new incoming call

On Mon, Nov 5, 2012 at 8:30 PM, ocset <ocset at the800group.com> wrote:
> <action application="set" data="ringback=local_stream://moh"/>

Is your "local_stream://moh" known to work in other cases? Can you set it up as MOH and try it without involving ring/transfer back?

Gabe

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

 

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

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

!DSPAM:5098af8532761933413868!



From gabe at gundy.org  Tue Nov  6 10:04:50 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 6 Nov 2012 00:04:50 -0700
Subject: [Freeswitch-users] Not able to connect mysql using Dbh
In-Reply-To: <1351504494438-7584071.post@n2.nabble.com>
References: <1351504494438-7584071.post@n2.nabble.com>
Message-ID: <CAMwZ-tJYpcLO7i07XoSBPcuFa1=SWXPU1FYf-_zLeBUtYtrWaw@mail.gmail.com>

On Mon, Oct 29, 2012 at 3:54 AM, satheeshukt <satheesh.ukt at gmail.com> wrote:
> I'm a newbie to the Freeswitch. While trying to configure freeswitch am
> getting the below error:
>
> [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/directory.lua:73:
> attempt to call field 'Dbh' (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/directory.lua:73: in main chunk


If you're new to it, I'd just start with the DEBs or RPMs out there.


BTW, I just looked at a fresh checkout and this is what I see in the git repo:

~/Source/freeswitch$ find -name *.lua

./src/mod/applications/mod_vmd/scripts/vmd.lua
./scripts/lua/api.lua
./scripts/lua/originate.lua
./scripts/lua/mwi_event.lua
./scripts/lua/helloworld.lua
./scripts/lua/sound_test.lua
./scripts/lua/callback.lua
./scripts/lua/zrtp_agent.lua
./scripts/lua/zrtp_proxy_media.lua

No mention of directory.lua. I don't know what you should take from
that, but there it is.



> Could you any of you kindly let me what I'm missing here in order to resolve
> it? Many thanks!

Update git and try again. If that fails, punt and grab someones binaries ;)


Good luck and let us know how it goes!


Best,
Gabe


From gabe at gundy.org  Tue Nov  6 10:07:06 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 6 Nov 2012 00:07:06 -0700
Subject: [Freeswitch-users] Strange problem with ringback (distorted
	sound)
In-Reply-To: <50891358.8090403@gmail.com>
References: <50891358.8090403@gmail.com>
Message-ID: <CAMwZ-tKHoZnX8u9vT8W9s-Z0YtjoewNKSfp6k6eYTWFXNQF0JQ@mail.gmail.com>

On Thu, Oct 25, 2012 at 4:24 AM, Carlo Dimaggio <jaasmailing at gmail.com> wrote:
> I have freeswitch configured with standard media mode (no proxy no bypass).
> When a call arrives (through external profile), FS sets ringback and sends
> 183 session progress.
> Unfortunately the calling party (i.e. a cellular phone), hears a distorted
> ring sound.
>
> With proxy mode, of course the calling party hears a correct ring sound.
>
>
> What could be the reason?

What do the relevant configs/variables look like?



Gabe


From avi at avimarcus.net  Tue Nov  6 10:25:17 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 6 Nov 2012 09:25:17 +0200
Subject: [Freeswitch-users] Play music while transferring new incoming
	call
In-Reply-To: <50988470.1040708@the800group.com>
References: <5097D36A.7010803@the800group.com>
	<CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>
	<50988470.1040708@the800group.com>
Message-ID: <CALNZuwmGJXoF8n0uauh3uS+-oEQ=QW0SmY=Q8rC0OAtD6k7DVg@mail.gmail.com>

Your second example didn't work because you took out the {ignore_early_media
*=true*} from the bridge.

-Avi


On Tue, Nov 6, 2012 at 5:30 AM, ocset <ocset at the800group.com> wrote:

>  Thanks Avi
>
> I have tried the following two dial plans but the phone just rings - no
> music.
>
>
> <extension name="ext_8888">
>         <condition field="destination_number" expression="^(8888)$">
>                 <action application="set"
> data="transfer_ringback=local_stream://moh"/>
>                 <action application="set"
> data="ringback=local_stream://moh"/>
>                 <action application="transfer" data="1005 XML default"/>
>         </condition>
>  </extension>
>
>
> <extension name="ext_8888">
>         <condition field="destination_number" expression="^(8888)$">
>                 <action application="set"
> data="transfer_ringback=local_stream://moh"/>
>                 <action application="set"
> data="ringback=local_stream://moh"/>
>                 <action application="bridge" data=
> "sofia/internal/1005 at 192.168.0.150" <sofia/internal/1005 at 192.168.0.150>/>
>         </condition>
>  </extension>
>
> What am I missing? Should there be an "answer" directive in there
> somewhere also?
>
> Thanks
> O.
>
>
> On 05/11/12 23:30, Avi Marcus wrote:
>
> You probably want ringback and/or transfer ringback. That's for playing
> something while the call is waiting to be answered.
>
> See:
> http://wiki.freeswitch.org/wiki/Time_of_Day_Routing#Example_for_office_open_09:00-16:00 there's
> something set there, using:
>
>
>        <action application="set" data="ringback=local_stream://moh"/>
>        <action application="set" data="transfer_ringback=local_stream://moh"/>
>
>  You're line isn't quite right -- "{ignore_early_media*=true*}", you need
> to set the variable data.
>
>  Playback is blocking -- it plays and then moves on. Transfer_ringback
> sets what to do when the transfer occurs.
>
>  I hope that helps and you can clean up the wiki when you've got it all
> figured out.
>
> -Avi
>
>
>
> On Mon, Nov 5, 2012 at 4:55 PM, ocset <ocset at the800group.com> wrote:
>
>> Hi
>>
>> I am trying to to configure FS to play music while transferring a new
>> incoming call. Like when you phone a business and you get the "your call
>> in important to use, please wait while we transfer..." message.
>>
>> I found this suggestion on the web but it does not work for me. I have
>> tested the two <action> command separately to make sure they work but
>> when they are put together, the call is never answered and no music is
>> played.
>>
>>
>> <include>
>>   <extension name="ext_8888">
>>          <condition field="destination_number" expression="^(8888)$">
>>                  <action application="playback" data="music/test.wav"/>
>>                  <action application="bridge"
>> data="{ignore_early_media}sofia/internal/1005 at 192.168.0.150"/>
>>          </condition>
>>   </extension>
>> </include>
>>
>> ps. I cannot find the answer in either of the two FS books. Would have
>> expected this to be there, at least in the cookbook? All I can find is
>> examples of playing music and then hanging up. Happy to be corrected if
>> I missed it:-)
>>
>> All help greatly appreciated
>> Thanks
>> O.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nandy1925 at gmail.com  Tue Nov  6 10:38:00 2012
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 6 Nov 2012 15:38:00 +0800
Subject: [Freeswitch-users] Play music while transferring new incoming
	call
In-Reply-To: <50988470.1040708@the800group.com>
References: <5097D36A.7010803@the800group.com>
	<CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>
	<50988470.1040708@the800group.com>
Message-ID: <CAHKqMb6crZvfDpb3aekct-j4CbzBW7uTf8F+375v30gYPs_SoA@mail.gmail.com>

Yes. Answer the call before the bridge so that MOH can be played on the
caller.

================================================
Lapulapu City, Phils
Phone: +63-32-3401807, (USA) +1-646-5471226
Mobile: +63-920-6373450
*Worldwide:*
Dial any 200+ Access Number in 37
Countries<http://www.snipurl.com/magicbox-access>,
then ...
              dial *011-63-32-3401807# afterwards.



On Tue, Nov 6, 2012 at 11:30 AM, ocset <ocset at the800group.com> wrote:

>  Thanks Avi
>
> I have tried the following two dial plans but the phone just rings - no
> music.
>
>
> <extension name="ext_8888">
>         <condition field="destination_number" expression="^(8888)$">
>                 <action application="set"
> data="transfer_ringback=local_stream://moh"/>
>                 <action application="set"
> data="ringback=local_stream://moh"/>
>                 <action application="transfer" data="1005 XML default"/>
>         </condition>
>  </extension>
>
>
> <extension name="ext_8888">
>         <condition field="destination_number" expression="^(8888)$">
>                 <action application="set"
> data="transfer_ringback=local_stream://moh"/>
>                 <action application="set"
> data="ringback=local_stream://moh"/>
>                 <action application="bridge" data=
> "sofia/internal/1005 at 192.168.0.150" <sofia/internal/1005 at 192.168.0.150>/>
>         </condition>
>  </extension>
>
> What am I missing? Should there be an "answer" directive in there
> somewhere also?
>
> Thanks
> O.
>
>
> On 05/11/12 23:30, Avi Marcus wrote:
>
> You probably want ringback and/or transfer ringback. That's for playing
> something while the call is waiting to be answered.
>
> See:
> http://wiki.freeswitch.org/wiki/Time_of_Day_Routing#Example_for_office_open_09:00-16:00 there's
> something set there, using:
>
>        <action application="set" data="ringback=local_stream://moh"/>
>        <action application="set" data="transfer_ringback=local_stream://moh"/>
>
>  You're line isn't quite right -- "{ignore_early_media*=true*}", you need
> to set the variable data.
>
>  Playback is blocking -- it plays and then moves on. Transfer_ringback
> sets what to do when the transfer occurs.
>
>  I hope that helps and you can clean up the wiki when you've got it all
> figured out.
>
> -Avi
>
>
>
> On Mon, Nov 5, 2012 at 4:55 PM, ocset <ocset at the800group.com> wrote:
>
>> Hi
>>
>> I am trying to to configure FS to play music while transferring a new
>> incoming call. Like when you phone a business and you get the "your call
>> in important to use, please wait while we transfer..." message.
>>
>> I found this suggestion on the web but it does not work for me. I have
>> tested the two <action> command separately to make sure they work but
>> when they are put together, the call is never answered and no music is
>> played.
>>
>>
>> <include>
>>   <extension name="ext_8888">
>>          <condition field="destination_number" expression="^(8888)$">
>>                  <action application="playback" data="music/test.wav"/>
>>                  <action application="bridge"
>> data="{ignore_early_media}sofia/internal/1005 at 192.168.0.150"/>
>>          </condition>
>>   </extension>
>> </include>
>>
>> ps. I cannot find the answer in either of the two FS books. Would have
>> expected this to be there, at least in the cookbook? All I can find is
>> examples of playing music and then hanging up. Happy to be corrected if
>> I missed it:-)
>>
>> All help greatly appreciated
>> Thanks
>> O.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From slickwilly2000 at gmx.de  Tue Nov  6 11:43:50 2012
From: slickwilly2000 at gmx.de (=?UTF-8?Q?Alex_M=C3=BCller?=)
Date: Tue, 6 Nov 2012 09:43:50 +0100
Subject: [Freeswitch-users] Pickup
In-Reply-To: <CAM21H-NP7oUyw_pqUggrSdC8y8CDjXTZ0wpVWp-DqbTOYy7XuA@mail.gmail.com>
References: <20121105172238.137730@gmx.net>
	<CAM21H-NP7oUyw_pqUggrSdC8y8CDjXTZ0wpVWp-DqbTOYy7XuA@mail.gmail.com>
Message-ID: <BCB1F374A5AF4112B5C700EB0D1C5764@iMac>

I have a homogeneous infrastructure of ?Siemens OpenStage 60G?. These phones are regular SIP-Phones.


From: Ognjen Seslija 
Sent: Monday, November 05, 2012 8:37 PM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Pickup

What phones do you have?




On Mon, Nov 5, 2012 at 6:22 PM, "Alex M?ller" <slickwilly2000 at gmx.de> wrote:

  Hi,

  today, I implemented the pickup feature with the "intercept-application". So I am able to pick up an internal call that isn't answered yet. That works perfect!

  The only problem is that I do not see the counterpart at the phone that picked up the call.


  An example to illustrate this situation: Alice calls Bob, so Bob's phone is ringing. Charlie picks up the call, so now is bridged to Alice. Charlie does not see that he is connected to Alice (name/number).

  Can anybody help how to get this information from the remaining leg after intercept?



  Thanks in advance!


  (Just for your information: I already know how so set these information (effective_callee_id_name/effective_callee_id_number) but do not know how to get this from the bleg)

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

  
  

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

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





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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From NuwanW at unifybusiness.co.uk  Tue Nov  6 12:09:26 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Tue, 6 Nov 2012 09:09:26 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695E99BC942F@BARXB0003.UnifyBusiness.local>

Hello,

I'm trying to broadcast audio on a bridged call. My requirement is to play audio on both legs at the same time. I used uuid_broadcast in following order,

Uuid_broadcast uuid 'path' both

Please note that I'm sending uuid_broadcast through an esl connection. So my actual request to freeswitch is as follows,

eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file' both");

(eslWriteConnection is an object of .Net ESLConnection)

The  issue I'm having is, freeswitch not playing the audio on both channels at the same time. FreeSwitch plays the audio on one leg first, then plays on the second leg (After it finished playing on first leg).

I don't have this issue with FreeSwitch 1.0.6, where it plays audio on both legs at the same time. I'm having this issue with FreeSwtich 1.2.3.

Could anyone please suggest any solution.

Thank you,



This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From tnsampaio at bsd.com.br  Tue Nov  6 14:37:30 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Tue, 06 Nov 2012 09:37:30 -0200
Subject: [Freeswitch-users] Core dump on CALL from a YEALINK on FreeBSD
Message-ID: <5098F67A.2020002@bsd.com.br>

Hi guys!

Im facing a strange problem (again!!!).
Im using my FS with xlite and zoiper very well, but when i try call a 
ipphone Yalink t22p on my desk, freeswitch just coredump and exit.
I tryed disable all codecs but alaw and ulaw, but no success.
I tryed build version 1.2 and 1.3 from git, but the problem persists.

I was running it on a freebsd 32 bits, today i put it on a 9.1-PRELEASE 
amd64 and FS from ports:
freeswitch at internal> version
FreeSWITCH Version 1.2.3 (1.2.3)
reeswitch at internal> /quit

When i start a call to the phone, freeswitch simple exit...

here was the logs and backtrace: http://pastebin.com/rGZRgn1H

When it say reading global variables it die

Did someone experienced same problem?


From kbdfck at gmail.com  Tue Nov  6 15:48:23 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 6 Nov 2012 16:48:23 +0400
Subject: [Freeswitch-users] Directory entry variables not set on channel
 when user authenticated by cidr and domains ACL?
Message-ID: <CA+GmxbZzteFO-5OyHbcT-rVovzvS2TO-CzF7H9Px4Jvw77j5ag@mail.gmail.com>

Hi all!

I use xml_curl to get subscriber information from database when they
authenticate with username/password. In my directory script I specify
<params> and <variables> for user entry, and they are set on channel.

Now I'm trying to achieve customer authentication by IP address instead of
username/password.

I have
 <list name="domains" default="deny">
       <node type="allow" domain="$${domain}"/>
  </list>

and apply-inbound-acl="domains" at users' profile, and passwordless
authentication works fine for users with "cidr" attribute, but FS doesn't
set channel variables I return in directory entries in response to
'network-list' request on 'reloadacl', for example.

Since my dialplan logic relies on these variables (like customer ID and so
on), dialplan scripts don't see these vars on incoming calls from users
authenticated by domains ACL and fail their calls :(

Is this normal behaviour or I'm missing something? How can I get user
variables from my DB set on channel when authenticating call by CIDR
attribute?



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From steveu at coppice.org  Tue Nov  6 16:08:26 2012
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 06 Nov 2012 21:08:26 +0800
Subject: [Freeswitch-users] duplicate fax pages
In-Reply-To: <CAEVfcaYyvO-tz145EY=QmYFpOCO4U4p-1UFy8nNGT6m2bh2-Tg@mail.gmail.com>
References: <CAEVfcaYyvO-tz145EY=QmYFpOCO4U4p-1UFy8nNGT6m2bh2-Tg@mail.gmail.com>
Message-ID: <50990BCA.3050307@coppice.org>

Hi Babak,

In the log you posed FS is working properly, but you appear to have 
communication problems. FS sent the first page at V.29 9600bps, and the 
far end replied RTN, which means the page was received badly. FS 
renegotiated to V.29 7200 and sent the page again. This also returned in 
RTN. FS tried again at V.27ter 4800, but before the page was finished 
the far end disconnected.

Steve


On 11/05/2012 01:59 AM, Babak Yakhchali wrote:
> Hi
> I'm using mod_spandsp to send fax to different targets. when 
> destination is another freeswitch or elastix everything works fine. 
> but when sending faxes to old fax machines if tiff file contains only 
> 1 page, sending will succeed. but if tiff contains more than 1 page FS 
> keeps sending first page multiple times till I issue hupall and it 
> stops sending(I should mention that sometimes it sends first page 3 
> times second page 3 times and keeps eating  remote fax machine paper 
> roll).
> I'm using this command:
> originate {fax_ident=$faxidnt,fax_header=$faxhdr,origination_caller_id_number=$callerid,tx_owner_id=$userid}sofia/gateway/fax_outbound/999 &txfax('path')
> in php to send faxes. I've set global t38_passthru to true and in 
> spandsp.conf:
> <fax-settings>
> <param name="use-ecm"value="true"/>
> <param name="verbose"value="true"/>
> <param name="disable-v17 "value="false"/>
> <param name="ident"value="test"/>
> <param name="header"value="test"/>
>
> <param name="spool-dir"value="/tmp"/>
> <param name="file-prefix"value="faxrx"/>
>     </fax-settings>
> and the logs for spandsp verbose mode:
> http://pastebin.freeswitch.org/20169
> and the version:
> FreeSWITCH Version 1.3.0+git~20120908T211235Z~36cee285b0 (1.3.0; git 
> at commit 36cee285b0 on Sat, 08 Sep 2012 21:12:35 Z)
> thanx
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ocset at the800group.com  Tue Nov  6 17:04:55 2012
From: ocset at the800group.com (ocset)
Date: Tue, 06 Nov 2012 22:04:55 +0800
Subject: [Freeswitch-users] Play music while transferring new incoming
 call
In-Reply-To: <CAHKqMb6crZvfDpb3aekct-j4CbzBW7uTf8F+375v30gYPs_SoA@mail.gmail.com>
References: <5097D36A.7010803@the800group.com>
	<CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>
	<50988470.1040708@the800group.com>
	<CAHKqMb6crZvfDpb3aekct-j4CbzBW7uTf8F+375v30gYPs_SoA@mail.gmail.com>
Message-ID: <50991907.1090006@the800group.com>

Well, after a long discussion with "MorkBork" on IRC, the riddle has 
been solved. (Thanks again for your patience MorkBork!)

The  <action application="bridge" 
data="sofia/internal/1005 at 192.168.0.150" 
<mailto:sofia/internal/1005 at 192.168.0.150>/> is causing the problem. 
192.168.0.150 is the FS IP address and is causing a loop-back though the 
dialplan.

In MorkBork's own words - "freeswitch sees this, handles it as a refer, 
it re-transverses the dialplan and misses all your dialplan logic on the 
second time around"

This works

<extension name="ext_8888">
           <condition field="destination_number" expression="^(8888)$">
                  <action application="export" 
data="ringback=local_stream://custom1"/>
                  <action application="bridge" data="user/1005"/>
           </condition>
</extension>

Thanks you all for your help and suggestions.


On 06/11/12 15:38, Nandy Dagondon wrote:
> Yes. Answer the call before the bridge so that MOH can be played on 
> the caller.
>
> ================================================
> Lapulapu City, Phils
> Phone: +63-32-3401807, (USA) +1-646-5471226
> Mobile: +63-920-6373450
> *Worldwide:*
> Dial any 200+ Access Number in 37 Countries 
> <http://www.snipurl.com/magicbox-access>, then ...
>               dial *011-63-32-3401807# afterwards.
>
>
>
> On Tue, Nov 6, 2012 at 11:30 AM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Thanks Avi
>
>     I have tried the following two dial plans but the phone just rings
>     - no music.
>
>
>     <extension name="ext_8888">
>             <condition field="destination_number" expression="^(8888)$">
>                     <action application="set"
>     data="transfer_ringback=local_stream://moh"/>
>                     <action application="set"
>     data="ringback=local_stream://moh"/>
>                     <action application="transfer" data="1005 XML
>     default"/>
>             </condition>
>      </extension>
>
>
>     <extension name="ext_8888">
>             <condition field="destination_number" expression="^(8888)$">
>                     <action application="set"
>     data="transfer_ringback=local_stream://moh"/>
>                     <action application="set"
>     data="ringback=local_stream://moh"/>
>                     <action application="bridge"
>     data="sofia/internal/1005 at 192.168.0.150"
>     <mailto:sofia/internal/1005 at 192.168.0.150>/>
>             </condition>
>      </extension>
>
>     What am I missing? Should there be an "answer" directive in there
>     somewhere also?
>
>     Thanks
>     O.
>
>
>     On 05/11/12 23:30, Avi Marcus wrote:
>>     You probably want ringback and/or transfer ringback. That's for
>>     playing something while the call is waiting to be answered.
>>
>>     See:
>>     http://wiki.freeswitch.org/wiki/Time_of_Day_Routing#Example_for_office_open_09:00-16:00 there's
>>     something set there, using:
>>             <action application="set" data="ringback=local_stream://moh"/>
>>             <action application="set" data="transfer_ringback=local_stream://moh"/>
>>     You're line isn't quite right -- "{ignore_early_media_=true_}",
>>     you need to set the variable data.
>>
>>     Playback is blocking -- it plays and then moves on.
>>     Transfer_ringback sets what to do when the transfer occurs.
>>
>>     I hope that helps and you can clean up the wiki when you've got
>>     it all figured out.
>>
>>     -Avi
>>
>>
>>
>>     On Mon, Nov 5, 2012 at 4:55 PM, ocset <ocset at the800group.com
>>     <mailto:ocset at the800group.com>> wrote:
>>
>>         Hi
>>
>>         I am trying to to configure FS to play music while
>>         transferring a new
>>         incoming call. Like when you phone a business and you get the
>>         "your call
>>         in important to use, please wait while we transfer..." message.
>>
>>         I found this suggestion on the web but it does not work for
>>         me. I have
>>         tested the two <action> command separately to make sure they
>>         work but
>>         when they are put together, the call is never answered and no
>>         music is
>>         played.
>>
>>
>>         <include>
>>           <extension name="ext_8888">
>>                  <condition field="destination_number"
>>         expression="^(8888)$">
>>                          <action application="playback"
>>         data="music/test.wav"/>
>>                          <action application="bridge"
>>         data="{ignore_early_media}sofia/internal/1005 at 192.168.0.150
>>         <mailto:1005 at 192.168.0.150>"/>
>>                  </condition>
>>           </extension>
>>         </include>
>>
>>         ps. I cannot find the answer in either of the two FS books.
>>         Would have
>>         expected this to be there, at least in the cookbook? All I
>>         can find is
>>         examples of playing music and then hanging up. Happy to be
>>         corrected if
>>         I missed it:-)
>>
>>         All help greatly appreciated
>>         Thanks
>>         O.
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-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
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Nov  6 17:08:20 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Nov 2012 09:08:20 -0500
Subject: [Freeswitch-users] Play music while transferring new incoming
	call
In-Reply-To: <CAHKqMb6crZvfDpb3aekct-j4CbzBW7uTf8F+375v30gYPs_SoA@mail.gmail.com>
References: <5097D36A.7010803@the800group.com>
	<CALNZuwmTqMd15M3rKWsPdEyBSQKqf==YGC3Nzv=Wimi_5Tqktw@mail.gmail.com>
	<50988470.1040708@the800group.com>
	<CAHKqMb6crZvfDpb3aekct-j4CbzBW7uTf8F+375v30gYPs_SoA@mail.gmail.com>
Message-ID: <6E451546-69DA-4832-8DE4-282D4EB88FE0@jerris.com>

This is incorrect.  You do not want to answer the call first.  Is this a sip call that is getting a 180 or 183 from the remote side?


On Nov 6, 2012, at 2:38 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Yes. Answer the call before the bridge so that MOH can be played on the caller.
> 
> ================================================
> Lapulapu City, Phils
> Phone: +63-32-3401807, (USA) +1-646-5471226
> Mobile: +63-920-6373450
> Worldwide:
> Dial any 200+ Access Number in 37 Countries, then ... 
>               dial *011-63-32-3401807# afterwards.
> 
> 
> 
> On Tue, Nov 6, 2012 at 11:30 AM, ocset <ocset at the800group.com> wrote:
> Thanks Avi
> 
> I have tried the following two dial plans but the phone just rings - no music.
> 
> 
> <extension name="ext_8888">
>         <condition field="destination_number" expression="^(8888)$">
>                 <action application="set" data="transfer_ringback=local_stream://moh"/>
>                 <action application="set" data="ringback=local_stream://moh"/>
>                 <action application="transfer" data="1005 XML default"/>
>         </condition>
>  </extension>
> 
> 
> <extension name="ext_8888">
>         <condition field="destination_number" expression="^(8888)$">
>                 <action application="set" data="transfer_ringback=local_stream://moh"/>
>                 <action application="set" data="ringback=local_stream://moh"/>
>                 <action application="bridge" data="sofia/internal/1005 at 192.168.0.150"/>
>         </condition>
>  </extension>
> 
> What am I missing? Should there be an "answer" directive in there somewhere also?
> 
> Thanks
> O.
> 
> 
> On 05/11/12 23:30, Avi Marcus wrote:
>> You probably want ringback and/or transfer ringback. That's for playing something while the call is waiting to be answered.
>> 
>> See: http://wiki.freeswitch.org/wiki/Time_of_Day_Routing#Example_for_office_open_09:00-16:00 there's something set there, using:
>>        <action application="set" data="ringback=local_stream://moh"/>
>>        <action application="set" data="transfer_ringback=local_stream://moh"/>
>> You're line isn't quite right -- "{ignore_early_media=true}", you need to set the variable data.
>> 
>> Playback is blocking -- it plays and then moves on. Transfer_ringback sets what to do when the transfer occurs.
>> 
>> I hope that helps and you can clean up the wiki when you've got it all figured out.
>> 
>> -Avi
>> 
>> 
>> 
>> On Mon, Nov 5, 2012 at 4:55 PM, ocset <ocset at the800group.com> wrote:
>> Hi
>> 
>> I am trying to to configure FS to play music while transferring a new
>> incoming call. Like when you phone a business and you get the "your call
>> in important to use, please wait while we transfer..." message.
>> 
>> I found this suggestion on the web but it does not work for me. I have
>> tested the two <action> command separately to make sure they work but
>> when they are put together, the call is never answered and no music is
>> played.
>> 
>> 
>> <include>
>>   <extension name="ext_8888">
>>          <condition field="destination_number" expression="^(8888)$">
>>                  <action application="playback" data="music/test.wav"/>
>>                  <action application="bridge"
>> data="{ignore_early_media}sofia/internal/1005 at 192.168.0.150"/>
>>          </condition>
>>   </extension>
>> </include>
>> 
>> ps. I cannot find the answer in either of the two FS books. Would have
>> expected this to be there, at least in the cookbook? All I can find is
>> examples of playing music and then hanging up. Happy to be corrected if
>> I missed it:-)
>> 
>> All help greatly appreciated
>> Thanks
>> O.
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Nov  6 17:09:42 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Nov 2012 09:09:42 -0500
Subject: [Freeswitch-users] Core dump on CALL from a YEALINK on FreeBSD
In-Reply-To: <5098F67A.2020002@bsd.com.br>
References: <5098F67A.2020002@bsd.com.br>
Message-ID: <D9BA6EC6-4424-42C8-84AF-20D801C7CA2C@jerris.com>

http://wiki.freeswitch.org/wiki/Reporting_Bugs

Please create a jira

On Nov 6, 2012, at 6:37 AM, Tiago N. Sampaio <tnsampaio at bsd.com.br> wrote:

> Hi guys!
> 
> Im facing a strange problem (again!!!).
> Im using my FS with xlite and zoiper very well, but when i try call a 
> ipphone Yalink t22p on my desk, freeswitch just coredump and exit.
> I tryed disable all codecs but alaw and ulaw, but no success.
> I tryed build version 1.2 and 1.3 from git, but the problem persists.
> 
> I was running it on a freebsd 32 bits, today i put it on a 9.1-PRELEASE 
> amd64 and FS from ports:
> freeswitch at internal> version
> FreeSWITCH Version 1.2.3 (1.2.3)
> reeswitch at internal> /quit
> 
> When i start a call to the phone, freeswitch simple exit...
> 
> here was the logs and backtrace: http://pastebin.com/rGZRgn1H
> 
> When it say reading global variables it die
> 
> Did someone experienced same problem?
> 
> _
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From mkovalenko at cybervisiontech.com  Tue Nov  6 17:08:11 2012
From: mkovalenko at cybervisiontech.com (Max Kovalenko)
Date: Tue, 6 Nov 2012 09:08:11 -0500 (EST)
Subject: [Freeswitch-users] media proxy
In-Reply-To: <CAAyX+KGSjJZ8ttMQ9WRFORUBzOoo=DXo2RiYsKKaqOJX4Qs-9g@mail.gmail.com>
Message-ID: <3414928.7.1352211231848.JavaMail.master@VoiceJuggler>

I'm getting the data from external DB containing subscribers' parameters.

Max Kovalenko
---------------
CyberVision Inc.

----- Original Message -----
From: "Ben Langfeld" <ben at langfeld.co.uk>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, November 4, 2012 10:46:00 PM
Subject: Re: [Freeswitch-users] media proxy


Can you point to other software that provides this distinction in media proxying? 

Regards, 
Ben Langfeld 



On 4 November 2012 11:57, Max Kovalenko < mkovalenko at cybervisiontech.com > wrote: 


Unfortunately, it isn't well documented, or else I would not ask about :)))))) 

Those parameters perhaps are exclusive of each other, by mechanics they affects to is almost the same. Or at least, I would call them two sides of ame thing. 

Theoretically, if you consider the situation of bypass_media=true, that means media flow bypass around Freeswitch, it also automatically annihilates settings of proxy_media variable. It doesn't matter, even you would set proxy_media=true for same channel, it should not provide you "light proxy" of media flow, because your settign of bypass_proxy comprises Freeswitch don't proxy media. 
Thus I guess in order to set "light" or "hard" proxy the variable bypass_media must be set to FALSE first, for same channel of course. 

My task requires either bypass media or light/hard proxy to be set for a channel depending of some conditions of a session. In the other words, I need to set up one of three modes per each channel. 

Perhaps I undestand something not so right - I will appreciate if you will correct me with the following: 

1. Default (full media proxy) : bypass_media=false ; proxy_media=false 

2. Transparent proxy (mine "light proxy"): bypass_media=false; proxy_media=true 

3. Bypass media: bupass_media=true (it's doesn't matter what currect value of proxy_media is) 

Thanks 


Max Kovalenko 
Team Leader 
VoIP & UC Team 
Managed Services Dept. 
CyberVision Inc. 
------------------------------------------------- 
tel. +1 (201) 585-9809 ext. 215 
Email: mkovalenko at cybervisiontech.com 
Skype: mkovalenko_cv, panzer_meister 
WWW: www.cybervisiontech.com 



----- ???????? ????????? ----- 
??: "Ken Rice" < krice at freeswitch.org > 
????: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????, ????????, ??????? 
????: Re: [Freeswitch-users] media proxy 


Default, proxy and bybass media are exclusive of each other, you can not 
combine them on a single call... 

Proxy media is just that proxy the media 

Bypass media is just that, bypass freeswitch and send the media direct 
between the end point... 

Setting this options is fairly well documented on the wiki... 




On 11/2/12 9:13 AM, "Max Kovalenko" < mkovalenko at cybervisiontech.com > wrote: 

> Hello, 
> 
> There are two modes of media stream to be proxied or bypassed. 
> 
> 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc. 
> 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT 
> traversal purposes. No media stream parsing is supported 
> 3. Media bypass - RTP streams are bypassing FreeSWITCH at all. 
> 
> There are also two channel parameters affecting above modes: bypass_media and 
> proxy_media. 
> 
> - Are these parameters independent? Meaning the combination: bypass_media=true 
> AND proxy_media=true is possible. What effect will be? 
> 
> - Does it means that if I want to turn on "light proxy" I would always need to 
> set bypass_media=false AND proxy_media=true? 
> 
> - How to set "hard proxy" (full media proxy) per channel before to bridge 
> legs? 
> 
> Waiting for your replay ASAP. Thank you in advance. 
> 

> Best Regards. 
> 
> Max Kovalenko 
> Team Leader 
> VoIP & UC Team 
> Managed Services Dept. 
> CyberVision Inc. 
> ------------------------------------------------- 
> tel. +1 (201) 585-9809 ext. 215 
> Email: mkovalenko at cybervisiontech.com 
> Skype: mkovalenko_cv, panzer_meister 
> WWW: www.cybervisiontech.com 
> 
> 
> 
> 

> _________________________________________________________________________ 
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org 
> http://www.freeswitchsolutions.com 
> 
>  
>  
> 
> Official FreeSWITCH Sites 
> http://www.freeswitch.org 
> http://wiki.freeswitch.org 
> http://www.cluecon.com 
> 
> FreeSWITCH-users mailing list 
> FreeSWITCH-users at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
> http://www.freeswitch.org 


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





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

 
 

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

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

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

 
 

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

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


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




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

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


From lists at kavun.ch  Tue Nov  6 17:26:56 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 6 Nov 2012 09:26:56 -0500
Subject: [Freeswitch-users] A-leg hangup cause is blank
In-Reply-To: <CAMwZ-tLJRodQ8Jbrm6uSrsM4OsKu7bWFAmYhqyWfFqDzMxB3AQ@mail.gmail.com>
References: <51EC5321-C965-4553-9DE4-F84B1FC460DD@kavun.ch>
	<6B9182CC-772E-4B56-80BB-A9A2DDC32425@kavun.ch>
	<CAMwZ-tLJRodQ8Jbrm6uSrsM4OsKu7bWFAmYhqyWfFqDzMxB3AQ@mail.gmail.com>
Message-ID: <F6D087C9-1D00-4567-A7E5-27A7E917E059@kavun.ch>

Hey Gabe, thanks for these suggestions. I actually found out that a LUA script that is called in a hangup hook will receive the the full landscape of the env variables, including the hangup_cause.

Thanks a bunch, I'm glad I escaped your famous replies. :P

Cheers,
Emrah
On Nov 6, 2012, at 1:33 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Nov 5, 2012 at 1:01 PM, Emrah <lists at kavun.ch> wrote:
>> It makes sense that hangup_cause is blank because at the time the variable is set, the channel hasn't been hanged-up?
> 
> Depending on what you're doing, you may have to look to the
> CS_REPORTING state. This is where you get your CDRs written out to
> disk, HTTP, DB or whatever.
> 
> http://wiki.freeswitch.org/wiki/Channel_States
> 
> Is listening on the event socket an option? How about watching for
> CDRs to be written to disk?
> 
> 
> Good luck,
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From sdevoy at bizfocused.com  Tue Nov  6 19:41:39 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 6 Nov 2012 11:41:39 -0500
Subject: [Freeswitch-users] Bad connection diagnostics?
Message-ID: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>

Hi,

I have a client who has been working fine for months, but as of late they
are reporting connections where the audio only works in one direction or
neither and "terrible echo".

 

I know they had some issues with their local Cable Connection, but they are
supposedly resolved.  Results from "ping -n 50 -l 256 " and "tracert" are
virtually identical to mine from here and I have no issues.   Are there some
diagnostics I could run to try and pin this down.  The phones are all cisco
504Gs, the server is a VPS from synapseglobal.

 

Thanks for any ideas you may have.

Sean

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From avi at avimarcus.net  Tue Nov  6 20:17:47 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 6 Nov 2012 19:17:47 +0200
Subject: [Freeswitch-users] Bad connection diagnostics?
In-Reply-To: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
References: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
Message-ID: <CALNZuwmDanvi0iOKKCnm9=pwQi8gRO4tpsohXszq_iBQLx7L5A@mail.gmail.com>

I would like an answer to this, too.

However, here's a place to start: you have 2 legs - you to their VPS, and
their VPS to their office phones.
Perhaps there's an issue with the VPS. Can you get their VPS to playback a
file and do an echo test? E.g. MOH or similar and call it directly and see
if any of the same issues persist. Either way, you've narrowed down where
the issue is...

-Avi


On Tue, Nov 6, 2012 at 6:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,****
>
> I have a client who has been working fine for months, but as of late they
> are reporting *connections where the audio only works in one direction or
> neither and ?terrible echo?.*****
>
> ** **
>
> I know they had some issues with their local Cable Connection, but they
> are supposedly resolved.  Results from ?ping ?n 50 ?l 256 ? and ?tracert?
> are virtually identical to mine from here and I have no issues.   Are there
> some diagnostics I could run to try and pin this down.  The phones are all
> cisco 504Gs, the server is a VPS from synapseglobal.****
>
> ** **
>
> Thanks for any ideas you may have.****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at gmail.com  Tue Nov  6 20:18:00 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 6 Nov 2012 18:18:00 +0100
Subject: [Freeswitch-users] mod_skypopen - when call routes to a skype
 voicemail there is no audio.
In-Reply-To: <CCAB0FF9.4223C%rhuddleston@gmail.com>
References: <00dd01cdb051$2ecae1e0$8c60a5a0$@com>
	<CCAB0FF9.4223C%rhuddleston@gmail.com>
Message-ID: <CALXCt0p5UoWBHhAdrZoCXU_rK3AhPj-U5rkti9nPS77oCJp31Q@mail.gmail.com>

Can you please test with latest git trunk ?

-giovanni

On Mon, Oct 22, 2012 at 9:00 PM, Robert <rhuddleston at gmail.com> wrote:

> Giovanni ? I sent you an offline email.. If you think it best I should
> code a patch to handle calls ringing to Skype voicemail ? let me know ? as
> this would be a simple hack for the time being
>
>
> From: Mike Patterson <mikecpatterson at yahoo.com>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 22 Oct 2012 08:31:37 -0400
> To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
>
> Subject: Re: [Freeswitch-users] mod_skypopen - when call routes to a
> skype voicemail there is no audio.
>
> I did open a Jira case.  It was moved to Freeswitch.  Thanks.****
>
> ** **
>
> [image:
> http://jira.freeswitch.org/s/en_USk8cqyf/783/4/_/jira-logo-scaled.png]****
>
> [image: http://jira.freeswitch.org/secure/useravatar?avatarId=10142]Jeff
> Lenk <http://jira.freeswitch.org/secure/ViewProfile.jspa?name=jlenk>assigned [image:
> Bug]FS-4742 <http://jira.freeswitch.org/browse/FS-4742> to Giovanni
> Maruzzelli<http://jira.freeswitch.org/secure/ViewProfile.jspa?name=gmaruzz>
> ****
>
> *mod_skypopen - when call routes to a skype voicemail there is no audio.*<http://jira.freeswitch.org/browse/FS-4742>
> ****
>
> Moved issue to proper project and reassigned to Giovanni****
>
> *Change By:* ****
>
> Jeff Lenk <http://jira.freeswitch.org/secure/ViewProfile.jspa?name=jlenk>(21/Oct/12 6:47 PM)
> ****
>
> *Assignee:* ****
>
> Michal Bielicki Giovanni Maruzzelli ****
>
> ** **
>
> ** **
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> On Behalf Of Giovanni Maruzzelli
> Sent: Monday, October 22, 2012 6:04 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] mod_skypopen - when call routes to a skype
> voicemail there is no audio.****
>
> ** **
>
> Hi Mike,****
>
> ** **
>
> can you please open a Jira, attach all the relevant issue?****
>
> ** **
>
>
> http://wiki.freeswitch.org/wiki/Skypopen#How_To_Report_BUGS_and_Feature_Requests
> ****
>
> ** **
>
> -giovanni****
>
> ** **
>
> On Sun, Oct 21, 2012 at 7:03 PM, Mike Patterson <mikecpatterson at yahoo.com>
> wrote:****
>
> > When using mod_skypopen we are able to route a call to the Skype user **
> **
>
> > and it works great if the user is online and picks up the call. ****
>
> > However, if the user allows the call to route to their Skype voice ****
>
> > mail, then we receive an error - VM_PLAYING_GREETING is not recognized.
> The entire debug is attached.****
>
> > I have a complete debug if needed.  Again, if the call is answered it **
> **
>
> > works fine.  If the call is routed to Skype voicemail we have about a **
> **
>
> > minute of dead air, and then a fast busy.****
>
> >** **
>
> >** **
>
> >** **
>
> > Using:****
>
> >** **
>
> > ubuntu 11.10, FreeSWITCH Version 1.3.0+git~20120919T180633Z~d22e0caf15**
> **
>
> > (1.3.0)****
>
> > AMD Athlon(tm) II X3 425 Processor, 3 cores****
>
> >** **
>
> >** **
>
> >** **
>
> >** **
>
> >** **
>
> > Log messages:****
>
> >** **
>
> >** **
>
> >** **
>
> > 2012-10-20 12:32:53.281306 [DEBUG] mod_skypopen.c:727****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  727  ][skype105       ][DIALING,UNPLACD]
> ****
>
> > skypopen/RR/skype105/mikec.patterson CHANNEL got SWITCH_SIG_BREAK****
>
> >** **
>
> > 2012-10-20 12:32:53.281306 [DEBUG] switch_core_state_machine.c:463****
>
> > (skypopen/RR/skype105/mikec.patterson) State ROUTING going to sleep****
>
> >** **
>
> > 2012-10-20 12:32:53.281306 [DEBUG] switch_core_state_machine.c:415****
>
> > (skypopen/RR/skype105/mikec.patterson) Running State Change ****
>
> > CS_CONSUME_MEDIA****
>
> >** **
>
> > 2012-10-20 12:32:53.281306 [DEBUG] switch_core_state_machine.c:482****
>
> > (skypopen/RR/skype105/mikec.patterson) State CONSUME_MEDIA****
>
> >** **
>
> > 2012-10-20 12:32:53.281306 [DEBUG] mod_skypopen.c:748****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  748  ][skype105       ][DIALING,UNPLACD]
> ****
>
> > skype105 CHANNEL CONSUME_MEDIA****
>
> >** **
>
> > 2012-10-20 12:32:53.281306 [DEBUG] switch_core_state_machine.c:482****
>
> > (skypopen/RR/skype105/mikec.patterson) State CONSUME_MEDIA going to ****
>
> > sleep****
>
> >** **
>
> > 2012-10-20 12:32:53.341169 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,UNPLACD]
> ****
>
> > READING: |||CALL 56 STATUS ROUTING|||****
>
> >** **
>
> > 2012-10-20 12:32:53.341169 [DEBUG] skypopen_protocol.c:758****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  758  ][skype105       ][DIALING,ROUTING]
> ****
>
> > skype_call: 56 is now ROUTING****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 TYPE OUTGOING|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 PARTNER_HANDLE mikec.patterson|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 PARTNER_DISPNAME Mike Patterson|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 ALLOWED_DURATION 600|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 SUBJECT |||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 TIMESTAMP 1350736374|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 0|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 STATUS UNKNOWN|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_ALLOWED_DURATION 600|||****
>
> >** **
>
> > 2012-10-20 12:32:54.921176 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 0|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 STATUS BLANK|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 TYPE CUSTOM_GREETING|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 PARTNER_HANDLE mikec.patterson|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 PARTNER_DISPNAME Mike Patterson|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 ALLOWED_DURATION 60|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 SUBJECT |||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 TIMESTAMP 1350736375|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 DURATION 0|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 STATUS NOTDOWNLOADED|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 STATUS PLAYING|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 STATUS BUFFERING|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 STATUS VM_PLAYING_GREETING|||****
>
> >** **
>
> > 2012-10-20 12:32:55.821160 [WARNING] skypopen_protocol.c:837****
>
> > [464155c|07bc7ba] [WARNINGA     837  ][skype105       ][DIALING,ROUTING]
> ****
>
> > skype_call: 56, STATUS: VM_PLAYING_GREETING is not recognized****
>
> >** **
>
> > 2012-10-20 12:32:56.021190 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 DURATION 6|||****
>
> >** **
>
> > 2012-10-20 12:32:56.021190 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 TYPE DEFAULT_GREETING|||****
>
> >** **
>
> > 2012-10-20 12:32:56.121217 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 STATUS PLAYING|||****
>
> >** **
>
> > 2012-10-20 12:33:02.121165 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 59 STATUS PLAYED|||****
>
> >** **
>
> > 2012-10-20 12:33:02.121165 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 TIMESTAMP 1350736382|||****
>
> >** **
>
> > 2012-10-20 12:33:02.121165 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 STATUS RECORDING|||****
>
> >** **
>
> > 2012-10-20 12:33:02.121165 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 STATUS VM_RECORDING|||****
>
> >** **
>
> > 2012-10-20 12:33:02.121165 [WARNING] skypopen_protocol.c:837****
>
> > [464155c|07bc7ba] [WARNINGA     837  ][skype105       ][DIALING,ROUTING]
> ****
>
> > skype_call: 56, STATUS: VM_RECORDING is not recognized****
>
> >** **
>
> > 2012-10-20 12:33:03.421160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 1|||****
>
> >** **
>
> > 2012-10-20 12:33:03.421160 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 1|||****
>
> >** **
>
> > 2012-10-20 12:33:03.821173 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 2|||****
>
> >** **
>
> > 2012-10-20 12:33:03.821173 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 2|||****
>
> >** **
>
> > 2012-10-20 12:33:04.821174 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 3|||****
>
> >** **
>
> > 2012-10-20 12:33:04.821174 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 3|||****
>
> >** **
>
> > 2012-10-20 12:33:05.821172 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 4|||****
>
> >** **
>
> > 2012-10-20 12:33:05.821172 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 4|||****
>
> >** **
>
> > 2012-10-20 12:33:06.921168 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 5|||****
>
> >** **
>
> > 2012-10-20 12:33:06.921168 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 5|||****
>
> >** **
>
> > 2012-10-20 12:33:07.821196 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 6|||****
>
> >** **
>
> > 2012-10-20 12:33:07.821196 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 6|||****
>
> >** **
>
> > 2012-10-20 12:33:08.821167 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 7|||****
>
> >** **
>
> > 2012-10-20 12:33:08.821167 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 7|||****
>
> >** **
>
> > 2012-10-20 12:33:09.821168 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 8|||****
>
> >** **
>
> > 2012-10-20 12:33:09.821168 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 8|||****
>
> >** **
>
> > 2012-10-20 12:33:10.921166 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 9|||****
>
> >** **
>
> > 2012-10-20 12:33:10.921166 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 9|||****
>
> >** **
>
> > 2012-10-20 12:33:11.821166 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 10|||****
>
> >** **
>
> > 2012-10-20 12:33:11.821166 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 10|||****
>
> >** **
>
> > 2012-10-20 12:33:12.921166 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||VOICEMAIL 58 DURATION 11|||****
>
> >** **
>
> > 2012-10-20 12:33:12.921166 [DEBUG] skypopen_protocol.c:207****
>
> > [464155c|07bc7ba] [DEBUG_SKYPE  207  ][skype105       ][DIALING,ROUTING]
> ****
>
> > READING: |||CALL 56 VM_DURATION 11|||****
>
> >** **
>
> > 2012-10-20 12:33:13.241213 [DEBUG] switch_core_session.c:969 Send ****
>
> > signal****
>
> > sofia/internal/15405078003 at 38.100.174.200 [BREAK]****
>
> >** **
>
> >** **
>
> > ______________________________________________________________________**
> **
>
> > ___ Professional FreeSWITCH Consulting Services:****
>
> > consulting at freeswitch.org****
>
> > http://www.freeswitchsolutions.com****
>
> >** **
>
> >  ****
>
> > ****
>
> >** **
>
> > Official FreeSWITCH Sites****
>
> > http://www.freeswitch.org****
>
> > http://wiki.freeswitch.org****
>
> > http://www.cluecon.com****
>
> >** **
>
> > FreeSWITCH-users mailing list****
>
> > FreeSWITCH-users at lists.freeswitch.org****
>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use**
> **
>
> > rs****
>
> > http://www.freeswitch.org****
>
> >** **
>
> ** **
>
> ** **
>
> ** **
>
> --****
>
> Sincerely,****
>
> ** **
>
> Giovanni Maruzzelli****
>
> Cell : +39-347-2665618****
>
> ** **
>
> _________________________________________________________________________*
> ***
>
> Professional FreeSWITCH Consulting Services:****
>
> consulting at freeswitch.org****
>
> http://www.freeswitchsolutions.com****
>
> ** **
>
> 
> ****
>
> ** **
>
> Official FreeSWITCH Sites****
>
> http://www.freeswitch.org****
>
> http://wiki.freeswitch.org****
>
> http://www.cluecon.com****
>
> ** **
>
> FreeSWITCH-users mailing list****
>
> FreeSWITCH-users at lists.freeswitch.org****
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**
> **
>
> http://www.freeswitch.org****
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  Official FreeSWITCH Sites
> http://www.freeswitch.org http://wiki.freeswitch.org
> http://www.cluecon.com FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From gmaruzz at gmail.com  Tue Nov  6 20:20:14 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 6 Nov 2012 18:20:14 +0100
Subject: [Freeswitch-users] GSMOPEN dongle port changes randomly
In-Reply-To: <509139E8.70107@infra-it.ru>
References: <CAO1DH1_-ht4w5Keu+mL5MeZgxYs-8z7XthFBLke273rex29jQw@mail.gmail.com>
	<509139E8.70107@infra-it.ru>
Message-ID: <CALXCt0rYt_R+9XaiV0bJRt9HsA9Ae+0H+ouNpU0J-ETONx3s7A@mail.gmail.com>

On Wed, Oct 31, 2012 at 3:47 PM, Ivan Mironov <ivan.mironov at infra-it.ru> wrote:
> Some USB dongles with some firmware versions can sometimes spontaneously
> reboot or even freeze. In case of reboot, for host system it looks like
> modem has been unplugged and then plugged back. You can check kernel log
> (usually /var/log/kern.log in ubuntu) for related messages from USB
> subsystem.
>
> Device files /dev/ttyUSB* doesn't have any stable numbering scheme,
> especially when multiple USB-Serial adapters (or modems) connected to
> system. You can use symlinks from /dev/serial/by-id or
> /dev/serial/by-path if you wish stable device paths. Unfortunately, most
> vendors doesn't assign serial numbers to their dongles, so by-id/ not
> usable when you have more than one dongle from one vendor with same
> model name. Symlinks in by-path/ should be stable when connecting
> devices directly to motherboard USB ports without any USB hubs, but I'm
> not quite sure in that... Also, mod_gsmopen from 1.2.3 doesn't work with
> long device names, here is the bugreport with patch:
> http://jira.freeswitch.org/browse/FS-4691.

Hello Ivan,

patch integrated in mainline.

Can you please share your udev by-path technique? would be very very useful!

-giovanni




>
> 31.10.2012 19:46, Yihui Li ?????:
>> Hi,
>>
>> I am using gsmopen on ubuntu 12.04LTS. With E169 dongle connected as
>> ttyUSB0, ttyUSB1 and ttyUSB2. It runs well at first. But after days, all
>> call failed. Checked the tty files, the ports on system is somehow
>> changed to ttyUSB0, ttyUSB1 and ttyUSB3.  ttyUSB2 just disappeared.
>> Does anyone meet the same problem? How can I make it stable? Thanks.
>>
>> Regards,
>> Eric
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> --
> ? ?????????, ???? ??????? <ivan.mironov at infra-it.ru>, ??? "?????-????".
> ????????? ???????: +7-912-479-2263
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at gmail.com  Tue Nov  6 20:22:24 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 6 Nov 2012 18:22:24 +0100
Subject: [Freeswitch-users] Problem: call ends when answered on xlite
 mod callcenter skypopen
In-Reply-To: <CAMHZFZ4za-WOHWY9X-BM5mgqF=EPNnW2BdQg2_WuySAosBeQQg@mail.gmail.com>
References: <CAMHZFZ6qEyRcjwWKevUkoWTuKP=vePj6u5Ukm3+qBwsvSm6nyw@mail.gmail.com>
	<50825AD6.1080805@elder.hu>
	<CAMHZFZ5k1DsCm+75rLFqQJv8gtrbu0t6-f4quidr0M7ANRF-yw@mail.gmail.com>
	<50827980.4050504@elder.hu>
	<CAMHZFZ5CkpVNejvJZEOUaYHQrX-Vrz19krNmZp0LWSyUaHvEyA@mail.gmail.com>
	<5082EDD2.30408@elder.hu>
	<CAMHZFZ4za-WOHWY9X-BM5mgqF=EPNnW2BdQg2_WuySAosBeQQg@mail.gmail.com>
Message-ID: <CALXCt0rGY5RAgiwmCy1CvUDH32-4jDH7XebTpviVMFmO6PiAnw@mail.gmail.com>

Can you please test with latest git trunk ?

-giovanni


On Sun, Oct 21, 2012 at 12:03 PM, Pankaj Belsare
<belsarepankaj at gmail.com> wrote:
> Hi,
>
> Thankyou for your help i have posted  the bug
> http://jira.freeswitch.org/browse/FS-4740
>
>
>
>
> On Sun, Oct 21, 2012 at 12:00 AM, Pusk?s Zsolt <errotan at elder.hu> wrote:
>>
>>
>> This is a bug indeed. I compiled the latest sources on my debian test box
>> and using the sample config from the wiki I made some test calls. Calls
>> coming from sofia users ( eg. softphone ) to the queue are good ( can answer
>> and talk ) but calls coming from skypopen is dropping as soon as I answer
>> the call on my desk ( Linksys SPA921 ).
>>
>> Please post a bug report on JIRA. I will dig into the source tomorrow and
>> hope to find some clue on this one :)
>>
>>
>> 2012-10-20 13:14 keltez?ssel, Pankaj Belsare ?rta:
>>
>> Hi,
>>
>> Yes i just want the call to be bridged to the xlite extensions a.k.a agent
>> which are in the mod callcenter  queue but the skype call hangs up,
>>
>> It works fine when even i transfer the active call while waiting on queue
>> to any xlite extension (via usionpbx ) or even an test ivr bridging works
>> charm.
>>
>> Its only when in the queue the call is answered the problem occurs even i
>> have tried with net4india sip account it went through with out any issue.
>>
>> I am not able to figure out is there any option or variable to be set that
>> i am missing.
>>
>> Thanks
>> Pankaj
>>
>> On Sat, Oct 20, 2012 at 3:44 PM, Pusk?s Zsolt <errotan at elder.hu> wrote:
>>>
>>>
>>>
>>> Interesting, this config should work AFAIK. Have you tried just bridging
>>> the incoming call to an X-Lite extension for testing ?
>>>
>>>
>>> 2012-10-20 11:34 keltez?ssel, Pankaj Belsare ?rta:
>>>
>>> Hi,
>>> Thanks for responding to my query. I am using Ubuntu 10.04 LTS freeswitch
>>> version 1.2.3 used interactive skype installer  for mod_skypopen
>>>
>>> http://pastebin.freeswitch.org/20102
>>>
>>> Thank you
>>> Pankaj
>>>
>>>
>>>
>>>
>>> On Sat, Oct 20, 2012 at 1:33 PM, Pusk?s Zsolt <errotan at elder.hu> wrote:
>>>>
>>>> Hi.
>>>>
>>>> Please use pastebin to upload your full console log from call start to
>>>> end and all config files. Which FS and skype version are you using ?
>>>>
>>>>
>>>>
>>>> 2012-10-19 22:32 keltez?ssel, Pankaj Belsare ?rta:
>>>>
>>>> Hi,
>>>>
>>>> I am trying to use mod callcenter for my FS installation with skypopen,
>>>> the problem is when then incoming call from skype gets hangup as soon as it
>>>> is answered by agent on queue, Can any body help me understand why this is
>>>> happening. here is my console log.
>>>>
>>>> 2012-10-20 01:44:52.925614 [NOTICE] sofia.c:7053 Channel
>>>> [sofia/internal/sip:100 at 192.168.0.254:30977] has been answered
>>>> 2012-10-20 01:44:52.925614 [DEBUG] switch_ivr_originate.c:3418 Originate
>>>> Resulted in Success: [sofia/internal/sip:100 at 192.168.0.254:30977]
>>>> 2012-10-20 01:44:52.925614 [DEBUG] switch_ivr_originate.c:3418 Originate
>>>> Resulted in Success: [sofia/internal/sip:100 at 192.168.0.254:30977]
>>>> 2012-10-20 01:44:52.925614 [DEBUG] mod_callcenter.c:1170 Updated tier:
>>>> Agent 100 at 192.168.0.102 in Queue support at 192.168.0.102 set state = Active
>>>> Inbound
>>>> 2012-10-20 01:44:52.945894 [DEBUG] mod_callcenter.c:1049 Updated Agent
>>>> 100 at 192.168.0.102 set state = In a queue call
>>>> 2012-10-20 01:44:52.945894 [DEBUG] mod_callcenter.c:1643 Agent
>>>> 100 at 192.168.0.102 answered "jeojit" <jeojit> from queue
>>>> support at 192.168.0.102
>>>> 2012-10-20 01:44:52.945894 [INFO] switch_channel.c:2806
>>>> sofia/internal/sip:100 at 192.168.0.254:30977 Flipping CID from "jeojit"
>>>> <jeojit> to "Outbound Call" <100>
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_ivr_bridge.c:1674
>>>> (skypopen/skype102) State Change CS_EXECUTE -> CS_HIBERNATE
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_session.c:1210 Send
>>>> signal skypopen/skype102 [BREAK]
>>>> 2012-10-20 01:44:52.945894 [DEBUG] mod_skypopen.c:727
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  727  ][skype102       ][UP,INPROGRS]
>>>> skypopen/skype102 CHANNEL got SWITCH_SIG_BREAK
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_ivr_bridge.c:1676
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) State Change CS_CONSUME_MEDIA
>>>> -> CS_HIBERNATE
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_session.c:1210 Send
>>>> signal sofia/internal/sip:100 at 192.168.0.254:30977 [BREAK]
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_session.c:821 Send signal
>>>> sofia/internal/sip:100 at 192.168.0.254:30977 [BREAK]
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_session.c:821 Send signal
>>>> skypopen/skype102 [BREAK]
>>>> 2012-10-20 01:44:52.945894 [DEBUG] mod_skypopen.c:727
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  727  ][skype102       ][UP,INPROGRS]
>>>> skypopen/skype102 CHANNEL got SWITCH_SIG_BREAK
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_state_machine.c:398
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) Running State Change
>>>> CS_HIBERNATE
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_state_machine.c:468
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) State HIBERNATE
>>>> 2012-10-20 01:44:52.945894 [DEBUG] mod_sofia.c:223
>>>> sofia/internal/sip:100 at 192.168.0.254:30977 SOFIA HIBERNATE
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_ivr_bridge.c:757
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) State Change CS_HIBERNATE ->
>>>> CS_RESET
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_session.c:1210 Send
>>>> signal sofia/internal/sip:100 at 192.168.0.254:30977 [BREAK]
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_state_machine.c:468
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) State HIBERNATE going to sleep
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_state_machine.c:398
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) Running State Change CS_RESET
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_state_machine.c:449
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) State RESET
>>>> 2012-10-20 01:44:52.945894 [DEBUG] mod_sofia.c:167
>>>> sofia/internal/sip:100 at 192.168.0.254:30977 SOFIA RESET
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_ivr_bridge.c:742
>>>> sofia/internal/sip:100 at 192.168.0.254:30977 CUSTOM RESET
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_state_machine.c:106
>>>> sofia/internal/sip:100 at 192.168.0.254:30977 Standard RESET
>>>> 2012-10-20 01:44:52.945894 [DEBUG] switch_core_state_machine.c:449
>>>> (sofia/internal/sip:100 at 192.168.0.254:30977) State RESET going to sleep
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:902
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  902  ][skype102       ][UP,INPROGRS] CHANNEL
>>>> READ FRAME goto CNG
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_ivr_play_say.c:1682 done
>>>> playing file local_stream://moh
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_callcenter.c:2684 Member jeojit
>>>> <jeojit> is answered by an agent in queue support at 192.168.0.102
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:1209
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  1209 ][skype102       ][UP,INPROGRS]
>>>> MSG_ID=27
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:453
>>>> (skypopen/skype102) State EXECUTE going to sleep
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:398
>>>> (skypopen/skype102) Running State Change CS_HIBERNATE
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:468
>>>> (skypopen/skype102) State HIBERNATE
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_ivr_bridge.c:757
>>>> (skypopen/skype102) State Change CS_HIBERNATE -> CS_RESET
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_session.c:1210 Send
>>>> signal skypopen/skype102 [BREAK]
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:727
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  727  ][skype102       ][UP,INPROGRS]
>>>> skypopen/skype102 CHANNEL got SWITCH_SIG_BREAK
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:468
>>>> (skypopen/skype102) State HIBERNATE going to sleep
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:398
>>>> (skypopen/skype102) Running State Change CS_RESET
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:449
>>>> (skypopen/skype102) State RESET
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:774
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  774  ][skype102       ][UP,INPROGRS]
>>>> skype102 CHANNEL RESET
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:783
>>>> (skypopen/skype102) State Change CS_RESET -> CS_HANGUP
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_session.c:1210 Send
>>>> signal skypopen/skype102 [BREAK]
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:727
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  727  ][skype102       ][UP,INPROGRS]
>>>> skypopen/skype102 CHANNEL got SWITCH_SIG_BREAK
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_ivr_bridge.c:742
>>>> skypopen/skype102 CUSTOM RESET
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:449
>>>> (skypopen/skype102) State RESET going to sleep
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:398
>>>> (skypopen/skype102) Running State Change CS_HANGUP
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_state_machine.c:638
>>>> (skypopen/skype102) State HANGUP
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:623
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  623  ][skype102       ][HANG_RQ,INPROGRS]
>>>> hanging up skype call: 119
>>>> 2012-10-20 01:44:52.965612 [DEBUG] skypopen_protocol.c:1675
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  1675 ][skype102       ][HANG_RQ,INPROGRS]
>>>> SENDING: |||ALTER CALL 119 END HANGUP||||
>>>> 2012-10-20 01:44:52.965612 [DEBUG] skypopen_protocol.c:1675
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  1675 ][skype102       ][HANG_RQ,INPROGRS]
>>>> SENDING: |||ALTER CALL 119 HANGUP||||
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:629
>>>> [464155c|07bc7ba] [DEBUG_SKYPE  629  ][skype102       ][HANG_RQ,INPROGRS]
>>>> skype102 CHANNEL HANGUP
>>>> 2012-10-20 01:44:52.965612 [DEBUG] mod_skypopen.c:648
>>>> (skypopen/skype102) State Change CS_HANGUP -> CS_DESTROY
>>>> 2012-10-20 01:44:52.965612 [DEBUG] switch_core_session.c:1210 Send
>>>> signal skypopen
>>>>
>>>> Pankaj
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gabe at gundy.org  Tue Nov  6 21:21:14 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 6 Nov 2012 11:21:14 -0700
Subject: [Freeswitch-users] A-leg hangup cause is blank
In-Reply-To: <F6D087C9-1D00-4567-A7E5-27A7E917E059@kavun.ch>
References: <51EC5321-C965-4553-9DE4-F84B1FC460DD@kavun.ch>
	<6B9182CC-772E-4B56-80BB-A9A2DDC32425@kavun.ch>
	<CAMwZ-tLJRodQ8Jbrm6uSrsM4OsKu7bWFAmYhqyWfFqDzMxB3AQ@mail.gmail.com>
	<F6D087C9-1D00-4567-A7E5-27A7E917E059@kavun.ch>
Message-ID: <CAMwZ-tJiY-sc2qU94qwb6-pmFKnFuR6arS=dy4eiKVRG589rKQ@mail.gmail.com>

On Tue, Nov 6, 2012 at 7:26 AM, Emrah <lists at kavun.ch> wrote:
> Thanks a bunch, I'm glad I escaped your famous replies. :P

Oh no... I didn't realize it's gone that far!  ;)

Happy hacking!


Gabe


From kbdfck at gmail.com  Tue Nov  6 22:01:59 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 6 Nov 2012 23:01:59 +0400
Subject: [Freeswitch-users] media proxy
In-Reply-To: <3414928.7.1352211231848.JavaMail.master@VoiceJuggler>
References: <CAAyX+KGSjJZ8ttMQ9WRFORUBzOoo=DXo2RiYsKKaqOJX4Qs-9g@mail.gmail.com>
	<3414928.7.1352211231848.JavaMail.master@VoiceJuggler>
Message-ID: <CA+GmxbbPpyymzuvECLPDd+8k-hwtPp1zBPQNMn4VrAWE5-eDuQ@mail.gmail.com>

Media proxy mode not only affects codecs, but any parameters negotiated by
SDP.
We use media proxy to handle NAT and still allow untouched SDP exchange
between customer endpoints and PSTN gateway for T38, for example

2012/11/6 Max Kovalenko <mkovalenko at cybervisiontech.com>

> I'm getting the data from external DB containing subscribers' parameters.
>
> Max Kovalenko
> ---------------
> CyberVision Inc.
>
> ----- Original Message -----
> From: "Ben Langfeld" <ben at langfeld.co.uk>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, November 4, 2012 10:46:00 PM
> Subject: Re: [Freeswitch-users] media proxy
>
>
> Can you point to other software that provides this distinction in media
> proxying?
>
> Regards,
> Ben Langfeld
>
>
>
> On 4 November 2012 11:57, Max Kovalenko < mkovalenko at cybervisiontech.com> wrote:
>
>
> Unfortunately, it isn't well documented, or else I would not ask about
> :))))))
>
> Those parameters perhaps are exclusive of each other, by mechanics they
> affects to is almost the same. Or at least, I would call them two sides of
> ame thing.
>
> Theoretically, if you consider the situation of bypass_media=true, that
> means media flow bypass around Freeswitch, it also automatically
> annihilates settings of proxy_media variable. It doesn't matter, even you
> would set proxy_media=true for same channel, it should not provide you
> "light proxy" of media flow, because your settign of bypass_proxy comprises
> Freeswitch don't proxy media.
> Thus I guess in order to set "light" or "hard" proxy the variable
> bypass_media must be set to FALSE first, for same channel of course.
>
> My task requires either bypass media or light/hard proxy to be set for a
> channel depending of some conditions of a session. In the other words, I
> need to set up one of three modes per each channel.
>
> Perhaps I undestand something not so right - I will appreciate if you will
> correct me with the following:
>
> 1. Default (full media proxy) : bypass_media=false ; proxy_media=false
>
> 2. Transparent proxy (mine "light proxy"): bypass_media=false;
> proxy_media=true
>
> 3. Bypass media: bupass_media=true (it's doesn't matter what currect value
> of proxy_media is)
>
> Thanks
>
>
> Max Kovalenko
> Team Leader
> VoIP & UC Team
> Managed Services Dept.
> CyberVision Inc.
> -------------------------------------------------
> tel. +1 (201) 585-9809 ext. 215
> Email: mkovalenko at cybervisiontech.com
> Skype: mkovalenko_cv, panzer_meister
> WWW: www.cybervisiontech.com
>
>
>
> ----- ???????? ????????? -----
> ??: "Ken Rice" < krice at freeswitch.org >
> ????: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> ????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????,
> ????????, ???????
> ????: Re: [Freeswitch-users] media proxy
>
>
> Default, proxy and bybass media are exclusive of each other, you can not
> combine them on a single call...
>
> Proxy media is just that proxy the media
>
> Bypass media is just that, bypass freeswitch and send the media direct
> between the end point...
>
> Setting this options is fairly well documented on the wiki...
>
>
>
>
> On 11/2/12 9:13 AM, "Max Kovalenko" < mkovalenko at cybervisiontech.com >
> wrote:
>
> > Hello,
> >
> > There are two modes of media stream to be proxied or bypassed.
> >
> > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
> etc.
> > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > traversal purposes. No media stream parsing is supported
> > 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> >
> > There are also two channel parameters affecting above modes:
> bypass_media and
> > proxy_media.
> >
> > - Are these parameters independent? Meaning the combination:
> bypass_media=true
> > AND proxy_media=true is possible. What effect will be?
> >
> > - Does it means that if I want to turn on "light proxy" I would always
> need to
> > set bypass_media=false AND proxy_media=true?
> >
> > - How to set "hard proxy" (full media proxy) per channel before to bridge
> > legs?
> >
> > Waiting for your replay ASAP. Thank you in advance.
> >
>
> > Best Regards.
> >
> > Max Kovalenko
> > Team Leader
> > VoIP & UC Team
> > Managed Services Dept.
> > CyberVision Inc.
> > -------------------------------------------------
> > tel. +1 (201) 585-9809 ext. 215
> > Email: mkovalenko at cybervisiontech.com
> > Skype: mkovalenko_cv, panzer_meister
> > WWW: www.cybervisiontech.com
> >
> >
> >
> >
>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From jaykris at gmail.com  Wed Nov  7 01:05:10 2012
From: jaykris at gmail.com (JP)
Date: Tue, 6 Nov 2012 14:05:10 -0800
Subject: [Freeswitch-users] mod_http_cache with mod_xml_curl
In-Reply-To: <CAHKQagGKuECyubTHX_UkfoSwNeqYE_xk3Y5a+aJsFm7CU_qCoQ@mail.gmail.com>
References: <CAMRwkEB-D9vaZqVDe4qrN_e4fdt98JzbgEWfktm8GW=76QQLQQ@mail.gmail.com>
	<CAHKQagGKuECyubTHX_UkfoSwNeqYE_xk3Y5a+aJsFm7CU_qCoQ@mail.gmail.com>
Message-ID: <CAMRwkEDkka0hKfKtL0XyWfxZ3ie8sE9ng==C6Y6eRiQbBdBAZA@mail.gmail.com>

Thanks Cal !
-JP

On Thu, Nov 1, 2012 at 1:16 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hi JP,
>
> This is actually mentioned in the documentation, but it wasn't very easy
> to find, so I have updated the wiki page to highlight this.
>
> Here is the updated URL;
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Caching_objects
>
> Please note, it would probably be less complex to place the caching logic
> into your web application instead.
>
> At a later stage - you could look at caching it directly within FreeSWITCH
> to get a bit more performance, but you probably won't need this level of
> performance to begin with.
>
> Hope this helps.
>
> Cal
>
> On Thu, Nov 1, 2012 at 6:54 PM, JP <jaykris at gmail.com> wrote:
>
>> Hi,
>> My project has a need to use dynamically generated dial plans(stored in
>> LDAP). But there are instances where I will need to fetch a dialplan that
>> is static most of the time but may change occasionally  I was wondering if
>> there is a way to use mod_xml_curl, but make it go through mod_http_cache
>> so that I can avoid unnecessary trips to the LDAP behind a web server when
>> not required? If you have any experience with something like this please
>> share it. Your help is greatly appreciated.
>>
>> -JP
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Wed Nov  7 01:09:26 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 6 Nov 2012 17:09:26 -0500
Subject: [Freeswitch-users] Bad connection diagnostics?
In-Reply-To: <CALNZuwmDanvi0iOKKCnm9=pwQi8gRO4tpsohXszq_iBQLx7L5A@mail.gmail.com>
References: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
	<CALNZuwmDanvi0iOKKCnm9=pwQi8gRO4tpsohXszq_iBQLx7L5A@mail.gmail.com>
Message-ID: <143301cdbc6b$6321bb50$296531f0$@bizfocused.com>

Thanks for the response AVI.  I should have been more clear.

 

This is a multi-tenant box.  We use it for my 3 offices too.  That is sort
of why I mentioned ping and tracert  results were the same, so I should have
filled in the rest.  The other tenants have no reported problems, and none
of our 3 sites (7 phones) seem to have these issues currently.  I have had
all of these issues at one time or another, but write it off to temporary
internet throughput variability issues/problems. 

 

I hate to use that excuse even when I truly think it is the problem.  But
when the problems are persistent (not constant) I am at a loss what to
capture/track/test.  They are looking to me to at least identify the problem
and hopefully get it resolved.

 

I do have the FS logs from at least one of these calls that had one way
audio, but I am not proficient enough to spot any audio channel problems.  I
can hang in there through dialplan issues and sofia connections/errors and
maybe even codec mismatches, but the audio connection information is still
one step too far for me.

 

Should I post those logs or can you offer another plan?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Tuesday, November 06, 2012 12:18 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bad connection diagnostics?

 

I would like an answer to this, too.

 

However, here's a place to start: you have 2 legs - you to their VPS, and
their VPS to their office phones.

Perhaps there's an issue with the VPS. Can you get their VPS to playback a
file and do an echo test? E.g. MOH or similar and call it directly and see
if any of the same issues persist. Either way, you've narrowed down where
the issue is...

 

-Avi

 

On Tue, Nov 6, 2012 at 6:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Hi,

I have a client who has been working fine for months, but as of late they
are reporting connections where the audio only works in one direction or
neither and "terrible echo".

 

I know they had some issues with their local Cable Connection, but they are
supposedly resolved.  Results from "ping -n 50 -l 256 " and "tracert" are
virtually identical to mine from here and I have no issues.   Are there some
diagnostics I could run to try and pin this down.  The phones are all cisco
504Gs, the server is a VPS from synapseglobal.

 

Thanks for any ideas you may have.

Sean


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




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

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

 

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From jaykris at gmail.com  Wed Nov  7 01:27:03 2012
From: jaykris at gmail.com (JP)
Date: Tue, 6 Nov 2012 14:27:03 -0800
Subject: [Freeswitch-users] Can I use mod_shout in a macro?
Message-ID: <CAMRwkEC6dD9=Uy4tJqk+AcQwXs64bfjSbwNKcGsw-Or3c8+cUQ@mail.gmail.com>

Is there a way that I can use mod_shout from a phrase macro?

Thanks
JP
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From spencer at 5ninesolutions.com  Wed Nov  7 02:09:42 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 6 Nov 2012 15:09:42 -0800
Subject: [Freeswitch-users] SIP to TDM t38 gateway
In-Reply-To: <5BA9AADC-BB63-4FF5-B1A6-1DEC1BE3E931@5ninesolutions.com>
References: <5BA9AADC-BB63-4FF5-B1A6-1DEC1BE3E931@5ninesolutions.com>
Message-ID: <CEB1E179-EB54-48F2-9298-8C8C1D403EA5@5ninesolutions.com>

Just curious if anyone has got this to work?  This is the dialplan that was on the wiki and also on Sangoma's wiki.  I've tried building several different versions going back to June all with the same behavior so I'm assuming there is a configuration error somewhere and not a bug.  Is there a was to get a higher level of debug to try to figure out why the media bug gets destroyed?

Thanks,
Spencer

On Nov 5, 2012, at 1:32 PM, Spencer Thomason wrote:

> Hello,
> I'm trying to use Freeswitch as a SIP to TDM gateway.  I'd like to use t38_gateway to detect fax tones and send a ReINVITE to t38.
> 
> Have a very minimal config with one profile that simply relays to FreeTDM
> 
> My dialplan is:
> <include>
>  <context name="sip">
> 
>    <extension name="dial-trunk">
>      <condition field="destination_number" expression="^(.*)$">
>        <action application="set" data="fax_enable_t38=true"/>
>        <action application="set" data="fax_enable_t38_request=true"/>
>        <action application="set" data="execute_on_answer=t38_gateway self"/>
>        <action application="bridge" data="freetdm/1/a/$1"/>
>      </condition>
>    </extension>
> 
>  </context>
> </include>
> 
> The problem is a media bug is created on the channel but almost immediately destroyed so fax tones are never detected. 
> 
> See:
> http://pastebin.freeswitch.org/20162
> 
> Thanks for any assistance,
> Spencer
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From philq at qsystemsengineering.com  Wed Nov  7 07:49:36 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Tue, 6 Nov 2012 20:49:36 -0800 (PST)
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCB96C43.5EDD5%krice@freeswitch.org>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
Message-ID: <1352263775999-7584364.post@n2.nabble.com>

This appears to have fixed the following issue that I never completely
finished supplying the diagnostic info for (sorry).  I noticed after a
recent update that the problem appears to have completely disappeared.  Now
I know why.  WooHoo!
http://jira.freeswitch.org/browse/FS-4152

Nice work.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Changes-to-how-ODBC-SQL-etc-works-tp7584221p7584364.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From philq at qsystemsengineering.com  Wed Nov  7 08:09:59 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Tue, 6 Nov 2012 21:09:59 -0800 (PST)
Subject: [Freeswitch-users] Inbound bypass media not working?
Message-ID: <1352264999635-7584365.post@n2.nabble.com>

FreeSwitch is still proxying media after adding param
name="inbound-bypass-media" value="true"/ (with enclosing <>) to
sofia.conf.xml.  When that didn't work, I added it to internal.xml as well. 
FS was restarted to ensure the changes were active but there was no change. 
I even tried setting inbound-no-media.  What could I be doing wrong?

I realize that not a lot of folks are using bypass-media (afaik) but if this
were a bug I'm sure that others would be complaining about it.

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inbound-bypass-media-not-working-tp7584365.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sparklezou at 163.com  Wed Nov  7 09:29:53 2012
From: sparklezou at 163.com (sparklezou)
Date: Wed, 7 Nov 2012 14:29:53 +0800
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
In-Reply-To: <4fad929.1ca9.13ad373167c.Coremail.sparklezou@163.com>
References: <CAM21H-PU=gk1x+yD34cOcxmV+=PuLHNHJZ__bHiAuO8H16z4XA@mail.gmail.com><711bb1b6.b10.13ad35a26bc.Coremail.sparklezou@163.com><4fad929.1ca9.13ad373167c.Coremail.sparklezou@163.com>
Message-ID: <a0ac8ad.46c8.13ad98f8e1e.Coremail.sparklezou@163.com>

Hi Ognjen,

I have checked the sip message log. There is NO sip "notification" message or any other kind message to notify all others in the same group.

How does FS implement it?

Please help on this case.

Thanks!

2012-11-07



sparklezou



????sparklezou
?????2012-11-06 10:01
???Re: Re: Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>,"Ognjen Seslija"<oseslija at gmail.com>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Ognjen,

There is a "Notify" message to inform all of the users in the Group? Correct?

How implment at FS side?

2012-11-06



sparklezou



????sparklezou
?????2012-11-06 09:33
???Re: Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>,"Ognjen Seslija"<oseslija at gmail.com>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Ognjen,

Could you please provide more info, which kind of phone support such feature?

Or it's tested on which kind of phones?

Thanks!

2012-11-06



sparklezou



????Ognjen Seslija
?????2012-11-06 02:24
???Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

That feature is phone dependent. Afaik it's working on snom, but not on Linksys/Cisco. 
FS already sets everything needed.


On Mon, Nov 5, 2012 at 9:54 AM, sparklezou <sparklezou at 163.com> wrote:

Hi Sir/Madam,

I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept

And also implement it on FS.

Here I want to know, does FS could implement such features? I know such features are working on some digital phone system.

1. "KAKA" & "GAGA" are in the same "Dial Groups".
2. When someone inside/outside call "KAKA", there will be visible sentens on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name), "12345678 call KAKA" (outside display the number). 

Is it posible to implement?

Thanks in advance!

2012-11-05



sparklezou

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From oseslija at gmail.com  Wed Nov  7 10:15:34 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 7 Nov 2012 08:15:34 +0100
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
In-Reply-To: <a0ac8ad.46c8.13ad98f8e1e.Coremail.sparklezou@163.com>
References: <CAM21H-PU=gk1x+yD34cOcxmV+=PuLHNHJZ__bHiAuO8H16z4XA@mail.gmail.com>
	<711bb1b6.b10.13ad35a26bc.Coremail.sparklezou@163.com>
	<4fad929.1ca9.13ad373167c.Coremail.sparklezou@163.com>
	<a0ac8ad.46c8.13ad98f8e1e.Coremail.sparklezou@163.com>
Message-ID: <CAM21H-OY-yPR3UrCXFZ=LzEF4G8Pt6xEWdGWrBgCPmc+1QfCqw@mail.gmail.com>

Afaik, Linksys SPA9xx series phones had possibility to subscribe to group
on their PBX, SPA9000.
When you press a group pickup soft key you would see a displayed info about
who's calling who, and you could intercept the call.

FS only send NOTIFY messages to phones that have subscribed to certain
extension, regardless of the call groups (standard rfc 3265 and 4235 type
presence). Phones generally light LED buttons in order to display the
status of the watched extension. With snom phones I have seen that it's
possible to get the display of the call in progress before picking the call
up (that's about what you want - A calling B).


On Wed, Nov 7, 2012 at 7:29 AM, sparklezou <sparklezou at 163.com> wrote:

> **
>  Hi Ognjen,
>
> I have checked the sip message log. There is NO sip "notification" message
> or any other kind message to notify all others in the same group.
>
> How does FS implement it?
>
> Please help on this case.
>
> Thanks!
>
> 2012-11-07
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-11-06 10:01
> *???*Re: Re: Re: [Freeswitch-users] About "Dial Groups" or "Callgroup
> intercept"
> *????*"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>,"Ognjen
> Seslija"<oseslija at gmail.com>
> *???*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
>
>  Hi Ognjen,
>
> There is a "Notify" message to inform all of the users in the Group?
> Correct?
>
> How implment at FS side?
>
> 2012-11-06
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-11-06 09:33
> *???*Re: Re: [Freeswitch-users] About "Dial Groups" or "Callgroup
> intercept"
> *????*"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>,"Ognjen
> Seslija"<oseslija at gmail.com>
> *???*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
>
>  Hi Ognjen,
>
> Could you please provide more info, which kind of phone support such
> feature?
>
> Or it's tested on which kind of phones?
>
> Thanks!
>
> 2012-11-06
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*Ognjen Seslija
> *?????*2012-11-06 02:24
> *???*Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
> *????*"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
> *???*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
>
> That feature is phone dependent. Afaik it's working on snom, but not on
> Linksys/Cisco.
> FS already sets everything needed.
>
> On Mon, Nov 5, 2012 at 9:54 AM, sparklezou <sparklezou at 163.com> wrote:
>
>> **
>> **
>> Hi Sir/Madam,
>>
>> I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept
>>
>> And also implement it on FS.
>>
>> Here I want to know, does FS could implement such features? I know such
>> features are working on some digital phone system.
>>
>> 1. "KAKA" & "GAGA" are in the same "Dial Groups".
>> 2. When someone inside/outside call "KAKA", there will be visible sentens
>> on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name),
>> "12345678 call KAKA" (outside display the number).
>>
>> Is it posible to implement?
>>
>> Thanks in advance!
>>
>> 2012-11-05
>> ------------------------------
>>  sparklezou
>> **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From vivlachaga at gmail.com  Wed Nov  7 08:47:32 2012
From: vivlachaga at gmail.com (=?iso-8859-1?Q?V=EDctor_Vladimir_Ch=E1vez_Gallardo?=)
Date: Tue, 6 Nov 2012 23:47:32 -0600
Subject: [Freeswitch-users] bypass_media between 2 outbound calls
In-Reply-To: <B838966E-C257-455D-8D68-92AE91B40795@gmail.com>
References: <B838966E-C257-455D-8D68-92AE91B40795@gmail.com>
Message-ID: <929CCE3B-4DE3-498D-8C52-0EA75C86B0C9@gmail.com>



Hi, i have an spidermonkey script, the script place two outbound calls, but i need to set the RTP directly between the legs (a-b) but i dont know hoy to set the parameter


i tried:
lega_session.setVariable('bypass_media', 'true');
bridge(lega_session,legb_session);	

and setting bypass_media also on both sessions before the bridge.. but i dont have the rtp in the correct way, instead my freeswitch it's behind the rtp flow

any idea?

V?ctor Vladimir Ch?vez Gallardo
vivlachaga at gmail.com

> 


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From rajprithiv88 at gmail.com  Wed Nov  7 11:03:22 2012
From: rajprithiv88 at gmail.com (Rajkumar K)
Date: Wed, 7 Nov 2012 13:33:22 +0530
Subject: [Freeswitch-users] FreeSWITCH High
	availability(Normal_temporary_failure problem)
Message-ID: <CAFYz+X=+BrXLChSegXxOw4hD+qFWdOM=V-_sOMhRYyt4A6Y1+A@mail.gmail.com>

Hi,

I am trying to achieve high availability in FreeSWITCH using heartbeat and
pacemaker and I am able to switch between the server whenever one of the
servers crashes. But the problem is one server is able to recover the calls
when im invoking sofia recover, but another server is recovering the call
one few times.(Mostly not able to recover).


I have primary and secondary server installed in two different
machines(centos), the FreeSWITCH instances are always running in both the
PCs. I am running heartbeat and pacemaker to monitor the IP or sofia
fail-over. A floating IP is configure in heartbeat to reach the active
server.
I succeeded in switching between the servers whenever IP or FreeSWITCH and
once it reaches the another server it invokes sofia recover to recover the
calls.

Both freeswtich instances are using the same configuration and database is
shared using ODBC connectivity.

Problem is:

Call is made using the primary server, and i did fsctl crash in primary
server cli. Heartbeat resource switches to secondary server and it invokes
"sofia profile internal restart" and "sofia recover". and it recovers the
call. The call gets recovered in 4-5 seconds.

At the same time i will start the freeswitch instance in the primary
server. Now if i crash the secondary server using fsctl crash, the
resources switches to primary server and it invokes "sofia profile internal
restart" and "sofia recover". Also the server sends invite request to the
clients But it ends in NORMAL_TEMPORARY_FAILURE. Wireshark log says Client
is responding with "Invalid CSeq" for the Server's INVITE request. This
happens always with the primary server and very few times primary server is
also able to recover the calls.

I have the same configurations in both the servers
And also i checked by stopping the heartbeat switching, and crashed the
primary server's freeswitch. Then if i start freeswitch again in the same
server and invoking sofia recover will recover the calls without any
problem.


I have also attached the cli logs of primary and secondary servers.
I am not able to identify the exact problem in this, Please help me out in
this problem.


Thanks
Rajkumar
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/* call recovered */

2012-10-31 09:19:22.639229 [NOTICE] switch_channel.c:941 New Channel sofia/internal/3606 at 10.1.3.50 [6a8db0f1-d26d-49a9-9498-093ec5174aae]
2012-10-31 09:19:22.639229 [NOTICE] switch_channel.c:939 Rename Channel sofia/internal/3606 at 10.1.3.50->sofia/internal/3606 at 10.1.3.50 [6a8db0f1-d26d-49a9-9498-093ec5174aae]
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/3606 at 10.1.3.50 PCMU/8000 20 ms 160 samples 64000 bits
2012-10-31 09:19:22.639229 [DEBUG] switch_core_codec.c:111 sofia/internal/3606 at 10.1.3.50 Original read codec set to PCMU:0
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3276 AUDIO RTP [sofia/internal/3606 at 10.1.3.50] 10.1.3.50 port 17242 -> 10.1.45.149 port 4000 codec: 0 ms: 20
2012-10-31 09:19:22.639229 [DEBUG] switch_rtp.c:1935 Not using a timer
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3540 Set 2833 dtmf send payload to 101
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3546 Set 2833 dtmf receive payload to 101
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3573 sofia/internal/3606 at 10.1.3.50 Set rtp dtmf delay to 40
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3579 Set comfort noise payload to 13
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:5965 (sofia/internal/3606 at 10.1.3.50) State Change CS_NEW -> CS_INIT
2012-10-31 09:19:22.639229 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3606 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.639229 [NOTICE] sofia_glue.c:5966 Resurrecting fallen channel sofia/internal/3606 at 10.1.3.50


2012-10-31 09:19:22.639229 [NOTICE] switch_channel.c:941 New Channel sofia/internal/3429 at 10.1.3.50 [b2da4eae-c696-434d-89ee-5ebcd978a668]
2012-10-31 09:19:22.639229 [NOTICE] switch_channel.c:939 Rename Channel sofia/internal/3429 at 10.1.3.50->sofia/internal/3429 at 10.1.3.50 [b2da4eae-c696-434d-89ee-5ebcd978a668]
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/3429 at 10.1.3.50 PCMU/8000 20 ms 160 samples 64000 bits
2012-10-31 09:19:22.639229 [DEBUG] switch_core_codec.c:111 sofia/internal/3429 at 10.1.3.50 Original read codec set to PCMU:0
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3606 at 10.1.3.50) Running State Change CS_INIT
2012-10-31 09:19:22.639229 [DEBUG] switch_channel.c:1936 (sofia/internal/3606 at 10.1.3.50) Callstate Change DOWN -> ACTIVE
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/3606 at 10.1.3.50) State INIT
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3276 AUDIO RTP [sofia/internal/3429 at 10.1.3.50] 10.1.3.50 port 25200 -> 10.1.45.148 port 4000 codec: 0 ms: 20
2012-10-31 09:19:22.639229 [DEBUG] mod_sofia.c:85 sofia/internal/3606 at 10.1.3.50 SOFIA INIT
2012-10-31 09:19:22.639229 [DEBUG] switch_rtp.c:1935 Not using a timer
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:2609 Local SDP:
v=0
o=FreeSWITCH 1351638120 1351638122 IN IP4 10.1.3.50
s=FreeSWITCH
c=IN IP4 10.1.3.50
t=0 0
m=audio 17242 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2012-10-31 09:19:22.639229 [DEBUG] mod_sofia.c:119 (sofia/internal/3606 at 10.1.3.50) State Change CS_INIT -> CS_RESET
2012-10-31 09:19:22.639229 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3606 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/3606 at 10.1.3.50) State INIT going to sleep
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3606 at 10.1.3.50) Running State Change CS_RESET
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3606 at 10.1.3.50) State RESET
2012-10-31 09:19:22.639229 [DEBUG] mod_sofia.c:166 sofia/internal/3606 at 10.1.3.50 SOFIA RESET
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:93 sofia/internal/3606 at 10.1.3.50 Standard RESET
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3606 at 10.1.3.50) State RESET going to sleep
2012-10-31 09:19:22.639229 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3606 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.639229 [DEBUG] sofia.c:6040 Channel sofia/internal/3606 at 10.1.3.50 entering state [calling][0]
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3540 Set 2833 dtmf send payload to 101
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3546 Set 2833 dtmf receive payload to 101
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3573 sofia/internal/3429 at 10.1.3.50 Set rtp dtmf delay to 40
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3579 Set comfort noise payload to 13
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:5965 (sofia/internal/3429 at 10.1.3.50) State Change CS_NEW -> CS_INIT
2012-10-31 09:19:22.639229 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.639229 [NOTICE] sofia_glue.c:5966 Resurrecting fallen channel sofia/internal/3429 at 10.1.3.50
2012-10-31 09:19:22.639229 [NOTICE] switch_channel.c:941 New Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [09de40c0-eba4-44a3-aa4d-d47ec4b2acf0]
2012-10-31 09:19:22.639229 [NOTICE] switch_channel.c:939 Rename Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292->sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [09de40c0-eba4-44a3-aa4d-d47ec4b2acf0]
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:ukytvsci at 10.1.19.195:1292 PCMU/8000 20 ms 160 samples 64000 bits
2012-10-31 09:19:22.639229 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Original read codec set to PCMU:0
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_INIT
2012-10-31 09:19:22.639229 [DEBUG] switch_channel.c:1936 (sofia/internal/3429 at 10.1.3.50) Callstate Change DOWN -> ACTIVE
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/3429 at 10.1.3.50) State INIT
2012-10-31 09:19:22.639229 [DEBUG] mod_sofia.c:85 sofia/internal/3429 at 10.1.3.50 SOFIA INIT
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3276 AUDIO RTP [sofia/internal/sip:ukytvsci at 10.1.19.195:1292] 10.1.3.50 port 18356 -> 10.1.19.195 port 50000 codec: 0 ms: 20
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:2609 Local SDP:
v=0
o=FreeSWITCH 1351630162 1351630164 IN IP4 10.1.3.50
s=FreeSWITCH
c=IN IP4 10.1.3.50
t=0 0
m=audio 25200 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2012-10-31 09:19:22.639229 [DEBUG] mod_sofia.c:119 (sofia/internal/3429 at 10.1.3.50) State Change CS_INIT -> CS_RESET
2012-10-31 09:19:22.639229 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/3429 at 10.1.3.50) State INIT going to sleep
2012-10-31 09:19:22.639229 [DEBUG] switch_rtp.c:1935 Not using a timer
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_RESET
2012-10-31 09:19:22.639229 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3429 at 10.1.3.50) State RESET
2012-10-31 09:19:22.639229 [DEBUG] mod_sofia.c:166 sofia/internal/3429 at 10.1.3.50 SOFIA RESET
2012-10-31 09:19:22.639229 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3540 Set 2833 dtmf send payload to 101
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3546 Set 2833 dtmf receive payload to 101
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3573 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Set rtp dtmf delay to 40
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:3579 Set comfort noise payload to 13
2012-10-31 09:19:22.639229 [DEBUG] sofia_glue.c:5965 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_NEW -> CS_INIT
2012-10-31 09:19:22.659361 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.659361 [NOTICE] sofia_glue.c:5966 Resurrecting fallen channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292
2012-10-31 09:19:22.659361 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_INIT
2012-10-31 09:19:22.659361 [DEBUG] switch_channel.c:1936 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Callstate Change DOWN -> ACTIVE
2012-10-31 09:19:22.659361 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State INIT
2012-10-31 09:19:22.659361 [DEBUG] mod_sofia.c:85 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 SOFIA INIT
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:2609 Local SDP:
v=0
o=FreeSWITCH 1351637006 1351637008 IN IP4 10.1.3.50
s=FreeSWITCH
c=IN IP4 10.1.3.50
t=0 0
m=audio 18356 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2012-10-31 09:19:22.659361 [DEBUG] mod_sofia.c:119 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_INIT -> CS_RESET
2012-10-31 09:19:22.659361 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.659361 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State INIT going to sleep
2012-10-31 09:19:22.659361 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_RESET
2012-10-31 09:19:22.659361 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State RESET
2012-10-31 09:19:22.659361 [DEBUG] mod_sofia.c:166 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 SOFIA RESET
2012-10-31 09:19:22.659361 [DEBUG] switch_core_state_machine.c:93 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Standard RESET
2012-10-31 09:19:22.659361 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State RESET going to sleep
2012-10-31 09:19:22.659361 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.659361 [DEBUG] sofia.c:6040 Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 entering state [calling][0]
2012-10-31 09:19:22.659361 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.659361 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.659361 [DEBUG] sofia.c:6040 Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 entering state [completing][200]
2012-10-31 09:19:22.659361 [DEBUG] sofia.c:6051 Remote SDP:
v=0^M
o=- 3560663913 3560663915 IN IP4 10.1.19.195^M
s=Blink 0.2.7 (Windows)^M
c=IN IP4 10.1.19.195^M
t=0 0^M
m=audio 50000 RTP/AVP 0 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=rtcp:50001^M

2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-10-31 09:19:22.659361 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.659361 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.659361 [DEBUG] sofia.c:6040 Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 entering state [ready][200]
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:2961 Already using PCMU
2012-10-31 09:19:22.659361 [DEBUG] sofia_glue.c:5158 Set 2833 dtmf send payload to 101
2012-10-31 09:19:22.679370 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.679370 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.739357 [INFO] switch_channel.c:2776 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Flipping CID from "Extension 3429" <3429> to "Outbound Call" <ukytvsci>
2012-10-31 09:19:22.739357 [DEBUG] switch_ivr_bridge.c:1672 (sofia/internal/3429 at 10.1.3.50) State Change CS_RESET -> CS_HIBERNATE
2012-10-31 09:19:22.739357 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.739357 [DEBUG] switch_ivr_bridge.c:1674 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_RESET -> CS_HIBERNATE
2012-10-31 09:19:22.739357 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.739357 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_HIBERNATE
2012-10-31 09:19:22.739357 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.739357 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State HIBERNATE
2012-10-31 09:19:22.739357 [DEBUG] mod_sofia.c:222 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 SOFIA HIBERNATE
2012-10-31 09:19:22.739357 [DEBUG] switch_ivr_bridge.c:757 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_HIBERNATE -> CS_RESET
2012-10-31 09:19:22.739357 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.739357 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State HIBERNATE going to sleep
2012-10-31 09:19:22.739357 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_RESET
2012-10-31 09:19:22.739357 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State RESET
2012-10-31 09:19:22.739357 [DEBUG] mod_sofia.c:166 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 SOFIA RESET
2012-10-31 09:19:22.739357 [DEBUG] switch_ivr_bridge.c:742 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 CUSTOM RESET
2012-10-31 09:19:22.739357 [DEBUG] switch_core_state_machine.c:93 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Standard RESET
2012-10-31 09:19:22.739357 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State RESET going to sleep
2012-10-31 09:19:22.739357 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.739357 [DEBUG] switch_ivr_bridge.c:742 sofia/internal/3429 at 10.1.3.50 CUSTOM RESET
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3429 at 10.1.3.50) State RESET going to sleep
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_HIBERNATE
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/3429 at 10.1.3.50) State HIBERNATE
2012-10-31 09:19:22.759770 [DEBUG] mod_sofia.c:222 sofia/internal/3429 at 10.1.3.50 SOFIA HIBERNATE
2012-10-31 09:19:22.759770 [DEBUG] switch_ivr_bridge.c:757 (sofia/internal/3429 at 10.1.3.50) State Change CS_HIBERNATE -> CS_RESET
2012-10-31 09:19:22.759770 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/3429 at 10.1.3.50) State HIBERNATE going to sleep
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_RESET
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3429 at 10.1.3.50) State RESET
2012-10-31 09:19:22.759770 [DEBUG] mod_sofia.c:166 sofia/internal/3429 at 10.1.3.50 SOFIA RESET
2012-10-31 09:19:22.759770 [DEBUG] switch_ivr_bridge.c:742 sofia/internal/3429 at 10.1.3.50 CUSTOM RESET
2012-10-31 09:19:22.759770 [DEBUG] switch_ivr_bridge.c:749 (sofia/internal/3429 at 10.1.3.50) State Change CS_RESET -> CS_SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3429 at 10.1.3.50) State RESET going to sleep
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/3429 at 10.1.3.50) State SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] mod_sofia.c:652 SOFIA SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_ivr_bridge.c:767 sofia/internal/3429 at 10.1.3.50 CUSTOM SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] sofia.c:6040 Channel sofia/internal/3429 at 10.1.3.50 entering state [calling][0]
2012-10-31 09:19:22.759770 [DEBUG] sofia.c:6040 Channel sofia/internal/3429 at 10.1.3.50 entering state [completing][200]
2012-10-31 09:19:22.759770 [DEBUG] sofia.c:6051 Remote SDP:
v=0^M
o=- 3560644143 3560644144 IN IP4 10.1.45.148^M
s=pjmedia^M
c=IN IP4 10.1.45.148^M
t=0 0^M
a=X-nat:0^M
m=audio 4000 RTP/AVP 0 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=rtcp:4001 IN IP4 10.1.45.148^M

2012-10-31 09:19:22.759770 [DEBUG] switch_ivr_bridge.c:799 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_RESET -> CS_SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.759770 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.759770 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] mod_sofia.c:652 SOFIA SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_ivr_bridge.c:767 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 CUSTOM SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:264 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Standard SOFT_EXECUTE
2012-10-31 09:19:22.759770 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State SOFT_EXECUTE going to sleep
2012-10-31 09:19:22.779360 [DEBUG] sofia.c:6040 Channel sofia/internal/3429 at 10.1.3.50 entering state [ready][200]
2012-10-31 09:19:22.779360 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-10-31 09:19:22.779360 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-10-31 09:19:22.779360 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-10-31 09:19:22.779360 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-10-31 09:19:22.779360 [DEBUG] sofia_glue.c:2961 Already using PCMU
2012-10-31 09:19:22.779360 [DEBUG] sofia_glue.c:5165 Set 2833 dtmf send/recv payload to 101
2012-10-31 09:19:22.779360 [DEBUG] switch_ivr_bridge.c:1257 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
2012-10-31 09:19:22.779360 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.779360 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_CONSUME_MEDIA
2012-10-31 09:19:22.779360 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State CONSUME_MEDIA
2012-10-31 09:19:22.779360 [DEBUG] switch_ivr_bridge.c:706 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 CUSTOM HOLD
2012-10-31 09:19:22.779360 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State CONSUME_MEDIA going to sleep
2012-10-31 09:19:22.779360 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.779360 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.779360 [DEBUG] switch_ivr_bridge.c:1359 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2012-10-31 09:19:22.779360 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.779360 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_EXCHANGE_MEDIA
2012-10-31 09:19:22.779360 [DEBUG] switch_core_state_machine.c:443 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State EXCHANGE_MEDIA
2012-10-31 09:19:22.779360 [DEBUG] mod_sofia.c:646 SOFIA EXCHANGE_MEDIA
2012-10-31 09:19:22.779360 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:19:22.779360 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:19:22.779360 [DEBUG] mod_sofia.c:2280 Not sending same id again "3429" <3429>
2012-10-31 09:19:22.779360 [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
2012-10-31 09:19:22.779360 [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
-------------- next part --------------
/*  call is not recovered */

2012-10-31 09:31:37.495317 [NOTICE] switch_channel.c:941 New Channel sofia/internal/3429 at 10.1.3.50 [b6f8eb4d-19d0-46ee-983d-81175e7f222b]
2012-10-31 09:31:37.495317 [NOTICE] switch_channel.c:939 Rename Channel sofia/internal/3429 at 10.1.3.50->sofia/internal/3429 at 10.1.3.50 [b6f8eb4d-19d0-46ee-983d-81175e7f222b]
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/3429 at 10.1.3.50 PCMU/8000 20 ms 160 samples 64000 bits
2012-10-31 09:31:37.495317 [DEBUG] switch_core_codec.c:111 sofia/internal/3429 at 10.1.3.50 Original read codec set to PCMU:0
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3276 AUDIO RTP [sofia/internal/3429 at 10.1.3.50] 10.1.3.50 port 25076 -> 10.1.45.148 port 4006 codec: 0 ms: 20
2012-10-31 09:31:37.495317 [DEBUG] switch_rtp.c:1935 Not using a timer
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3540 Set 2833 dtmf send payload to 101
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3546 Set 2833 dtmf receive payload to 101
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3573 sofia/internal/3429 at 10.1.3.50 Set rtp dtmf delay to 40
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3579 Set comfort noise payload to 13
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:5965 (sofia/internal/3429 at 10.1.3.50) State Change CS_NEW -> CS_INIT
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.495317 [NOTICE] sofia_glue.c:5966 Resurrecting fallen channel sofia/internal/3429 at 10.1.3.50
2012-10-31 09:31:37.495317 [NOTICE] switch_channel.c:941 New Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [048fd28b-765f-4444-9eb4-90487d72d191]
2012-10-31 09:31:37.495317 [NOTICE] switch_channel.c:939 Rename Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292->sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [048fd28b-765f-4444-9eb4-90487d72d191]
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:ukytvsci at 10.1.19.195:1292 PCMU/8000 20 ms 160 samples 64000 bits
2012-10-31 09:31:37.495317 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Original read codec set to PCMU:0
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_INIT
2012-10-31 09:31:37.495317 [DEBUG] switch_channel.c:1936 (sofia/internal/3429 at 10.1.3.50) Callstate Change DOWN -> ACTIVE
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3276 AUDIO RTP [sofia/internal/sip:ukytvsci at 10.1.19.195:1292] 10.1.3.50 port 26368 -> 10.1.19.195 port 50006 codec: 0 ms: 20
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/3429 at 10.1.3.50) State INIT
2012-10-31 09:31:37.495317 [DEBUG] mod_sofia.c:85 sofia/internal/3429 at 10.1.3.50 SOFIA INIT
2012-10-31 09:31:37.495317 [DEBUG] switch_rtp.c:1935 Not using a timer
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:2609 Local SDP:
v=0
o=FreeSWITCH 1351631021 1351631023 IN IP4 10.1.3.50
s=FreeSWITCH
c=IN IP4 10.1.3.50
t=0 0
m=audio 25076 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2012-10-31 09:31:37.495317 [DEBUG] mod_sofia.c:119 (sofia/internal/3429 at 10.1.3.50) State Change CS_INIT -> CS_RESET
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/3429 at 10.1.3.50) State INIT going to sleep
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_RESET
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3429 at 10.1.3.50) State RESET
2012-10-31 09:31:37.495317 [DEBUG] mod_sofia.c:166 sofia/internal/3429 at 10.1.3.50 SOFIA RESET
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3540 Set 2833 dtmf send payload to 101
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3546 Set 2833 dtmf receive payload to 101
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3573 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Set rtp dtmf delay to 40
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:3579 Set comfort noise payload to 13
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:5965 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_NEW -> CS_INIT
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.495317 [NOTICE] sofia_glue.c:5966 Resurrecting fallen channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_INIT
2012-10-31 09:31:37.495317 [DEBUG] switch_channel.c:1936 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Callstate Change DOWN -> ACTIVE
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State INIT
2012-10-31 09:31:37.495317 [DEBUG] mod_sofia.c:85 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 SOFIA INIT
2012-10-31 09:31:37.495317 [DEBUG] sofia_glue.c:2609 Local SDP:
v=0
o=FreeSWITCH 1351629729 1351629731 IN IP4 10.1.3.50
s=FreeSWITCH
c=IN IP4 10.1.3.50
t=0 0
m=audio 26368 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2012-10-31 09:31:37.495317 [DEBUG] mod_sofia.c:119 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_INIT -> CS_RESET
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State INIT going to sleep
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_RESET
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State RESET
2012-10-31 09:31:37.495317 [DEBUG] mod_sofia.c:166 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 SOFIA RESET
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:93 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Standard RESET
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State RESET going to sleep
2012-10-31 09:31:37.495317 [DEBUG] sofia.c:6045 Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 entering state [calling][0]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] sofia.c:6045 Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 entering state [terminated][500]
2012-10-31 09:31:37.495317 [DEBUG] switch_channel.c:2914 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Callstate Change ACTIVE -> HANGUP
2012-10-31 09:31:37.495317 [NOTICE] sofia.c:6837 Hangup sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [CS_RESET] [NORMAL_TEMPORARY_FAILURE]
2012-10-31 09:31:37.495317 [DEBUG] switch_channel.c:2937 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [KILL]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.495317 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_HANGUP
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State HANGUP
2012-10-31 09:31:37.515319 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 hanging up, cause: NORMAL_TEMPORARY_FAILURE
2012-10-31 09:31:37.515319 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.515319 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.515319 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State HANGUP going to sleep
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_HANGUP -> CS_REPORTING
2012-10-31 09:31:37.515319 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_REPORTING
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State REPORTING
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State REPORTING going to sleep
2012-10-31 09:31:37.515319 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State Change CS_REPORTING -> CS_DESTROY
2012-10-31 09:31:37.515319 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [BREAK]
2012-10-31 09:31:37.515319 [DEBUG] switch_core_session.c:1429 Session 2 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Locked, Waiting on external entities
2012-10-31 09:31:37.595338 [DEBUG] switch_ivr_bridge.c:1685 originatee uuid 048fd28b-765f-4444-9eb4-90487d72d191 is not present
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:93 sofia/internal/3429 at 10.1.3.50 Standard RESET
2012-10-31 09:31:37.595338 [NOTICE] switch_core_session.c:1447 Session 2 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Ended
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/3429 at 10.1.3.50) State RESET going to sleep
2012-10-31 09:31:37.595338 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:ukytvsci at 10.1.19.195:1292 [CS_DESTROY]
2012-10-31 09:31:37.595338 [DEBUG] sofia.c:6045 Channel sofia/internal/3429 at 10.1.3.50 entering state [calling][0]
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Callstate Change HANGUP -> DOWN
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) Running State Change CS_DESTROY
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State DESTROY
2012-10-31 09:31:37.595338 [DEBUG] mod_sofia.c:374 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 SOFIA DESTROY
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:ukytvsci at 10.1.19.195:1292 Standard DESTROY
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:ukytvsci at 10.1.19.195:1292) State DESTROY going to sleep
2012-10-31 09:31:37.595338 [DEBUG] sofia.c:6045 Channel sofia/internal/3429 at 10.1.3.50 entering state [terminated][500]
2012-10-31 09:31:37.595338 [DEBUG] switch_channel.c:2914 (sofia/internal/3429 at 10.1.3.50) Callstate Change ACTIVE -> HANGUP
2012-10-31 09:31:37.595338 [NOTICE] sofia.c:6837 Hangup sofia/internal/3429 at 10.1.3.50 [CS_RESET] [NORMAL_TEMPORARY_FAILURE]
2012-10-31 09:31:37.595338 [DEBUG] switch_channel.c:2937 Send signal sofia/internal/3429 at 10.1.3.50 [KILL]
2012-10-31 09:31:37.595338 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.595338 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_HANGUP
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/3429 at 10.1.3.50) State HANGUP
2012-10-31 09:31:37.615316 [DEBUG] mod_sofia.c:463 sofia/internal/3429 at 10.1.3.50 Overriding SIP cause 503 with 500 from the other leg
2012-10-31 09:31:37.615316 [DEBUG] mod_sofia.c:469 Channel sofia/internal/3429 at 10.1.3.50 hanging up, cause: NORMAL_TEMPORARY_FAILURE
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:47 sofia/internal/3429 at 10.1.3.50 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/3429 at 10.1.3.50) State HANGUP going to sleep
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/3429 at 10.1.3.50) State Change CS_HANGUP -> CS_REPORTING
2012-10-31 09:31:37.615316 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_REPORTING
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/3429 at 10.1.3.50) State REPORTING
2012-10-31 09:31:37.615316 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to DB: INSERT INTO cdr VALUES ("3429","3429","3606","MobEx_IP","2012-10-31 09:31:37","2012-10-31 09:31:51","2012-10-31 09:31:37",0,-14,"NORMAL_TEMPORARY_FAILURE","b6f8eb4d-19d0-46ee-983d-81175e7f222b","","3429")
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:79 sofia/internal/3429 at 10.1.3.50 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/3429 at 10.1.3.50) State REPORTING going to sleep
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/3429 at 10.1.3.50) State Change CS_REPORTING -> CS_DESTROY
2012-10-31 09:31:37.615316 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/3429 at 10.1.3.50 [BREAK]
2012-10-31 09:31:37.615316 [DEBUG] switch_core_session.c:1429 Session 1 (sofia/internal/3429 at 10.1.3.50) Locked, Waiting on external entities
2012-10-31 09:31:37.615316 [NOTICE] switch_core_session.c:1447 Session 1 (sofia/internal/3429 at 10.1.3.50) Ended
2012-10-31 09:31:37.615316 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/3429 at 10.1.3.50 [CS_DESTROY]
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/3429 at 10.1.3.50) Callstate Change HANGUP -> DOWN
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/3429 at 10.1.3.50) Running State Change CS_DESTROY
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/3429 at 10.1.3.50) State DESTROY
2012-10-31 09:31:37.615316 [DEBUG] mod_sofia.c:374 sofia/internal/3429 at 10.1.3.50 SOFIA DESTROY
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:86 sofia/internal/3429 at 10.1.3.50 Standard DESTROY
2012-10-31 09:31:37.615316 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/3429 at 10.1.3.50) State DESTROY going to sleep

From gabe at gundy.org  Wed Nov  7 11:32:53 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 7 Nov 2012 01:32:53 -0700
Subject: [Freeswitch-users] Directory entry variables not set on channel
 when user authenticated by cidr and domains ACL?
In-Reply-To: <CA+GmxbZzteFO-5OyHbcT-rVovzvS2TO-CzF7H9Px4Jvw77j5ag@mail.gmail.com>
References: <CA+GmxbZzteFO-5OyHbcT-rVovzvS2TO-CzF7H9Px4Jvw77j5ag@mail.gmail.com>
Message-ID: <CAMwZ-tK1mPP+fe1bH=50t-unW-bJ6zBadW=PQd-iohj5hPwUVg@mail.gmail.com>

On Tue, Nov 6, 2012 at 5:48 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Is this normal behaviour or I'm missing something? How can I get user
> variables from my DB set on channel when authenticating call by CIDR
> attribute?

When you're using the CIDR, you're not technically auth-ing, right?
Just a guess here (since I don't run an environment like yours), but
it's not going to be hitting the directory, right? Do you even see a
POST the to HTTP server asking for the directory in this case?

Anyway, this should do what you need:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user


Let us know what you find out.


Best,
Gabe


From mkovalenko at cybervisiontech.com  Wed Nov  7 11:47:19 2012
From: mkovalenko at cybervisiontech.com (Max Kovalenko)
Date: Wed, 7 Nov 2012 03:47:19 -0500 (EST)
Subject: [Freeswitch-users] media proxy
In-Reply-To: <CA+GmxbbPpyymzuvECLPDd+8k-hwtPp1zBPQNMn4VrAWE5-eDuQ@mail.gmail.com>
Message-ID: <18321355.161.1352278383986.JavaMail.master@VoiceJuggler>

Lets return to root direction of the topic.

How to set Full media proxy mode per channel?  

Meaning, what values should be set for channel variables: bypass_media & proxy_media ?

Max Kovalenko
---------------
CyberVision Inc.

----- Original Message -----
From: "Dmitry Sytchev" <kbdfck at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, November 6, 2012 9:01:59 PM
Subject: Re: [Freeswitch-users] media proxy


Media proxy mode not only affects codecs, but any parameters negotiated by SDP. 
We use media proxy to handle NAT and still allow untouched SDP exchange between customer endpoints and PSTN gateway for T38, for example 


2012/11/6 Max Kovalenko < mkovalenko at cybervisiontech.com > 


I'm getting the data from external DB containing subscribers' parameters. 

Max Kovalenko 
--------------- 
CyberVision Inc. 



----- Original Message ----- 
From: "Ben Langfeld" < ben at langfeld.co.uk > 
To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
Sent: Sunday, November 4, 2012 10:46:00 PM 
Subject: Re: [Freeswitch-users] media proxy 


Can you point to other software that provides this distinction in media proxying? 

Regards, 
Ben Langfeld 



On 4 November 2012 11:57, Max Kovalenko < mkovalenko at cybervisiontech.com > wrote: 


Unfortunately, it isn't well documented, or else I would not ask about :)))))) 

Those parameters perhaps are exclusive of each other, by mechanics they affects to is almost the same. Or at least, I would call them two sides of ame thing. 

Theoretically, if you consider the situation of bypass_media=true, that means media flow bypass around Freeswitch, it also automatically annihilates settings of proxy_media variable. It doesn't matter, even you would set proxy_media=true for same channel, it should not provide you "light proxy" of media flow, because your settign of bypass_proxy comprises Freeswitch don't proxy media. 
Thus I guess in order to set "light" or "hard" proxy the variable bypass_media must be set to FALSE first, for same channel of course. 

My task requires either bypass media or light/hard proxy to be set for a channel depending of some conditions of a session. In the other words, I need to set up one of three modes per each channel. 

Perhaps I undestand something not so right - I will appreciate if you will correct me with the following: 

1. Default (full media proxy) : bypass_media=false ; proxy_media=false 

2. Transparent proxy (mine "light proxy"): bypass_media=false; proxy_media=true 

3. Bypass media: bupass_media=true (it's doesn't matter what currect value of proxy_media is) 

Thanks 


Max Kovalenko 
Team Leader 
VoIP & UC Team 
Managed Services Dept. 
CyberVision Inc. 
------------------------------------------------- 
tel. +1 (201) 585-9809 ext. 215 
Email: mkovalenko at cybervisiontech.com 
Skype: mkovalenko_cv, panzer_meister 
WWW: www.cybervisiontech.com 



----- ???????? ????????? ----- 
??: "Ken Rice" < krice at freeswitch.org > 
????: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????, ????????, ??????? 
????: Re: [Freeswitch-users] media proxy 


Default, proxy and bybass media are exclusive of each other, you can not 
combine them on a single call... 

Proxy media is just that proxy the media 

Bypass media is just that, bypass freeswitch and send the media direct 
between the end point... 

Setting this options is fairly well documented on the wiki... 




On 11/2/12 9:13 AM, "Max Kovalenko" < mkovalenko at cybervisiontech.com > wrote: 

> Hello, 
> 
> There are two modes of media stream to be proxied or bypassed. 
> 
> 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc. 
> 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT 
> traversal purposes. No media stream parsing is supported 
> 3. Media bypass - RTP streams are bypassing FreeSWITCH at all. 
> 
> There are also two channel parameters affecting above modes: bypass_media and 
> proxy_media. 
> 
> - Are these parameters independent? Meaning the combination: bypass_media=true 
> AND proxy_media=true is possible. What effect will be? 
> 
> - Does it means that if I want to turn on "light proxy" I would always need to 
> set bypass_media=false AND proxy_media=true? 
> 
> - How to set "hard proxy" (full media proxy) per channel before to bridge 
> legs? 
> 
> Waiting for your replay ASAP. Thank you in advance. 
> 

> Best Regards. 
> 
> Max Kovalenko 
> Team Leader 
> VoIP & UC Team 
> Managed Services Dept. 
> CyberVision Inc. 
> ------------------------------------------------- 
> tel. +1 (201) 585-9809 ext. 215 
> Email: mkovalenko at cybervisiontech.com 
> Skype: mkovalenko_cv, panzer_meister 
> WWW: www.cybervisiontech.com 
> 
> 
> 
> 

> _________________________________________________________________________ 
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org 
> http://www.freeswitchsolutions.com 
> 
>  
>  
> 
> Official FreeSWITCH Sites 
> http://www.freeswitch.org 
> http://wiki.freeswitch.org 
> http://www.cluecon.com 
> 
> FreeSWITCH-users mailing list 
> FreeSWITCH-users at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
> http://www.freeswitch.org 


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





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

 
 

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

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

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

 
 

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

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


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

 
 

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

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

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

 
 

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

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




-- 
Best regards, 

Dmitry Sytchev, 
IT Engineer 

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




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

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


From sparklezou at 163.com  Wed Nov  7 12:43:19 2012
From: sparklezou at 163.com (sparklezou)
Date: Wed, 7 Nov 2012 17:43:19 +0800
Subject: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
In-Reply-To: <CAM21H-OY-yPR3UrCXFZ=LzEF4G8Pt6xEWdGWrBgCPmc+1QfCqw@mail.gmail.com>
References: <CAM21H-OY-yPR3UrCXFZ=LzEF4G8Pt6xEWdGWrBgCPmc+1QfCqw@mail.gmail.com>
Message-ID: <1bd8a758.6ebf.13ada40ac41.Coremail.sparklezou@163.com>

Hi Ognjen,

Correct. It seems the function I need.

"B", "C" in the same group.  In another word, "B" "C" subscribed the status of each other.

During the calling "A" calling "B", there should a disply on "C"'s phone, "A calling B". It should be in the ringing, "C" see "A calling B", then descided to pick up or NOT.

It should be useful for the Secretary.

OR, "B", "C", "D", "E",... belongs to the SIP group account "GP1". The member only need subscribe the SIP group account "GP1" status. Then could get the NOTIFY message.

Does it implement at FS side?

Thanks!

2012-11-07



sparklezou



????Ognjen Seslija
?????2012-11-07 15:15
???Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
???

Afaik, Linksys SPA9xx series phones had possibility to subscribe to group on their PBX, SPA9000.
When you press a group pickup soft key you would see a displayed info about who's calling who, and you could intercept the call.


FS only send NOTIFY messages to phones that have subscribed to certain extension, regardless of the call groups (standard rfc 3265 and 4235 type presence). Phones generally light LED buttons in order to display the status of the watched extension. With snom phones I have seen that it's possible to get the display of the call in progress before picking the call up (that's about what you want - A calling B).




On Wed, Nov 7, 2012 at 7:29 AM, sparklezou <sparklezou at 163.com> wrote:

Hi Ognjen,

I have checked the sip message log. There is NO sip "notification" message or any other kind message to notify all others in the same group.

How does FS implement it?

Please help on this case.

Thanks!

2012-11-07



sparklezou



????sparklezou
?????2012-11-06 10:01
???Re: Re: Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>,"Ognjen Seslija"<oseslija at gmail.com>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Ognjen,

There is a "Notify" message to inform all of the users in the Group? Correct?

How implment at FS side?

2012-11-06



sparklezou



????sparklezou
?????2012-11-06 09:33
???Re: Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>,"Ognjen Seslija"<oseslija at gmail.com>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Ognjen,

Could you please provide more info, which kind of phone support such feature?

Or it's tested on which kind of phones?

Thanks!

2012-11-06



sparklezou



????Ognjen Seslija
?????2012-11-06 02:24
???Re: [Freeswitch-users] About "Dial Groups" or "Callgroup intercept"
????"FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

That feature is phone dependent. Afaik it's working on snom, but not on Linksys/Cisco. 
FS already sets everything needed.


On Mon, Nov 5, 2012 at 9:54 AM, sparklezou <sparklezou at 163.com> wrote:

Hi Sir/Madam,

I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept

And also implement it on FS.

Here I want to know, does FS could implement such features? I know such features are working on some digital phone system.

1. "KAKA" & "GAGA" are in the same "Dial Groups".
2. When someone inside/outside call "KAKA", there will be visible sentens on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name), "12345678 call KAKA" (outside display the number). 

Is it posible to implement?

Thanks in advance!

2012-11-05



sparklezou

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From fdelawarde at wirelessmundi.com  Wed Nov  7 13:49:36 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 07 Nov 2012 11:49:36 +0100
Subject: [Freeswitch-users] Possible mod_smpp
In-Reply-To: <AANLkTimjx4Ggj5Aw2zM3eU0PQTOcy2ShY_2u1JDZFLhN@mail.gmail.com>
References: <AANLkTiksHkyBnVnbRamuNDK=kxfnE6jCoQdV1eWhXgbr@mail.gmail.com>
	<4D37457A.6040607@coppice.org>
	<AANLkTimjx4Ggj5Aw2zM3eU0PQTOcy2ShY_2u1JDZFLhN@mail.gmail.com>
Message-ID: <1352285376.19936.76.camel@luna.madrid.commsmundi.com>

I'm interested in a mod_smpp. Is anyone else?



On Thu, 2011-01-20 at 10:52 -0800, Michael Collins wrote:
> Our IRC buddy Delphiworld wants it for something. Math quoted him a
> price to do it and he decided that he doesn't want to pay it all
> himself so he's fishing around for other interested parties. Judging
> by the lack of response to my message I'd have to say that interest is
> still very weak...
> 
> 
> -MC
> 
> On Wed, Jan 19, 2011 at 12:11 PM, Steve Underwood <steveu at coppice.org>
> wrote:
>         
>         On 01/20/2011 03:15 AM, Michael Collins wrote:
>         > Hello all,
>         >
>         > If you are interested in giving monetary support to have a
>         > professional software firm create mod_smpp then please
>         contact me off
>         > list and I will give you more details.
>         >
>         > Thanks,
>         > Michael
>         
>         SMPP as in Short Message Peer to Peer? I have floated the idea
>         of open
>         sourcing an implementation of that a couple of times, but
>         interest
>         seemed weak and I never bothered.
>         
>         Steve
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sos at sokhapkin.dyndns.org  Wed Nov  7 14:25:49 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 07 Nov 2012 06:25:49 -0500
Subject: [Freeswitch-users] media proxy
In-Reply-To: <18321355.161.1352278383986.JavaMail.master@VoiceJuggler>
References: <18321355.161.1352278383986.JavaMail.master@VoiceJuggler>
Message-ID: <20194796.DB6nv3XX4f@sos>

 Set bypass_media=true and do not set proxy_media variable.

On Wednesday 07 November 2012 03:47:19 Max Kovalenko wrote:
> Lets return to root direction of the topic.
> 
> How to set Full media proxy mode per channel?
> 
> Meaning, what values should be set for channel variables: bypass_media &
> proxy_media ?
> 
> Max Kovalenko
> ---------------
> CyberVision Inc.
> 
> ----- Original Message -----
> From: "Dmitry Sytchev" <kbdfck at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, November 6, 2012 9:01:59 PM
> Subject: Re: [Freeswitch-users] media proxy
> 
> 
> Media proxy mode not only affects codecs, but any parameters negotiated by
> SDP. We use media proxy to handle NAT and still allow untouched SDP
> exchange between customer endpoints and PSTN gateway for T38, for example
> 
> 
> 2012/11/6 Max Kovalenko < mkovalenko at cybervisiontech.com >
> 
> 
> I'm getting the data from external DB containing subscribers' parameters.
> 
> Max Kovalenko
> ---------------
> CyberVision Inc.
> 
> 
> 
> ----- Original Message -----
> From: "Ben Langfeld" < ben at langfeld.co.uk >
> To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> Sent: Sunday, November 4, 2012 10:46:00 PM
> Subject: Re: [Freeswitch-users] media proxy
> 
> 
> Can you point to other software that provides this distinction in media
> proxying?
> 
> Regards,
> Ben Langfeld
> 
> 
> 
> On 4 November 2012 11:57, Max Kovalenko < mkovalenko at cybervisiontech.com >
> wrote:
> 
> 
> Unfortunately, it isn't well documented, or else I would not ask about
> :))))))
> 
> Those parameters perhaps are exclusive of each other, by mechanics they
> affects to is almost the same. Or at least, I would call them two sides of
> ame thing.
> 
> Theoretically, if you consider the situation of bypass_media=true, that
> means media flow bypass around Freeswitch, it also automatically
> annihilates settings of proxy_media variable. It doesn't matter, even you
> would set proxy_media=true for same channel, it should not provide you
> "light proxy" of media flow, because your settign of bypass_proxy comprises
> Freeswitch don't proxy media. Thus I guess in order to set "light" or
> "hard" proxy the variable bypass_media must be set to FALSE first, for same
> channel of course.
> 
> My task requires either bypass media or light/hard proxy to be set for a
> channel depending of some conditions of a session. In the other words, I
> need to set up one of three modes per each channel.
> 
> Perhaps I undestand something not so right - I will appreciate if you will
> correct me with the following:
> 
> 1. Default (full media proxy) : bypass_media=false ; proxy_media=false
> 
> 2. Transparent proxy (mine "light proxy"): bypass_media=false;
> proxy_media=true
> 
> 3. Bypass media: bupass_media=true (it's doesn't matter what currect value
> of proxy_media is)
> 
> Thanks
> 
> 
> Max Kovalenko
> Team Leader
> VoIP & UC Team
> Managed Services Dept.
> CyberVision Inc.
> -------------------------------------------------
> tel. +1 (201) 585-9809 ext. 215
> Email: mkovalenko at cybervisiontech.com
> Skype: mkovalenko_cv, panzer_meister
> WWW: www.cybervisiontech.com
> 
> 
> 
> ----- ???????? ????????? -----
> ??: "Ken Rice" < krice at freeswitch.org >
> ????: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> ????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????, ????????,
> ??????? ????: Re: [Freeswitch-users] media proxy
> 
> 
> Default, proxy and bybass media are exclusive of each other, you can not
> combine them on a single call...
> 
> Proxy media is just that proxy the media
> 
> Bypass media is just that, bypass freeswitch and send the media direct
> between the end point...
> 
> Setting this options is fairly well documented on the wiki...
> 
> On 11/2/12 9:13 AM, "Max Kovalenko" < mkovalenko at cybervisiontech.com > 
wrote:
> > Hello,
> > 
> > There are two modes of media stream to be proxied or bypassed.
> > 
> > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc.
> > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > traversal purposes. No media stream parsing is supported
> > 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> > 
> > There are also two channel parameters affecting above modes: bypass_media
> > and proxy_media.
> > 
> > - Are these parameters independent? Meaning the combination:
> > bypass_media=true AND proxy_media=true is possible. What effect will be?
> > 
> > - Does it means that if I want to turn on "light proxy" I would always
> > need to set bypass_media=false AND proxy_media=true?
> > 
> > - How to set "hard proxy" (full media proxy) per channel before to bridge
> > legs?
> > 
> > Waiting for your replay ASAP. Thank you in advance.
> > 
> > 
> > Best Regards.
> > 
> > Max Kovalenko
> > Team Leader
> > VoIP & UC Team
> > Managed Services Dept.
> > CyberVision Inc.
> > -------------------------------------------------
> > tel. +1 (201) 585-9809 ext. 215
> > Email: mkovalenko at cybervisiontech.com
> > Skype: mkovalenko_cv, panzer_meister
> > WWW: www.cybervisiontech.com
> > 
> > 
> > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 sos at sokhapkin.dyndns.org  Wed Nov  7 14:59:07 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 07 Nov 2012 06:59:07 -0500
Subject: [Freeswitch-users] media proxy
In-Reply-To: <20194796.DB6nv3XX4f@sos>
References: <18321355.161.1352278383986.JavaMail.master@VoiceJuggler>
	<20194796.DB6nv3XX4f@sos>
Message-ID: <1926824.3x4chkyD2N@sos>

 Oops, Full proxy is when neither variable is set to "true".

On Wednesday 07 November 2012 06:25:49 Sergey Okhapkin wrote:
>  Set bypass_media=true and do not set proxy_media variable.
> 
> On Wednesday 07 November 2012 03:47:19 Max Kovalenko wrote:
> > Lets return to root direction of the topic.
> > 
> > How to set Full media proxy mode per channel?
> > 
> > Meaning, what values should be set for channel variables: bypass_media &
> > proxy_media ?
> > 
> > Max Kovalenko
> > ---------------
> > CyberVision Inc.
> > 
> > ----- Original Message -----
> > From: "Dmitry Sytchev" <kbdfck at gmail.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Tuesday, November 6, 2012 9:01:59 PM
> > Subject: Re: [Freeswitch-users] media proxy
> > 
> > 
> > Media proxy mode not only affects codecs, but any parameters negotiated by
> > SDP. We use media proxy to handle NAT and still allow untouched SDP
> > exchange between customer endpoints and PSTN gateway for T38, for example
> > 
> > 
> > 2012/11/6 Max Kovalenko < mkovalenko at cybervisiontech.com >
> > 
> > 
> > I'm getting the data from external DB containing subscribers' parameters.
> > 
> > Max Kovalenko
> > ---------------
> > CyberVision Inc.
> > 
> > 
> > 
> > ----- Original Message -----
> > From: "Ben Langfeld" < ben at langfeld.co.uk >
> > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> > Sent: Sunday, November 4, 2012 10:46:00 PM
> > Subject: Re: [Freeswitch-users] media proxy
> > 
> > 
> > Can you point to other software that provides this distinction in media
> > proxying?
> > 
> > Regards,
> > Ben Langfeld
> > 
> > 
> > 
> > On 4 November 2012 11:57, Max Kovalenko < mkovalenko at cybervisiontech.com >
> > wrote:
> > 
> > 
> > Unfortunately, it isn't well documented, or else I would not ask about
> > 
> > :))))))
> > 
> > Those parameters perhaps are exclusive of each other, by mechanics they
> > affects to is almost the same. Or at least, I would call them two sides of
> > ame thing.
> > 
> > Theoretically, if you consider the situation of bypass_media=true, that
> > means media flow bypass around Freeswitch, it also automatically
> > annihilates settings of proxy_media variable. It doesn't matter, even you
> > would set proxy_media=true for same channel, it should not provide you
> > "light proxy" of media flow, because your settign of bypass_proxy
> > comprises
> > Freeswitch don't proxy media. Thus I guess in order to set "light" or
> > "hard" proxy the variable bypass_media must be set to FALSE first, for
> > same
> > channel of course.
> > 
> > My task requires either bypass media or light/hard proxy to be set for a
> > channel depending of some conditions of a session. In the other words, I
> > need to set up one of three modes per each channel.
> > 
> > Perhaps I undestand something not so right - I will appreciate if you will
> > correct me with the following:
> > 
> > 1. Default (full media proxy) : bypass_media=false ; proxy_media=false
> > 
> > 2. Transparent proxy (mine "light proxy"): bypass_media=false;
> > proxy_media=true
> > 
> > 3. Bypass media: bupass_media=true (it's doesn't matter what currect value
> > of proxy_media is)
> > 
> > Thanks
> > 
> > 
> > Max Kovalenko
> > Team Leader
> > VoIP & UC Team
> > Managed Services Dept.
> > CyberVision Inc.
> > -------------------------------------------------
> > tel. +1 (201) 585-9809 ext. 215
> > Email: mkovalenko at cybervisiontech.com
> > Skype: mkovalenko_cv, panzer_meister
> > WWW: www.cybervisiontech.com
> > 
> > 
> > 
> > ----- ???????? ????????? -----
> > ??: "Ken Rice" < krice at freeswitch.org >
> > ????: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> > ????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????,
> > ????????,
> > ??????? ????: Re: [Freeswitch-users] media proxy
> > 
> > 
> > Default, proxy and bybass media are exclusive of each other, you can not
> > combine them on a single call...
> > 
> > Proxy media is just that proxy the media
> > 
> > Bypass media is just that, bypass freeswitch and send the media direct
> > between the end point...
> > 
> > Setting this options is fairly well documented on the wiki...
> > 
> > On 11/2/12 9:13 AM, "Max Kovalenko" < mkovalenko at cybervisiontech.com >
> 
> wrote:
> > > Hello,
> > > 
> > > There are two modes of media stream to be proxied or bypassed.
> > > 
> > > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
> > > etc.
> > > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > > traversal purposes. No media stream parsing is supported
> > > 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> > > 
> > > There are also two channel parameters affecting above modes:
> > > bypass_media
> > > and proxy_media.
> > > 
> > > - Are these parameters independent? Meaning the combination:
> > > bypass_media=true AND proxy_media=true is possible. What effect will be?
> > > 
> > > - Does it means that if I want to turn on "light proxy" I would always
> > > need to set bypass_media=false AND proxy_media=true?
> > > 
> > > - How to set "hard proxy" (full media proxy) per channel before to
> > > bridge
> > > legs?
> > > 
> > > Waiting for your replay ASAP. Thank you in advance.
> > > 
> > > 
> > > Best Regards.
> > > 
> > > Max Kovalenko
> > > Team Leader
> > > VoIP & UC Team
> > > Managed Services Dept.
> > > CyberVision Inc.
> > > -------------------------------------------------
> > > tel. +1 (201) 585-9809 ext. 215
> > > Email: mkovalenko at cybervisiontech.com
> > > Skype: mkovalenko_cv, panzer_meister
> > > WWW: www.cybervisiontech.com
> > > 
> > > 
> > > 
> > > 
> > > 
> > > ________________________________________________________________________
> > > _
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > > 
> > > 
> > > 
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 kbdfck at gmail.com  Wed Nov  7 16:04:51 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 7 Nov 2012 17:04:51 +0400
Subject: [Freeswitch-users] media proxy
In-Reply-To: <20194796.DB6nv3XX4f@sos>
References: <18321355.161.1352278383986.JavaMail.master@VoiceJuggler>
	<20194796.DB6nv3XX4f@sos>
Message-ID: <CA+GmxbbFnKw4=x9eTPE=5oKiFhewidKBWR20hJiBo=PP-PpumA@mail.gmail.com>

This probably will not work if you have NATed endpoints
At least, without proper NAT handling on these devices to make them
advertise their external host/port in SDP

BTW, there is a variable *bypass_media_after_bridge* and it should make FS
send re-invite to endpoints connecting them directly

2012/11/7 Sergey Okhapkin <sos at sokhapkin.dyndns.org>

>  Set bypass_media=true and do not set proxy_media variable.
>
> On Wednesday 07 November 2012 03:47:19 Max Kovalenko wrote:
> > Lets return to root direction of the topic.
> >
> > How to set Full media proxy mode per channel?
> >
> > Meaning, what values should be set for channel variables: bypass_media &
> > proxy_media ?
> >
> > Max Kovalenko
> > ---------------
> > CyberVision Inc.
> >
> > ----- Original Message -----
> > From: "Dmitry Sytchev" <kbdfck at gmail.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Tuesday, November 6, 2012 9:01:59 PM
> > Subject: Re: [Freeswitch-users] media proxy
> >
> >
> > Media proxy mode not only affects codecs, but any parameters negotiated
> by
> > SDP. We use media proxy to handle NAT and still allow untouched SDP
> > exchange between customer endpoints and PSTN gateway for T38, for example
> >
> >
> > 2012/11/6 Max Kovalenko < mkovalenko at cybervisiontech.com >
> >
> >
> > I'm getting the data from external DB containing subscribers' parameters.
> >
> > Max Kovalenko
> > ---------------
> > CyberVision Inc.
> >
> >
> >
> > ----- Original Message -----
> > From: "Ben Langfeld" < ben at langfeld.co.uk >
> > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> > Sent: Sunday, November 4, 2012 10:46:00 PM
> > Subject: Re: [Freeswitch-users] media proxy
> >
> >
> > Can you point to other software that provides this distinction in media
> > proxying?
> >
> > Regards,
> > Ben Langfeld
> >
> >
> >
> > On 4 November 2012 11:57, Max Kovalenko < mkovalenko at cybervisiontech.com>
> > wrote:
> >
> >
> > Unfortunately, it isn't well documented, or else I would not ask about
> > :))))))
> >
> > Those parameters perhaps are exclusive of each other, by mechanics they
> > affects to is almost the same. Or at least, I would call them two sides
> of
> > ame thing.
> >
> > Theoretically, if you consider the situation of bypass_media=true, that
> > means media flow bypass around Freeswitch, it also automatically
> > annihilates settings of proxy_media variable. It doesn't matter, even you
> > would set proxy_media=true for same channel, it should not provide you
> > "light proxy" of media flow, because your settign of bypass_proxy
> comprises
> > Freeswitch don't proxy media. Thus I guess in order to set "light" or
> > "hard" proxy the variable bypass_media must be set to FALSE first, for
> same
> > channel of course.
> >
> > My task requires either bypass media or light/hard proxy to be set for a
> > channel depending of some conditions of a session. In the other words, I
> > need to set up one of three modes per each channel.
> >
> > Perhaps I undestand something not so right - I will appreciate if you
> will
> > correct me with the following:
> >
> > 1. Default (full media proxy) : bypass_media=false ; proxy_media=false
> >
> > 2. Transparent proxy (mine "light proxy"): bypass_media=false;
> > proxy_media=true
> >
> > 3. Bypass media: bupass_media=true (it's doesn't matter what currect
> value
> > of proxy_media is)
> >
> > Thanks
> >
> >
> > Max Kovalenko
> > Team Leader
> > VoIP & UC Team
> > Managed Services Dept.
> > CyberVision Inc.
> > -------------------------------------------------
> > tel. +1 (201) 585-9809 ext. 215
> > Email: mkovalenko at cybervisiontech.com
> > Skype: mkovalenko_cv, panzer_meister
> > WWW: www.cybervisiontech.com
> >
> >
> >
> > ----- ???????? ????????? -----
> > ??: "Ken Rice" < krice at freeswitch.org >
> > ????: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> > ????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????,
> ????????,
> > ??????? ????: Re: [Freeswitch-users] media proxy
> >
> >
> > Default, proxy and bybass media are exclusive of each other, you can not
> > combine them on a single call...
> >
> > Proxy media is just that proxy the media
> >
> > Bypass media is just that, bypass freeswitch and send the media direct
> > between the end point...
> >
> > Setting this options is fairly well documented on the wiki...
> >
> > On 11/2/12 9:13 AM, "Max Kovalenko" < mkovalenko at cybervisiontech.com >
> wrote:
> > > Hello,
> > >
> > > There are two modes of media stream to be proxied or bypassed.
> > >
> > > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
> etc.
> > > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > > traversal purposes. No media stream parsing is supported
> > > 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> > >
> > > There are also two channel parameters affecting above modes:
> bypass_media
> > > and proxy_media.
> > >
> > > - Are these parameters independent? Meaning the combination:
> > > bypass_media=true AND proxy_media=true is possible. What effect will
> be?
> > >
> > > - Does it means that if I want to turn on "light proxy" I would always
> > > need to set bypass_media=false AND proxy_media=true?
> > >
> > > - How to set "hard proxy" (full media proxy) per channel before to
> bridge
> > > legs?
> > >
> > > Waiting for your replay ASAP. Thank you in advance.
> > >
> > >
> > > Best Regards.
> > >
> > > Max Kovalenko
> > > Team Leader
> > > VoIP & UC Team
> > > Managed Services Dept.
> > > CyberVision Inc.
> > > -------------------------------------------------
> > > tel. +1 (201) 585-9809 ext. 215
> > > Email: mkovalenko at cybervisiontech.com
> > > Skype: mkovalenko_cv, panzer_meister
> > > WWW: www.cybervisiontech.com
> > >
> > >
> > >
> > >
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From kbdfck at gmail.com  Wed Nov  7 16:11:16 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 7 Nov 2012 17:11:16 +0400
Subject: [Freeswitch-users] Directory entry variables not set on channel
 when user authenticated by cidr and domains ACL?
In-Reply-To: <CAMwZ-tK1mPP+fe1bH=50t-unW-bJ6zBadW=PQd-iohj5hPwUVg@mail.gmail.com>
References: <CA+GmxbZzteFO-5OyHbcT-rVovzvS2TO-CzF7H9Px4Jvw77j5ag@mail.gmail.com>
	<CAMwZ-tK1mPP+fe1bH=50t-unW-bJ6zBadW=PQd-iohj5hPwUVg@mail.gmail.com>
Message-ID: <CA+GmxbZ+r_oiWgejNmw_yQGhnB0bJCsax0xNYvmAvN1nRX1ZQQ@mail.gmail.com>

Thanks, I'll try this

Technically ACL is not auth, but when I respond to directory request on
'reloadacl' and freeswitch startup, I pass all variables etc.
Seems freeswitch takes only user id and domain from this response, and then
sets 'variable_acl_token' and 'variable_sip_acl_token' to 'user_id at domain'
for matching cidr value.

So question is will FS make directory request on set_user or not :) I'll
try and share my results


2012/11/7 Gabriel Gunderson <gabe at gundy.org>

> On Tue, Nov 6, 2012 at 5:48 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > Is this normal behaviour or I'm missing something? How can I get user
> > variables from my DB set on channel when authenticating call by CIDR
> > attribute?
>
> When you're using the CIDR, you're not technically auth-ing, right?
> Just a guess here (since I don't run an environment like yours), but
> it's not going to be hitting the directory, right? Do you even see a
> POST the to HTTP server asking for the directory in this case?
>
> Anyway, this should do what you need:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
>
>
> Let us know what you find out.
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From abdlquadri at googlemail.com  Wed Nov  7 15:05:42 2012
From: abdlquadri at googlemail.com (Mumuney Abdlquadri)
Date: Wed, 7 Nov 2012 13:05:42 +0100
Subject: [Freeswitch-users] outbound socket connection erro
Message-ID: <CAGDHizk0Q-q3NasEMJ4QEQJQgMNdhgnz9PffzKmBEkppcNRu7A@mail.gmail.com>

Hi All,

I am using node-esl to setup an outbound connection.

I am on ubuntu. I have a softphone connected to  freeswitch from a win7
system on Virtualbox.

my dialplan is such:

<include>
  <extension name="abdlquadri.com">
    <condition field="destination_number" expression="^(1982)$">
<action application="sleep" data="1000"/>
<action application="set" data="name=abdlquadri"/>
      <action application="log" data="INFO ${name} is online!"/>
     <action application="socket" data="127.0.0.1:8022 syn full"/>
    </condition>
  </extension>
</include>

When I dial 1982 the output from freeswitch is such:

2012-11-07 12:40:04.403324 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/1000 at 10.0.2.2 [df4ad290-28cf-11e2-a3ea-bdbddea31077]
2012-11-07 12:40:04.403324 [INFO] mod_dialplan_xml.c:485 Processing
abdlquadri <1000>->1982 in context default
2012-11-07 12:40:05.403318 [INFO] mod_dptools.c:1420 abdlquadri is online!
2012-11-07 12:40:05.403318 [INFO] switch_core_session.c:2137 Sending early
media
2012-11-07 12:40:05.423325 [NOTICE] mod_sofia.c:2585 Pre-Answer
sofia/internal/1000 at 10.0.2.2!
2012-11-07 12:40:29.563345 [ERR] mod_event_socket.c:458 Socket Error!
2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:226
sofia/internal/1000 at 10.0.2.2 has executed the last dialplan instruction,
hanging up.
2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:228 Hangup
sofia/internal/1000 at 10.0.2.2 [CS_EXECUTE] [NORMAL_CLEARING]
2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1400 Session 6
(sofia/internal/1000 at 10.0.2.2) Ended
2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1402 Close
Channel sofia/internal/1000 at 10.0.2.2 [CS_DESTROY]


My node-esl code does not get called. Here is it:

var  esl = require('modesl');
var conn = new esl.Server({
    port:'8022',
    host:'127.0.0.1'
},function(){

    conn.on('connection::open', function(){
        conn.execute("answer");
        conn.getInfo();
        console.log('Connection Open');
    });
    conn.on('connection::ready', function(conne){
        conne.execute("answer");
        conne.getInfo();
        console.log('Connection Ready');
    });
    conn.on('connection::close', function(){
        conn.execute("answer");
        conn.getInfo();
        console.log('Connection Closed');
    });
    console.log(conn);
});


I guess the problem is this line: 2012-11-07 12:40:29.563345 [ERR]
mod_event_socket.c:458 Socket Error!

Please is there anything am doing wrong.

Thanks. All.
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From shayne.alone at gmail.com  Wed Nov  7 15:44:45 2012
From: shayne.alone at gmail.com (shayne.alone at gmail.com)
Date: Wed, 7 Nov 2012 16:14:45 +0330
Subject: [Freeswitch-users] SIP client authentication via ActiveDirectory
Message-ID: <CAK627Ozpb8USB4OVq1RZy7d0AwhxAEcUa5j9UwyGBsXH_7msyg@mail.gmail.com>

Hi all;
I'm looking for a way to authentication sip client with AD.
here is a simple wiki <http://wiki.freeswitch.org/wiki/Mod_ldap> and i had
enabled the module to load.
but there is no traffic toward to AD at all! how can i debug or better is
this all for LDAP authentication of sip clients?
should configure directory to use LDAP API anywhere else?


-- 
Regards,
Ali R. Taleghani <a.taleghani at aut.ac.ir>
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From shayne.alone at gmail.com  Wed Nov  7 16:04:08 2012
From: shayne.alone at gmail.com (shayne.alone at gmail.com)
Date: Wed, 7 Nov 2012 16:34:08 +0330
Subject: [Freeswitch-users] SIP client authentication via ActiveDirectory
In-Reply-To: <CAK627Ozpb8USB4OVq1RZy7d0AwhxAEcUa5j9UwyGBsXH_7msyg@mail.gmail.com>
References: <CAK627Ozpb8USB4OVq1RZy7d0AwhxAEcUa5j9UwyGBsXH_7msyg@mail.gmail.com>
Message-ID: <CAK627OzqgAOux52GuTjk=aCj+dKa3hCLRt+-5-mnv-Qfjnec=A@mail.gmail.com>

am I working in a wrong way at all?! I mean there is no more doc,wiki for
this purpose!
is this a wrong way with I am trying to auth with?
should I try radius instead or some what else :-/

On Wed, Nov 7, 2012 at 4:14 PM, shayne.alone at gmail.com <
shayne.alone at gmail.com> wrote:

> Hi all;
> I'm looking for a way to authentication sip client with AD.
> here is a simple wiki <http://wiki.freeswitch.org/wiki/Mod_ldap> and i
> had enabled the module to load.
> but there is no traffic toward to AD at all! how can i debug or better is
> this all for LDAP authentication of sip clients?
> should configure directory to use LDAP API anywhere else?
>
>
> --
> Regards,
> Ali R. Taleghani <a.taleghani at aut.ac.ir>
>
>


-- 
Regards,
Ali R. Taleghani <a.taleghani at aut.ac.ir>
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From vbvbrj at gmail.com  Wed Nov  7 17:45:42 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 07 Nov 2012 16:45:42 +0200
Subject: [Freeswitch-users] mod_callcenter missed calls.
Message-ID: <509A7416.7050509@gmail.com>

Hello.

I was asked about missed calls increasing when the agent was talking to 
some client from queue. The strange thing is that mod_callcenter does 
not attempt to contact an agent which is in state of answering a call 
("in a queue call" state) as I understand from log. But despite this 
fact and the fact that the agent does not get its missing call counter 
increased in mod_callcenter's database, the phone still somehow get an 
attempt and increases its internal count of missed call. I don't 
understand how it is, because if the phone were receiving the second 
call, it will show on the display as "calling extension..." and if the 
second parallel call were not answered then it must increase the count. 
But no second line is shown.

Is this a problem of FS trying to contact an active answering agent or 
it is the phone which messes up? The phone is DPH-150S.


-- 
Mimiko desu.


From steveayre at gmail.com  Wed Nov  7 18:40:46 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Nov 2012 15:40:46 +0000
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCBABF54.5EF0B%krice@freeswitch.org>
References: <CAMwZ-t+9jFFtmPZfyTvFLdTJRGgjJEcAeZnzQ5a7bmBBXEmj4Q@mail.gmail.com>
	<CCBABF54.5EF0B%krice@freeswitch.org>
Message-ID: <CAFiqYumDsfQ5N9bLfTVLZmYMWAqZ02OGbo6x_qZfkrqUdZ6LHA@mail.gmail.com>

>
> B) There is that whole threadsafe vs non-threadsafe mysql client lib issue,


In recent versions they're the same (ie it's all threadsafe now) so this
should only be an issue on older systems..

libmysqlclient_r still exists for legacy code, but is now just a symlink to
libmysqlclient. Possibly it'd disappear at some point.

I can't remember if that change was in 5.5 or 5.1 though, and there would
be plenty of 5.0/5.1 systems around still. But since libmysqlclient18 (5.5)
is threadsafe and can connect to older servers a simple way to enforce it
would be to require that as a minimum version perhaps.

There's also the mysql_config binary, which has a --libs_r option. Only
using that one should ensure on all versions that only the thread-safe
library is used.

-Steve



On 3 November 2012 17:32, Ken Rice <krice at freeswitch.org> wrote:

> While this may solve the issue... This does not address a number of other
> issues
>
> A) 90+% of the primary FreeSWITCH developers use PostgreSQL primarily...
>
> B) There is that whole threadsafe vs non-threadsafe mysql client lib issue,
> so we would have to come up with a way in the build system to detect which
> we have, and never use the non-thread safe one... (if we don't do this, the
> code would never be stable as we could never know for sure which one
> actually got detected and linked against)
>
>
>
>
>
> On 11/3/12 1:40 AM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>
> > On Fri, Nov 2, 2012 at 11:55 PM, curriegrad2004
> > <curriegrad2004 at gmail.com> wrote:
> >> Quite impossible. The reason is because of the license that the MySQL
> >> libs are licensed under. iirc, they are GPL licensed and they are not
> >> compatible with the MPL that FreeSWITCH uses.
> >
> > I've got PostgreSQL, so I've got all I need :) But, it seems like this
> > legal issue is easily resolved:
> >
> > http://www.mysql.com/about/legal/licensing/foss-exception/
> >
> > Anyway, I'm not a lawyer or even a very thorough reader ;)
> >
> >
> > Best,
> > Gabe
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From luis.daniel.lucio at gmail.com  Wed Nov  7 19:31:50 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 7 Nov 2012 10:31:50 -0600
Subject: [Freeswitch-users] Connecting delays too much sometims
Message-ID: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>

I have FS 1.2.3 with updated 1 week ago vbilling installed,

some destinations delays to much to start connecting, others are too quick.

Wondering if you tell me some topics i shall review to find the bootleneck.

Using mod_memcache will help?


Regards,

LD


From steveayre at gmail.com  Wed Nov  7 19:49:29 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Nov 2012 16:49:29 +0000
Subject: [Freeswitch-users] Bad connection diagnostics?
In-Reply-To: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
References: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
Message-ID: <7739B897-151C-415D-9A10-130AB2AE2F83@gmail.com>

One way audio will usually be due to NAT problems. Eg the IP in the SDP is an internal IP so it can send audio but the remote end doesn't know where to send audio to in the other direction.

Echo on a VoIP call would be due to one of the endpoints.

Sent from my iPad



On 6 Nov 2012, at 16:41, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Hi,
> I have a client who has been working fine for months, but as of late they are reporting connections where the audio only works in one direction or neither and ?terrible echo?.
>  
> I know they had some issues with their local Cable Connection, but they are supposedly resolved.  Results from ?ping ?n 50 ?l 256 ? and ?tracert? are virtually identical to mine from here and I have no issues.   Are there some diagnostics I could run to try and pin this down.  The phones are all cisco 504Gs, the server is a VPS from synapseglobal.
>  
> Thanks for any ideas you may have.
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Wed Nov  7 19:49:29 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Nov 2012 16:49:29 +0000
Subject: [Freeswitch-users] Bad connection diagnostics?
In-Reply-To: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
References: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
Message-ID: <7739B897-151C-415D-9A10-130AB2AE2F83@gmail.com>

One way audio will usually be due to NAT problems. Eg the IP in the SDP is an internal IP so it can send audio but the remote end doesn't know where to send audio to in the other direction.

Echo on a VoIP call would be due to one of the endpoints.

Sent from my iPad



On 6 Nov 2012, at 16:41, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Hi,
> I have a client who has been working fine for months, but as of late they are reporting connections where the audio only works in one direction or neither and ?terrible echo?.
>  
> I know they had some issues with their local Cable Connection, but they are supposedly resolved.  Results from ?ping ?n 50 ?l 256 ? and ?tracert? are virtually identical to mine from here and I have no issues.   Are there some diagnostics I could run to try and pin this down.  The phones are all cisco 504Gs, the server is a VPS from synapseglobal.
>  
> Thanks for any ideas you may have.
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Wed Nov  7 19:52:48 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Nov 2012 16:52:48 +0000
Subject: [Freeswitch-users] Bad connection diagnostics?
In-Reply-To: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
References: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
Message-ID: <66DFCA29-90E1-4B48-8FB3-66E774A49FDE@gmail.com>

Incidentally you may also get timing issues on a VPS since other guests/the host can unpredictably steal CPU time away from your guest, so it's harder to get reliable timings. That'd most likely present itself as lags or audio glitches where frames are dropped.

Shouldn't cause one way audio or echo though.

Sent from my iPad



On 6 Nov 2012, at 16:41, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Hi,
> I have a client who has been working fine for months, but as of late they are reporting connections where the audio only works in one direction or neither and ?terrible echo?.
>  
> I know they had some issues with their local Cable Connection, but they are supposedly resolved.  Results from ?ping ?n 50 ?l 256 ? and ?tracert? are virtually identical to mine from here and I have no issues.   Are there some diagnostics I could run to try and pin this down.  The phones are all cisco 504Gs, the server is a VPS from synapseglobal.
>  
> Thanks for any ideas you may have.
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Wed Nov  7 19:52:48 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Nov 2012 16:52:48 +0000
Subject: [Freeswitch-users] Bad connection diagnostics?
In-Reply-To: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
References: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
Message-ID: <66DFCA29-90E1-4B48-8FB3-66E774A49FDE@gmail.com>

Incidentally you may also get timing issues on a VPS since other guests/the host can unpredictably steal CPU time away from your guest, so it's harder to get reliable timings. That'd most likely present itself as lags or audio glitches where frames are dropped.

Shouldn't cause one way audio or echo though.

Sent from my iPad



On 6 Nov 2012, at 16:41, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Hi,
> I have a client who has been working fine for months, but as of late they are reporting connections where the audio only works in one direction or neither and ?terrible echo?.
>  
> I know they had some issues with their local Cable Connection, but they are supposedly resolved.  Results from ?ping ?n 50 ?l 256 ? and ?tracert? are virtually identical to mine from here and I have no issues.   Are there some diagnostics I could run to try and pin this down.  The phones are all cisco 504Gs, the server is a VPS from synapseglobal.
>  
> Thanks for any ideas you may have.
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From avi at avimarcus.net  Wed Nov  7 19:53:56 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 7 Nov 2012 18:53:56 +0200
Subject: [Freeswitch-users] Connecting delays too much sometims
In-Reply-To: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>
References: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>
Message-ID: <CALNZuw=yjuh+ME9t+2e0UMDk8_642pQEEBYhH=7Otev==3n9+w@mail.gmail.com>

What's too much of a delay? something in vbilling? The LCR query? the PDD
on your gateway?
And what could possibly be too quick?

If it's solely v-billing please contact them and not the FS list in general.

Please ask a more descriptive question. A pastebin'ed log showing an
example of your issue might help.

-Avi


On Wed, Nov 7, 2012 at 6:31 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> I have FS 1.2.3 with updated 1 week ago vbilling installed,
>
> some destinations delays to much to start connecting, others are too quick.
>
> Wondering if you tell me some topics i shall review to find the bootleneck.
>
> Using mod_memcache will help?
>
>
> Regards,
>
> LD
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.venugopan at mundio.com  Wed Nov  7 20:11:51 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Wed, 7 Nov 2012 17:11:51 +0000
Subject: [Freeswitch-users] Calling number lookup
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2328259@Mail-Kilo.squay.com>

Hi,

Which script will lookup the destination number?

Say for example if we are dialling 750**** number, which script will lookup corresponding database and dial this number

Regards,
Archana

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From krice at freeswitch.org  Wed Nov  7 20:11:56 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 07 Nov 2012 11:11:56 -0600
Subject: [Freeswitch-users] Weekly News and Notes or something
Message-ID: <CCBFF27C.5F18D%krice@freeswitch.org>

Hey Guys,

Mr Collins is under the weather this week, so I?m trying to fill in for him.

Don?t forget today is the Weekly FreeSWITCH Community Conference Call at 1PM
Easter, 10AM Pacific or 1700 UTC.
Join us on the bridge.
And of course, notes for this week along with how to connect can be found
via the Web at http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_07

Coming today FreeSWITCH 1.2.4, git has already been tagged and source
tarballs are available at http://files.freeswitch.org/

And this Friday don?t forget to join us for the Friday Free For All, 888
will be open, join us unmute and have some fun.

If you have a project or topic you feel would be of interest for the
FreeSWITCH community in general and would like to do a presentation on the
weekly conference call contact me offlist. And as always suggestions for
topics on the weekly conference call are always welcomed.


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

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From sdevoy at bizfocused.com  Wed Nov  7 20:37:56 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 7 Nov 2012 12:37:56 -0500
Subject: [Freeswitch-users] Connecting delays too much sometims
In-Reply-To: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>
References: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>
Message-ID: <002301cdbd0e$9f7020d0$de506270$@bizfocused.com>

We had this issue as well.  Unfortunately at least in our case, this is
entirely a problem with your provider.  We changed to "Premium Rates"  and
got MUCH faster connections.  "Least Cost Routing" can be VERY SLOW.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis
Daniel Lucio Quiroz
Sent: Wednesday, November 07, 2012 11:32 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Connecting delays too much sometims

I have FS 1.2.3 with updated 1 week ago vbilling installed,

some destinations delays to much to start connecting, others are too quick.

Wondering if you tell me some topics i shall review to find the bootleneck.

Using mod_memcache will help?


Regards,

LD

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




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

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




From nbhatti at gmail.com  Wed Nov  7 20:37:52 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Wed, 7 Nov 2012 20:37:52 +0300
Subject: [Freeswitch-users] Connecting delays too much sometims
In-Reply-To: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>
References: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>
Message-ID: <845DA134-EAF7-46E9-9C8D-DEA0FC5B7D9C@gmail.com>


Could be an issue with your  providers? Did you try swap the providers? 

Thanks,
--
Muhammad Naseer Bhatti



On Nov 7, 2012, at 7:31 PM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com> wrote:

> I have FS 1.2.3 with updated 1 week ago vbilling installed,
> 
> some destinations delays to much to start connecting, others are too quick.
> 
> Wondering if you tell me some topics i shall review to find the bootleneck.
> 
> Using mod_memcache will help?
> 
> 
> Regards,
> 
> LD
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From avi at avimarcus.net  Wed Nov  7 20:46:02 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 7 Nov 2012 19:46:02 +0200
Subject: [Freeswitch-users] Calling number lookup
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2328259@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2328259@Mail-Kilo.squay.com>
Message-ID: <CALNZuwnpHS8qa-buB+1+A0axmnkVEru8JPLKxN7sXc1zcikhJQ@mail.gmail.com>

When you dial anything, it "hits the dialplan" for that context and follows
those instructions.

This might help explain the dialplan:
http://wiki.freeswitch.org/wiki/Dialplan_XML -- but it sounds like the FS
book would help more.

Or are you asking about mod_lcr to do least cost routing with several
gateways?

-Avi


On Wed, Nov 7, 2012 at 7:11 PM, Archana Venugopan <a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> Which script will lookup the destination number?****
>
> ** **
>
> Say for example if we are dialling 750**** number, which script will
> lookup corresponding database and dial this number****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Wed Nov  7 20:46:01 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 07 Nov 2012 11:46:01 -0600
Subject: [Freeswitch-users] Calling number lookup
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2328259@Mail-Kilo.squay.com>
Message-ID: <CCBFFA79.5F196%krice@freeswitch.org>

There are many ways to do this from mod_xml_curl to mod_easyroute to custom
scripts written in any of the languages embedded in freeswitch such as Lua,
Java or Mono...

I would suggest doing some research via the FreeSWITCH Wiki. (its heavily
indexed by google) to narrow down a method that?s good for you. Also the
FreeSwitch books available via multiple online sources as eBooks are handy
for this type of research also

K


On 11/7/12 11:11 AM, "Archana Venugopan" <a.venugopan at mundio.com> wrote:

> Hi,
>  
> Which script will lookup the destination number?
>  
> Say for example if we are dialling 750**** number, which script will lookup
> corresponding database and dial this number
>  
> Regards,
> Archana
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From drk at drkngs.net  Wed Nov  7 20:48:57 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Wed, 07 Nov 2012 09:48:57 -0800
Subject: [Freeswitch-users] Possible mod_smpp
In-Reply-To: <1352285376.19936.76.camel@luna.madrid.commsmundi.com>
Message-ID: <20121107174857.9be49bc4@mail.tritonwest.net>

I would be, however is FS the right place to put this? Since most SMS backbone providers would only give you 1 connection, it would make sense to have some type of SMPP/XMPP gateway for multiple FS's to connect to.   
   
--Dave
      _____  

  From: Fran?ois Delawarde [mailto:fdelawarde at wirelessmundi.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Wed, 07 Nov 2012 02:49:36 -0800
Subject: Re: [Freeswitch-users] Possible mod_smpp

I'm interested in a mod_smpp. Is anyone else?



On Thu, 2011-01-20 at 10:52 -0800, Michael Collins wrote:
> Our IRC buddy Delphiworld wants it for something. Math quoted him a
> price to do it and he decided that he doesn't want to pay it all
> himself so he's fishing around for other interested parties. Judging
> by the lack of response to my message I'd have to say that interest is
> still very weak...
> 
> 
> -MC
> 
> On Wed, Jan 19, 2011 at 12:11 PM, Steve Underwood <steveu at coppice.org>
> wrote:
> 
> On 01/20/2011 03:15 AM, Michael Collins wrote:
> > Hello all,
> >
> > If you are interested in giving monetary support to have a
> > professional software firm create mod_smpp then please
> contact me off
> > list and I will give you more details.
> >
> > Thanks,
> > Michael
> 
> SMPP as in Short Message Peer to Peer? I have floated the idea
> of open
> sourcing an implementation of that a couple of times, but
> interest
> seemed weak and I never bothered.
> 
> Steve
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
      
   
 
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From philq at qsystemsengineering.com  Wed Nov  7 20:58:45 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Wed, 07 Nov 2012 12:58:45 -0500
Subject: [Freeswitch-users] Connecting delays too much sometimes
Message-ID: <027b01cdbd11$88800f40$99802dc0$@com>

Are you using Alcazar for termination, by chance?  They're having an issue
with that right now, where calls to certain destinations take around 12
seconds to proceed, while others proceed in a normal fashion.  It appears to
be occuring when they attempt to terminate the calls through Broadvox.  I
provided them with some diagnostic info and they said they were looking into
the issue.  That was a couple of days ago.

Phil Quesinberry
Q Systems Engineering, Inc.
Embedded Systems Development and VoIP Business Telephone Hosting
Improve your business telephone services and save money
(410) 969-8002
http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/> 


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From philq at qsystemsengineering.com  Wed Nov  7 21:24:07 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Wed, 07 Nov 2012 13:24:07 -0500
Subject: [Freeswitch-users] Connecting delays too much sometimes
Message-ID: <028c01cdbd15$13fb1f80$3bf15e80$@com>

I was wrong about that . just tried again to a different number and it was
terminated through CoreTel this time with the same delay.  So it's
definitely an Alcazar problem.

The rep told me that they use FreeSWITCH and Kamilio when we opened an
account, perhaps the FS database change has something to do with it. not
sure how the integration with Kamilio is set up for them and whether or not
it would involve the database.  They've been having this problem for close
to a couple of weeks now though, I think.

- Phil
_____________________________________________
From: Phil Quesinberry
Sent: Wednesday, November 07, 2012 12:59 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Connecting delays too much sometimes


Are you using Alcazar for termination, by chance?  They're having an issue
with that right now, where calls to certain destinations take around 12
seconds to proceed, while others proceed in a normal fashion.  It appears to
be occuring when they attempt to terminate the calls through Broadvox.  I
provided them with some diagnostic info and they said they were looking into
the issue.  That was a couple of days ago.

Phil Quesinberry
Q Systems Engineering, Inc.
Embedded Systems Development and VoIP Business Telephone Hosting
Improve your business telephone services and save money
(410) 969-8002
http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/> 


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From steveayre at gmail.com  Wed Nov  7 21:26:49 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Nov 2012 18:26:49 +0000
Subject: [Freeswitch-users] outbound socket connection erro
In-Reply-To: <CAGDHizk0Q-q3NasEMJ4QEQJQgMNdhgnz9PffzKmBEkppcNRu7A@mail.gmail.com>
References: <CAGDHizk0Q-q3NasEMJ4QEQJQgMNdhgnz9PffzKmBEkppcNRu7A@mail.gmail.com>
Message-ID: <CAFiqYu=LdYaz1pW2h2iYfXsVJc7LJWa3inAmcJcfdO6Pvv5pKw@mail.gmail.com>

>
> <action application="socket" data="127.0.0.1:8022 syn full"/>


Try "127.0.0.1:8022 syn*c* full"

Also verify your program is indeed listening on 127.0.0.1:8022 (use
netstat).

-Steve





On 7 November 2012 12:05, Mumuney Abdlquadri <abdlquadri at googlemail.com>wrote:

> Hi All,
>
> I am using node-esl to setup an outbound connection.
>
> I am on ubuntu. I have a softphone connected to  freeswitch from a win7
> system on Virtualbox.
>
> my dialplan is such:
>
> <include>
>   <extension name="abdlquadri.com">
>     <condition field="destination_number" expression="^(1982)$">
> <action application="sleep" data="1000"/>
> <action application="set" data="name=abdlquadri"/>
>       <action application="log" data="INFO ${name} is online!"/>
>      <action application="socket" data="127.0.0.1:8022 syn full"/>
>     </condition>
>   </extension>
> </include>
>
> When I dial 1982 the output from freeswitch is such:
>
> 2012-11-07 12:40:04.403324 [NOTICE] switch_channel.c:926 New Channel
> sofia/internal/1000 at 10.0.2.2 [df4ad290-28cf-11e2-a3ea-bdbddea31077]
> 2012-11-07 12:40:04.403324 [INFO] mod_dialplan_xml.c:485 Processing
> abdlquadri <1000>->1982 in context default
> 2012-11-07 12:40:05.403318 [INFO] mod_dptools.c:1420 abdlquadri is online!
> 2012-11-07 12:40:05.403318 [INFO] switch_core_session.c:2137 Sending early
> media
> 2012-11-07 12:40:05.423325 [NOTICE] mod_sofia.c:2585 Pre-Answer
> sofia/internal/1000 at 10.0.2.2!
> 2012-11-07 12:40:29.563345 [ERR] mod_event_socket.c:458 Socket Error!
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:226
> sofia/internal/1000 at 10.0.2.2 has executed the last dialplan instruction,
> hanging up.
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:228 Hangup
> sofia/internal/1000 at 10.0.2.2 [CS_EXECUTE] [NORMAL_CLEARING]
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1400 Session 6
> (sofia/internal/1000 at 10.0.2.2) Ended
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1402 Close
> Channel sofia/internal/1000 at 10.0.2.2 [CS_DESTROY]
>
>
> My node-esl code does not get called. Here is it:
>
> var  esl = require('modesl');
> var conn = new esl.Server({
>     port:'8022',
>     host:'127.0.0.1'
> },function(){
>
>     conn.on('connection::open', function(){
>         conn.execute("answer");
>         conn.getInfo();
>         console.log('Connection Open');
>     });
>     conn.on('connection::ready', function(conne){
>         conne.execute("answer");
>         conne.getInfo();
>         console.log('Connection Ready');
>     });
>     conn.on('connection::close', function(){
>         conn.execute("answer");
>         conn.getInfo();
>         console.log('Connection Closed');
>     });
>     console.log(conn);
> });
>
>
> I guess the problem is this line: 2012-11-07 12:40:29.563345 [ERR]
> mod_event_socket.c:458 Socket Error!
>
> Please is there anything am doing wrong.
>
> Thanks. All.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Nov  7 21:54:58 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Nov 2012 18:54:58 +0000
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCB96C43.5EDD5%krice@freeswitch.org>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
Message-ID: <CAFiqYumsDyQ1s_9wj6H571coNeD=s5G=Pb9gu3_28snfGzU61w@mail.gmail.com>

Since I can see this becoming a popular question (it got me today), I've
put some notes up on the Wiki.

FAQ entry:
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_I.27ve_upgraded_and_my_ODBC_DSNs_no_longer_work.21

New "Release Notes" page where we can stick notes on behaviour changing
stuff for people to review when upgrading:
http://wiki.freeswitch.org/wiki/Release_Notes

Which is linked to from the Which_Version_Should_I_Use template:
http://wiki.freeswitch.org/wiki/<http://wiki.freeswitch.org/wiki/Which_Version_Should_I_Use>
Which_Version_Should_I_Use<http://wiki.freeswitch.org/wiki/Which_Version_Should_I_Use>
Which puts it on the front page and Download/Installation pages.

A note link is also at the top of the Download/Installation page to make it
a little more prominent.

-Steve



On 2 November 2012 17:25, Ken Rice <krice at freeswitch.org> wrote:

>  Hey Guys,
>
> There?s some new Database Goodness in the core of FreeSWITCH that can lead
> to some unexpected things for you guys updating existing installations
> using ODBC.
>
> We now have Native PostgreSQL support in the core, and along with this
> comes some changes to the various ?odbc-dsn? settings around the tree.
>
> If you are using the format ?dsn:username:password? you wont be affected,
> however if you are just specifying a DSN as ?dsn? you will need to listen up
>
> The settings for this field have changed.
>
> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> options='-c client_min_messages=NOTICE for postgresql (the stuff after
> pgsql:// is a standard libpq connect string for you programmer types)
> odbc://dns:username:password  for ODBC   ( dsn:: should also work or
> dns:username: )
> sqlite://filename   for sqlite different SQLite Databases
>
> I think we still need to doc this up good, but its there and its coming
> strong...
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yiftah at choochee.com  Wed Nov  7 21:56:51 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Wed, 7 Nov 2012 10:56:51 -0800
Subject: [Freeswitch-users] Calling number lookup
In-Reply-To: <CALNZuwnpHS8qa-buB+1+A0axmnkVEru8JPLKxN7sXc1zcikhJQ@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2328259@Mail-Kilo.squay.com>
	<CALNZuwnpHS8qa-buB+1+A0axmnkVEru8JPLKxN7sXc1zcikhJQ@mail.gmail.com>
Message-ID: <CAFVUaV_YLBNmsggZ8x2Q_uTkmqskVbrJbeeqj+HA8dNYEdEkTA@mail.gmail.com>

I think he means shortcut numbers translation to real numbers


On Wed, Nov 7, 2012 at 9:46 AM, Avi Marcus <avi at avimarcus.net> wrote:

> When you dial anything, it "hits the dialplan" for that context and
> follows those instructions.
>
> This might help explain the dialplan:
> http://wiki.freeswitch.org/wiki/Dialplan_XML -- but it sounds like the FS
> book would help more.
>
> Or are you asking about mod_lcr to do least cost routing with several
> gateways?
>
> -Avi
>
>
> On Wed, Nov 7, 2012 at 7:11 PM, Archana Venugopan <a.venugopan at mundio.com>wrote:
>
>>  Hi,****
>>
>> ** **
>>
>> Which script will lookup the destination number?****
>>
>> ** **
>>
>> Say for example if we are dialling 750**** number, which script will
>> lookup corresponding database and dial this number****
>>
>> ** **
>>
>> Regards,****
>>
>> Archana****
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vivlachaga at gmail.com  Wed Nov  7 22:25:20 2012
From: vivlachaga at gmail.com (=?iso-8859-1?Q?V=EDctor_Vladimir_Ch=E1vez_Gallardo?=)
Date: Wed, 7 Nov 2012 13:25:20 -0600
Subject: [Freeswitch-users] Connecting RTP between leg A and B directly
Message-ID: <7C47175E-5B4E-43E7-A869-184F8BD81900@gmail.com>

Hi, i have an spidermonkey script, the script place two outbound calls, but i need to set the RTP directly between the legs (a-b) but i dont know hoy to set the parameter bypass_media

i tried:
lega_session.setVariable('bypass_media', 'true');
bridge(lega_session,legb_session);	

also 

lega_session.setVariable('bypass_media', 'true');
legb_session.setVariable('bypass_media', 'true');
bridge(lega_session,legb_session);	

and also setting another variable bypass_media_after_bridge

lega_session.setVariable('bypass_media_after_bridge', 'true');
legb_session.setVariable('bypass_media_after_bridge', 'true');
lega_session.setVariable('bypass_media', 'true');
legb_session.setVariable('bypass_media', 'true');
bridge(lega_session,legb_session);	


but i dont have the RTP in the correct way, instead my freeswitch it's behind, the rtp flow:  LEG-A -----> FREESWITCH -------> LEG-B

any idea?


From djsnickles at yahoo.com  Thu Nov  8 00:41:23 2012
From: djsnickles at yahoo.com (tehfink)
Date: Wed, 7 Nov 2012 13:41:23 -0800 (PST)
Subject: [Freeswitch-users]  I could really use help with my ITSP outage,
	translate asterisk to freeswitch
Message-ID: <1352324483.79292.yext-apple-iphone@web160602.mail.bf1.yahoo.com>

Hi Mario, what Freeswitch settings did you use to restore service?

I agree with you, the FS wiki shows a parm to turn DNSRV off so I am pretty sure it's on by default. Anyway, we're back up. Thanks for the note!




From sdevoy at bizfocused.com  Thu Nov  8 01:52:45 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 7 Nov 2012 17:52:45 -0500
Subject: [Freeswitch-users] Bad connection diagnostics?
In-Reply-To: <66DFCA29-90E1-4B48-8FB3-66E774A49FDE@gmail.com>
References: <11be01cdbc3d$9850ee00$c8f2ca00$@bizfocused.com>
	<66DFCA29-90E1-4B48-8FB3-66E774A49FDE@gmail.com>
Message-ID: <02d501cdbd3a$9a2521d0$ce6f6570$@bizfocused.com>

AH HA!  Thanks Steve.  They also reported ?huge laaaaaaaagggg? on one instance.  That makes perfect sense for that.

 

However, I find NAT issue questionable in this case since they have been working for so long and the network hardware has not changed.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, November 07, 2012 11:53 AM
To: FreeSWITCH Users Help
Cc: <FreeSWITCH-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Bad connection diagnostics?

 

Incidentally you may also get timing issues on a VPS since other guests/the host can unpredictably steal CPU time away from your guest, so it's harder to get reliable timings. That'd most likely present itself as lags or audio glitches where frames are dropped.

 

Shouldn't cause one way audio or echo though.

Sent from my iPad

 

 


On 6 Nov 2012, at 16:41, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

Hi,

I have a client who has been working fine for months, but as of late they are reporting connections where the audio only works in one direction or neither and ?terrible echo?.

 

I know they had some issues with their local Cable Connection, but they are supposedly resolved.  Results from ?ping ?n 50 ?l 256 ? and ?tracert? are virtually identical to mine from here and I have no issues.   Are there some diagnostics I could run to try and pin this down.  The phones are all cisco 504Gs, the server is a VPS from synapseglobal.

 

Thanks for any ideas you may have.

Sean

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




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

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

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From geddes.jeffrey at gmail.com  Thu Nov  8 02:31:30 2012
From: geddes.jeffrey at gmail.com (Jeff Geddes)
Date: Wed, 7 Nov 2012 19:31:30 -0400
Subject: [Freeswitch-users] Newbie routing question
Message-ID: <CALAD2r7VhQQv_=XNC-3ceDuQWugaRmyJkDtyys6jDED4Rjfd6g@mail.gmail.com>

I've been able to receive a call from a landline thru a gateway and route
it to my softphone.

can also make an outbound call from my softphone to a landline.

I need to 'tie' the two together, receive an inbound call and using a
dialplan route it as an outbound call to the number in the dialplan.

what would be the proper approach to do this?

thanks
jeff
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From mani at tungli.net  Thu Nov  8 02:44:03 2012
From: mani at tungli.net (=?ISO-8859-1?Q?Gunnlaugur_M=E1ni_Hr=F3lfsson?=)
Date: Wed, 7 Nov 2012 23:44:03 +0000
Subject: [Freeswitch-users] Problem with digest authentication
Message-ID: <CAAupo5QzEg42+g30sWH2j-QFCxweT6XS9C0QYqZNPRy1mkR1BQ@mail.gmail.com>

Hello all,

I'm having a problem with a particular device, that I'm hoping some one on
this list can help me with. The device is a Telsey IAD.

At first the device registers just fine. But when the registration expires
and the device re-registers, freeswitch responds with a 403 Forbidden.

I realized that this must be because either the device og Freeswitch wasn't
calculating the digest response correctly. So I tried to calculate it
myself. This is a sample sent from the device, that Freeswitch responded
with a 403 to:

REGISTER sip:test.sip.is SIP/2.0
Call-ID: 5f880-1485c-aab33cd at 10.171.32.9
From: <sip:4151664 at test.sip.is;user=phone>;tag=2f5c0-1485c
To: <sip:4151664 at test.sip.is;user=phone>
CSeq: 104 REGISTER
Via: SIP/2.0/UDP 10.171.32.9;branch=z9hG4bK-2f5c0-101315
Contact: <sip:4151664 at 10.171.32.9;user=phone>
Max-Forwards: 70
User-Agent: Telsey ver4_49_25
Expires: 30
Supported: timer
Authorization: Digest username="4151664",realm="test.sip.is",uri="sip:
test.sip.is
",response="c3b5cd8147fb883fbbe9acb183838717",algorithm=MD5,nonce="9202dbb4-292d-11e2-bac2-399d85a4015c",qop=auth,cnonce="0002f818",nc=00000001
Content-Length: 0

Assuming a password of "1234", the digest response should
be 041323244fc4bdac8d49872bf29d9a3a. But the device seems to increment the
nc value to 00000002 and calculates the response
as c3b5cd8147fb883fbbe9acb183838717.

The problem is that the device sends nc=00000001 so freeswitch uses that
value to calculate the digest, while the device uses 00000002.

I think it is obviously the device that's broken but not Freeswitch. But,
in this case I need to get this device to work as is. Is there any way I
can have freeswitch authenticate the device without using digest method
that requires a nonce count, like rfc2069?

Best regards,
Mani Hrolfsson
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From lists at kavun.ch  Thu Nov  8 03:46:52 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 7 Nov 2012 19:46:52 -0500
Subject: [Freeswitch-users] Multiple profiles on a single domain
Message-ID: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>

Hi Freeswitchers,

I had to start an additional profile on port 5070 because my Verizon MiFi blocks 5060 completely.
For now, 5070 allows me to have a call for 120 seconds? Than the RTP session drops.

Here is how I started my new profile. Sounds too simple to be true!
cp internal.xml internal-5070.xml
Edit internal-5070.xml to change the port number
Run reloadxml followed by sofia profile internal-5070 start?

I can register and make short calls, but I do not see my registration when I issue a  sofia_contact username at domain and incoming calls do not work.

I did not edit anything else in internal-5070.xml and do not force a registration domain either. Sofia status does not show me an additional domain name being created.

The documentation is a bit unclear to me on how FS finds it's domains? I am sure it is something with the automatic parsing parameter at the top of my sofia profiles, but it still doesn't click? 

any hint will be enormously appreciated.

Thanks!
E

From sdevoy at bizfocused.com  Thu Nov  8 04:37:02 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 7 Nov 2012 20:37:02 -0500
Subject: [Freeswitch-users] Newbie routing question
In-Reply-To: <CALAD2r7VhQQv_=XNC-3ceDuQWugaRmyJkDtyys6jDED4Rjfd6g@mail.gmail.com>
References: <CALAD2r7VhQQv_=XNC-3ceDuQWugaRmyJkDtyys6jDED4Rjfd6g@mail.gmail.com>
Message-ID: <038401cdbd51$8dd94700$a98bd500$@bizfocused.com>

Without seeing your dial plan, the answer I would give is take the dialplan
for you outbound call and put it in the dialplan where it currently  matches
your inbound call.

 

That is to say, match the inbound call number and run the plan to dial out
to a land line.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Geddes
Sent: Wednesday, November 07, 2012 6:31 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Newbie routing question

 

I've been able to receive a call from a landline thru a gateway and route it
to my softphone.

 

can also make an outbound call from my softphone to a landline.

 

I need to 'tie' the two together, receive an inbound call and using a
dialplan route it as an outbound call to the number in the dialplan.

 

what would be the proper approach to do this?

 

thanks

jeff

 

 

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From william.king at quentustech.com  Thu Nov  8 04:51:47 2012
From: william.king at quentustech.com (William King)
Date: Wed, 07 Nov 2012 17:51:47 -0800
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCB96C43.5EDD5%krice@freeswitch.org>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
Message-ID: <509B1033.3050006@quentustech.com>

Mod_lua now is able to take advantage of pgsql native support.

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

On 11/02/2012 10:25 AM, Ken Rice wrote:
> Hey Guys,
> 
> There?s some new Database Goodness in the core of FreeSWITCH that can
> lead to some unexpected things for you guys updating existing
> installations using ODBC.
> 
> We now have Native PostgreSQL support in the core, and along with this
> comes some changes to the various ?odbc-dsn? settings around the tree.
> 
> If you are using the format ?dsn:username:password? you wont be
> affected, however if you are just specifying a DSN as ?dsn? you will
> need to listen up
> 
> The settings for this field have changed.
> 
> pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> options='-c client_min_messages=NOTICE for postgresql (the stuff after
> pgsql:// is a standard libpq connect string for you programmer types)
> odbc://dns:username:password  for ODBC   ( dsn:: should also work or
> dns:username: )
> sqlite://filename   for sqlite different SQLite Databases
> 
> I think we still need to doc this up good, but its there and its coming
> strong...
> 
> -- 
> Ken
> _http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> _irc.freenode.net #freeswitch
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From luis.daniel.lucio at gmail.com  Thu Nov  8 07:59:18 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 7 Nov 2012 22:59:18 -0600
Subject: [Freeswitch-users] Connecting delays too much sometims
In-Reply-To: <002301cdbd0e$9f7020d0$de506270$@bizfocused.com>
References: <CAFLo2QwuGghnyAkxnjjzdkyuP0hdbs2m-AV6B0ZEnwwHJKeNGQ@mail.gmail.com>
	<002301cdbd0e$9f7020d0$de506270$@bizfocused.com>
Message-ID: <CAFLo2Qy_baMf+dEEC02ftG=-L5N14rZpkp-uwpK_=kvXvg_53Q@mail.gmail.com>

what trunk are you using Sean?

I did a tcpdumpcapture, i realize that the delay is between the INVITE
sent by me  and the RINGIN answer from vendor.

2012/11/7 Sean Devoy <sdevoy at bizfocused.com>:
> We had this issue as well.  Unfortunately at least in our case, this is
> entirely a problem with your provider.  We changed to "Premium Rates"  and
> got MUCH faster connections.  "Least Cost Routing" can be VERY SLOW.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis
> Daniel Lucio Quiroz
> Sent: Wednesday, November 07, 2012 11:32 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Connecting delays too much sometims
>
> I have FS 1.2.3 with updated 1 week ago vbilling installed,
>
> some destinations delays to much to start connecting, others are too quick.
>
> Wondering if you tell me some topics i shall review to find the bootleneck.
>
> Using mod_memcache will help?
>
>
> Regards,
>
> LD
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From regis.freeswitch.org at tornad.net  Thu Nov  8 10:53:39 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Thu, 8 Nov 2012 08:53:39 +0100
Subject: [Freeswitch-users] Connecting RTP between leg A and B directly
In-Reply-To: <7C47175E-5B4E-43E7-A869-184F8BD81900@gmail.com>
References: <7C47175E-5B4E-43E7-A869-184F8BD81900@gmail.com>
Message-ID: <CAHWtr1TKvrFFSk-EFM72AGy71YOoUiYykHs47Kf6mSKfTGB97g@mail.gmail.com>

Hi,

IMHO, variables are used only in bridge in dialplan.
When doing bridge manualy via JS, you must call api* uuid_media off
<uuid> *after
the bridge of the call, maybe with an execute_on_brigde variable.

Take care that your A and B legs can "see" each other without funky nat,
uuid_media not renegociate RTP port.

Hope i'm right and it helps you ;)

regards



2012/11/7 V?ctor Vladimir Ch?vez Gallardo <vivlachaga at gmail.com>

> Hi, i have an spidermonkey script, the script place two outbound calls,
> but i need to set the RTP directly between the legs (a-b) but i dont know
> hoy to set the parameter bypass_media
>
> i tried:
> lega_session.setVariable('bypass_media', 'true');
> bridge(lega_session,legb_session);
>
> also
>
> lega_session.setVariable('bypass_media', 'true');
> legb_session.setVariable('bypass_media', 'true');
> bridge(lega_session,legb_session);
>
> and also setting another variable bypass_media_after_bridge
>
> lega_session.setVariable('bypass_media_after_bridge', 'true');
> legb_session.setVariable('bypass_media_after_bridge', 'true');
> lega_session.setVariable('bypass_media', 'true');
> legb_session.setVariable('bypass_media', 'true');
> bridge(lega_session,legb_session);
>
>
> but i dont have the RTP in the correct way, instead my freeswitch it's
> behind, the rtp flow:  LEG-A -----> FREESWITCH -------> LEG-B
>
> any idea?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.venugopan at mundio.com  Thu Nov  8 14:05:00 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 8 Nov 2012 11:05:00 +0000
Subject: [Freeswitch-users] c script modification
Message-ID: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>

Hi,

This is my first change in C script in freeswitch. If I modify mod_voicemail.c file how to compile and get the latest changes? I see Makefile in many folders which 1 should I run exactly?
Could you please guide me in this with the steps needed to be run  after changing C file in freeswitch.

Regards,
Archana

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From abdlquadri at googlemail.com  Thu Nov  8 14:15:47 2012
From: abdlquadri at googlemail.com (Mumuney Abdlquadri)
Date: Thu, 8 Nov 2012 12:15:47 +0100
Subject: [Freeswitch-users] freeswitch as a service
Message-ID: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>

Hi All,

Is there anyone provide service modeled after SaaS PaaS for freeswitch?

Regards
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From abdlquadri at googlemail.com  Thu Nov  8 14:23:15 2012
From: abdlquadri at googlemail.com (Mumuney Abdlquadri)
Date: Thu, 8 Nov 2012 12:23:15 +0100
Subject: [Freeswitch-users] outbound connection using node-esl socket error
	more detail added
Message-ID: <CAGDHizkE46=KVJ_QccyYP0uzAR79XYc0YGni=k=_idUzAxc6pw@mail.gmail.com>

Hi All,

I am using node-esl to setup an outbound connection.

I am on ubuntu. I have a softphone connected to  freeswitch from a win7
system on Virtualbox.

my dialplan is such:

<include>
  <extension name="abdlquadri.com">
    <condition field="destination_number" expression="^(1982)$">
<action application="sleep" data="1000"/>
<action application="set" data="name=abdlquadri"/>
      <action application="log" data="INFO ${name} is online!"/>
     <action application="socket" data="127.0.0.1:8022 sync full"/>
    </condition>
  </extension>
</include>

When I dial 1982 the output from freeswitch is such:

2012-11-07 12:40:04.403324 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/1000 at 10.0.2.2 [df4ad290-28cf-11e2-a3ea-bdbddea31077]
2012-11-07 12:40:04.403324 [INFO] mod_dialplan_xml.c:485 Processing
abdlquadri <1000>->1982 in context default
2012-11-07 12:40:05.403318 [INFO] mod_dptools.c:1420 abdlquadri is online!
2012-11-07 12:40:05.403318 [INFO] switch_core_session.c:2137 Sending early
media
2012-11-07 12:40:05.423325 [NOTICE] mod_sofia.c:2585 Pre-Answer
sofia/internal/1000 at 10.0.2.2!
2012-11-07 12:40:29.563345 [ERR] mod_event_socket.c:458 Socket Error!
2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:226
sofia/internal/1000 at 10.0.2.2 has executed the last dialplan instruction,
hanging up.
2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:228 Hangup
sofia/internal/1000 at 10.0.2.2 [CS_EXECUTE] [NORMAL_CLEARING]
2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1400 Session 6
(sofia/internal/1000 at 10.0.2.2) Ended
2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1402 Close
Channel sofia/internal/1000 at 10.0.2.2 [CS_DESTROY]


My node-esl code does not get called. Here is it:

var  esl = require('modesl');
var conn = new esl.Server({
    port:'8022',
    host:'127.0.0.1'
},function(){

    conn.on('connection::open', function(){
        conn.execute("answer");
        conn.getInfo();
        console.log('Connection Open');
    });
    conn.on('connection::ready', function(conne){
        conne.execute("answer");
        conne.getInfo();
        console.log('Connection Ready');
    });
    conn.on('connection::close', function(){
        conn.execute("answer");
        conn.getInfo();
        console.log('Connection Closed');
    });
    console.log(conn);
});


I guess the problem is this line: 2012-11-07 12:40:29.563345 [ERR]
mod_event_socket.c:458 Socket Error!

My does not give any log.

I also checked to make sure I have the socket up with lsof -i :8022

COMMAND  PID       USER   FD   TYPE DEVICE SIZE/OFF NODE NAME
node    7911 abdlquadri    7u  IPv4 225303      0t0  TCP localhost:8022
(LISTEN)

Please is there anything am doing wrong.

Thanks. All.
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From a.venugopan at mundio.com  Thu Nov  8 14:46:19 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 8 Nov 2012 11:46:19 +0000
Subject: [Freeswitch-users] voicemail
Message-ID: <592A9CF93E12394E8472A6CC66E66BF232835B@Mail-Kilo.squay.com>

Hi,

When I dial voicemail number I initially get 'please enter password'. Which script will be read so that this phrase comes first? I want to move this 'please enter password' after say 'please enter id' then where I should put this?
Please help.

Regards,
Archana

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From rmorin at blie-ent.com  Thu Nov  8 14:48:47 2012
From: rmorin at blie-ent.com (Rob Morin)
Date: Thu, 8 Nov 2012 06:48:47 -0500
Subject: [Freeswitch-users] Newbie routing question
In-Reply-To: <038401cdbd51$8dd94700$a98bd500$@bizfocused.com>
References: <CALAD2r7VhQQv_=XNC-3ceDuQWugaRmyJkDtyys6jDED4Rjfd6g@mail.gmail.com>
	<038401cdbd51$8dd94700$a98bd500$@bizfocused.com>
Message-ID: <210001cdbda7$03d3e960$0b7bbc20$@blie-ent.com>

I agree with Sean, and it's what I do as well.

 

The catch is that most likely you are incurring costs both ways, receiving
and sending. Not that it's a bad thing, just that you should know.

 

In my dialplan, I'm routing to the extension and simultaneously calling a my
cell:

<action application="bridge"
data="user/1031@$${domain},sofia/gateway/hunt.flowroute/1xxxxxxxxx"/>

 

I'll only incur the outbound cost if I answer the cell. It also counts
against my monthly minutes with the cell carrier.

 

Good luck,

Rob

 

From: Sean Devoy [mailto:sdevoy at bizfocused.com] 
Sent: Wednesday, November 07, 2012 8:37 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Newbie routing question

 

Without seeing your dial plan, the answer I would give is take the dialplan
for you outbound call and put it in the dialplan where it currently  matches
your inbound call.

 

That is to say, match the inbound call number and run the plan to dial out
to a land line.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Geddes
Sent: Wednesday, November 07, 2012 6:31 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Newbie routing question

 

I've been able to receive a call from a landline thru a gateway and route it
to my softphone.

 

can also make an outbound call from my softphone to a landline.

 

I need to 'tie' the two together, receive an inbound call and using a
dialplan route it as an outbound call to the number in the dialplan.

 

what would be the proper approach to do this?

 

thanks

jeff

 

 

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From cal.leeming at simplicitymedialtd.co.uk  Thu Nov  8 15:30:23 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 8 Nov 2012 12:30:23 +0000
Subject: [Freeswitch-users] freeswitch as a service
In-Reply-To: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
References: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
Message-ID: <CAHKQagGXmnktoRUB=BOMjqEUDPLq5PuOfeJ9jujea13qeMSM+g@mail.gmail.com>

Hi Mumuney,

There are several products that use FreeSWITCH for its core, and then offer
that product as a platform - although 2600hz is the only ones that springs
to mind.

You might be able to talk directly with one of these companies (say ITSPTec
or Cudatel) to see if they would offer a bespoke managed platform for you.

Speaking personally, I spent quite a while reviewing different FreeSWITCH
GUI/API platforms and they were all lacking in one area or another.

Currently myself and a few other developers are building a platform that
sits on top of FreeSWITCH, which then provides all the typical features
you'd expect to see in a wholesale/multi-tenant platform, accessible via a
sane API and web interface - however this is several months away from
completion, open sourcing it will inevitably take longer, and the first
release would only be considered a 'stable beta'.

Hope this helps!

Cal

On Thu, Nov 8, 2012 at 11:15 AM, Mumuney Abdlquadri <
abdlquadri at googlemail.com> wrote:

> Hi All,
>
> Is there anyone provide service modeled after SaaS PaaS for freeswitch?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fdelawarde at wirelessmundi.com  Thu Nov  8 15:42:17 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 08 Nov 2012 13:42:17 +0100
Subject: [Freeswitch-users] Possible mod_smpp
In-Reply-To: <20121107174857.9be49bc4@mail.tritonwest.net>
References: <20121107174857.9be49bc4@mail.tritonwest.net>
Message-ID: <1352378537.21382.2.camel@luna.madrid.commsmundi.com>

Right now for SMPP there is kannel which works quite well, but native
integration with FS and mod_sms would be ideal (of course there is ESL,
but it could be simpler that way).

FS is "designed to route and interconnect popular communication
protocols using audio, video, text or any other form of media", so
definitely the right place to put this! :-)

Fran?ois.

On Wed, 2012-11-07 at 09:48 -0800, Dave R. Kompel wrote: 
> I would be, however is FS the right place to put this? Since most SMS
> backbone providers would only give you 1 connection, it would make
> sense to have some type of SMPP/XMPP gateway for multiple FS's to
> connect to. 
>  
> --Dave
> 
>         
>         ______________________________________________________________
>         
>         From: Fran?ois Delawarde [mailto:fdelawarde at wirelessmundi.com]
>         To: FreeSWITCH Users Help
>         [mailto:freeswitch-users at lists.freeswitch.org]
>         Sent: Wed, 07 Nov 2012 02:49:36 -0800
>         Subject: Re: [Freeswitch-users] Possible mod_smpp
>         
>         I'm interested in a mod_smpp. Is anyone else?
>         
>         
>         
>         On Thu, 2011-01-20 at 10:52 -0800, Michael Collins wrote:
>         > Our IRC buddy Delphiworld wants it for something. Math
>         quoted him a
>         > price to do it and he decided that he doesn't want to pay it
>         all
>         > himself so he's fishing around for other interested parties.
>         Judging
>         > by the lack of response to my message I'd have to say that
>         interest is
>         > still very weak...
>         > 
>         > 
>         > -MC
>         > 
>         > On Wed, Jan 19, 2011 at 12:11 PM, Steve Underwood
>         <steveu at coppice.org>
>         > wrote:
>         > 
>         > On 01/20/2011 03:15 AM, Michael Collins wrote:
>         > > Hello all,
>         > >
>         > > If you are interested in giving monetary support to have a
>         > > professional software firm create mod_smpp then please
>         > contact me off
>         > > list and I will give you more details.
>         > >
>         > > Thanks,
>         > > Michael
>         > 
>         > SMPP as in Short Message Peer to Peer? I have floated the
>         idea
>         > of open
>         > sourcing an implementation of that a couple of times, but
>         > interest
>         > seemed weak and I never bothered.
>         > 
>         > Steve
>         > 
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org 
>         > 
>         > 
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org 
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Nov  8 15:45:53 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Thu, 8 Nov 2012 13:45:53 +0100
Subject: [Freeswitch-users] media proxy
In-Reply-To: <18321355.161.1352278383986.JavaMail.master@VoiceJuggler>
References: <CA+GmxbbPpyymzuvECLPDd+8k-hwtPp1zBPQNMn4VrAWE5-eDuQ@mail.gmail.com>
	<18321355.161.1352278383986.JavaMail.master@VoiceJuggler>
Message-ID: <CAM21H-OzXLMSN95d-J17njdepLj9guFTZ9GnEy8G5LmjT+A9Rg@mail.gmail.com>

None. It's default.


On Wed, Nov 7, 2012 at 9:47 AM, Max Kovalenko <
mkovalenko at cybervisiontech.com> wrote:

> Lets return to root direction of the topic.
>
> How to set Full media proxy mode per channel?
>
> Meaning, what values should be set for channel variables: bypass_media &
> proxy_media ?
>
> Max Kovalenko
> ---------------
> CyberVision Inc.
>
> ----- Original Message -----
> From: "Dmitry Sytchev" <kbdfck at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, November 6, 2012 9:01:59 PM
> Subject: Re: [Freeswitch-users] media proxy
>
>
> Media proxy mode not only affects codecs, but any parameters negotiated by
> SDP.
> We use media proxy to handle NAT and still allow untouched SDP exchange
> between customer endpoints and PSTN gateway for T38, for example
>
>
> 2012/11/6 Max Kovalenko < mkovalenko at cybervisiontech.com >
>
>
> I'm getting the data from external DB containing subscribers' parameters.
>
> Max Kovalenko
> ---------------
> CyberVision Inc.
>
>
>
> ----- Original Message -----
> From: "Ben Langfeld" < ben at langfeld.co.uk >
> To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> Sent: Sunday, November 4, 2012 10:46:00 PM
> Subject: Re: [Freeswitch-users] media proxy
>
>
> Can you point to other software that provides this distinction in media
> proxying?
>
> Regards,
> Ben Langfeld
>
>
>
> On 4 November 2012 11:57, Max Kovalenko < mkovalenko at cybervisiontech.com> wrote:
>
>
> Unfortunately, it isn't well documented, or else I would not ask about
> :))))))
>
> Those parameters perhaps are exclusive of each other, by mechanics they
> affects to is almost the same. Or at least, I would call them two sides of
> ame thing.
>
> Theoretically, if you consider the situation of bypass_media=true, that
> means media flow bypass around Freeswitch, it also automatically
> annihilates settings of proxy_media variable. It doesn't matter, even you
> would set proxy_media=true for same channel, it should not provide you
> "light proxy" of media flow, because your settign of bypass_proxy comprises
> Freeswitch don't proxy media.
> Thus I guess in order to set "light" or "hard" proxy the variable
> bypass_media must be set to FALSE first, for same channel of course.
>
> My task requires either bypass media or light/hard proxy to be set for a
> channel depending of some conditions of a session. In the other words, I
> need to set up one of three modes per each channel.
>
> Perhaps I undestand something not so right - I will appreciate if you will
> correct me with the following:
>
> 1. Default (full media proxy) : bypass_media=false ; proxy_media=false
>
> 2. Transparent proxy (mine "light proxy"): bypass_media=false;
> proxy_media=true
>
> 3. Bypass media: bupass_media=true (it's doesn't matter what currect value
> of proxy_media is)
>
> Thanks
>
>
> Max Kovalenko
> Team Leader
> VoIP & UC Team
> Managed Services Dept.
> CyberVision Inc.
> -------------------------------------------------
> tel. +1 (201) 585-9809 ext. 215
> Email: mkovalenko at cybervisiontech.com
> Skype: mkovalenko_cv, panzer_meister
> WWW: www.cybervisiontech.com
>
>
>
> ----- ???????? ????????? -----
> ??: "Ken Rice" < krice at freeswitch.org >
> ????: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> ????????????: ???????, 2 ?????? 2012 ? 17:12:11 GMT +02:00 ?????,
> ????????, ???????
> ????: Re: [Freeswitch-users] media proxy
>
>
> Default, proxy and bybass media are exclusive of each other, you can not
> combine them on a single call...
>
> Proxy media is just that proxy the media
>
> Bypass media is just that, bypass freeswitch and send the media direct
> between the end point...
>
> Setting this options is fairly well documented on the wiki...
>
>
>
>
> On 11/2/12 9:13 AM, "Max Kovalenko" < mkovalenko at cybervisiontech.com >
> wrote:
>
> > Hello,
> >
> > There are two modes of media stream to be proxied or bypassed.
> >
> > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations,
> etc.
> > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT
> > traversal purposes. No media stream parsing is supported
> > 3. Media bypass - RTP streams are bypassing FreeSWITCH at all.
> >
> > There are also two channel parameters affecting above modes:
> bypass_media and
> > proxy_media.
> >
> > - Are these parameters independent? Meaning the combination:
> bypass_media=true
> > AND proxy_media=true is possible. What effect will be?
> >
> > - Does it means that if I want to turn on "light proxy" I would always
> need to
> > set bypass_media=false AND proxy_media=true?
> >
> > - How to set "hard proxy" (full media proxy) per channel before to bridge
> > legs?
> >
> > Waiting for your replay ASAP. Thank you in advance.
> >
>
> > Best Regards.
> >
> > Max Kovalenko
> > Team Leader
> > VoIP & UC Team
> > Managed Services Dept.
> > CyberVision Inc.
> > -------------------------------------------------
> > tel. +1 (201) 585-9809 ext. 215
> > Email: mkovalenko at cybervisiontech.com
> > Skype: mkovalenko_cv, panzer_meister
> > WWW: www.cybervisiontech.com
> >
> >
> >
> >
>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abdlquadri at googlemail.com  Thu Nov  8 15:47:58 2012
From: abdlquadri at googlemail.com (Mumuney Abdlquadri)
Date: Thu, 8 Nov 2012 13:47:58 +0100
Subject: [Freeswitch-users] freeswitch as a service
In-Reply-To: <CAHWtr1SFqBxpC-aQPMbu5JaWC8R3q9ir2m4U3gMcnxTpcYNW=A@mail.gmail.com>
References: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
	<CAHWtr1SFqBxpC-aQPMbu5JaWC8R3q9ir2m4U3gMcnxTpcYNW=A@mail.gmail.com>
Message-ID: <CAGDHizkmr=0RboicBmdgxdh7ow7qQCdFKLQ+ZDzbpuSHjtfOLw@mail.gmail.com>

Hi Regis,

What I am thinking of is something like Heruko http://www.heroku.com/.
Where you only pay for what you use. Are you familiar with Heroku,
http://nodejitsu.com/ etc?



Regards.


On Thu, Nov 8, 2012 at 1:35 PM, Regis M <regis.freeswitch.org at tornad.net>wrote:

> Hi,
>
> I don't do it, but I ever think of it but I'm not sure of the market
> interrest...
>
> Typically, how many are you ready to put per month for a model like this ?
> What's service do you could want with it ? Just freeswitch installed and
> ready to run with access to config file and maintenance ?
>
> Anyone else could be interested by something like this ?
>
> Regards,
>
> 2012/11/8 Mumuney Abdlquadri <abdlquadri at googlemail.com>
>
>> Hi All,
>>
>> Is there anyone provide service modeled after SaaS PaaS for freeswitch?
>>
>> Regards
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From bsullivan at farmradio.org  Thu Nov  8 13:06:23 2012
From: bsullivan at farmradio.org (farmradio)
Date: Thu, 8 Nov 2012 02:06:23 -0800 (PST)
Subject: [Freeswitch-users] GSMOPEN dongle port changes randomly
In-Reply-To: <1352368858797-7584409.post@n2.nabble.com>
References: <CAO1DH1_-ht4w5Keu+mL5MeZgxYs-8z7XthFBLke273rex29jQw@mail.gmail.com>
	<509139E8.70107@infra-it.ru>
	<CALXCt0rYt_R+9XaiV0bJRt9HsA9Ae+0H+ouNpU0J-ETONx3s7A@mail.gmail.com>
	<1352368858797-7584409.post@n2.nabble.com>
Message-ID: <1352369183092-7584410.post@n2.nabble.com>

Hi Giovanni and Ivan - 

we would really like to know what this workaround is as we are encountering
the same problem using the USB dongle (Huawei E173) with Freedom Fone.

Thanks.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/GSMOPEN-dongle-port-changes-randomly-tp7584150p7584410.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From regis.freeswitch.org at tornad.net  Thu Nov  8 15:35:39 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Thu, 8 Nov 2012 13:35:39 +0100
Subject: [Freeswitch-users] freeswitch as a service
In-Reply-To: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
References: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
Message-ID: <CAHWtr1SFqBxpC-aQPMbu5JaWC8R3q9ir2m4U3gMcnxTpcYNW=A@mail.gmail.com>

Hi,

I don't do it, but I ever think of it but I'm not sure of the market
interrest...

Typically, how many are you ready to put per month for a model like this ?
What's service do you could want with it ? Just freeswitch installed and
ready to run with access to config file and maintenance ?

Anyone else could be interested by something like this ?

Regards,

2012/11/8 Mumuney Abdlquadri <abdlquadri at googlemail.com>

> Hi All,
>
> Is there anyone provide service modeled after SaaS PaaS for freeswitch?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vipkilla at gmail.com  Thu Nov  8 16:11:51 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 8 Nov 2012 08:11:51 -0500
Subject: [Freeswitch-users] c script modification
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>
Message-ID: <CAC-LwPM32GKj-Vd9=bjXMwdweBOBU9SJaOtZJ8Z7jBwxFHpERQ@mail.gmail.com>

After making changes to the C file. In the FS src root folder (IE
/usr/src/freeswitch)
run this:
make mod_voicemail-install

On Thu, Nov 8, 2012 at 6:05 AM, Archana Venugopan
<a.venugopan at mundio.com> wrote:
> Hi,
>
>
>
> This is my first change in C script in freeswitch. If I modify
> mod_voicemail.c file how to compile and get the latest changes? I see
> Makefile in many folders which 1 should I run exactly?
>
> Could you please guide me in this with the steps needed to be run  after
> changing C file in freeswitch.
>
>
>
> Regards,
>
> Archana
>
>
>
>


From gerald.weber at besharp.at  Thu Nov  8 16:22:07 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Thu, 8 Nov 2012 13:22:07 +0000
Subject: [Freeswitch-users] c script modification
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2C62A3@exchange.besharp.at>

You can do a make mod_voicemail-install from the root oft he source folder  to compile && install the module in its binary path

If you change something in the core, do a make && make install

See http://wiki.freeswitch.org/wiki/Installation_Guide#Upgrading_and_Re-installation for mor details

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Archana Venugopan
Gesendet: Donnerstag, 08. November 2012 12:05
An: FreeSWITCH Users Help
Betreff: [Freeswitch-users] c script modification

Hi,

This is my first change in C script in freeswitch. If I modify mod_voicemail.c file how to compile and get the latest changes? I see Makefile in many folders which 1 should I run exactly?
Could you please guide me in this with the steps needed to be run  after changing C file in freeswitch.

Regards,
Archana

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From avi at avimarcus.net  Thu Nov  8 16:39:37 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 8 Nov 2012 15:39:37 +0200
Subject: [Freeswitch-users] freeswitch as a service
In-Reply-To: <CAHKQagGXmnktoRUB=BOMjqEUDPLq5PuOfeJ9jujea13qeMSM+g@mail.gmail.com>
References: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
	<CAHKQagGXmnktoRUB=BOMjqEUDPLq5PuOfeJ9jujea13qeMSM+g@mail.gmail.com>
Message-ID: <CALNZuwmhEfiPdHDoE82UJATU8ezrMLxqXejpzm4VX=d9Q+Q3UA@mail.gmail.com>

As mentioned, there's  2600hz and a second one that I know of: Plivio.
Plivio started as a framework to abstract out the API, somewhat similar to
Twilio, that you can run on your own server. Then they started offering
their own hosted version for you to use.

Pricing is here: http://www.plivo.com/pricing/

-Avi




On Thu, Nov 8, 2012 at 2:30 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hi Mumuney,
>
> There are several products that use FreeSWITCH for its core, and then
> offer that product as a platform - although 2600hz is the only ones that
> springs to mind.
>
> You might be able to talk directly with one of these companies
> (say ITSPTec or Cudatel) to see if they would offer a bespoke managed
> platform for you.
>
> Speaking personally, I spent quite a while reviewing different FreeSWITCH
> GUI/API platforms and they were all lacking in one area or another.
>
> Currently myself and a few other developers are building a platform that
> sits on top of FreeSWITCH, which then provides all the typical features
> you'd expect to see in a wholesale/multi-tenant platform, accessible via a
> sane API and web interface - however this is several months away from
> completion, open sourcing it will inevitably take longer, and the first
> release would only be considered a 'stable beta'.
>
> Hope this helps!
>
> Cal
>
> On Thu, Nov 8, 2012 at 11:15 AM, Mumuney Abdlquadri <
> abdlquadri at googlemail.com> wrote:
>
>> Hi All,
>>
>> Is there anyone provide service modeled after SaaS PaaS for freeswitch?
>>
>> Regards
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From abdlquadri at googlemail.com  Thu Nov  8 16:40:28 2012
From: abdlquadri at googlemail.com (Mumuney Abdlquadri)
Date: Thu, 8 Nov 2012 14:40:28 +0100
Subject: [Freeswitch-users] thanks Cal Leeming
Message-ID: <CAGDHiz=puYYjxFpjdpzyG+AV3Ee=wJdAzQGLizK3oPogGe5GYg@mail.gmail.com>

thanks Cal Leeming
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From darcy at Vex.Net  Thu Nov  8 16:46:29 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Thu, 8 Nov 2012 08:46:29 -0500
Subject: [Freeswitch-users] voicemail
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF232835B@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF232835B@Mail-Kilo.squay.com>
Message-ID: <20121108084629.7711a6ff@dilbert>

On Thu, 8 Nov 2012 11:46:19 +0000
Archana Venugopan <a.venugopan at mundio.com> wrote:
> When I dial voicemail number I initially get 'please enter password'.
> Which script will be read so that this phrase comes first? I want to
> move this 'please enter password' after say 'please enter id' then
> where I should put this? Please help.

What does your dialplan look like?  I had the exact opposite problem.
People were dialing *98 from their phone and it asked them for their
account (phone number.)  I had to change this:

 <action application="voicemail" data="check default ${domain_name}"/>

to this:

 <action application="voicemail" data="check default ${domain_name} ${caller_id_number}"/>

Maybe you have to do something similar in reverse.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From abaci64 at gmail.com  Thu Nov  8 16:48:05 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 08 Nov 2012 08:48:05 -0500
Subject: [Freeswitch-users] c script modification
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>
Message-ID: <509BB815.80202@gmail.com>

if you modified mod_voicemail.c and would like to recompile it with the 
changes you made, run the following command from your freeswitch src dir 
(/usr/src/freeswitch)
make mod_voicemail-install
from the restart freeswitch or 'reload mod_voicemail' to get the new 
version loaded in freeswitch.

On 11/8/2012 6:05 AM, Archana Venugopan wrote:
>
> Hi,
>
> This is my first change in C script in freeswitch. If I modify 
> mod_voicemail.c file how to compile and get the latest changes? I see 
> Makefile in many folders which 1 should I run exactly?
>
> Could you please guide me in this with the steps needed to be run 
>  after changing C file in freeswitch.
>
> Regards,
>
> Archana
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From Chad.Engler at patlive.com  Thu Nov  8 16:50:35 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Thu, 8 Nov 2012 08:50:35 -0500
Subject: [Freeswitch-users] outbound connection using node-esl socket
	errormore detail added
In-Reply-To: <CAGDHizkE46=KVJ_QccyYP0uzAR79XYc0YGni=k=_idUzAxc6pw@mail.gmail.com>
References: <CAGDHizkE46=KVJ_QccyYP0uzAR79XYc0YGni=k=_idUzAxc6pw@mail.gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606E7CA76@mail2.patlive.local>

Interesting, are you able to reach the node program using telnet?

 

                telnet localhost 8022

 

-Chad 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Mumuney Abdlquadri
Sent: Thursday, November 08, 2012 6:23 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] outbound connection using node-esl socket
errormore detail added

 

Hi All,

 

I am using node-esl to setup an outbound connection.

 

I am on ubuntu. I have a softphone connected to  freeswitch from a win7
system on Virtualbox.

 

my dialplan is such:

 

<include>

  <extension name="abdlquadri.com <http://abdlquadri.com/> ">

    <condition field="destination_number" expression="^(1982)$">

<action application="sleep" data="1000"/>

<action application="set" data="name=abdlquadri"/>

      <action application="log" data="INFO ${name} is online!"/>

     <action application="socket" data="127.0.0.1:8022
<http://127.0.0.1:8022/>  sync full"/>

    </condition>

  </extension>

</include>

 

When I dial 1982 the output from freeswitch is such:

 

2012-11-07 12:40:04.403324 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/1000 at 10.0.2.2 [df4ad290-28cf-11e2-a3ea-bdbddea31077]

2012-11-07 12:40:04.403324 [INFO] mod_dialplan_xml.c:485 Processing
abdlquadri <1000>->1982 in context default

2012-11-07 12:40:05.403318 [INFO] mod_dptools.c:1420 abdlquadri is
online!

2012-11-07 12:40:05.403318 [INFO] switch_core_session.c:2137 Sending
early media

2012-11-07 12:40:05.423325 [NOTICE] mod_sofia.c:2585 Pre-Answer
sofia/internal/1000 at 10.0.2.2!

2012-11-07 12:40:29.563345 [ERR] mod_event_socket.c:458 Socket Error!

2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:226
sofia/internal/1000 at 10.0.2.2 has executed the last dialplan instruction,
hanging up.

2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:228
Hangup sofia/internal/1000 at 10.0.2.2 [CS_EXECUTE] [NORMAL_CLEARING]

2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1400 Session 6
(sofia/internal/1000 at 10.0.2.2) Ended

2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1402 Close
Channel sofia/internal/1000 at 10.0.2.2 [CS_DESTROY]

 

 

My node-esl code does not get called. Here is it:

 

var  esl = require('modesl');

var conn = new esl.Server({

    port:'8022', 

    host:'127.0.0.1'

},function(){

    

    conn.on('connection::open', function(){

        conn.execute("answer");

        conn.getInfo();

        console.log('Connection Open');

    });

    conn.on('connection::ready', function(conne){

        conne.execute("answer");

        conne.getInfo();

        console.log('Connection Ready');

    });

    conn.on('connection::close', function(){

        conn.execute("answer");

        conn.getInfo();

        console.log('Connection Closed');

    });

    console.log(conn);

});

 

 

I guess the problem is this line: 2012-11-07 12:40:29.563345 [ERR]
mod_event_socket.c:458 Socket Error!

 

My does not give any log.

 

I also checked to make sure I have the socket up with lsof -i :8022

 

COMMAND  PID       USER   FD   TYPE DEVICE SIZE/OFF NODE NAME

node    7911 abdlquadri    7u  IPv4 225303      0t0  TCP localhost:8022
(LISTEN)

 

Please is there anything am doing wrong.

 

Thanks. All.

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From adam.kelloway at newpace.ca  Thu Nov  8 16:57:03 2012
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Thu, 08 Nov 2012 09:57:03 -0400
Subject: [Freeswitch-users] outbound connection using node-esl socket
 error more detail added
In-Reply-To: <CAGDHizkE46=KVJ_QccyYP0uzAR79XYc0YGni=k=_idUzAxc6pw@mail.gmail.com>
References: <CAGDHizkE46=KVJ_QccyYP0uzAR79XYc0YGni=k=_idUzAxc6pw@mail.gmail.com>
Message-ID: <509BBA2F.1000606@newpace.ca>

127.0.0.1:8022 happens to be the default configuration for an inbound 
esl connection. Your server may not be listening on this port at all; 
FreeSWITCH itself might be. Check your event_socket.conf.xml and make 
sure the inbound listen port is different from your server's port.

Adam

On 08/11/2012 7:23 AM, Mumuney Abdlquadri wrote:
> Hi All,
>
> I am using node-esl to setup an outbound connection.
>
> I am on ubuntu. I have a softphone connected to  freeswitch from a 
> win7 system on Virtualbox.
>
> my dialplan is such:
>
> <include>
>   <extension name="abdlquadri.com <http://abdlquadri.com/>">
>     <condition field="destination_number" expression="^(1982)$">
> <action application="sleep" data="1000"/>
> <action application="set" data="name=abdlquadri"/>
>       <action application="log" data="INFO ${name} is online!"/>
>      <action application="socket" data="127.0.0.1:8022 
> <http://127.0.0.1:8022/> sync full"/>
>     </condition>
>   </extension>
> </include>
>
> When I dial 1982 the output from freeswitch is such:
>
> 2012-11-07 12:40:04.403324 [NOTICE] switch_channel.c:926 New Channel 
> sofia/internal/1000 at 10.0.2.2 
> <mailto:1000 at 10.0.2.2> [df4ad290-28cf-11e2-a3ea-bdbddea31077]
> 2012-11-07 12:40:04.403324 [INFO] mod_dialplan_xml.c:485 Processing 
> abdlquadri <1000>->1982 in context default
> 2012-11-07 12:40:05.403318 [INFO] mod_dptools.c:1420 abdlquadri is online!
> 2012-11-07 12:40:05.403318 [INFO] switch_core_session.c:2137 Sending 
> early media
> 2012-11-07 12:40:05.423325 [NOTICE] mod_sofia.c:2585 Pre-Answer 
> sofia/internal/1000 at 10.0.2.2 <mailto:1000 at 10.0.2.2>!
> 2012-11-07 12:40:29.563345 [ERR] mod_event_socket.c:458 Socket Error!
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:226 
> sofia/internal/1000 at 10.0.2.2 <mailto:1000 at 10.0.2.2> has executed the 
> last dialplan instruction, hanging up.
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:228 
> Hangup sofia/internal/1000 at 10.0.2.2 
> <mailto:1000 at 10.0.2.2> [CS_EXECUTE] [NORMAL_CLEARING]
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1400 Session 
> 6 (sofia/internal/1000 at 10.0.2.2 <mailto:1000 at 10.0.2.2>) Ended
> 2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1402 Close 
> Channel sofia/internal/1000 at 10.0.2.2 <mailto:1000 at 10.0.2.2> [CS_DESTROY]
>
>
> My node-esl code does not get called. Here is it:
>
> var  esl = require('modesl');
> var conn = new esl.Server({
>     port:'8022',
>     host:'127.0.0.1'
> },function(){
>     conn.on('connection::open', function(){
>         conn.execute("answer");
>         conn.getInfo();
>         console.log('Connection Open');
>     });
>     conn.on('connection::ready', function(conne){
>         conne.execute("answer");
>         conne.getInfo();
>         console.log('Connection Ready');
>     });
>     conn.on('connection::close', function(){
>         conn.execute("answer");
>         conn.getInfo();
>         console.log('Connection Closed');
>     });
>     console.log(conn);
> });
>
>
> I guess the problem is this line: 2012-11-07 12:40:29.563345 [ERR] 
> mod_event_socket.c:458 Socket Error!
>
> My does not give any log.
>
> I also checked to make sure I have the socket up with lsof -i :8022
>
> COMMAND  PID       USER   FD   TYPE DEVICE SIZE/OFF NODE NAME
> node    7911 abdlquadri  7u  IPv4 225303      0t0  TCP localhost:8022 
> (LISTEN)
>
> Please is there anything am doing wrong.
>
> Thanks. All.


	
	
	
	

	

	

	

	

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From a.venugopan at mundio.com  Thu Nov  8 17:11:46 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 8 Nov 2012 14:11:46 +0000
Subject: [Freeswitch-users] voicemail
In-Reply-To: <20121108084629.7711a6ff@dilbert>
References: <592A9CF93E12394E8472A6CC66E66BF232835B@Mail-Kilo.squay.com>
	<20121108084629.7711a6ff@dilbert>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF23283AA@Mail-Kilo.squay.com>

Hi,
Thanks. But in my default.xml file I see like this for voicemail and I don't see password is being called by default.

      <action application="voicemail" data="default ${domain_name} ${user_id}"/>

    <!-- Voicemail -->
    <extension name="voicemail">
      <condition field="${svc_application}" expression="^voicemail$">
        <action application="info"/>
        <action application="answer"/>
        <action application="start_dtmf"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default ${domain_name} ${user_name}"/>

Now when we dial 20* its saying immediately 'Please enter password' which I want to remove. Thanks

Regards,
Archana

-----Original Message-----
From: D'Arcy J.M. Cain [mailto:darcy at Vex.Net] 
Sent: 08 November 2012 13:46
To: FreeSWITCH Users Help
Cc: Archana Venugopan
Subject: Re: [Freeswitch-users] voicemail

On Thu, 8 Nov 2012 11:46:19 +0000
Archana Venugopan <a.venugopan at mundio.com> wrote:
> When I dial voicemail number I initially get 'please enter password'.
> Which script will be read so that this phrase comes first? I want to
> move this 'please enter password' after say 'please enter id' then
> where I should put this? Please help.

What does your dialplan look like?  I had the exact opposite problem.
People were dialing *98 from their phone and it asked them for their
account (phone number.)  I had to change this:

 <action application="voicemail" data="check default ${domain_name}"/>

to this:

 <action application="voicemail" data="check default ${domain_name} ${caller_id_number}"/>

Maybe you have to do something similar in reverse.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From avi at avimarcus.net  Thu Nov  8 17:25:24 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 8 Nov 2012 16:25:24 +0200
Subject: [Freeswitch-users] c script modification
In-Reply-To: <509BB815.80202@gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF232834F@Mail-Kilo.squay.com>
	<509BB815.80202@gmail.com>
Message-ID: <CALNZuwk8hUDqzyU_h3uc9_mgO2Qgdnujfastscm32AX+CAWc-A@mail.gmail.com>

If you make a good general use patch, then submit the diff to a jira for
merging.

And even if it's not ideal for everyone... for your own maintainability it
would be helpful to make it an *option *then it can probably be merged too.

-Avi
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From abdlquadri at googlemail.com  Thu Nov  8 17:54:31 2012
From: abdlquadri at googlemail.com (Mumuney Abdlquadri)
Date: Thu, 8 Nov 2012 15:54:31 +0100
Subject: [Freeswitch-users] outbound connection using node-esl socket
 error telnet output added
Message-ID: <CAGDHiz=vjbgKPzL0rbxMAMQfp6Dou-XK-Gxavz5vnFC_wmiXXw@mail.gmail.com>

Hi chad,

Output of telnet localhost 8022

Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
connect

Connection closed by foreign host.

On my application side I got this error:

/home/abdlquadri/node_modules/modesl/lib/esl/connection.js:386
uuid = self.getInfo().getHeader('unique-id');
                      ^
TypeError: Cannot call method 'getHeader' of null
    at Connection.execute
(/home/abdlquadri/node_modules/modesl/lib/esl/connection.js:386:24)
    at null.<anonymous>
(/home/abdlquadri/public_html/coWork_nodeServer/server.js:9:15)
    at EventEmitter.emit
(/home/abdlquadri/node_modules/modesl/node_modules/eventemitter2/lib/eventemitter2.js:313:21)
    at Server._onConnection
(/home/abdlquadri/node_modules/modesl/lib/esl/server.js:46:10)
    at Server.EventEmitter.emit (events.js:96:17)
    at TCP.onconnection (net.js:1038:8)


I guess it connected that is why the application can crash. Probably
becuase telnet did not send an esl connection object.

Regards.
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From freeswitch at peely.com  Thu Nov  8 18:56:14 2012
From: freeswitch at peely.com (peely)
Date: Thu, 8 Nov 2012 07:56:14 -0800 (PST)
Subject: [Freeswitch-users] Continue dialplan execution on 503,
	NOT on early media.
Message-ID: <1352390174915-7584423.post@n2.nabble.com>

Hi,

I have a problem with a specific carrier which I have three interconnects
with where they are rejecting with 503 on a CAPs limit. To work around this
after all interconnects have been attempted I am trying to pause for 1s then
attempt the three interconnects out to this carrier again:

    <extension name="to carrier">
        <condition field="${sip_to_params}" expression="tgrp=carrier"/>
        <condition field="destination_number" expression="^(.+)">
            <action application="set" data="bypass_media=true"/>
            <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/>
            <action application="set"
data="fail_on_single_reject=!NORMAL_TEMPORARY_FAILURE"/>
            <action application="bridge"
data="sofia/gateway/carrier1/+$1|sofia/gateway/carrier2/+$1|sofia/gateway/carrier3/+$1"/>
            <action application="sleep" data="2000"/>
            <action application="bridge"
data="sofia/gateway/carrier1/+$1|sofia/gateway/carrier2/+$1|sofia/gateway/carrier3/+$1"/>
        </condition>
    </extension>

This all works well, except I get a large number of call attempts which
result in a 183 (+ SDP) followed by a 480 and these cause the dialplan to
continue with the failure reason NO_USER_RESPONSE which is not listed in the
continue_on_fail reasons. I want to give up immediately on these calls.

I tried the ignore_early_media=true, but the problem with this approach is
that FreeSWITCH considers calls which ring for longer than 60 seconds as a
timeout and clears back, which is not desirable.

Is there any other approach to reroute to alternate carriers, and sleep if
necessary only on 503 and not retry on all other causes, including 18x?



Thanks,



Neil.



--
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Sent from the freeswitch-users mailing list archive at Nabble.com.


From freeswitch at peely.com  Thu Nov  8 19:07:01 2012
From: freeswitch at peely.com (peely)
Date: Thu, 8 Nov 2012 08:07:01 -0800 (PST)
Subject: [Freeswitch-users] Continue dialplan execution on 503,
	NOT on early media.
In-Reply-To: <1352390174915-7584423.post@n2.nabble.com>
References: <1352390174915-7584423.post@n2.nabble.com>
Message-ID: <1352390821192-7584428.post@n2.nabble.com>

Sorry, I also meant to say that I had tried "hangup_after_Bridge=true" as
well and this has no effect, 183 > 480 calls are still retried after 2
seconds.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Continue-dialplan-execution-on-503-NOT-on-early-media-tp7584423p7584428.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From slickwilly2000 at gmx.de  Thu Nov  8 20:21:54 2012
From: slickwilly2000 at gmx.de (=?UTF-8?Q?Alex_M=C3=BCller?=)
Date: Thu, 8 Nov 2012 18:21:54 +0100
Subject: [Freeswitch-users] MOH on valet_park
Message-ID: <1B571DABF46D4488BE145EEF2D6DB2BC@iMac>

Hey guys,

maybe I am wrong but I am bit confused about music-on-hold (MOH) and valet_park-application, executed from the dialplan.

The problem in short: valet_park starts music-on-hold on remote-party AFTER parking-slot-annoucement not BEFORE.


The problem in detail:

I am using the valet_park-application to park (park in) an established call in order to pickup (park out) the call later from another phone. I do this by calling #4 during call (bind_digit_action) and all works fine.

FreeSWITCH announces me the parking-slot. While FreeSWITCH is doing this, the remote-party gets immediately the ?hold_music?. In my case, this is a simple wav-file and no endless-loop-music, because the wav-file annouces ?Your connection is hold? and this text has always start from the beginning of the file (see below).

When the parking-slot-application has finished the parking-slot-annoucement, it starts again the MOH on the remote-party-leg. So my MOH starts twice and that?s the problem.


As i figured out, the ?first? playback has nothing to do with valet_park itself, it is just the MOH when the remote-party-leg is ?on-hold?. The ?second? playback is originated by valet_park.


Any solutions?


Alex


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

Here are my exzerpts from the dialplan:

hold_music=$${sounds_dir}/de/de/marilda/generic/hold.wav


<action application="bind_digit_action" data="feature-digits,#4,exec:execute_extension,park-in XML features,both,self"/>


<extension name="park-in">
    <condition field="${destination_number}" expression="^park-in$">
        <action application="answer"/>
        <action application="valet_park" data="parking-slots auto in 800 899"/>
    </condition>
</extension>
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From devmgr1 at yahoo.com  Thu Nov  8 16:43:05 2012
From: devmgr1 at yahoo.com (Brandon Lee)
Date: Thu, 8 Nov 2012 05:43:05 -0800 (PST)
Subject: [Freeswitch-users] freeswitch as a service
In-Reply-To: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
References: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
Message-ID: <1352382185.16267.YahooMailNeo@web140801.mail.bf1.yahoo.com>

Hello Mumuney,

We provide enterprise FreeSWITCH server hosting, along with wholesale VoIP toll free termination, origination and termination coupled with today's most innovative technological advanced IT Cloud Suite of management tools.? Moreover, we offer FreeSWITCH hosting in 3 of our geographical different datacenters in the world's most connected facilities (e.g. Los Angeles, California "One Wilshire,"? Dallas, Texas "Infomark," and New Jersey) with 100% uptime SLA.? 

While we offer FreeSWITCH hosting, we currently do not offer FreeSWITCH consulting.? However, we do have extensive knowledge developing and deploying web applications that integrate with FreeSWITCH.? Below I've listed a few URLs for you to review should you be interested.

You can view our FreeSWITCH offerings at: http://www.circuitid.com/voip-pbx-servers.php

You can also view our backend system at:? http://www.circuitid.com/video-tutorials.php





________________________________
 From: Mumuney Abdlquadri <abdlquadri at googlemail.com>
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, November 8, 2012 3:15 AM
Subject: [Freeswitch-users] freeswitch as a service
 

Hi All,

Is there anyone provide service modeled after SaaS PaaS for freeswitch?

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From Chad.Engler at patlive.com  Thu Nov  8 20:58:47 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Thu, 8 Nov 2012 12:58:47 -0500
Subject: [Freeswitch-users] outbound connection using node-esl socket
	error telnet output added
In-Reply-To: <CAGDHiz=vjbgKPzL0rbxMAMQfp6Dou-XK-Gxavz5vnFC_wmiXXw@mail.gmail.com>
References: <CAGDHiz=vjbgKPzL0rbxMAMQfp6Dou-XK-Gxavz5vnFC_wmiXXw@mail.gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606E7CAAE@mail2.patlive.local>

Right, it looks like you did connect to the node application, but I
think Adam brought up a good point. 8022 is freeswitch's inbound port.
Try changing that port in your application and restarting both
freeswitch and your app to see if that helps.

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Mumuney Abdlquadri
Sent: Thursday, November 08, 2012 9:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] outbound connection using node-esl
socket error telnet output added

 

Hi chad,

 

Output of telnet localhost 8022

 

Trying 127.0.0.1...

Connected to localhost.

Escape character is '^]'.

connect

 

Connection closed by foreign host.

 

On my application side I got this error:

 

/home/abdlquadri/node_modules/modesl/lib/esl/connection.js:386

            uuid = self.getInfo().getHeader('unique-id');

                                 ^

TypeError: Cannot call method 'getHeader' of null

    at Connection.execute
(/home/abdlquadri/node_modules/modesl/lib/esl/connection.js:386:24)

    at null.<anonymous>
(/home/abdlquadri/public_html/coWork_nodeServer/server.js:9:15)

    at EventEmitter.emit
(/home/abdlquadri/node_modules/modesl/node_modules/eventemitter2/lib/eve
ntemitter2.js:313:21)

    at Server._onConnection
(/home/abdlquadri/node_modules/modesl/lib/esl/server.js:46:10)

    at Server.EventEmitter.emit (events.js:96:17)

    at TCP.onconnection (net.js:1038:8)

 

 

I guess it connected that is why the application can crash. Probably
becuase telnet did not send an esl connection object.

 

Regards.

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From yiftah at choochee.com  Thu Nov  8 21:00:15 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Thu, 8 Nov 2012 10:00:15 -0800
Subject: [Freeswitch-users] freeswitch as a service
In-Reply-To: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
References: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
Message-ID: <CAFVUaV-cOQARQux3MFbGTM87QPwX2hmznsz22Wxf0QkpVSV5_A@mail.gmail.com>

We do it

On Thu, Nov 8, 2012 at 3:15 AM, Mumuney Abdlquadri <
abdlquadri at googlemail.com> wrote:

> Hi All,
>
> Is there anyone provide service modeled after SaaS PaaS for freeswitch?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Nov  8 21:21:32 2012
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Nov 2012 12:21:32 -0600
Subject: [Freeswitch-users] freeswitch as a service
In-Reply-To: <1352382185.16267.YahooMailNeo@web140801.mail.bf1.yahoo.com>
References: <CAGDHizkmFcYO73y+kENv9iWqaLRB4rv2L0i_ZVVf9H5TsO3Bpg@mail.gmail.com>
	<1352382185.16267.YahooMailNeo@web140801.mail.bf1.yahoo.com>
Message-ID: <C6B606A1-1A3D-4847-A47F-E4CE1B2F80AB@freeswitch.org>

But we do.  You can email consulting at freeswitch.org if you need assistance.  Not only does it help support the project it helps us pay our bills. (Not sure you know this bug Geeks have LARGER than average bills)  :P
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 8, 2012, at 7:43 AM, Brandon Lee <devmgr1 at yahoo.com> wrote:

> While we offer FreeSWITCH hosting, we currently do not offer FreeSWITCH consulting. 



From sterned at xakep.ru  Thu Nov  8 21:19:54 2012
From: sterned at xakep.ru (sterned)
Date: Thu, 8 Nov 2012 10:19:54 -0800 (PST)
Subject: [Freeswitch-users] problem with receiving and sending faxes
Message-ID: <1352398794952-7584436.post@n2.nabble.com>

Hello Please help me with the fax. I created a user 500 and 501. created a
dialplan for 500 and am trying to send a fax from the server command:
freeswitch @ internal> originate loopback/500 & txfax (/ tmp /
txfax-sample.tiff)
use: soft fax Venta Fax&Voice 
dialplan:
<extension name="fax">
  <condition>
     <action application="set" data="fax_enable_t38=true"/> 
    <action application="set" data="execute_on_answer=t38_gateway self"/>
    <action application="set" data="proxy_media=true" />
      <action application="bridge" data="user/500 at 192.168.13.18"/>
  </condition>
</extension>
log:
freeswitch at internal> originate loopback/500 &txfax(/tmp/txfax-sample.tiff)
-ERR NORMAL_CLEARING

2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:2005 Parsing
global variables
2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:951 New Channel
loopback/500-a [c2807b1f-acac-4e17-9271-6cf5c61000f2]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:154 loopback/500-a setup
codec L16/8000/20
2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:949 Rename Channel
loopback/500-a->loopback/500-a [c2807b1f-acac-4e17-9271-6cf5c61000f2]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:1069 (loopback/500-a)
State Change CS_NEW -> CS_INIT
2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-a [BREAK]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-a CHANNEL
KILL
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
(loopback/500-a) Running State Change CS_INIT
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
(loopback/500-a) State INIT
2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:951 New Channel
loopback/500-b [d3de53aa-7a3e-483b-bd3f-434eba87815b]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:154 loopback/500-b setup
codec L16/8000/20
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:268 (loopback/500-b) State
Change CS_NEW -> CS_INIT
2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-b [BREAK]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-b CHANNEL
KILL
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:324 (loopback/500-a) State
Change CS_INIT -> CS_ROUTING
2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-a [BREAK]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-a CHANNEL
KILL
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
(loopback/500-a) State INIT going to sleep
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
(loopback/500-a) Running State Change CS_ROUTING
2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1964 (loopback/500-a)
Callstate Change DOWN -> RINGING
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
(loopback/500-a) State ROUTING
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:356 loopback/500-a CHANNEL
ROUTING
2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:67
(loopback/500-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-a [BREAK]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-a CHANNEL
KILL
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
(loopback/500-a) State ROUTING going to sleep
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
(loopback/500-a) Running State Change CS_CONSUME_MEDIA
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:465
(loopback/500-a) State CONSUME_MEDIA
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:561 CHANNEL CONSUME_MEDIA
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:465
(loopback/500-a) State CONSUME_MEDIA going to sleep
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
(loopback/500-b) Running State Change CS_INIT
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
(loopback/500-b) State INIT
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:324 (loopback/500-b) State
Change CS_INIT -> CS_ROUTING
2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-b [BREAK]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-b CHANNEL
KILL
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
(loopback/500-b) State INIT going to sleep
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
(loopback/500-b) Running State Change CS_ROUTING
2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1964 (loopback/500-b)
Callstate Change DOWN -> RINGING
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
(loopback/500-b) State ROUTING
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:356 loopback/500-b CHANNEL
ROUTING
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:117
loopback/500-b Standard ROUTING
2012-11-08 21:21:08.409369 [INFO] mod_dialplan_xml.c:485 Processing 
<0000000000>->500 in context default
Dialplan: loopback/500-b parsing [default->unloop] continue=false
Dialplan: loopback/500-b Regex (PASS) [unloop] ${unroll_loops}(true) =~
/^true$/ break=on-false
Dialplan: loopback/500-b Regex (FAIL) [unloop] ${sip_looped_call}() =~
/^true$/ break=on-false
Dialplan: loopback/500-b parsing [default->tod_example] continue=true
Dialplan: loopback/500-b Date/TimeMatch (FAIL) [tod_example] break=on-false
Dialplan: loopback/500-b parsing [default->holiday_example] continue=true
Dialplan: loopback/500-b Date/TimeMatch (FAIL) [holiday_example]
break=on-false
Dialplan: loopback/500-b parsing [default->global-intercept] continue=false
Dialplan: loopback/500-b Regex (FAIL) [global-intercept]
destination_number(500) =~ /^886$/ break=on-false
Dialplan: loopback/500-b parsing [default->group-intercept] continue=false
Dialplan: loopback/500-b Regex (FAIL) [group-intercept]
destination_number(500) =~ /^\*8$/ break=on-false
Dialplan: loopback/500-b parsing [default->intercept-ext] continue=false
Dialplan: loopback/500-b Regex (FAIL) [intercept-ext]
destination_number(500) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: loopback/500-b parsing [default->redial] continue=false
Dialplan: loopback/500-b Regex (FAIL) [redial] destination_number(500) =~
/^(redial|870)$/ break=on-false
Dialplan: loopback/500-b parsing [default->global] continue=true
Dialplan: loopback/500-b Regex (FAIL) [global] ${call_debug}(false) =~
/^true$/ break=never
Dialplan: loopback/500-b Regex (FAIL) [global] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: loopback/500-b Absolute Condition [global]
Dialplan: loopback/500-b Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: loopback/500-b Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: loopback/500-b Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: loopback/500-b Action export(RFC2822_DATE=${strftime(%a, %d %b %Y
%T %z)})
Dialplan: loopback/500-b parsing [default->snom-demo-2] continue=false
Dialplan: loopback/500-b Regex (FAIL) [snom-demo-2] destination_number(500)
=~ /^9001$/ break=on-false
Dialplan: loopback/500-b parsing [default->snom-demo-1] continue=false
Dialplan: loopback/500-b Regex (FAIL) [snom-demo-1] destination_number(500)
=~ /^9000$/ break=on-false
Dialplan: loopback/500-b parsing [default->eavesdrop] continue=false
Dialplan: loopback/500-b Regex (FAIL) [eavesdrop] destination_number(500) =~
/^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: loopback/500-b parsing [default->eavesdrop] continue=false
Dialplan: loopback/500-b Regex (FAIL) [eavesdrop] destination_number(500) =~
/^779$/ break=on-false
Dialplan: loopback/500-b parsing [default->call_return] continue=false
Dialplan: loopback/500-b Regex (FAIL) [call_return] destination_number(500)
=~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: loopback/500-b parsing [default->del-group] continue=false
Dialplan: loopback/500-b Regex (FAIL) [del-group] destination_number(500) =~
/^80(\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->add-group] continue=false
Dialplan: loopback/500-b Regex (FAIL) [add-group] destination_number(500) =~
/^81(\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->call-group-simo] continue=false
Dialplan: loopback/500-b Regex (FAIL) [call-group-simo]
destination_number(500) =~ /^82(\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->call-group-order] continue=false
Dialplan: loopback/500-b Regex (FAIL) [call-group-order]
destination_number(500) =~ /^83(\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->extension-intercom]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [extension-intercom]
destination_number(500) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: loopback/500-b parsing [default->Local_Extension] continue=false
Dialplan: loopback/500-b Regex (FAIL) [Local_Extension]
destination_number(500) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: loopback/500-b parsing [default->Local_Extension_Skinny]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [Local_Extension_Skinny]
destination_number(500) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: loopback/500-b parsing [default->group_dial_sales] continue=false
Dialplan: loopback/500-b Regex (FAIL) [group_dial_sales]
destination_number(500) =~ /^2000$/ break=on-false
Dialplan: loopback/500-b parsing [default->group_dial_support]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [group_dial_support]
destination_number(500) =~ /^2001$/ break=on-false
Dialplan: loopback/500-b parsing [default->group_dial_billing]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [group_dial_billing]
destination_number(500) =~ /^2002$/ break=on-false
Dialplan: loopback/500-b parsing [default->operator] continue=false
Dialplan: loopback/500-b Regex (FAIL) [operator] destination_number(500) =~
/^(operator|0)$/ break=on-false
Dialplan: loopback/500-b parsing [default->vmain] continue=false
Dialplan: loopback/500-b Regex (FAIL) [vmain] destination_number(500) =~
/^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: loopback/500-b parsing [default->sip_uri] continue=false
Dialplan: loopback/500-b Regex (FAIL) [sip_uri] destination_number(500) =~
/^sip:(.*)$/ break=on-false
Dialplan: loopback/500-b parsing [default->nb_conferences] continue=false
Dialplan: loopback/500-b Regex (FAIL) [nb_conferences]
destination_number(500) =~ /^(30\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->wb_conferences] continue=false
Dialplan: loopback/500-b Regex (FAIL) [wb_conferences]
destination_number(500) =~ /^(31\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->uwb_conferences] continue=false
Dialplan: loopback/500-b Regex (FAIL) [uwb_conferences]
destination_number(500) =~ /^(32\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->cdquality_conferences]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [cdquality_conferences]
destination_number(500) =~ /^(33\d{2})$/ break=on-false
Dialplan: loopback/500-b parsing [default->freeswitch_public_conf_via_sip]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [freeswitch_public_conf_via_sip]
destination_number(500) =~ /^9(888|8888|1616|3232)$/ break=on-false
Dialplan: loopback/500-b parsing [default->mad_boss_intercom] continue=false
Dialplan: loopback/500-b Regex (FAIL) [mad_boss_intercom]
destination_number(500) =~ /^0911$/ break=on-false
Dialplan: loopback/500-b parsing [default->mad_boss_intercom] continue=false
Dialplan: loopback/500-b Regex (FAIL) [mad_boss_intercom]
destination_number(500) =~ /^0912$/ break=on-false
Dialplan: loopback/500-b parsing [default->mad_boss] continue=false
Dialplan: loopback/500-b Regex (FAIL) [mad_boss] destination_number(500) =~
/^0913$/ break=on-false
Dialplan: loopback/500-b parsing [default->ivr_demo] continue=false
Dialplan: loopback/500-b Regex (FAIL) [ivr_demo] destination_number(500) =~
/^5000$/ break=on-false
Dialplan: loopback/500-b parsing [default->dynamic_conference]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [dynamic_conference]
destination_number(500) =~ /^5001$/ break=on-false
Dialplan: loopback/500-b parsing [default->rtp_multicast_page]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [rtp_multicast_page]
destination_number(500) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: loopback/500-b parsing [default->park] continue=false
Dialplan: loopback/500-b Regex (FAIL) [park] destination_number(500) =~
/^5900$/ break=on-false
Dialplan: loopback/500-b parsing [default->unpark] continue=false
Dialplan: loopback/500-b Regex (FAIL) [unpark] destination_number(500) =~
/^5901$/ break=on-false
Dialplan: loopback/500-b parsing [default->valet_park] continue=false
Dialplan: loopback/500-b Regex (FAIL) [valet_park] destination_number(500)
=~ /^(6000)$/ break=on-false
Dialplan: loopback/500-b parsing [default->valet_park] continue=false
Dialplan: loopback/500-b Regex (FAIL) [valet_park] destination_number(500)
=~ /^(60\d[1-9])$/ break=on-false
Dialplan: loopback/500-b parsing [default->park] continue=false
Dialplan: loopback/500-b Regex (FAIL) [park] source(mod_loopback) =~
/mod_sofia/ break=on-false
Dialplan: loopback/500-b parsing [default->unpark] continue=false
Dialplan: loopback/500-b Regex (FAIL) [unpark] source(mod_loopback) =~
/mod_sofia/ break=on-false
Dialplan: loopback/500-b parsing [default->park] continue=false
Dialplan: loopback/500-b Regex (FAIL) [park] source(mod_loopback) =~
/mod_sofia/ break=on-false
Dialplan: loopback/500-b parsing [default->unpark] continue=false
Dialplan: loopback/500-b Regex (FAIL) [unpark] source(mod_loopback) =~
/mod_sofia/ break=on-false
Dialplan: loopback/500-b parsing [default->wait] continue=false
Dialplan: loopback/500-b Regex (FAIL) [wait] destination_number(500) =~
/^wait$/ break=on-false
Dialplan: loopback/500-b parsing [default->fax_receive] continue=false
Dialplan: loopback/500-b Regex (FAIL) [fax_receive] destination_number(500)
=~ /^9178$/ break=on-false
Dialplan: loopback/500-b parsing [default->fax_transmit] continue=false
Dialplan: loopback/500-b Regex (FAIL) [fax_transmit] destination_number(500)
=~ /^9179$/ break=on-false
Dialplan: loopback/500-b parsing [default->ringback_180] continue=false
Dialplan: loopback/500-b Regex (FAIL) [ringback_180] destination_number(500)
=~ /^9180$/ break=on-false
Dialplan: loopback/500-b parsing [default->ringback_183_uk_ring]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [ringback_183_uk_ring]
destination_number(500) =~ /^9181$/ break=on-false
Dialplan: loopback/500-b parsing [default->ringback_183_music_ring]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [ringback_183_music_ring]
destination_number(500) =~ /^9182$/ break=on-false
Dialplan: loopback/500-b parsing [default->ringback_post_answer_uk_ring]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [ringback_post_answer_uk_ring]
destination_number(500) =~ /^9183$/ break=on-false
Dialplan: loopback/500-b parsing [default->ringback_post_answer_music]
continue=false
Dialplan: loopback/500-b Regex (FAIL) [ringback_post_answer_music]
destination_number(500) =~ /^9184$/ break=on-false
Dialplan: loopback/500-b parsing [default->ClueCon] continue=false
Dialplan: loopback/500-b Regex (FAIL) [ClueCon] destination_number(500) =~
/^9191$/ break=on-false
Dialplan: loopback/500-b parsing [default->show_info] continue=false
Dialplan: loopback/500-b Regex (FAIL) [show_info] destination_number(500) =~
/^9192$/ break=on-false
Dialplan: loopback/500-b parsing [default->video_record] continue=false
Dialplan: loopback/500-b Regex (FAIL) [video_record] destination_number(500)
=~ /^9193$/ break=on-false
Dialplan: loopback/500-b parsing [default->video_playback] continue=false
Dialplan: loopback/500-b Regex (FAIL) [video_playback]
destination_number(500) =~ /^9194$/ break=on-false
Dialplan: loopback/500-b parsing [default->delay_echo] continue=false
Dialplan: loopback/500-b Regex (FAIL) [delay_echo] destination_number(500)
=~ /^9195$/ break=on-false
Dialplan: loopback/500-b parsing [default->echo] continue=false
Dialplan: loopback/500-b Regex (FAIL) [echo] destination_number(500) =~
/^9196$/ break=on-false
Dialplan: loopback/500-b parsing [default->milliwatt] continue=false
Dialplan: loopback/500-b Regex (FAIL) [milliwatt] destination_number(500) =~
/^9197$/ break=on-false
Dialplan: loopback/500-b parsing [default->tone_stream] continue=false
Dialplan: loopback/500-b Regex (FAIL) [tone_stream] destination_number(500)
=~ /^9198$/ break=on-false
Dialplan: loopback/500-b parsing [default->zrtp_enrollement] continue=false
Dialplan: loopback/500-b Regex (FAIL) [zrtp_enrollement]
destination_number(500) =~ /^9787$/ break=on-false
Dialplan: loopback/500-b parsing [default->hold_music] continue=false
Dialplan: loopback/500-b Regex (FAIL) [hold_music] destination_number(500)
=~ /^9664$/ break=on-false
Dialplan: loopback/500-b parsing [default->laugh break] continue=false
Dialplan: loopback/500-b Regex (FAIL) [laugh break] destination_number(500)
=~ /^9386$/ break=on-false
Dialplan: loopback/500-b parsing [default->5555555] continue=false
Dialplan: loopback/500-b Regex (FAIL) [5555555] destination_number(500) =~
/^(1000)$/ break=on-false
Dialplan: loopback/500-b parsing [default->fax] continue=false
Dialplan: loopback/500-b Absolute Condition [fax]
Dialplan: loopback/500-b Action set(fax_enable_t38=true)
Dialplan: loopback/500-b Action set(execute_on_answer=t38_gateway self)
Dialplan: loopback/500-b Action set(proxy_media=true)
Dialplan: loopback/500-b Action bridge(user/500 at 192.168.13.18)
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:167
(loopback/500-b) State Change CS_ROUTING -> CS_EXECUTE
2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-b [BREAK]
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-b CHANNEL
KILL
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
(loopback/500-b) State ROUTING going to sleep
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
(loopback/500-b) Running State Change CS_EXECUTE
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:453
(loopback/500-b) State EXECUTE
2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:395 loopback/500-b CHANNEL
EXECUTE
2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:209
loopback/500-b Standard EXECUTE
EXECUTE loopback/500-b
hash(insert/192.168.13.18-spymap/0000000000/d3de53aa-7a3e-483b-bd3f-434eba87815b)
EXECUTE loopback/500-b hash(insert/192.168.13.18-last_dial/0000000000/500)
EXECUTE loopback/500-b
hash(insert/192.168.13.18-last_dial/global/d3de53aa-7a3e-483b-bd3f-434eba87815b)
EXECUTE loopback/500-b export(RFC2822_DATE=Thu, 08 Nov 2012 21:21:08 +0400)
2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1118 EXPORT
(export_vars) [RFC2822_DATE]=[Thu, 08 Nov 2012 21:21:08 +0400]
EXECUTE loopback/500-b set(fax_enable_t38=true)
2012-11-08 21:21:08.409369 [DEBUG] mod_dptools.c:1319 loopback/500-b SET
[fax_enable_t38]=[true]
EXECUTE loopback/500-b set(execute_on_answer=t38_gateway self)
2012-11-08 21:21:08.409369 [DEBUG] mod_dptools.c:1319 loopback/500-b SET
[execute_on_answer]=[t38_gateway self]
EXECUTE loopback/500-b set(proxy_media=true)
2012-11-08 21:21:08.409369 [DEBUG] mod_dptools.c:1319 loopback/500-b SET
[proxy_media]=[true]
EXECUTE loopback/500-b bridge(user/500 at 192.168.13.18)
2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1072 loopback/500-b
EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Nov 2012 21:21:08 +0400] to
event
2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:2005 Parsing
global variables
2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1072 loopback/500-b
EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Nov 2012 21:21:08 +0400] to
event
2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:2005 Parsing
global variables
2012-11-08 21:21:08.409369 [DEBUG] switch_event.c:1569 Parsing variable
[sip_invite_domain]=[192.168.13.18]
2012-11-08 21:21:08.409369 [DEBUG] switch_event.c:1569 Parsing variable
[presence_id]=[500 at 192.168.13.18]
2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:951 New Channel
sofia/internal/sip:500 at 192.168.13.233:5060
[f617d5ed-1e77-4a22-85a1-d0db3da0fc4f]
2012-11-08 21:21:08.409369 [DEBUG] mod_sofia.c:4879
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_NEW -> CS_INIT
2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_INIT
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:437
(sofia/internal/sip:500 at 192.168.13.233:5060) State INIT
2012-11-08 21:21:08.429387 [DEBUG] mod_sofia.c:86
sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA INIT
2012-11-08 21:21:08.429387 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:08.429387 [DEBUG] mod_sofia.c:126
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_INIT ->
CS_ROUTING
2012-11-08 21:21:08.429387 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:437
(sofia/internal/sip:500 at 192.168.13.233:5060) State INIT going to sleep
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_ROUTING
2012-11-08 21:21:08.429387 [DEBUG] switch_channel.c:1964
(sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change DOWN ->
RINGING
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING
2012-11-08 21:21:08.429387 [DEBUG] mod_sofia.c:149
sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA ROUTING
2012-11-08 21:21:08.429387 [DEBUG] switch_ivr_originate.c:67
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2012-11-08 21:21:08.429387 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING going to sleep
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
CS_CONSUME_MEDIA
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:465
(sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA
2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:465
(sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA going to
sleep
2012-11-08 21:21:08.429387 [DEBUG] sofia.c:6282 Channel
sofia/internal/sip:500 at 192.168.13.233:5060 entering state [calling][0]
2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:29.609239 [DEBUG] switch_channel.c:2950
(sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change RINGING ->
HANGUP
2012-11-08 21:21:29.609239 [NOTICE] sofia.c:711 Hangup
sofia/internal/sip:500 at 192.168.13.233:5060 [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2012-11-08 21:21:29.609239 [DEBUG] switch_channel.c:2973 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [KILL]
2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_HANGUP
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:638
(sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP
2012-11-08 21:21:29.609239 [DEBUG] mod_sofia.c:483 Channel
sofia/internal/sip:500 at 192.168.13.233:5060 hanging up, cause:
NORMAL_CLEARING
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:48
sofia/internal/sip:500 at 192.168.13.233:5060 Standard HANGUP, cause:
NORMAL_CLEARING
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:638
(sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP going to sleep
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:429
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_HANGUP ->
CS_REPORTING
2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
CS_REPORTING
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:703
(sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:92
sofia/internal/sip:500 at 192.168.13.233:5060 Standard REPORTING, cause:
NORMAL_CLEARING
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:703
(sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING going to sleep
2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:423
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_REPORTING ->
CS_DESTROY
2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1415 Session 35
(sofia/internal/sip:500 at 192.168.13.233:5060) Locked, Waiting on external
entities
2012-11-08 21:21:29.629589 [DEBUG] switch_ivr_originate.c:3508 Originate
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2012-11-08 21:21:29.629589 [NOTICE] switch_ivr_originate.c:2591 Cannot
create outgoing channel of type [user] cause: [NORMAL_CLEARING]
2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1433 Session 35
(sofia/internal/sip:500 at 192.168.13.233:5060) Ended
2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1437 Close Channel
sofia/internal/sip:500 at 192.168.13.233:5060 [CS_DESTROY]
2012-11-08 21:21:29.629589 [DEBUG] switch_ivr_originate.c:3508 Originate
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2012-11-08 21:21:29.629589 [INFO] mod_dptools.c:3027 Originate Failed. 
Cause: NORMAL_CLEARING
2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2950 (loopback/500-b)
Callstate Change RINGING -> HANGUP
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:527
(sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change HANGUP -> DOWN
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:530
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
(sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY
2012-11-08 21:21:29.629589 [DEBUG] mod_sofia.c:376
sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:99
sofia/internal/sip:500 at 192.168.13.233:5060 Standard DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
(sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY going to sleep
2012-11-08 21:21:29.629589 [NOTICE] mod_dptools.c:3147 Hangup loopback/500-b
[CS_EXECUTE] [NORMAL_CLEARING]
2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2973 Send signal
loopback/500-b [KILL]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-b [BREAK]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:2553 loopback/500-b
skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:453
(loopback/500-b) State EXECUTE going to sleep
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
(loopback/500-b) Running State Change CS_HANGUP
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
(loopback/500-b) State HANGUP
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:453 loopback/500-b CHANNEL
HANGUP
2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2950 (loopback/500-a)
Callstate Change RINGING -> HANGUP
2012-11-08 21:21:29.629589 [NOTICE] mod_loopback.c:464 Hangup loopback/500-a
[CS_CONSUME_MEDIA] [NORMAL_CLEARING]
2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2973 Send signal
loopback/500-a [KILL]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
(loopback/500-a) Running State Change CS_HANGUP
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
(loopback/500-a) State HANGUP
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:453 loopback/500-a CHANNEL
HANGUP
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:48
loopback/500-a Standard HANGUP, cause: NORMAL_CLEARING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
(loopback/500-a) State HANGUP going to sleep
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:429
(loopback/500-a) State Change CS_HANGUP -> CS_REPORTING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-a [BREAK]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
(loopback/500-a) Running State Change CS_REPORTING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
(loopback/500-a) State REPORTING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:92
loopback/500-a Standard REPORTING, cause: NORMAL_CLEARING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
(loopback/500-a) State REPORTING going to sleep
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:423
(loopback/500-a) State Change CS_REPORTING -> CS_DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-a [BREAK]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1415 Session 33
(loopback/500-a) Locked, Waiting on external entities
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-a [BREAK]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:48
loopback/500-b Standard HANGUP, cause: NORMAL_CLEARING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
(loopback/500-b) State HANGUP going to sleep
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:429
(loopback/500-b) State Change CS_HANGUP -> CS_REPORTING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-b [BREAK]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
(loopback/500-b) Running State Change CS_REPORTING
2012-11-08 21:21:29.629589 [DEBUG] switch_ivr_originate.c:3508 Originate
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1433 Session 33
(loopback/500-a) Ended
2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1437 Close Channel
loopback/500-a [CS_DESTROY]
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:527
(loopback/500-a) Callstate Change HANGUP -> DOWN
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:530
(loopback/500-a) Running State Change CS_DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
(loopback/500-a) State DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:99
loopback/500-a Standard DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
(loopback/500-a) State DESTROY going to sleep
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
(loopback/500-b) State REPORTING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:92
loopback/500-b Standard REPORTING, cause: NORMAL_CLEARING
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
(loopback/500-b) State REPORTING going to sleep
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:423
(loopback/500-b) State Change CS_REPORTING -> CS_DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
loopback/500-b [BREAK]
2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b CHANNEL
KILL
2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1415 Session 34
(loopback/500-b) Locked, Waiting on external entities
2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1433 Session 34
(loopback/500-b) Ended
2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1437 Close Channel
loopback/500-b [CS_DESTROY]
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:527
(loopback/500-b) Callstate Change HANGUP -> DOWN
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:530
(loopback/500-b) Running State Change CS_DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
(loopback/500-b) State DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:99
loopback/500-b Standard DESTROY
2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
(loopback/500-b) State DESTROY going to sleep

so I'm trying to send a fax to the fax 501 -> 500
log:
2012-11-08 21:18:21.109351 [NOTICE] switch_channel.c:951 New Channel
sofia/internal/501 at 192.168.13.18 [ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d]
2012-11-08 21:18:21.109351 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.129757 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/501 at 192.168.13.18) Running State Change CS_NEW
2012-11-08 21:18:21.129757 [DEBUG] switch_core_state_machine.c:416
(sofia/internal/501 at 192.168.13.18) State NEW
2012-11-08 21:18:21.129757 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.149399 [DEBUG] sofia.c:8412 IP 192.168.13.71 Rejected by
acl "domains". Falling back to Digest auth.
2012-11-08 21:18:21.149399 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.149399 [DEBUG] sofia.c:1728 detaching session
ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d
2012-11-08 21:18:21.149399 [DEBUG] sofia.c:1820 Re-attaching to session
ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d
2012-11-08 21:18:21.149399 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.149399 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] sofia.c:8412 IP 192.168.13.71 Rejected by
acl "domains". Falling back to Digest auth.
2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6282 Channel
sofia/internal/501 at 192.168.13.18 entering state [received][100]
2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6293 Remote SDP:
v=0
o=- 1352395099 1 IN IP4 192.168.13.71
s=VFT38M/v.25.08.2012
c=IN IP4 192.168.13.71
t=0 0
m=audio 5018 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:240

2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6506
(sofia/internal/501 at 192.168.13.18) State Change CS_NEW -> CS_INIT
2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/501 at 192.168.13.18) Running State Change CS_INIT
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
(sofia/internal/501 at 192.168.13.18) State INIT
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:86
sofia/internal/501 at 192.168.13.18 SOFIA INIT
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:126
(sofia/internal/501 at 192.168.13.18) State Change CS_INIT -> CS_ROUTING
2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
(sofia/internal/501 at 192.168.13.18) State INIT going to sleep
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/501 at 192.168.13.18) Running State Change CS_ROUTING
2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1964
(sofia/internal/501 at 192.168.13.18) Callstate Change DOWN -> RINGING
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/501 at 192.168.13.18) State ROUTING
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:149
sofia/internal/501 at 192.168.13.18 SOFIA ROUTING
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:117
sofia/internal/501 at 192.168.13.18 Standard ROUTING
2012-11-08 21:18:21.169346 [INFO] mod_dialplan_xml.c:485 Processing 501
<501>->500 in context default
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->tod_example]
continue=true
Dialplan: sofia/internal/501 at 192.168.13.18 Date/TimeMatch (FAIL)
[tod_example] break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->holiday_example] continue=true
Dialplan: sofia/internal/501 at 192.168.13.18 Date/TimeMatch (FAIL)
[holiday_example] break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->global-intercept] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [global-intercept]
destination_number(500) =~ /^886$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [group-intercept]
destination_number(500) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->intercept-ext]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [intercept-ext]
destination_number(500) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->redial]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [redial]
destination_number(500) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->global]
continue=true
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [global]
${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
break=never
Dialplan: sofia/internal/501 at 192.168.13.18 Absolute Condition [global]
Dialplan: sofia/internal/501 at 192.168.13.18 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/501 at 192.168.13.18 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/501 at 192.168.13.18 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/501 at 192.168.13.18 Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->snom-demo-2]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [snom-demo-2]
destination_number(500) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->snom-demo-1]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [snom-demo-1]
destination_number(500) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [eavesdrop]
destination_number(500) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->eavesdrop]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [eavesdrop]
destination_number(500) =~ /^779$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->call_return]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [call_return]
destination_number(500) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->del-group]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [del-group]
destination_number(500) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->add-group]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [add-group]
destination_number(500) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [call-group-simo]
destination_number(500) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [call-group-order]
destination_number(500) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->extension-intercom] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [extension-intercom]
destination_number(500) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [Local_Extension]
destination_number(500) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->Local_Extension_Skinny] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
[Local_Extension_Skinny] destination_number(500) =~ /^(11[01][0-9])$/
break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->group_dial_sales] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [group_dial_sales]
destination_number(500) =~ /^2000$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->group_dial_support] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [group_dial_support]
destination_number(500) =~ /^2001$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->group_dial_billing] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [group_dial_billing]
destination_number(500) =~ /^2002$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->operator]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [operator]
destination_number(500) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->vmain]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [vmain]
destination_number(500) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->sip_uri]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [sip_uri]
destination_number(500) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->nb_conferences]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [nb_conferences]
destination_number(500) =~ /^(30\d{2})$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->wb_conferences]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [wb_conferences]
destination_number(500) =~ /^(31\d{2})$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->uwb_conferences] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [uwb_conferences]
destination_number(500) =~ /^(32\d{2})$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->cdquality_conferences] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
[cdquality_conferences] destination_number(500) =~ /^(33\d{2})$/
break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(500) =~
/^9(888|8888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [mad_boss_intercom]
destination_number(500) =~ /^0911$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [mad_boss_intercom]
destination_number(500) =~ /^0912$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->mad_boss]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [mad_boss]
destination_number(500) =~ /^0913$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->ivr_demo]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [ivr_demo]
destination_number(500) =~ /^5000$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->dynamic_conference] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [dynamic_conference]
destination_number(500) =~ /^5001$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->rtp_multicast_page] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [rtp_multicast_page]
destination_number(500) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->park]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [park]
destination_number(500) =~ /^5900$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unpark]
destination_number(500) =~ /^5901$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->valet_park]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [valet_park]
destination_number(500) =~ /^(6000)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->valet_park]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [valet_park]
destination_number(500) =~ /^(60\d[1-9])$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->park]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [park]
destination_number(500) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unpark]
destination_number(500) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->park]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [park]
destination_number(500) =~ /callpark/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unpark]
destination_number(500) =~ /pickup/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->wait]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [wait]
destination_number(500) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->fax_receive]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [fax_receive]
destination_number(500) =~ /^9178$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->fax_transmit]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [fax_transmit]
destination_number(500) =~ /^9179$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->ringback_180]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [ringback_180]
destination_number(500) =~ /^9180$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->ringback_183_uk_ring] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
[ringback_183_uk_ring] destination_number(500) =~ /^9181$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->ringback_183_music_ring] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
[ringback_183_music_ring] destination_number(500) =~ /^9182$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(500) =~ /^9183$/
break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->ringback_post_answer_music] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
[ringback_post_answer_music] destination_number(500) =~ /^9184$/
break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->ClueCon]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [ClueCon]
destination_number(500) =~ /^9191$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->show_info]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [show_info]
destination_number(500) =~ /^9192$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->video_record]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [video_record]
destination_number(500) =~ /^9193$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->video_playback]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [video_playback]
destination_number(500) =~ /^9194$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->delay_echo]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [delay_echo]
destination_number(500) =~ /^9195$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->echo]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [echo]
destination_number(500) =~ /^9196$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->milliwatt]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [milliwatt]
destination_number(500) =~ /^9197$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->tone_stream]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [tone_stream]
destination_number(500) =~ /^9198$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing
[default->zrtp_enrollement] continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [zrtp_enrollement]
destination_number(500) =~ /^9787$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->hold_music]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [hold_music]
destination_number(500) =~ /^9664$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->laugh break]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [laugh break]
destination_number(500) =~ /^9386$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->5555555]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [5555555]
destination_number(500) =~ /^(1000)$/ break=on-false
Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->fax]
continue=false
Dialplan: sofia/internal/501 at 192.168.13.18 Absolute Condition [fax]
Dialplan: sofia/internal/501 at 192.168.13.18 Action set(fax_enable_t38=true)
Dialplan: sofia/internal/501 at 192.168.13.18 Action
set(execute_on_answer=t38_gateway self)
Dialplan: sofia/internal/501 at 192.168.13.18 Action set(proxy_media=true)
Dialplan: sofia/internal/501 at 192.168.13.18 Action
bridge(user/500 at 192.168.13.18)
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:167
(sofia/internal/501 at 192.168.13.18) State Change CS_ROUTING -> CS_EXECUTE
2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/501 at 192.168.13.18) State ROUTING going to sleep
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/501 at 192.168.13.18) Running State Change CS_EXECUTE
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:453
(sofia/internal/501 at 192.168.13.18) State EXECUTE
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:242
sofia/internal/501 at 192.168.13.18 SOFIA EXECUTE
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:209
sofia/internal/501 at 192.168.13.18 Standard EXECUTE
EXECUTE sofia/internal/501 at 192.168.13.18
hash(insert/192.168.13.18-spymap/501/ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d)
EXECUTE sofia/internal/501 at 192.168.13.18
hash(insert/192.168.13.18-last_dial/501/500)
EXECUTE sofia/internal/501 at 192.168.13.18
hash(insert/192.168.13.18-last_dial/global/ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d)
EXECUTE sofia/internal/501 at 192.168.13.18 export(RFC2822_DATE=Thu, 08 Nov
2012 21:18:21 +0400)
2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1118 EXPORT
(export_vars) [RFC2822_DATE]=[Thu, 08 Nov 2012 21:18:21 +0400]
EXECUTE sofia/internal/501 at 192.168.13.18 set(fax_enable_t38=true)
2012-11-08 21:18:21.169346 [DEBUG] mod_dptools.c:1319
sofia/internal/501 at 192.168.13.18 SET [fax_enable_t38]=[true]
EXECUTE sofia/internal/501 at 192.168.13.18 set(execute_on_answer=t38_gateway
self)
2012-11-08 21:18:21.169346 [DEBUG] mod_dptools.c:1319
sofia/internal/501 at 192.168.13.18 SET [execute_on_answer]=[t38_gateway self]
EXECUTE sofia/internal/501 at 192.168.13.18 set(proxy_media=true)
2012-11-08 21:18:21.169346 [DEBUG] mod_dptools.c:1319
sofia/internal/501 at 192.168.13.18 SET [proxy_media]=[true]
EXECUTE sofia/internal/501 at 192.168.13.18 bridge(user/500 at 192.168.13.18)
2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1072
sofia/internal/501 at 192.168.13.18 EXPORTING[export_vars] [RFC2822_DATE]=[Thu,
08 Nov 2012 21:18:21 +0400] to event
2012-11-08 21:18:21.169346 [DEBUG] switch_ivr_originate.c:2005 Parsing
global variables
2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1072
sofia/internal/501 at 192.168.13.18 EXPORTING[export_vars] [RFC2822_DATE]=[Thu,
08 Nov 2012 21:18:21 +0400] to event
2012-11-08 21:18:21.169346 [DEBUG] switch_ivr_originate.c:2005 Parsing
global variables
2012-11-08 21:18:21.169346 [DEBUG] switch_event.c:1569 Parsing variable
[sip_invite_domain]=[192.168.13.18]
2012-11-08 21:18:21.169346 [DEBUG] switch_event.c:1569 Parsing variable
[presence_id]=[500 at 192.168.13.18]
2012-11-08 21:18:21.169346 [NOTICE] switch_channel.c:951 New Channel
sofia/internal/sip:500 at 192.168.13.233:5060
[a318696b-d309-4724-abc6-1f0aca4f0dd3]
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:4879
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_NEW -> CS_INIT
2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:4954 [zrtp_passthru] Setting
a-leg inherit_codec=true
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:4957 [zrtp_passthru] Setting
b-leg absolute_codec_string='PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b'
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_INIT
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
(sofia/internal/sip:500 at 192.168.13.233:5060) State INIT
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:86
sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA INIT
2012-11-08 21:18:21.169346 [DEBUG] sofia_glue.c:1920
sofia/internal/sip:500 at 192.168.13.233:5060 Patched SDP
---
v=0
o=- 1352395099 1 IN IP4 192.168.13.71
s=VFT38M/v.25.08.2012
c=IN IP4 192.168.13.71
t=0 0
m=audio 5018 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:240

+++
v=0
o=FreeSWITCH 3287184146 3287184147 IN IP4 192.168.13.18
s=FreeSWITCH
c=IN IP4 192.168.13.18
t=0 0
m=audio 16966 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:240

2012-11-08 21:18:21.169346 [DEBUG] sofia_glue.c:2637 Local SDP:
v=0
o=FreeSWITCH 3287184146 3287184147 IN IP4 192.168.13.18
s=FreeSWITCH
c=IN IP4 192.168.13.18
t=0 0
m=audio 16966 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:240

2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:126
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_INIT ->
CS_ROUTING
2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
(sofia/internal/sip:500 at 192.168.13.233:5060) State INIT going to sleep
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_ROUTING
2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1964
(sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change DOWN ->
RINGING
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING
2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:149
sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA ROUTING
2012-11-08 21:18:21.169346 [DEBUG] switch_ivr_originate.c:67
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING going to sleep
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
CS_CONSUME_MEDIA
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:465
(sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA
2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:465
(sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA going to
sleep
2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6282 Channel
sofia/internal/sip:500 at 192.168.13.233:5060 entering state [calling][0]
2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:905 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:53.169624 [DEBUG] sofia.c:6282 Channel
sofia/internal/sip:500 at 192.168.13.233:5060 entering state [terminated][408]
2012-11-08 21:18:53.169624 [DEBUG] switch_channel.c:2950
(sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change RINGING ->
HANGUP
2012-11-08 21:18:53.169624 [NOTICE] sofia.c:7082 Hangup
sofia/internal/sip:500 at 192.168.13.233:5060 [CS_CONSUME_MEDIA]
[RECOVERY_ON_TIMER_EXPIRE]
2012-11-08 21:18:53.169624 [DEBUG] switch_channel.c:2973 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [KILL]
2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_HANGUP
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:638
(sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP
2012-11-08 21:18:53.169624 [DEBUG] mod_sofia.c:483 Channel
sofia/internal/sip:500 at 192.168.13.233:5060 hanging up, cause:
RECOVERY_ON_TIMER_EXPIRE
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:48
sofia/internal/sip:500 at 192.168.13.233:5060 Standard HANGUP, cause:
RECOVERY_ON_TIMER_EXPIRE
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:638
(sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP going to sleep
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:429
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_HANGUP ->
CS_REPORTING
2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
CS_REPORTING
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:703
(sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:92
sofia/internal/sip:500 at 192.168.13.233:5060 Standard REPORTING, cause:
RECOVERY_ON_TIMER_EXPIRE
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:703
(sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING going to sleep
2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:423
(sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_REPORTING ->
CS_DESTROY
2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1415 Session 30
(sofia/internal/sip:500 at 192.168.13.233:5060) Locked, Waiting on external
entities
2012-11-08 21:18:53.190125 [DEBUG] switch_ivr_originate.c:3508 Originate
Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2012-11-08 21:18:53.190125 [NOTICE] switch_ivr_originate.c:2591 Cannot
create outgoing channel of type [user] cause: [RECOVERY_ON_TIMER_EXPIRE]
2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1433 Session 30
(sofia/internal/sip:500 at 192.168.13.233:5060) Ended
2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1437 Close Channel
sofia/internal/sip:500 at 192.168.13.233:5060 [CS_DESTROY]
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:527
(sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change HANGUP -> DOWN
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:530
(sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_DESTROY
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
(sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY
2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:376
sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA DESTROY
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:99
sofia/internal/sip:500 at 192.168.13.233:5060 Standard DESTROY
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
(sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY going to sleep
2012-11-08 21:18:53.190125 [DEBUG] switch_ivr_originate.c:3508 Originate
Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2012-11-08 21:18:53.190125 [INFO] mod_dptools.c:3027 Originate Failed. 
Cause: RECOVERY_ON_TIMER_EXPIRE
2012-11-08 21:18:53.190125 [DEBUG] switch_channel.c:2950
(sofia/internal/501 at 192.168.13.18) Callstate Change RINGING -> HANGUP
2012-11-08 21:18:53.190125 [NOTICE] mod_dptools.c:3147 Hangup
sofia/internal/501 at 192.168.13.18 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE]
2012-11-08 21:18:53.190125 [DEBUG] switch_channel.c:2973 Send signal
sofia/internal/501 at 192.168.13.18 [KILL]
2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:2553
sofia/internal/501 at 192.168.13.18 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:453
(sofia/internal/501 at 192.168.13.18) State EXECUTE going to sleep
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/501 at 192.168.13.18) Running State Change CS_HANGUP
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:638
(sofia/internal/501 at 192.168.13.18) State HANGUP
2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:477
sofia/internal/501 at 192.168.13.18 Overriding SIP cause 504 with 408 from the
other leg
2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:483 Channel
sofia/internal/501 at 192.168.13.18 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:613 Responding to INVITE
with: 408
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:48
sofia/internal/501 at 192.168.13.18 Standard HANGUP, cause:
RECOVERY_ON_TIMER_EXPIRE
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:638
(sofia/internal/501 at 192.168.13.18) State HANGUP going to sleep
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:429
(sofia/internal/501 at 192.168.13.18) State Change CS_HANGUP -> CS_REPORTING
2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/501 at 192.168.13.18) Running State Change CS_REPORTING
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:703
(sofia/internal/501 at 192.168.13.18) State REPORTING
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:92
sofia/internal/501 at 192.168.13.18 Standard REPORTING, cause:
RECOVERY_ON_TIMER_EXPIRE
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:703
(sofia/internal/501 at 192.168.13.18) State REPORTING going to sleep
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:423
(sofia/internal/501 at 192.168.13.18) State Change CS_REPORTING -> CS_DESTROY
2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1210 Send signal
sofia/internal/501 at 192.168.13.18 [BREAK]
2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1415 Session 29
(sofia/internal/501 at 192.168.13.18) Locked, Waiting on external entities
2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1433 Session 29
(sofia/internal/501 at 192.168.13.18) Ended
2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1437 Close Channel
sofia/internal/501 at 192.168.13.18 [CS_DESTROY]
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:527
(sofia/internal/501 at 192.168.13.18) Callstate Change HANGUP -> DOWN
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:530
(sofia/internal/501 at 192.168.13.18) Running State Change CS_DESTROY
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
(sofia/internal/501 at 192.168.13.18) State DESTROY
2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:376
sofia/internal/501 at 192.168.13.18 SOFIA DESTROY
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:99
sofia/internal/501 at 192.168.13.18 Standard DESTROY
2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
(sofia/internal/501 at 192.168.13.18) State DESTROY going to sleep





--
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From frederick at targointernet.com  Thu Nov  8 21:55:50 2012
From: frederick at targointernet.com (Frederick Pruneau)
Date: Thu, 08 Nov 2012 13:55:50 -0500
Subject: [Freeswitch-users] Parked calls cannot be retrieved
In-Reply-To: <509298ED.5010705@targointernet.com>
References: <509298ED.5010705@targointernet.com>
Message-ID: <509C0036.1010000@targointernet.com>

Finally, I changed fifo for valet_parking and It is working. It is also 
working with BLF. Here is the code for those who want the same information:

<extension name="parkinglot">
       <condition field="destination_number" expression="^(18[0-2])$">
         <action application="valet_park" data="lot $1"/>
       </condition>
     </extension>
     <extension name="parkingkey">
       <condition field="destination_number" 
expression="^(park\+)(18[0-2])$">
         <action application="transfer" data="$2 XML context_1"/>
       </condition>
     </extension>
  The parkingkey extension is important for, in my case, Granstream 
phones working with BLF. I put park+180 in multi-function key.

Everything is working fine.
On 2012-11-01 11:44, Frederick Pruneau wrote:
> Hi all,
>
> I am configuring a blue.box server with freeswitch. I am tweaking 
> blue.box to add fifo call park. I followed instructions from this wiki 
> page http://wiki.freeswitch.org/wiki/Park_%26_Retrieve .I can park the 
> call but cannot retrieve it. When I dial the same extension where I 
> parked the call (Usually, it is supposed to pick up the parked call), 
> the call is parked in another park slot. Here is my configuration:
>
>     <extension name="parking_slots">
>         <condition field="destination_number" expression="^(18[0-2])$" 
> break="on-false">
>         <action inline="true" application="set" 
> data="slot_count=${fifo(count $1@${domain_name})}"/>
>         <action inline="true" application="set" 
> data="slot_count=${slot_count:-3:2}"/>
>     </condition>
>     <condition field="${slot_count}" expression="^\:0$" break="always">
>         <action application="unset" data="fifo_chime_list"/>
>         <action application="set" data="fifo_chime_freq=0"/>
>         <action application="fifo" 
> data="${destination_number}@${domain_name} in undef local_stream://moh"/>
>         <anti-action application="fifo" 
> data="${destination_number}@${domain_name} out nowait"/>
>     </condition>
>     </extension>
>
> I have found this in the log file:
>
> EXECUTE sofia/sipinterface_1/ext at X.X.X.X 
> set(slot_count=180 at X.X.X.X:0:*1*:0:0:0)
>
> The number 1 is the parking slot number. It changes to the next 
> available parking slot (In this case,  number 2) everytime I dial 180.
>
> Am i missing something?
>
> Thanks for your help!
>
> Fred


-- 
Fr?d?rick Pruneau
Administrateur r?seau | Network administrator
Targo Communications
Ste-Clotilde : (450) 826-0031
Montr?al : (514) 448-0773

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From freeswitch at peely.com  Thu Nov  8 19:49:35 2012
From: freeswitch at peely.com (peely)
Date: Thu, 8 Nov 2012 08:49:35 -0800 (PST)
Subject: [Freeswitch-users] Continue dialplan execution on 503,
	NOT on early media.
In-Reply-To: <1352390174915-7584423.post@n2.nabble.com>
References: <1352390174915-7584423.post@n2.nabble.com>
Message-ID: <1352393375608-7584433.post@n2.nabble.com>

So after widening my forum searching, and suspecting I might need to script
something I copied a similar issue with USER_BUSY responses.

I've created a lua script which hangs up if the content disposition is
early_media:

endpoint_disposition = session:getVariable("endpoint_disposition");

if (endpoint_disposition == "EARLY MEDIA") then
    session:hangup();
end


Then, added it to my dial plan:


    <extension name="to carrier">
        <condition field="${sip_to_params}" expression="tgrp=carrier"/>
        <condition field="destination_number" expression="^(.+)">
            <action application="set" data="bypass_media=true"/>
            <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/>
            <action application="set"
data="fail_on_single_reject=!NORMAL_TEMPORARY_FAILURE"/>
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="bridge"
data="sofia/gateway/carrier1/+$1|sofia/gateway/carrier2/+$1|sofia/gateway/carrier3/+$1"/>
            *<action application="lua" data="killearlymedia.lua"/>*
            <action application="sleep" data="2000"/>
            <action application="bridge"
data="sofia/gateway/carrier1/+$1|sofia/gateway/carrier2/+$1|sofia/gateway/carrier3/+$1"/>
        </condition>
    </extension>


This seems to do what I need.


Neil.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Continue-dialplan-execution-on-503-NOT-on-early-media-tp7584423p7584433.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Thu Nov  8 22:42:29 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 8 Nov 2012 19:42:29 +0000
Subject: [Freeswitch-users] Continue dialplan execution on 503,
 NOT on early media.
In-Reply-To: <1352390821192-7584428.post@n2.nabble.com>
References: <1352390174915-7584423.post@n2.nabble.com>
	<1352390821192-7584428.post@n2.nabble.com>
Message-ID: <CAFiqYu=6xPCRRDkbJ0TN6VKP-bXGmrPm0vDWPcK7LzwshaxDew@mail.gmail.com>

This should work, but only if ignore_early_media is false (or not set)

On 8 November 2012 16:07, peely <freeswitch at peely.com> wrote:

> Sorry, I also meant to say that I had tried "hangup_after_Bridge=true" as
> well and this has no effect, 183 > 480 calls are still retried after 2
> seconds.
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Continue-dialplan-execution-on-503-NOT-on-early-media-tp7584423p7584428.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Chad.Engler at patlive.com  Fri Nov  9 00:41:30 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Thu, 8 Nov 2012 16:41:30 -0500
Subject: [Freeswitch-users] Wiki Call Recording Torrents - 404
In-Reply-To: <CCB6BD34.5ECE9%krice@freeswitch.org>
References: <CA+NjZ2ut-Ozk=nEMwkF9S4FYr0U2-H_CyLi4Xx-nHqSngQPWXg@mail.gmail.com>
	<CCB6BD34.5ECE9%krice@freeswitch.org>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606E7CAF5@mail2.patlive.local>

Has there been any movement on this? Or can I grab the 10/17 recording
from your backups Ken?

 

Thanks,

 

Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Rice
Sent: Wednesday, October 31, 2012 12:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Wiki Call Recording Torrents - 404

 

Some of these might actually not have recordings... 

Collins I still have a pile of backups for many of these in mp3
format... Get with me after todays call and we'll look to see if any of
them match up


On 10/31/12 11:28 AM, "Nick Vines" <jnvines at gmail.com> wrote:

Just a little reminder in case this slipped out of mind. Some of the
conference recordings have broken links. 

Nick 

On Tue, Oct 23, 2012 at 12:19 PM, Michael Collins <msc at freeswitch.org>
wrote:

Thanks for the heads up. Raymond and I are checking into it.
-MC


On Tue, Oct 23, 2012 at 7:57 AM, Nick Vines <jnvines at gmail.com> wrote:

Still getting 404 for the following conference recordings.

2012_10_17
2012_10_10
and
2012_08_15

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

Nick

On Fri, Oct 19, 2012 at 9:12 AM, Stuart Gilbertson | Consider IT Limited
<stuart.gilbertson at considerit.co.uk> wrote:

I get 404's for those two torrent links.


Stuart

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




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

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



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




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

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

 


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

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From yiftah at choochee.com  Fri Nov  9 01:22:04 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Thu, 8 Nov 2012 14:22:04 -0800
Subject: [Freeswitch-users] Wiki Call Recording Torrents - 404
In-Reply-To: <BD87D3B1554F614190EB6E54CE95DDB606E7CAF5@mail2.patlive.local>
References: <CA+NjZ2ut-Ozk=nEMwkF9S4FYr0U2-H_CyLi4Xx-nHqSngQPWXg@mail.gmail.com>
	<CCB6BD34.5ECE9%krice@freeswitch.org>
	<BD87D3B1554F614190EB6E54CE95DDB606E7CAF5@mail2.patlive.local>
Message-ID: <CAFVUaV_N6DofnnECymde-6QUyr=jZ5vMC9zswWUdvNL+Tn7VGg@mail.gmail.com>

BTW, Ken how big is the conference files ?
Do you need a webserver to download those files?
We have big servers in the cloud with public ip addresses we can probably
store some of those files for you and people can download it from us
through HTTP

On Thu, Nov 8, 2012 at 1:41 PM, Chad Engler <Chad.Engler at patlive.com> wrote:

> Has there been any movement on this? Or can I grab the 10/17 recording
> from your backups Ken?****
>
> ** **
>
> Thanks,****
>
> ** **
>
> Chad****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
> *Sent:* Wednesday, October 31, 2012 12:34 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Wiki Call Recording Torrents - 404****
>
> ** **
>
> Some of these might actually not have recordings...
>
> Collins I still have a pile of backups for many of these in mp3 format...
> Get with me after todays call and we?ll look to see if any of them match up
>
>
> On 10/31/12 11:28 AM, "Nick Vines" <jnvines at gmail.com> wrote:****
>
> Just a little reminder in case this slipped out of mind. Some of the
> conference recordings have broken links.
>
> Nick
>
> On Tue, Oct 23, 2012 at 12:19 PM, Michael Collins <msc at freeswitch.org>
> wrote:****
>
> Thanks for the heads up. Raymond and I are checking into it.
> -MC
>
>
> On Tue, Oct 23, 2012 at 7:57 AM, Nick Vines <jnvines at gmail.com> wrote:****
>
> Still getting 404 for the following conference recordings.
>
> 2012_10_17
> 2012_10_10
> and
> 2012_08_15
>
> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call
>
> Nick
>
> On Fri, Oct 19, 2012 at 9:12 AM, Stuart Gilbertson | Consider IT Limited <
> stuart.gilbertson at considerit.co.uk> wrote:****
>
> I get 404's for those two torrent links.
>
>
> Stuart
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Fri Nov  9 01:05:45 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 8 Nov 2012 17:05:45 -0500
Subject: [Freeswitch-users] Bad connection diagnostics? -BUMP
Message-ID: <0a9201cdbdfd$342c5ab0$9c851010$@bizfocused.com>

BUMP.  Any other thoughts anyone?

What can I monitor/track/store/ect on the Server?  It is Centos 5.x and I am
not a unix guy.

Can I store the CPU usage level stats?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Tuesday, November 06, 2012 5:09 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Bad connection diagnostics?

 

Thanks for the response AVI.  I should have been more clear.

 

This is a multi-tenant box.  We use it for my 3 offices too.  That is sort
of why I mentioned ping and tracert  results were the same, so I should have
filled in the rest.  The other tenants have no reported problems, and none
of our 3 sites (7 phones) seem to have these issues currently.  I have had
all of these issues at one time or another, but write it off to temporary
internet throughput variability issues/problems. 

 

I hate to use that excuse even when I truly think it is the problem.  But
when the problems are persistent (not constant) I am at a loss what to
capture/track/test.  They are looking to me to at least identify the problem
and hopefully get it resolved.

 

I do have the FS logs from at least one of these calls that had one way
audio, but I am not proficient enough to spot any audio channel problems.  I
can hang in there through dialplan issues and sofia connections/errors and
maybe even codec mismatches, but the audio connection information is still
one step too far for me.

 

Should I post those logs or can you offer another plan?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Tuesday, November 06, 2012 12:18 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bad connection diagnostics?

 

I would like an answer to this, too.

 

However, here's a place to start: you have 2 legs - you to their VPS, and
their VPS to their office phones.

Perhaps there's an issue with the VPS. Can you get their VPS to playback a
file and do an echo test? E.g. MOH or similar and call it directly and see
if any of the same issues persist. Either way, you've narrowed down where
the issue is...

 

-Avi

 

On Tue, Nov 6, 2012 at 6:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Hi,

I have a client who has been working fine for months, but as of late they
are reporting connections where the audio only works in one direction or
neither and "terrible echo".

 

I know they had some issues with their local Cable Connection, but they are
supposedly resolved.  Results from "ping -n 50 -l 256 " and "tracert" are
virtually identical to mine from here and I have no issues.   Are there some
diagnostics I could run to try and pin this down.  The phones are all cisco
504Gs, the server is a VPS from synapseglobal.

 

Thanks for any ideas you may have.

Sean


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




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

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

 

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From krice at freeswitch.org  Fri Nov  9 01:53:16 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 08 Nov 2012 16:53:16 -0600
Subject: [Freeswitch-users] Wiki Call Recording Torrents - 404
In-Reply-To: <CAFVUaV_N6DofnnECymde-6QUyr=jZ5vMC9zswWUdvNL+Tn7VGg@mail.gmail.com>
Message-ID: <CCC193FC.608C2%krice@freeswitch.org>

I have a webserver... I?m just in the middle of re-arranging a few things..
.opening a new data center... For hosting VoIP servers... Give me a few and
I?ll checkj


On 11/8/12 4:22 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:

> BTW, Ken how big is the conference files ?
> Do you need a webserver to download those files?
> We have big servers in the cloud with public ip addresses we can probably
> store some of those files for you and people can download it from us through
> HTTP 
> ?
> On Thu, Nov 8, 2012 at 1:41 PM, Chad Engler <Chad.Engler at patlive.com> wrote:
>> Has there been any movement on this? Or can I grab the 10/17 recording from
>> your backups Ken?
>> ?
>> Thanks,
>> ?
>> Chad
>> ?
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
>> Sent: Wednesday, October 31, 2012 12:34 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Wiki Call Recording Torrents - 404
>> ?
>> Some of these might actually not have recordings...
>> 
>> Collins I still have a pile of backups for many of these in mp3 format... Get
>> with me after todays call and we?ll look to see if any of them match up
>> 
>> 
>> On 10/31/12 11:28 AM, "Nick Vines" <jnvines at gmail.com
>> <http://jnvines at gmail.com> > wrote:
>> Just a little reminder in case this slipped out of mind. Some of the
>> conference recordings have broken links.?
>> 
>> Nick?
>> 
>> On Tue, Oct 23, 2012 at 12:19 PM, Michael Collins <msc at freeswitch.org
>> <http://msc at freeswitch.org> > wrote:
>> Thanks for the heads up. Raymond and I are checking into it.
>> -MC
>> 
>> 
>> On Tue, Oct 23, 2012 at 7:57 AM, Nick Vines <jnvines at gmail.com
>> <http://jnvines at gmail.com> > wrote:
>> Still getting 404 for the following conference recordings.
>> 
>> 2012_10_17
>> 2012_10_10
>> and
>> 2012_08_15
>> 
>> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call
>> 
>> Nick
>> 
>> On Fri, Oct 19, 2012 at 9:12 AM, Stuart Gilbertson | Consider IT Limited
>> <stuart.gilbertson at considerit.co.uk
>> <http://stuart.gilbertson at considerit.co.uk> > wrote:
>> I get 404's for those two torrent links.
>> 
>> 
>> Stuart
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> ?

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

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From geddes.jeffrey at gmail.com  Fri Nov  9 02:04:14 2012
From: geddes.jeffrey at gmail.com (Jeff Geddes)
Date: Thu, 8 Nov 2012 19:04:14 -0400
Subject: [Freeswitch-users] Newbie routing question
Message-ID: <CALAD2r4dqQN5zzkQAEU1o-KaZ9g4vN3Y3J-JU9syvbNPaar6SA@mail.gmail.com>

Thanks Sean/Rob

your suggestions helped me to get it working.

appreciate it!

jeff
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From yiftah at choochee.com  Fri Nov  9 03:34:35 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Thu, 8 Nov 2012 16:34:35 -0800
Subject: [Freeswitch-users] Wiki Call Recording Torrents - 404
In-Reply-To: <CCC193FC.608C2%krice@freeswitch.org>
References: <CAFVUaV_N6DofnnECymde-6QUyr=jZ5vMC9zswWUdvNL+Tn7VGg@mail.gmail.com>
	<CCC193FC.608C2%krice@freeswitch.org>
Message-ID: <CAFVUaV-NtuJxy+MF+PjjOSw3NsRCkmCnUGKuZzZJn0vDKa+rCA@mail.gmail.com>

OK let me know if you need any help with the hosting

On Thu, Nov 8, 2012 at 2:53 PM, Ken Rice <krice at freeswitch.org> wrote:

>  I have a webserver... I?m just in the middle of re-arranging a few
> things.. .opening a new data center... For hosting VoIP servers... Give me
> a few and I?ll checkj
>
>
> On 11/8/12 4:22 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>
> BTW, Ken how big is the conference files ?
> Do you need a webserver to download those files?
> We have big servers in the cloud with public ip addresses we can probably
> store some of those files for you and people can download it from us
> through HTTP
>
> On Thu, Nov 8, 2012 at 1:41 PM, Chad Engler <Chad.Engler at patlive.com>
> wrote:
>
> Has there been any movement on this? Or can I grab the 10/17 recording
> from your backups Ken?
>
> Thanks,
>
> Chad
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *On Behalf Of *Ken Rice
> *Sent:* Wednesday, October 31, 2012 12:34 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Wiki Call Recording Torrents - 404
>
> Some of these might actually not have recordings...
>
> Collins I still have a pile of backups for many of these in mp3 format...
> Get with me after todays call and we?ll look to see if any of them match up
>
>
> On 10/31/12 11:28 AM, "Nick Vines" <jnvines at gmail.com <
> http://jnvines at gmail.com> > wrote:
> Just a little reminder in case this slipped out of mind. Some of the
> conference recordings have broken links.
>
> Nick
>
> On Tue, Oct 23, 2012 at 12:19 PM, Michael Collins <msc at freeswitch.org <
> http://msc at freeswitch.org> > wrote:
> Thanks for the heads up. Raymond and I are checking into it.
> -MC
>
>
> On Tue, Oct 23, 2012 at 7:57 AM, Nick Vines <jnvines at gmail.com <
> http://jnvines at gmail.com> > wrote:
> Still getting 404 for the following conference recordings.
>
> 2012_10_17
> 2012_10_10
> and
> 2012_08_15
>
> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call
>
> Nick
>
> On Fri, Oct 19, 2012 at 9:12 AM, Stuart Gilbertson | Consider IT Limited <
> stuart.gilbertson at considerit.co.uk <
> http://stuart.gilbertson at considerit.co.uk> > wrote:
> I get 404's for those two torrent links.
>
>
> Stuart
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mi.ke at null.net  Fri Nov  9 04:24:20 2012
From: mi.ke at null.net (Mi Ke)
Date: Thu, 08 Nov 2012 20:24:20 -0500
Subject: [Freeswitch-users] Error 606 (USER_NOT_REGISTERED) but in fact he
	is...
Message-ID: <20121109012420.38330@gmx.com>

Hi,

 I have an IP phone behind NAT registered to also NATed FreeSWITCH (EC2):

 10.10.10.107 (phone) -> 111.111.111.111 (NAT) -> internet -> 222.222.222.222 (NAT) -> 10.210.250.135 (FS)

 Registration part of sip trace looks OK (?):

 tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fdcf8004430): events IN
 tport.c:2845 tport_recv_event() tport_recv_event(0x7fdcf8004430)
 tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x7fdcf8004430) msg 0x7fdcf8010980 from (udp/10.210.250.135:5060) has 702 bytes, veclen = 1
 tport.c:3004 tport_deliver() tport_deliver(0x7fdcf8004430): msg 0x7fdcf8010980 (702 bytes) from udp/111.111.111.111:5060/sip next=(nil)
 nta.c:2803 agent_recv_request() nta: received REGISTER sip:222.222.222.222 SIP/2.0 (CSeq 49512)
 nta.c:3090 agent_check_request_via() nta: Via check: received=111.111.111.111
 nta.c:3161 agent_aliases() nta: canonizing sip:222.222.222.222 with contact
 nta.c:3002 agent_recv_request() nta: REGISTER (49512) going to a default leg
 2012-11-09 01:14:34.833161 [ERR] sofia_reg.c:1613 DELETE PRESENCE SQL: delete from sip_presence where sip_user='1000' and sip_host='222.222.222.222' and profile_name='external' and open_closed='closed'
 nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_respond
 nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_destroy
 tport.c:3238 tport_tsend() tport_tsend(0x7fdcf8004430) tpn = UDP/111.111.111.111:5060
 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x7fdcf8004430): not found by name UDP/111.111.111.111:5060
 nta.c:6678 incoming_reply() nta: sent 200 OK for REGISTER (49512)

 Here registration LED on the phone goes green, phone gives me a dialing tone i.e. behaves like registered.

 show registrations looks OK:

 reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
 1000,222.222.222.222,6a7735fa-6f81e64c at 10.10.10.107,sofia/external/sip:1000 at 10.10.10.107:5060;fs_nat=yes;fs_path=sip%3A1000%40111.111.111.111%3A5060,1352420589,111.111.111.111,5060,udp,domU-12-31-39-09-F5-79,

 sofia status profile external
 ...
 REGISTRATIONS 1



 Meanwhile when incoming call hits my dialplan (simple bridge to user/1000 at domain) I'm getting ERR 606 (User not registered)

 and here is what list_users gives to me:

 userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number
 1000|default|10.210.250.135|default|error/user_not_registered||Mike|1000

 The question is - what am I doing wrong?

 Thanks in advance for all yours hints

 Mike
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From anton.jugatsu at gmail.com  Fri Nov  9 07:12:16 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 9 Nov 2012 08:12:16 +0400
Subject: [Freeswitch-users] Error 606 (USER_NOT_REGISTERED) but in fact
	he is...
In-Reply-To: <20121109012420.38330@gmx.com>
References: <20121109012420.38330@gmx.com>
Message-ID: <CANBhyPB7ainkvLavRawwD=57X+W_nPVjQcTNx1DHRn3kJGgroA@mail.gmail.com>

Which soifa profile do you use for registrering this user. Can you post
sofia_contact */thatuser at domain. Also, do not forget abount dial_string.

2012/11/9 Mi Ke <mi.ke at null.net>

> Hi,
>
> I have an IP phone behind NAT registered to also NATed FreeSWITCH (EC2):
>
> 10.10.10.107 (phone) -> 111.111.111.111 (NAT) -> internet ->
> 222.222.222.222 (NAT) -> 10.210.250.135 (FS)
>
> Registration part of sip trace looks OK (?):
>
> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fdcf8004430): events IN
> tport.c:2845 tport_recv_event() tport_recv_event(0x7fdcf8004430)
> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x7fdcf8004430) msg
> 0x7fdcf8010980 from (udp/10.210.250.135:5060) has 702 bytes, veclen = 1
> tport.c:3004 tport_deliver() tport_deliver(0x7fdcf8004430): msg
> 0x7fdcf8010980 (702 bytes) from udp/111.111.111.111:5060/sip next=(nil)
> nta.c:2803 agent_recv_request() nta: received REGISTER sip:222.222.222.222
> SIP/2.0 (CSeq 49512)
> nta.c:3090 agent_check_request_via() nta: Via check:
> received=111.111.111.111
> nta.c:3161 agent_aliases() nta: canonizing sip:222.222.222.222 with contact
> nta.c:3002 agent_recv_request() nta: REGISTER (49512) going to a default
> leg
> 2012-11-09 01:14:34.833161 [ERR] sofia_reg.c:1613 DELETE PRESENCE SQL:
> delete from sip_presence where sip_user='1000' and
> sip_host='222.222.222.222' and profile_name='external' and
> open_closed='closed'
> nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_respond
> nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_destroy
> tport.c:3238 tport_tsend() tport_tsend(0x7fdcf8004430) tpn = UDP/
> 111.111.111.111:5060
> tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x7fdcf8004430): not
> found by name UDP/111.111.111.111:5060
> nta.c:6678 incoming_reply() nta: sent 200 OK for REGISTER (49512)
>
> Here registration LED on the phone goes green, phone gives me a dialing
> tone i.e. behaves like registered.
>
> show registrations looks OK:
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
> 1000,222.222.222.222,6a7735fa-6f81e64c at 10.10.10.107
> ,sofia/external/sip:1000 at 10.10.10.107:5060
> ;fs_nat=yes;fs_path=sip%3A1000%40111.111.111.111%3A5060,1352420589,111.111.111.111,5060,udp,domU-12-31-39-09-F5-79,
>
> sofia status profile external
> ...
> REGISTRATIONS           1
>
>
>
> Meanwhile when incoming call hits my dialplan (simple bridge to
> user/1000 at domain) I'm getting ERR 606 (User not registered)
>
> and here is what list_users gives to me:
>
>
> userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number
> 1000|default|10.210.250.135|default|error/user_not_registered||Mike|1000
>
> The question is - what am I doing wrong?
>
> Thanks in advance for all yours hints
>
> Mike
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gabe at gundy.org  Fri Nov  9 08:53:57 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 8 Nov 2012 22:53:57 -0700
Subject: [Freeswitch-users] Bad connection diagnostics? -BUMP
In-Reply-To: <0a9201cdbdfd$342c5ab0$9c851010$@bizfocused.com>
References: <0a9201cdbdfd$342c5ab0$9c851010$@bizfocused.com>
Message-ID: <CAMwZ-tKFby6ww6pak0dz0OdMHZwEQctupmVy0U_L_-J9E5wvoQ@mail.gmail.com>

On Thu, Nov 8, 2012 at 3:05 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> BUMP.  Any other thoughts anyone?

It's best if you don't start a new email thread for an existing email.
Now we're required to read two threads for context ;)



> What can I monitor/track/store/ect on the Server?  It is Centos 5.x and I am
> not a unix guy.

Here is a list of common tools to monitor your servers:
http://www.linuxscrew.com/2012/03/22/linux-monitoring-tools/

The problem is that if you're not comfortable with *nix, you might not
be able to install them :/

Here are a lot of interesting commandline tools that you can run (no
setup required):
http://www.cyberciti.biz/tips/top-linux-monitoring-tools.html

If nothing else, they'll give you a feel for what's available to monitor.

Good luck!


Gabe


From jim.daley1223 at yahoo.com  Fri Nov  9 03:31:23 2012
From: jim.daley1223 at yahoo.com (Jim Daley)
Date: Thu, 8 Nov 2012 16:31:23 -0800 (PST)
Subject: [Freeswitch-users] (no subject)
Message-ID: <1352421083.47663.YahooMailNeo@web120002.mail.ne1.yahoo.com>

I hope this is the right place to ask this question.
?
I am running freeswitch using LUA. I'm doing outbound calling for a load tester.
?
I have a number that when you call it you need to enter 2654# before the call get's answered. So I need a way of dialing DTMF digits before the other end answers. I tried to just sleep and send_dtmf after the session.ready but that doesn't work.
?
Is there a way to do this. Sample code below...thanks
?if (session:ready()) then
session:answer();
session:setVariable("CallCmp", "Script_dcetest1")
session:setVariable("PromptFail", "Script_dcetest1") ;
api = freeswitch.API();
session:sleep(1000);
--status = session:getVariable("SilenceStatus");
session:execute("wait_for_silence", "200 30 20 20000"); --
-- Continue in english, press 2
status = session:getVariable("SilenceStatus"); -- This will get past langauge and main menu prompts
if((status == "unset")and(session:ready())) then 
session:execute("send_dtmf", "2 at 100"); --@ determines dtmf length, enter NPI
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From regis.freeswitch.org at tornad.net  Fri Nov  9 10:00:45 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Fri, 9 Nov 2012 08:00:45 +0100
Subject: [Freeswitch-users] Connecting RTP between leg A and B directly
In-Reply-To: <89C0923E-F050-46EA-B5AA-599280A82BD2@gmail.com>
References: <89C0923E-F050-46EA-B5AA-599280A82BD2@gmail.com>
Message-ID: <CAHWtr1RJ=9YrLdMGn6R8m=z_nJCwKQRrN1OTRDzLAzV_qQHNnw@mail.gmail.com>

Hi,

You must try to execute the api command uuid_media by setting a variable
that coupled with an event.
Try looking at the execute_on family :
http://wiki.freeswitch.org/wiki/Channel_Variables#The_execute_on_family

In your case, I think
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
or
api_after_bridge... maybe more but not documented :).
or
bridge_pre_execute_bleg_app

All this variable/event handler can help you to run command outside script.
but, i think the uuid_media off must be called after the media up this FS.

Hope it helps.


2012/11/9 V?ctor Vladimir Ch?vez Gallardo <vivlachaga at gmail.com>

> Hi Regis, how you doing?
>
> see, i tried your suggestion, setting the uuid_media on both sessions
> AFTER the bridge, also seting bypass_media_after_bridge = true in the call
> parameters, if i put the apiExecute, the script is stalled on bridge action
> until one leg trigger up an release code.. in that moment the instructions
> apiExecute("uuid_media","off") are executed (the script flow continue), but
> if i put those instructions BEFORE, i have an a forced release on leg A
> "inline:1  Session is not active!"
>
> you know something about this issue? there's something wrong with my app
> logic? will be very useffuly your help, thanks a lot
>
> originate_options_lega =
> "bypass_media_after_bridge=true,ignore_early_media=true,origination_caller_id_number=1004,originate_timeout=60,leg=1";
> originate_options_legb =
> "bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=60,leg=2";
>
> bridge(lega_session,legb_session);
>                                 apiExecute("uuid_media", "off " +
> lega_uuid);
>                                 apiExecute("uuid_media", "off " +
> legb_uuid);
>
>
> Hi,
> IMHO, variables are used only in bridge in dialplan.
> When doing bridge manualy via JS, you must call api uuid_media off
> <uuid> after the bridge of the call, maybe with an execute_on_brigde
> variable.
> Take care that your A and B legs can "see" each other without funky nat,
> uuid_media not renegociate RTP port.
> Hope i'm right and it helps you ;)
> regards
>
>
> 2012/11/7 V?ctor Vladimir Ch?vez Gallardo <vivlachaga at gmail.com>
>
> Hi, i have an spidermonkey script, the script place two outbound calls,
> but i need to set the RTP directly between the legs (a-b) but i dont know
> hoy to set the parameter bypass_media
>
> i tried:
>
> lega_session.setVariable('bypass_media', 'true');
>
> bridge(lega_session,legb_session);
>
> also
>
> lega_session.setVariable('bypass_media', 'true');
>
> legb_session.setVariable('bypass_media', 'true');
>
> bridge(lega_session,legb_session);
>
> and also setting another variable bypass_media_after_bridge
>
> lega_session.setVariable('bypass_media_after_bridge', 'true');
>
> legb_session.setVariable('bypass_media_after_bridge', 'true');
>
> lega_session.setVariable('bypass_media', 'true');
>
> legb_session.setVariable('bypass_media', 'true');
>
> bridge(lega_session,legb_session);
>
>
> but i dont have the RTP in the correct way, instead my freeswitch it's
> behind, the rtp flow:  LEG-A -----> FREESWITCH -------> LEG-B
>
> any idea?
>
>
>
>
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From freeswitch at peely.com  Fri Nov  9 12:56:12 2012
From: freeswitch at peely.com (peely)
Date: Fri, 9 Nov 2012 01:56:12 -0800 (PST)
Subject: [Freeswitch-users] Continue dialplan execution on 503,
	NOT on early media.
In-Reply-To: <CAFiqYu=6xPCRRDkbJ0TN6VKP-bXGmrPm0vDWPcK7LzwshaxDew@mail.gmail.com>
References: <1352390174915-7584423.post@n2.nabble.com>
	<1352390821192-7584428.post@n2.nabble.com>
	<CAFiqYu=6xPCRRDkbJ0TN6VKP-bXGmrPm0vDWPcK7LzwshaxDew@mail.gmail.com>
Message-ID: <1352454972308-7584455.post@n2.nabble.com>

None of the combinations seemed to do what I needed without
ignore_early_media, and ignore_early_media had its own consequences. It
seems there's no way of stopping the dialplan from proceeding after receipt
of media, but continue for other reasons.

The lua script seems to work like a charm however.

If my observations are correct and final, shall I add this to the
continue_on_fail wiki?



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Continue-dialplan-execution-on-503-NOT-on-early-media-tp7584423p7584455.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From abdlquadri at googlemail.com  Fri Nov  9 12:58:34 2012
From: abdlquadri at googlemail.com (Mumuney Abdlquadri)
Date: Fri, 9 Nov 2012 10:58:34 +0100
Subject: [Freeswitch-users] only port 8040 works for outbound esl connection?
Message-ID: <CAGDHiznkD56SNODZbn-VertPmvHXcscgjWPhat03+2hGNw7YfQ@mail.gmail.com>

I have posted a challenge I was having with outbound esl connection. I was
using port 8022 because that was what is in the source of node-esl.

Adam Kelloway suggested freeswitch is listening on that port but freeswitch
listens on 8021. But I decided to change the port anyway it did not work. I
then decided to change it to port 8040 like in the book and it worked.

I tried other ports but they did not work.

My question, is there anything that says freeswitch must only send outbound
esl connection to only port 8040 and nothing else?

Regards
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From NuwanW at unifybusiness.co.uk  Fri Nov  9 13:39:48 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Fri, 9 Nov 2012 10:39:48 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695E99BC9A8F@BARXB0003.UnifyBusiness.local>

Hello all,

I'm resending this request as I haven't had any reply yet. Any help would be greatly appreciated.

I'm trying to broadcast audio on a bridged call. My requirement is to play audio on both legs at the same time. I used uuid_broadcast in following order,

Uuid_broadcast uuid 'path' both

Please note that I'm sending uuid_broadcast through an esl connection. So my actual request to freeswitch is as follows,

eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file' both");

(eslWriteConnection is an object of .Net ESLConnection)

The  issue I'm having is, freeswitch not playing the audio on both channels at the same time. FreeSwitch plays the audio on one leg first, then plays on the second leg (After it finished playing on first leg).

I don't have this issue with FreeSwitch 1.0.6, where it plays audio on both legs at the same time. I'm having this issue with FreeSwtich 1.2.3.

Could anyone please suggest any solution.

Thank you,



This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From miha at softnet.si  Fri Nov  9 15:14:41 2012
From: miha at softnet.si (Miha)
Date: Fri, 09 Nov 2012 13:14:41 +0100
Subject: [Freeswitch-users] User busy detect
Message-ID: <509CF3B1.2050508@softnet.si>

Hi,

I would like to implement in my biling function that the FS will do call 
fwd if user is busy.

How can I detect user busy? I am doing dialplan with curl.


br,
Miha


From steveayre at gmail.com  Fri Nov  9 15:28:30 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 9 Nov 2012 12:28:30 +0000
Subject: [Freeswitch-users] only port 8040 works for outbound esl
	connection?
In-Reply-To: <CAGDHiznkD56SNODZbn-VertPmvHXcscgjWPhat03+2hGNw7YfQ@mail.gmail.com>
References: <CAGDHiznkD56SNODZbn-VertPmvHXcscgjWPhat03+2hGNw7YfQ@mail.gmail.com>
Message-ID: <CAFiqYukCH4mDqfxb-Pr8QXiEmFsSkQ2WPj19Rd5+Lx1jXJt6Og@mail.gmail.com>

That suggests the application you were connecting to (not FS, the other)
was listening on port 8040 not 8022.



On 9 November 2012 09:58, Mumuney Abdlquadri <abdlquadri at googlemail.com>wrote:

> I have posted a challenge I was having with outbound esl connection. I was
> using port 8022 because that was what is in the source of node-esl.
>
> Adam Kelloway suggested freeswitch is listening on that port but
> freeswitch listens on 8021. But I decided to change the port anyway it did
> not work. I then decided to change it to port 8040 like in the book and it
> worked.
>
> I tried other ports but they did not work.
>
> My question, is there anything that says freeswitch must only send
> outbound esl connection to only port 8040 and nothing else?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From a.venugopan at mundio.com  Fri Nov  9 15:49:57 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 9 Nov 2012 12:49:57 +0000
Subject: [Freeswitch-users] mod_voicemail.c
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2328597@Mail-Kilo.squay.com>

Hi,

I have got a doubt. In mod_voicemail.c script I see the below query

"insert into voicemail_msgs(created_epoch, read_epoch, username, domain, uuid, cid_name, "
                                                          "cid_number, in_folder, file_path, message_len, flags, read_flags, forwarded_by) "
                                                          "values(%ld,0,'%q','%q','%q','%q','%q','%q','%q','%u','','%q','%q')", (long) switch_epoch_time_now(NULL),
                                                          myid, domain_name, use_uuid, caller_id_name, caller_id_number,
                                                          myfolder, file_path, message_len, read_flags, switch_str_nil(forwarded_by));

Want to know from where myid and domain_name are picked up. Initially I thought it was picking from voicemail_prefs but even if there are no entries for a caller in voicemail_prefs table it still insert the value.
Please let me know on this.

Mod_voicemail.c code is the same as present online.

Regards,
Archana

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From Chad.Engler at patlive.com  Fri Nov  9 16:08:07 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Fri, 9 Nov 2012 08:08:07 -0500
Subject: [Freeswitch-users] New node-esl Version
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606E7CB24@mail2.patlive.local>

node-esl just got a version bump to v0.0.7 with some great bug fixes,
added tests, and more examples.

 

As always you can find the repository at:
https://github.com/englercj/node-esl

 

Thanks to lxfontes <https://github.com/lxfontes>  for his great pull
requests, keep the bug reports and pull requests coming guys; need to
get test coverage to 100%!

 

Here is the full changelog:

 

Version v0.0.7 Changelog

==================================

 

* Added tests for esl, esl.Event, esl.Parser, esl.Server, esl.Connection

* Fixed esl.setLogLevel to work correctly

* Fixed Parser to correctly identify Content-Length

* Made Connection callback truly asynchronous all the time

* Updated Server ID generator

* Connection now only automatically subscribes to events required for it
to work properly

* Added some "Modesl" headers to quickly check if a reply is OK or ERR

* Fixed esl.Server not being able to handle empty options

* Fixed an issue getting unique ID of outbound connection, issue #4
(thanks lxfontes)

* Fixed an issue with execute creating invalid events, issue #5 (thanks
lxfontes)

* Added "myevents" option to esl.Server (thanks lxfontes)

* Added more examples, including an Outbound socket example (thanks
lxfontes)

* Only supporting v0.8.x of node until v0.1.0

 

Test Coverage Report Summary (as of v0.0.7)

==================================

Overall

--------

516 SLOC, 66% coverage

 

By File

--------

/esl.js:             20 SLOC, 100% coverage

/esl/event.js:       83 SLOC, 100% coverage

/esl/connection.js: 283 SLOC,  40% coverage

/esl/server.js:      45 SLOC,  91% coverage

/esl/parser.js:      85 SLOC,  96% coverage

 

Thanks everyone for their continued support, and I look forward to
hearing how people fair with the new version!

 

Thanks,

 

Chad Engler

Web Programmer

PATLive

1.800.775.7790 x746

Chad.Engler at patlive.com <mailto:Chad.Engler at patlive.com> 

 

Hosted Communications | Friendly Service    www.patlive.com
<http://www.patlive.com/> 

 

 

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From steveayre at gmail.com  Fri Nov  9 17:20:57 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 9 Nov 2012 14:20:57 +0000
Subject: [Freeswitch-users] User busy detect
In-Reply-To: <509CF3B1.2050508@softnet.si>
References: <509CF3B1.2050508@softnet.si>
Message-ID: <CAFiqYukm_mC3pbxXqiPVMFrSWFW5u81P8be2U8-K-YMdJyWvoQ@mail.gmail.com>

There are two ways...

1) Use the Limit functionality. This only works if you're the only person
sending calls, since you can only count your own calls.
http://wiki.freeswitch.org/wiki/Limit

2) Send the call to the user, and see if they reject the call with 486 Busy
Here if busy. Simply handle that hangup reason.
This will work if they are receiving a call from other users, but doesn't
work with all phones - eg some will have multiple lines so allow a call to
ring even if they've already answered another call.

Steve



On 9 November 2012 12:14, Miha <miha at softnet.si> wrote:

> Hi,
>
> I would like to implement in my biling function that the FS will do call
> fwd if user is busy.
>
> How can I detect user busy? I am doing dialplan with curl.
>
>
> br,
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jim.daley1223 at yahoo.com  Fri Nov  9 17:07:23 2012
From: jim.daley1223 at yahoo.com (Jim Daley)
Date: Fri, 9 Nov 2012 06:07:23 -0800 (PST)
Subject: [Freeswitch-users] Need to dial dtmf digits before call is
	answered.
In-Reply-To: <1352421083.47663.YahooMailNeo@web120002.mail.ne1.yahoo.com>
References: <1352421083.47663.YahooMailNeo@web120002.mail.ne1.yahoo.com>
Message-ID: <1352470043.50209.YahooMailNeo@web120005.mail.ne1.yahoo.com>





________________________________
From: Jim Daley <jim.daley1223 at yahoo.com>
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>; "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, November 8, 2012 6:31 PM
Subject: [Freeswitch-users] (no subject)


I hope this is the right place to ask this question.

I am running freeswitch using LUA. I'm doing outbound calling for a load tester.

I have a number that when you call it you need to enter 2654# before the call get's answered. So I need a way of dialing DTMF digits before the other end answers. I tried to just sleep and send_dtmf after the session.ready but that doesn't work.

Is there a way to do this. Sample code below...thanks
if (session:ready()) then
session:answer();
session:setVariable("CallCmp", "Script_dcetest1")
session:setVariable("PromptFail", "Script_dcetest1") ;
api = freeswitch.API();
session:sleep(1000);
--status = session:getVariable("SilenceStatus");
session:execute("wait_for_silence", "200 30 20 20000"); --
-- Continue in english, press 2
status = session:getVariable("SilenceStatus"); -- This will get past langauge and main menu prompts
if((status == "unset")and(session:ready())) then 
session:execute("send_dtmf", "2 at 100"); --@ determines dtmf length, enter NPI
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com/


/

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org/
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From lynn.nielson at greenseedtechnologies.com  Fri Nov  9 19:14:33 2012
From: lynn.nielson at greenseedtechnologies.com (Lynn Nielson)
Date: Fri, 9 Nov 2012 09:14:33 -0700
Subject: [Freeswitch-users] No Media on bridging two external calls
Message-ID: <CANGqFgMEiGyXtAw4wBp7WW8pu5r8eMGVYp2eUOmEncMPCh3WBA@mail.gmail.com>

Running FreeSwitch on Ubuntu 12.04 FreeSWITCH Version
1.3.0+git~20120823T225232Z~fbc83cb0ea

I'm having a problem with no media/audio when bridging two external calls,
however, if one of the legs is an internal extension the the audio works
fine. Also, if I add bypass_media=true, the audio works for the two
external call bridge as well.  I would really like to understand why this
works this way and if there is a configuration option that would enable the
internal and external bridging to work the same.  Here are two examples of
the problem from the command line using sofia.

This fails on audio but makes a connection (external to external):
originate sofia/gateway/flowroute/1801xxxxxxx
&bridge(sofia/gateway/flowroute/1801yyyyyyy)

This works (external to external):
originate {bypass_media=true}sofia/gateway/flowroute/1801xxxxxxx
&bridge(sofia/gateway/flowroute/1801yyyyyyy)

This works (internal to external):
originate sofia/internal/1010%IPADDRESS
&bridge(sofia/gateway/flowroute/1801yyyyyyy)

This also works (internal to external):
originate {bypass_media=true}sofia/internal/1010%IPADDRESS
&bridge(sofia/gateway/flowroute/1801yyyyyyy)


Thanks for any explanation,

Lynn
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From sdevoy at bizfocused.com  Fri Nov  9 20:07:01 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 9 Nov 2012 12:07:01 -0500
Subject: [Freeswitch-users] Bad connection diagnostics? -BUMP
In-Reply-To: <CAMwZ-tKFby6ww6pak0dz0OdMHZwEQctupmVy0U_L_-J9E5wvoQ@mail.gmail.com>
References: <0a9201cdbdfd$342c5ab0$9c851010$@bizfocused.com>
	<CAMwZ-tKFby6ww6pak0dz0OdMHZwEQctupmVy0U_L_-J9E5wvoQ@mail.gmail.com>
Message-ID: <10f901cdbe9c$a2919d40$e7b4d7c0$@bizfocused.com>

Thanks Gabe.

I didn't realize a reply would start a new thread.

I am not *nix illiterate, it is just more like an "English as a second
language" relationship.  I just need someone to point the direction, as you
did.

Thanks very much.
Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel
Gunderson
Sent: Friday, November 09, 2012 12:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bad connection diagnostics? -BUMP

On Thu, Nov 8, 2012 at 3:05 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> BUMP.  Any other thoughts anyone?

It's best if you don't start a new email thread for an existing email.
Now we're required to read two threads for context ;)



> What can I monitor/track/store/ect on the Server?  It is Centos 5.x 
> and I am not a unix guy.

Here is a list of common tools to monitor your servers:
http://www.linuxscrew.com/2012/03/22/linux-monitoring-tools/

The problem is that if you're not comfortable with *nix, you might not be
able to install them :/

Here are a lot of interesting commandline tools that you can run (no setup
required):
http://www.cyberciti.biz/tips/top-linux-monitoring-tools.html

If nothing else, they'll give you a feel for what's available to monitor.

Good luck!


Gabe

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




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

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




From Chad.Engler at patlive.com  Fri Nov  9 21:12:43 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Fri, 9 Nov 2012 13:12:43 -0500
Subject: [Freeswitch-users] Bad connection diagnostics? -BUMP
In-Reply-To: <10f901cdbe9c$a2919d40$e7b4d7c0$@bizfocused.com>
References: <0a9201cdbdfd$342c5ab0$9c851010$@bizfocused.com><CAMwZ-tKFby6ww6pak0dz0OdMHZwEQctupmVy0U_L_-J9E5wvoQ@mail.gmail.com>
	<10f901cdbe9c$a2919d40$e7b4d7c0$@bizfocused.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606E7CB94@mail2.patlive.local>

A different subject starts a new thread, to clear up confusion.

-Chad

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Friday, November 09, 2012 12:07 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Bad connection diagnostics? -BUMP

Thanks Gabe.

I didn't realize a reply would start a new thread.

I am not *nix illiterate, it is just more like an "English as a second
language" relationship.  I just need someone to point the direction, as
you did.

Thanks very much.
Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Gabriel Gunderson
Sent: Friday, November 09, 2012 12:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bad connection diagnostics? -BUMP

On Thu, Nov 8, 2012 at 3:05 PM, Sean Devoy <sdevoy at bizfocused.com>
wrote:
> BUMP.  Any other thoughts anyone?

It's best if you don't start a new email thread for an existing email.
Now we're required to read two threads for context ;)



> What can I monitor/track/store/ect on the Server?  It is Centos 5.x 
> and I am not a unix guy.

Here is a list of common tools to monitor your servers:
http://www.linuxscrew.com/2012/03/22/linux-monitoring-tools/

The problem is that if you're not comfortable with *nix, you might not
be able to install them :/

Here are a lot of interesting commandline tools that you can run (no
setup
required):
http://www.cyberciti.biz/tips/top-linux-monitoring-tools.html

If nothing else, they'll give you a feel for what's available to
monitor.

Good luck!


Gabe

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




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

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



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




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

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


From krice at freeswitch.org  Fri Nov  9 23:11:53 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 09 Nov 2012 14:11:53 -0600
Subject: [Freeswitch-users] Friday FreeForAll
Message-ID: <CCC2BFA9.60922%krice@freeswitch.org>

Hey Guys

The FreeForAll is running

Join us at sip:888 at conference.freeswitch.org

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

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From frederick at targointernet.com  Fri Nov  9 23:26:33 2012
From: frederick at targointernet.com (Frederick Pruneau)
Date: Fri, 09 Nov 2012 15:26:33 -0500
Subject: [Freeswitch-users] conference_set_auto_outcall with Grandstream
	phones
Message-ID: <509D66F9.6090000@targointernet.com>

Hi all,

I have setup a paging like solution with freeswitch. I am working with 
Grandstream GXP2100 phones. I have enabled "allow auto answer by 
Call-Info" option on the phones. Upgraded phones to the latest firmware. 
When i dial the paging extension, the phones freeze and I cannot do 
anything with the phones. I have to wait 2 or 3 minutes and phones 
hangup. Here is my paging config in my dialplan:

    <extension name="paging">
      <condition field="destination_number" expression="^(12345)$">

        <action application="answer"/>

        <action application="set" data="conference_auto_outcall_timeout=5"/>
        <action application="set" 
data="conference_auto_outcall_flags=mute"/>
        <action application="set" 
data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
        <action application="set" 
data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
        <action application="set" 
data="conference_auto_outcall_profile=default"/>
        <action application="conference_set_auto_outcall" 
data="[sip_auto_answer=true]user/300@$${domain}"/>
        <action application="conference_set_auto_outcall" 
data="[sip_auto_answer=true]user/301@$${domain}"/>

        <action application="conference" data="$1 at default"/>
        <action application="conference" data="$1 kick all"/>
      </condition>
    </extension>

Is this a bug between freeswitch and Grandstream phones?

Thanks!
Fred


From sos at sokhapkin.dyndns.org  Fri Nov  9 23:41:15 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 09 Nov 2012 15:41:15 -0500
Subject: [Freeswitch-users] Friday FreeForAll
In-Reply-To: <CCC2BFA9.60922%krice@freeswitch.org>
References: <CCC2BFA9.60922%krice@freeswitch.org>
Message-ID: <1417318.UFQ9Jsnr3x@sos>

 conference.freeswitch.org doesn't respond to SIP requests. Am I the only 
unlucky?

On Friday 09 November 2012 14:11:53 Ken Rice wrote:
> Hey Guys
> 
> The FreeForAll is running
> 
> Join us at sip:888 at conference.freeswitch.org
> 
> K


From bpriddy at bryantschools.org  Fri Nov  9 23:57:31 2012
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Fri, 9 Nov 2012 14:57:31 -0600
Subject: [Freeswitch-users] Friday FreeForAll
In-Reply-To: <CCC2BFA9.60922%krice@freeswitch.org>
References: <CCC2BFA9.60922%krice@freeswitch.org>
Message-ID: <CADK-BjtMid6RkDbXDH1srpV5awjBWXr8ejPRh0GKmVQepC2DVQ@mail.gmail.com>

Do you have a list of ports that are needed for audio for this conference?


On Fri, Nov 9, 2012 at 2:11 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Hey Guys
>
> The FreeForAll is running
>
> Join us at sip:888 at conference.freeswitch.org
>
> K
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

*Blakelund Priddy*
Network Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
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From egable+freeswitch at gmail.com  Sat Nov 10 01:52:25 2012
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Fri, 9 Nov 2012 14:52:25 -0800
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <509B1033.3050006@quentustech.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<509B1033.3050006@quentustech.com>
Message-ID: <CAD-6L_Wr-qGzYaBJ9mHp-sLGP_rz7cJkc8gWVn_GCxaLhyd84w@mail.gmail.com>

It should probably also be pointed out that I pushed some fixes for
core_pgsql support over the past couple days for better handling
reconnecting to the DB if it loses connectivity. You will definitely want
those fixes in place before you lose connectivity on a real, production
server. At this time, I have been able to bounce Postgres over and over
about 30 times in my tests without any issues getting reconnected and
resuming operations.


On Wed, Nov 7, 2012 at 5:51 PM, William King
<william.king at quentustech.com>wrote:

> Mod_lua now is able to take advantage of pgsql native support.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 11/02/2012 10:25 AM, Ken Rice wrote:
> > Hey Guys,
> >
> > There?s some new Database Goodness in the core of FreeSWITCH that can
> > lead to some unexpected things for you guys updating existing
> > installations using ODBC.
> >
> > We now have Native PostgreSQL support in the core, and along with this
> > comes some changes to the various ?odbc-dsn? settings around the tree.
> >
> > If you are using the format ?dsn:username:password? you wont be
> > affected, however if you are just specifying a DSN as ?dsn? you will
> > need to listen up
> >
> > The settings for this field have changed.
> >
> > pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> > options='-c client_min_messages=NOTICE for postgresql (the stuff after
> > pgsql:// is a standard libpq connect string for you programmer types)
> > odbc://dns:username:password  for ODBC   ( dsn:: should also work or
> > dns:username: )
> > sqlite://filename   for sqlite different SQLite Databases
> >
> > I think we still need to doc this up good, but its there and its coming
> > strong...
> >
> > --
> > Ken
> > _http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> > _irc.freenode.net #freeswitch
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing from
our children, we're stealing from them--and it's not even considered to be
a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
not live to eat.) ~Marcus Tullius Cicero
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From egable+freeswitch at gmail.com  Sat Nov 10 02:03:15 2012
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Fri, 9 Nov 2012 15:03:15 -0800
Subject: [Freeswitch-users] FreeSWITCH High
 availability(Normal_temporary_failure problem)
In-Reply-To: <CAFYz+X=+BrXLChSegXxOw4hD+qFWdOM=V-_sOMhRYyt4A6Y1+A@mail.gmail.com>
References: <CAFYz+X=+BrXLChSegXxOw4hD+qFWdOM=V-_sOMhRYyt4A6Y1+A@mail.gmail.com>
Message-ID: <CAD-6L_Xf8eD2=MTKscrZxwkpPBafe3-j5dSXV9RM2jjbrgZc2Q@mail.gmail.com>

I have not looked at your invalid CSeq error, but since you are using
pacemaker, you might find this resource agent useful. Here is an example of
how to use it:

primitive FreeSWITCH ocf:fssolutions:FreeSWITCH \

        params ips="eth0/217.160.21.196/26:eth1/217.160.21.197/26"
user="freeswitch" group="freeswitch" \

        op monitor interval="3s" role="Master" on-fail="restart" depth="0" \

        op monitor interval="10s" role="Slave" on-fail="restart" depth="0" \

        op start interval="0" timeout="30" \

        op stop interval="0" timeout="60"

ms FreeSWITCH-MS FreeSWITCH \

        meta master-max="1" master-node-max="1" clone-max="2"
clone-node-max="1" \

                notify="false" target-role="Master"

location FreeSWITCH-MS-on-ha02 FreeSWITCH-MS 50: ha02.bkw.org

location FreeSWITCH-MS-on-ha01 FreeSWITCH-MS 50: ha01.bkw.org

property $id="cib-bootstrap-options" \

        dc-version="1.1.7-6.el6-148fccfd5985c5590cc601123c6c16e966b85d14" \

        cluster-infrastructure="corosync" \

        expected-quorum-votes="2" \

        stonith-enabled="false" \

        no-quorum-policy="ignore" \

        last-lrm-refresh="1347305291"

rsc_defaults $id="rsc-options" \

        resource-stickiness="100"


On Wed, Nov 7, 2012 at 12:03 AM, Rajkumar K <rajprithiv88 at gmail.com> wrote:

> Hi,
>
> I am trying to achieve high availability in FreeSWITCH using heartbeat and
> pacemaker and I am able to switch between the server whenever one of the
> servers crashes. But the problem is one server is able to recover the calls
> when im invoking sofia recover, but another server is recovering the call
> one few times.(Mostly not able to recover).
>
>
> I have primary and secondary server installed in two different
> machines(centos), the FreeSWITCH instances are always running in both the
> PCs. I am running heartbeat and pacemaker to monitor the IP or sofia
> fail-over. A floating IP is configure in heartbeat to reach the active
> server.
> I succeeded in switching between the servers whenever IP or FreeSWITCH and
> once it reaches the another server it invokes sofia recover to recover the
> calls.
>
> Both freeswtich instances are using the same configuration and database is
> shared using ODBC connectivity.
>
> Problem is:
>
> Call is made using the primary server, and i did fsctl crash in primary
> server cli. Heartbeat resource switches to secondary server and it invokes
> "sofia profile internal restart" and "sofia recover". and it recovers the
> call. The call gets recovered in 4-5 seconds.
>
> At the same time i will start the freeswitch instance in the primary
> server. Now if i crash the secondary server using fsctl crash, the
> resources switches to primary server and it invokes "sofia profile internal
> restart" and "sofia recover". Also the server sends invite request to the
> clients But it ends in NORMAL_TEMPORARY_FAILURE. Wireshark log says Client
> is responding with "Invalid CSeq" for the Server's INVITE request. This
> happens always with the primary server and very few times primary server is
> also able to recover the calls.
>
> I have the same configurations in both the servers
> And also i checked by stopping the heartbeat switching, and crashed the
> primary server's freeswitch. Then if i start freeswitch again in the same
> server and invoking sofia recover will recover the calls without any
> problem.
>
>
> I have also attached the cli logs of primary and secondary servers.
> I am not able to identify the exact problem in this, Please help me out in
> this problem.
>
>
> Thanks
> Rajkumar
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing from
our children, we're stealing from them--and it's not even considered to be
a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
not live to eat.) ~Marcus Tullius Cicero
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From gabe at gundy.org  Sat Nov 10 03:08:07 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 9 Nov 2012 17:08:07 -0700
Subject: [Freeswitch-users] Bad connection diagnostics? -BUMP
In-Reply-To: <10f901cdbe9c$a2919d40$e7b4d7c0$@bizfocused.com>
References: <0a9201cdbdfd$342c5ab0$9c851010$@bizfocused.com>
	<CAMwZ-tKFby6ww6pak0dz0OdMHZwEQctupmVy0U_L_-J9E5wvoQ@mail.gmail.com>
	<10f901cdbe9c$a2919d40$e7b4d7c0$@bizfocused.com>
Message-ID: <CAMwZ-tJKtAxsAz9rRNheTw4b0zfy4mk-yqAaZiYm8zGQCOrBxg@mail.gmail.com>

On Fri, Nov 9, 2012 at 10:07 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> I didn't realize a reply would start a new thread.

Not a biggie.


> I am not *nix illiterate, it is just more like an "English as a second
> language" relationship.  I just need someone to point the direction, as you
> did.

We're always happy to help!


Best,
Gabe


From prasd.d.b at gmail.com  Sat Nov 10 06:25:23 2012
From: prasd.d.b at gmail.com (Prasd D)
Date: Fri, 9 Nov 2012 19:25:23 -0800
Subject: [Freeswitch-users] Voicemail to email with exim
Message-ID: <CAPZnxCQ3v2Kwm-13ocG+CGjA525UJNV9-WzZ=8-Nr46uU5LU8g@mail.gmail.com>

I am using using exim with eximcompat.sh way (according to)
http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
but i am not getting email.
I don't see anything referring to email in the debug log also. Can
anyone tell how to get this working ?

Is there a way to make it work such that there's a reminder sent to
the recipient with no attachment but with a link to the voicemail file
?

-- 
Thanks,
Prasd


From kheimerl at cs.berkeley.edu  Sat Nov 10 07:42:19 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 9 Nov 2012 20:42:19 -0800
Subject: [Freeswitch-users] Return code from ESL Message Sending
Message-ID: <CACyT-3mexr4O9D1jSqM94U50x4TacjDCHi37UNh2Y6sEmHaoCQ@mail.gmail.com>

Hello Freeswitch Users:

We're currently trying to get the return code from a MESSAGE we send using
ESL. The closest we've found is this jira:
http://jira.freeswitch.org/browse/FS-4453 which seems to provide similar
functionality for the chat command, but nothing for ESL.

Here's a pastebin of our current code: http://pastebin.freeswitch.org/20201

The server we are hitting is returning a "415 Unsupported Content Type"
(which is correct) and we're trying to discover that in freeswitch, instead
of assuming the message was received correctly. Right now, we get that the
recvEventTimed is returning None. This is all done on the a pull of FS from
yesterday.

Any suggestions?
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From colemichae at gmail.com  Sat Nov 10 13:59:39 2012
From: colemichae at gmail.com (Michael Cole)
Date: Sat, 10 Nov 2012 18:59:39 +0800
Subject: [Freeswitch-users] mod_voicemail error
Message-ID: <1483970.EesMfQR03j@michael-laptop>

Nov 10 10:00:46 37 freeswitch[1857]: 2012-11-10 10:00:46.019811 [ERR] 
mod_voicemail.c:3311 Error creating 
/usr/storage/voicemail/default/137.220.92.81/1002

The user freeswitch has access to the directory read and write.

I even changed the ACL on the directory to be RWX for all users but still 
getting the error.

If I run the entire freeswitch as root then no problems but that is a complete 
no no .. 

Does anyone have an idea where i'm going wrong? 

Regards Michael



From philq at qsystemsengineering.com  Sat Nov 10 19:26:12 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Sat, 10 Nov 2012 11:26:12 -0500
Subject: [Freeswitch-users] Mod_opal & PTLIB
Message-ID: <006001cdbf60$1c408b10$54c1a130$@com>

I?m trying to compile mod_opal here but I can?t get it to work.  I?ve tried
the buildopal.sh script, the manual install, sacrificial offerings, etc. to
no avail.  I was getting an error about PTLib being too old, so I grabbed
PTLib 2.10.7 source here:
http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html

Trying to compile it, I get a ?No operating system selected? error.  I?ve
pasted in the make attempt below, Distro is CentOS 5.8.

It looks like a lot of other folks have had/are having problems with
mod_opal and ptlib, but I?m not seeing much in the way of solutions.  Has
anyone else gotten past this problem?  Any pointers?  (no pun intended)

[root at Tyrion ptlib]# make
Setting default PTLIBDIR to /root/ptlib
make[1]: Entering directory `/root/ptlib'
make[2]: Entering directory `/root/ptlib'
make[3]: Entering directory `/root/ptlib'
make[3]: Leaving directory `/root/ptlib'
make[2]: Leaving directory `/root/ptlib'
set -e; if test -d /root/ptlib/src ; then make -C /root/ptlib/src optshared;
fi;  if test -d /root/ptlib/plugins ; then make -C/root/ptlib/plugins
optshared; fi;
make[2]: Entering directory `/root/ptlib/src'
make[3]: Entering directory `/root/ptlib/src'
[CC] ptclib/psasl.cxx
In file included from /root/ptlib/include/ptlib/object.h:44,
                 from /root/ptlib/include/ptlib.h:47,
                 from ptclib/psasl.cxx:35:
/root/ptlib/include/ptlib/unix/ptlib/platform.h:555:2: error: #error No
operating system selected.
/root/ptlib/include/ptlib/mutex.h:109: error: ?PThreadIdentifier? does not
name a type
/root/ptlib/include/ptlib/thread.h:292: error: ?PThreadIdentifier? does not
name a type
/root/ptlib/include/ptlib/thread.h:293: error: ?PThreadIdentifier? does not
name a type
/root/ptlib/include/ptlib/thread.h:403: error: ?PThreadIdentifier? does not
name a type
/root/ptlib/include/ptlib/syncthrd.h:326: error: ?PThreadIdentifier? was not
declared in this scope
/root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 1 is
invalid
/root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 3 is
invalid
/root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 4 is
invalid
/root/ptlib/include/ptclib/cypher.h:382: error: ?PUInt32l? does not name a
type
make[3]: *** [/root/ptlib/lib_linux_x86_64/obj/psasl.o] Error 1
make[3]: Leaving directory `/root/ptlib/src'
make[2]: *** [optshared] Error 2
make[2]: Leaving directory `/root/ptlib/src'
make[1]: *** [optshared] Error 2
make[1]: Leaving directory `/root/ptlib'
make: *** [default] Error 2

Many thanks,

Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.com


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From sdame at 207me.com  Sat Nov 10 20:59:22 2012
From: sdame at 207me.com (Stephen Dame)
Date: Sat, 10 Nov 2012 12:59:22 -0500
Subject: [Freeswitch-users] mod_opus codec name.
Message-ID: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com>

I have mod_opus loaded.

 

Trying to find the documentation as to what the name of the CODEC is for
including it in codec preferences.

 

I tried OPUS,   but that doesn't seem to work.

 

<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h,
G7221 at 16000h,G722,PCMU,PCMA,GSM" />

<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=speex at 16000h@20i,PCMU,PCMA,GSM" />

 

Trying to get Jitsi to negotiate 

 

2012-11-10 12:54:22.039330 [DEBUG] sofia.c:5475 Remote SDP:

v=0

o=1000 0 0 IN IP4 192.168.99.198

s=-

c=IN IP4 192.168.99.198

t=0 0

m=audio 5020 RTP/AVP 96

a=rtpmap:96 opus/48000

a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level

a=zrtp-hash:1.10
29740c437bd6d867c08a8854b65368c5052c70ec0b463a06f748062657529239

 

 

thanks in advance,   I googled the list, and docs don't seem to find the
answer.

 

Regards,

Stephen

                                                                     

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From sunzhimailbox at gmail.com  Sat Nov 10 16:46:40 2012
From: sunzhimailbox at gmail.com (zhi sun)
Date: Sat, 10 Nov 2012 21:46:40 +0800
Subject: [Freeswitch-users] are there any high performance rtmp dispatcher
 solution, comparing to the opensips/Kamailo (for SIP)?
Message-ID: <CAA0cQJe9uyAWL9rfh02Bmme52qYYna90h-gBRUgMexcb6xa9TA@mail.gmail.com>

hi guys,

i have an opensips as SIP dispatcher and loadbalance of several freeswitch
(behind). For now, if the end-users use flash rtmp web phone to make a
call, the opensip cannot handle the rtmp request, although the freeswitch
has mod_rtmp.

in this situation, could i use a freeswitch at the front of opensip, and
woks only as rtmp proxy as below?
FS (rtmp dispatcher) --> opensips --> FSs

if not, should i use the 3rd party rtmp gateway/dispatcher to replace the
opensips?

are there any high performance rtmp dispatcher solution, comparing to the
opensips/Kamailo (for SIP)?

i am new to this area, thanks for any suggestions.
-iamsyt
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From anton.jugatsu at gmail.com  Sat Nov 10 23:33:59 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sun, 11 Nov 2012 00:33:59 +0400
Subject: [Freeswitch-users] mod_opus codec name.
In-Reply-To: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com>
References: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com>
Message-ID: <CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com>

As we see in sdp - it's definitely works, but I'm not shure about jitsi.
When I was testing jitsi nightly build for OPUS support it was not working
at all.


2012/11/10 Stephen Dame <sdame at 207me.com>

> I have mod_opus loaded.****
>
> ** **
>
> Trying to find the documentation as to what the name of the CODEC is for
> including it in codec preferences.****
>
> ** **
>
> I tried OPUS,   but that doesn?t seem to work.****
>
> ** **
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,speex at 16000h
> @20i,speex at 8000h@20i,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM" />****
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 16000h@20i,PCMU,PCMA,GSM"
> />****
>
> ** **
>
> Trying to get Jitsi to negotiate ****
>
> ** **
>
> 2012-11-10 12:54:22.039330 [DEBUG] sofia.c:5475 Remote SDP:****
>
> v=0****
>
> o=1000 0 0 IN IP4 192.168.99.198****
>
> s=-****
>
> c=IN IP4 192.168.99.198****
>
> t=0 0****
>
> m=audio 5020 RTP/AVP 96****
>
> a=rtpmap:96 opus/48000****
>
> a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level****
>
> a=zrtp-hash:1.10
> 29740c437bd6d867c08a8854b65368c5052c70ec0b463a06f748062657529239****
>
> ** **
>
> ** **
>
> thanks in advance,   I googled the list, and docs don?t seem to find the
> answer.****
>
> ** **
>
> Regards,****
>
> Stephen****
>
>                                                                      ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at kavun.ch  Sun Nov 11 00:56:20 2012
From: lists at kavun.ch (Emrah)
Date: Sat, 10 Nov 2012 16:56:20 -0500
Subject: [Freeswitch-users] Multiple profiles on a single domain
In-Reply-To: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>
References: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>
Message-ID: <FFD7A3F2-1F67-4C46-944F-024715BE7BF0@kavun.ch>

Hi all,

This is not urgent and I am now tunneling my traffic through a VPN. However if it still something I am interested in and any insight about how domains are aliased would be greatly appreciated.

Cheers!
On Nov 7, 2012, at 7:46 PM, Emrah <lists at kavun.ch> wrote:

> Hi Freeswitchers,
> 
> I had to start an additional profile on port 5070 because my Verizon MiFi blocks 5060 completely.
> For now, 5070 allows me to have a call for 120 seconds? Than the RTP session drops.
> 
> Here is how I started my new profile. Sounds too simple to be true!
> cp internal.xml internal-5070.xml
> Edit internal-5070.xml to change the port number
> Run reloadxml followed by sofia profile internal-5070 start?
> 
> I can register and make short calls, but I do not see my registration when I issue a  sofia_contact username at domain and incoming calls do not work.
> 
> I did not edit anything else in internal-5070.xml and do not force a registration domain either. Sofia status does not show me an additional domain name being created.
> 
> The documentation is a bit unclear to me on how FS finds it's domains? I am sure it is something with the automatic parsing parameter at the top of my sofia profiles, but it still doesn't click? 
> 
> any hint will be enormously appreciated.
> 
> Thanks!
> E



From sdame at 207me.com  Sun Nov 11 03:14:44 2012
From: sdame at 207me.com (Stephen Dame)
Date: Sat, 10 Nov 2012 19:14:44 -0500
Subject: [Freeswitch-users] mod_opus codec name.
In-Reply-To: <CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com>
References: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com>
	<CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com>
Message-ID: <003b01cdbfa1$8e6a3f60$ab3ebe20$@207me.com>

Anton,   thanks for info,  I get call not acceptable here. So no codec seems
to be matching.   is OPUS the right value,  it never shows when its
searching.

 

c=IN IP4 192.168.99.198

t=0 0

m=audio 5032 RTP/AVP 96

a=rtpmap:96 opus/48000

a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level

a=zrtp-hash:1.10
6e63145d4d42de5a83db9967b61e247360ccf093e5d3299eb3605b976901f71
e

m=video 5034 RTP/AVP 97 99

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1

a=rtpmap:99 H264/90000

a=fmtp:99 profile-level-id=4DE01f

a=recvonly

a=imageattr:97 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]

a=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]

a=zrtp-hash:1.10
14c0fd7b5b9c4386a10aa646894fdf8f03710de838f71e8241fca33b79a6592
9

 

2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:362
(sofia/intern
al/1000 at 107.22.240.239) Running State Change CS_NEW

2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:380
(sofia/intern
al/1000 at 107.22.240.239) State NEW

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G726-24:123:8000:20:24000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G729:18:8000:20:8000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[SPEEX:99:32000:20:44000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[SPEEX:99:16000:20:42200]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[SPEEX:99:8000:20:24600]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G7221:115:32000:20:48000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G7221:107:16000:20:32000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G722:9:8000:20:64000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[PCMU:0:8000:20:64000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[PCMA:8:8000:20:64000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[GSM:3:8000:20:13200]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4930 No 2833 in SDP.
Disable 28                                         33 dtmf and switch to
INFO

2012-11-10 18:56:32.159321 [DEBUG] switch_channel.c:2846
(sofia/internal/1000 at 10
7.22.240.239) Callstate Change DOWN -> HANGUP

2012-11-10 18:56:32.159321 [NOTICE] sofia.c:5743 Hangup
sofia/internal/1000 at 107.                                         22.240.239
[CS_NEW] [INCOMPATIBLE_DESTINATION]

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton
Kvashenkin
Sent: Saturday, November 10, 2012 3:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_opus codec name.

 

As we see in sdp - it's definitely works, but I'm not shure about jitsi.
When I was testing jitsi nightly build for OPUS support it was not working
at all.

 

2012/11/10 Stephen Dame <sdame at 207me.com>

I have mod_opus loaded.

 

Trying to find the documentation as to what the name of the CODEC is for
including it in codec preferences.

 

I tried OPUS,   but that doesn't seem to work.

 

<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h,
G7221 at 16000h,G722,PCMU,PCMA,GSM" />

<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=speex at 16000h@20i,PCMU,PCMA,GSM" />

 

Trying to get Jitsi to negotiate 

 

2012-11-10 12:54:22.039330 [DEBUG] sofia.c:5475 Remote SDP:

v=0

o=1000 0 0 IN IP4 192.168.99.198

s=-

c=IN IP4 192.168.99.198

t=0 0

m=audio 5020 RTP/AVP 96

a=rtpmap:96 opus/48000

a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level

a=zrtp-hash:1.10
29740c437bd6d867c08a8854b65368c5052c70ec0b463a06f748062657529239

 

 

thanks in advance,   I googled the list, and docs don't seem to find the
answer.

 

Regards,

Stephen

                                                                     


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




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

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

 

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From mi.ke at null.net  Sun Nov 11 03:33:29 2012
From: mi.ke at null.net (Mi Ke)
Date: Sat, 10 Nov 2012 19:33:29 -0500
Subject: [Freeswitch-users] Error 606 (USER_NOT_REGISTERED) but in
 fact	he is...
Message-ID: <20121111003330.38340@gmx.com>

Anton, thanks for your help - the problem was pre-process directive in config pointing to internal profile by default while I was using external. Everything works now.



----- Original Message -----
From: Anton Kvashenkin
Sent: 11/09/12 06:12 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Error 606 (USER_NOT_REGISTERED) but in fact he is...

 Which soifa profile do you use for registrering this user. Can you post sofia_contact */thatuser at domain. Also, do not forget abount dial_string.

 2012/11/9 Mi Ke < mi.ke at null.net >
Hi,

 I have an IP phone behind NAT registered to also NATed FreeSWITCH (EC2):

 10.10.10.107 (phone) -> 111.111.111.111 (NAT) -> internet -> 222.222.222.222 (NAT) -> 10.210.250.135 (FS)

 Registration part of sip trace looks OK (?):

 tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fdcf8004430): events IN
 tport.c:2845 tport_recv_event() tport_recv_event(0x7fdcf8004430)
 tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x7fdcf8004430) msg 0x7fdcf8010980 from (udp/http://10.210.250.135:5060 ) has 702 bytes, veclen = 1
 tport.c:3004 tport_deliver() tport_deliver(0x7fdcf8004430): msg 0x7fdcf8010980 (702 bytes) from udp/http://111.111.111.111:5060/sip  next=(nil)
 nta.c:2803 agent_recv_request() nta: received REGISTER sip:222.222.222.222 SIP/2.0 (CSeq 49512)
 nta.c:3090 agent_check_request_via() nta: Via check: received=111.111.111.111
 nta.c:3161 agent_aliases() nta: canonizing sip:222.222.222.222 with contact
 nta.c:3002 agent_recv_request() nta: REGISTER (49512) going to a default leg
 2012-11-09 01:14:34.833161 [ERR] sofia_reg.c:1613 DELETE PRESENCE SQL: delete from sip_presence where sip_user='1000' and sip_host='222.222.222.222' and profile_name='external' and open_closed='closed'
 nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_respond
 nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_destroy
 tport.c:3238 tport_tsend() tport_tsend(0x7fdcf8004430) tpn = UDP/http://111.111.111.111:5060 
 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x7fdcf8004430): not found by name UDP/http://111.111.111.111:5060 
 nta.c:6678 incoming_reply() nta: sent 200 OK for REGISTER (49512)

 Here registration LED on the phone goes green, phone gives me a dialing tone i.e. behaves like registered.

 show registrations looks OK:

 reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
 1000,222.222.222.222, 6a7735fa-6f81e64c at 10.10.10.107 ,sofia/external/sip:1000 at 10.10.10.107:5060;fs_nat=yes;fs_path=sip%3A1000%40111.111.111.111%3A5060,1352420589,111.111.111.111,5060,udp,domU-12-31-39-09-F5-79,

 sofia status profile external
 ...
 REGISTRATIONS 1



 Meanwhile when incoming call hits my dialplan (simple bridge to user/1000 at domain) I'm getting ERR 606 (User not registered)

 and here is what list_users gives to me:

 userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number
 1000|default|10.210.250.135|default|error/user_not_registered||Mike|1000

 The question is - what am I doing wrong?

 Thanks in advance for all yours hints

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

 
 

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

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sun Nov 11 07:00:59 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 10 Nov 2012 22:00:59 -0600
Subject: [Freeswitch-users] Multiple profiles on a single domain
In-Reply-To: <FFD7A3F2-1F67-4C46-944F-024715BE7BF0@kavun.ch>
References: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>
	<FFD7A3F2-1F67-4C46-944F-024715BE7BF0@kavun.ch>
Message-ID: <CAKbxfG-yLHQ7bAnEHhzwHF01xssjn+fYEr3AL9TqPL=m2KPW_A@mail.gmail.com>

You forgot to change the name of the profile too in the name attribute.
You can't have 2 profiles with the same name....
On Nov 10, 2012 4:53 PM, "Emrah" <lists at kavun.ch> wrote:

> Hi all,
>
> This is not urgent and I am now tunneling my traffic through a VPN.
> However if it still something I am interested in and any insight about how
> domains are aliased would be greatly appreciated.
>
> Cheers!
> On Nov 7, 2012, at 7:46 PM, Emrah <lists at kavun.ch> wrote:
>
> > Hi Freeswitchers,
> >
> > I had to start an additional profile on port 5070 because my Verizon
> MiFi blocks 5060 completely.
> > For now, 5070 allows me to have a call for 120 seconds? Than the RTP
> session drops.
> >
> > Here is how I started my new profile. Sounds too simple to be true!
> > cp internal.xml internal-5070.xml
> > Edit internal-5070.xml to change the port number
> > Run reloadxml followed by sofia profile internal-5070 start?
> >
> > I can register and make short calls, but I do not see my registration
> when I issue a  sofia_contact username at domain and incoming calls do not
> work.
> >
> > I did not edit anything else in internal-5070.xml and do not force a
> registration domain either. Sofia status does not show me an additional
> domain name being created.
> >
> > The documentation is a bit unclear to me on how FS finds it's domains? I
> am sure it is something with the automatic parsing parameter at the top of
> my sofia profiles, but it still doesn't click?
> >
> > any hint will be enormously appreciated.
> >
> > Thanks!
> > E
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anton.jugatsu at gmail.com  Sun Nov 11 09:26:16 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sun, 11 Nov 2012 10:26:16 +0400
Subject: [Freeswitch-users] mod_opus codec name.
In-Reply-To: <003b01cdbfa1$8e6a3f60$ab3ebe20$@207me.com>
References: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com>
	<CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com>
	<003b01cdbfa1$8e6a3f60$ab3ebe20$@207me.com>
Message-ID: <CANBhyPDhHcc1_Gw+Q3BQS=mL0nUv0_-U2ACbg=WyfK+4_ymGKw@mail.gmail.com>

Don't forget to make reloadxml (F6). Can you paste full debug log with
<sofia profile internla siptrace on> or

ngrep -d <your iface> -qt -W byline port 5060 and host <which making call>


2012/11/11 Stephen Dame <sdame at 207me.com>

> Anton,   thanks for info,  I get call not acceptable here. So no codec
> seems to be matching?   is OPUS the right value,  it never shows when its
> searching.****
>
> ** **
>
> c=IN IP4 192.168.99.198****
>
> t=0 0****
>
> m=audio 5032 RTP/AVP 96****
>
> a=rtpmap:96 opus/48000****
>
> a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level****
>
> a=zrtp-hash:1.10
> 6e63145d4d42de5a83db9967b61e247360ccf093e5d3299eb3605b976901f71
> e****
>
> m=video 5034 RTP/AVP 97 99****
>
> a=rtpmap:97 H264/90000****
>
> a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1****
>
> a=rtpmap:99 H264/90000****
>
> a=fmtp:99 profile-level-id=4DE01f****
>
> a=recvonly****
>
> a=imageattr:97 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]****
>
> a=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]****
>
> a=zrtp-hash:1.10
> 14c0fd7b5b9c4386a10aa646894fdf8f03710de838f71e8241fca33b79a6592
> 9****
>
> ** **
>
> 2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:362
> (sofia/intern                                         al/
> 1000 at 107.22.240.239) Running State Change CS_NEW****
>
> 2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:380
> (sofia/intern                                         al/
> 1000 at 107.22.240.239) State NEW****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[G726-24:123:8000:20:24000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
>                          6:48000:20:0]/[G729:18:8000:20:8000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[SPEEX:99:32000:20:44000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[SPEEX:99:16000:20:42200]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[SPEEX:99:8000:20:24600]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[G7221:115:32000:20:48000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[G7221:107:16000:20:32000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[G722:9:8000:20:64000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[PCMU:0:8000:20:64000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
>                          6:48000:20:0]/[PCMA:8:8000:20:64000]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [opus:9
> 6:48000:20:0]/[GSM:3:8000:20:13200]****
>
> 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4930 No 2833 in SDP.
> Disable 28                                         33 dtmf and switch to
> INFO****
>
> 2012-11-10 18:56:32.159321 [DEBUG] switch_channel.c:2846
> (sofia/internal/1000 at 10
> 7.22.240.239) Callstate Change DOWN -> HANGUP****
>
> 2012-11-10 18:56:32.159321 [NOTICE] sofia.c:5743 Hangup
> sofia/internal/1000 at 107.
> 22.240.239 [CS_NEW] [INCOMPATIBLE_DESTINATION]****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anton
> Kvashenkin
> *Sent:* Saturday, November 10, 2012 3:34 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_opus codec name.****
>
> ** **
>
> As we see in sdp - it's definitely works, but I'm not shure about jitsi.
> When I was testing jitsi nightly build for OPUS support it was not working
> at all.****
>
> ** **
>
> 2012/11/10 Stephen Dame <sdame at 207me.com>****
>
> I have mod_opus loaded.****
>
>  ****
>
> Trying to find the documentation as to what the name of the CODEC is for
> including it in codec preferences.****
>
>  ****
>
> I tried OPUS,   but that doesn?t seem to work.****
>
>  ****
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,speex at 16000h
> @20i,speex at 8000h@20i,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM" />****
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 16000h@20i,PCMU,PCMA,GSM"
> />****
>
>  ****
>
> Trying to get Jitsi to negotiate ****
>
>  ****
>
> 2012-11-10 12:54:22.039330 [DEBUG] sofia.c:5475 Remote SDP:****
>
> v=0****
>
> o=1000 0 0 IN IP4 192.168.99.198****
>
> s=-****
>
> c=IN IP4 192.168.99.198****
>
> t=0 0****
>
> m=audio 5020 RTP/AVP 96****
>
> a=rtpmap:96 opus/48000****
>
> a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level****
>
> a=zrtp-hash:1.10
> 29740c437bd6d867c08a8854b65368c5052c70ec0b463a06f748062657529239****
>
>  ****
>
>  ****
>
> thanks in advance,   I googled the list, and docs don?t seem to find the
> answer.****
>
>  ****
>
> Regards,****
>
> Stephen****
>
>                                                                      ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at kavun.ch  Sun Nov 11 10:13:14 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 11 Nov 2012 02:13:14 -0500
Subject: [Freeswitch-users] Multiple profiles on a single domain
In-Reply-To: <CAKbxfG-yLHQ7bAnEHhzwHF01xssjn+fYEr3AL9TqPL=m2KPW_A@mail.gmail.com>
References: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>
	<FFD7A3F2-1F67-4C46-944F-024715BE7BF0@kavun.ch>
	<CAKbxfG-yLHQ7bAnEHhzwHF01xssjn+fYEr3AL9TqPL=m2KPW_A@mail.gmail.com>
Message-ID: <F7FD48A0-C55A-4A09-B979-8CEB027FAC0F@kavun.ch>

You're good, I actually did change the name but forgot to mention it in the email.
The profile is up but registrations do not list in sofia_contact.

Can you point me out to some documentation on how FS uses domains? I thoroughly checked the Wiki? I understand all that is there to understand. But yet I am not sure how it all connects together.
I know how it works technically and can operate a multi domain set up, but how does a profile identify itself to a domain?
I am not forcing a domain in the params and let it parse by itself. O at least I think it parses it?

Thanks!
On Nov 10, 2012, at 11:00 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> You forgot to change the name of the profile too in the name attribute.  You can't have 2 profiles with the same name....
> 
> On Nov 10, 2012 4:53 PM, "Emrah" <lists at kavun.ch> wrote:
> Hi all,
> 
> This is not urgent and I am now tunneling my traffic through a VPN. However if it still something I am interested in and any insight about how domains are aliased would be greatly appreciated.
> 
> Cheers!
> On Nov 7, 2012, at 7:46 PM, Emrah <lists at kavun.ch> wrote:
> 
> > Hi Freeswitchers,
> >
> > I had to start an additional profile on port 5070 because my Verizon MiFi blocks 5060 completely.
> > For now, 5070 allows me to have a call for 120 seconds? Than the RTP session drops.
> >
> > Here is how I started my new profile. Sounds too simple to be true!
> > cp internal.xml internal-5070.xml
> > Edit internal-5070.xml to change the port number
> > Run reloadxml followed by sofia profile internal-5070 start?
> >
> > I can register and make short calls, but I do not see my registration when I issue a  sofia_contact username at domain and incoming calls do not work.
> >
> > I did not edit anything else in internal-5070.xml and do not force a registration domain either. Sofia status does not show me an additional domain name being created.
> >
> > The documentation is a bit unclear to me on how FS finds it's domains? I am sure it is something with the automatic parsing parameter at the top of my sofia profiles, but it still doesn't click?
> >
> > any hint will be enormously appreciated.
> >
> > Thanks!
> > E
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From eugene.prokopiev at gmail.com  Sat Nov 10 22:02:09 2012
From: eugene.prokopiev at gmail.com (Eugene Prokopiev)
Date: Sat, 10 Nov 2012 22:02:09 +0300
Subject: [Freeswitch-users] Check user state in dialplan
Message-ID: <CANM4RuigJ6CRDgZSYBzH3nrMYUGAkO-fT67V6U48ng+Cg2W_6A@mail.gmail.com>

Hi,

I need to perform different actions in response to the following conditions:

* The called number is not exists in the directory (check id and
number-alias)
* The called number is known, but not registered
* The called number is busy
* The called number is not answered due timeout
* The called number is answered

In the first case I need to send the call through a particular gateway, in
the next two cases I need to play a file, in two remaining cases I need to
call user and play a file only on timeout.

What is the right to do it in the dialplan?

-- 
Regards,
Eugene Prokopiev
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From eugene.prokopiev at gmail.com  Sat Nov 10 22:03:28 2012
From: eugene.prokopiev at gmail.com (Eugene Prokopiev)
Date: Sat, 10 Nov 2012 22:03:28 +0300
Subject: [Freeswitch-users] Nested contexts emulation
Message-ID: <CANM4RuhYnJV29PvdvffNBij0H2FsFWqJrd=H-h4EYcyQW-3Svw@mail.gmail.com>

Hi,

I can include one context to another. It is very convenient to avoid
duplication of the configuration code. With FreeSWITCH it is also quite in
demand feature for the same extensions in different contexts for different
profiles. But the only way to avoid duplication is to have one big context
and block certain extensions by conditions for different profile. Can I do
this by another way?


-- 
Regards,
Eugene Prokopiev
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From anton.jugatsu at gmail.com  Sun Nov 11 12:32:04 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sun, 11 Nov 2012 13:32:04 +0400
Subject: [Freeswitch-users] Multiple profiles on a single domain
In-Reply-To: <F7FD48A0-C55A-4A09-B979-8CEB027FAC0F@kavun.ch>
References: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>
	<FFD7A3F2-1F67-4C46-944F-024715BE7BF0@kavun.ch>
	<CAKbxfG-yLHQ7bAnEHhzwHF01xssjn+fYEr3AL9TqPL=m2KPW_A@mail.gmail.com>
	<F7FD48A0-C55A-4A09-B979-8CEB027FAC0F@kavun.ch>
Message-ID: <CANBhyPC9nKZGQK4puX4Q5BUdozEh-yEq08_S8OyvvDHjKDPfPg@mail.gmail.com>

sofia_contact */user at domain

also check dial_string


2012/11/11 Emrah <lists at kavun.ch>

> You're good, I actually did change the name but forgot to mention it in
> the email.
> The profile is up but registrations do not list in sofia_contact.
>
> Can you point me out to some documentation on how FS uses domains? I
> thoroughly checked the Wiki? I understand all that is there to understand.
> But yet I am not sure how it all connects together.
> I know how it works technically and can operate a multi domain set up, but
> how does a profile identify itself to a domain?
> I am not forcing a domain in the params and let it parse by itself. O at
> least I think it parses it?
>
> Thanks!
> On Nov 10, 2012, at 11:00 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> > You forgot to change the name of the profile too in the name attribute.
>  You can't have 2 profiles with the same name....
> >
> > On Nov 10, 2012 4:53 PM, "Emrah" <lists at kavun.ch> wrote:
> > Hi all,
> >
> > This is not urgent and I am now tunneling my traffic through a VPN.
> However if it still something I am interested in and any insight about how
> domains are aliased would be greatly appreciated.
> >
> > Cheers!
> > On Nov 7, 2012, at 7:46 PM, Emrah <lists at kavun.ch> wrote:
> >
> > > Hi Freeswitchers,
> > >
> > > I had to start an additional profile on port 5070 because my Verizon
> MiFi blocks 5060 completely.
> > > For now, 5070 allows me to have a call for 120 seconds? Than the RTP
> session drops.
> > >
> > > Here is how I started my new profile. Sounds too simple to be true!
> > > cp internal.xml internal-5070.xml
> > > Edit internal-5070.xml to change the port number
> > > Run reloadxml followed by sofia profile internal-5070 start?
> > >
> > > I can register and make short calls, but I do not see my registration
> when I issue a  sofia_contact username at domain and incoming calls do not
> work.
> > >
> > > I did not edit anything else in internal-5070.xml and do not force a
> registration domain either. Sofia status does not show me an additional
> domain name being created.
> > >
> > > The documentation is a bit unclear to me on how FS finds it's domains?
> I am sure it is something with the automatic parsing parameter at the top
> of my sofia profiles, but it still doesn't click?
> > >
> > > any hint will be enormously appreciated.
> > >
> > > Thanks!
> > > E
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Sun Nov 11 13:58:40 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 11 Nov 2012 18:58:40 +0800
Subject: [Freeswitch-users] Nested contexts emulation
In-Reply-To: <CANM4RuhYnJV29PvdvffNBij0H2FsFWqJrd=H-h4EYcyQW-3Svw@mail.gmail.com>
References: <CANM4RuhYnJV29PvdvffNBij0H2FsFWqJrd=H-h4EYcyQW-3Svw@mail.gmail.com>
Message-ID: <5D527A2BA8704F5D8632365453282D04@gmail.com>

you can 

1) transfer from one context to another
2) put all common extensions in one file and "include" into multi- contexes.


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


On Sunday, November 11, 2012 at 3:03 AM, Eugene Prokopiev wrote:

> Hi,
> 
> I can include one context to another. It is very convenient to avoid duplication of the configuration code. With FreeSWITCH it is also quite in demand feature for the same extensions in different contexts for different profiles. But the only way to avoid duplication is to have one big context and block certain extensions by conditions for different profile. Can I do this by another way?
> 
> 
> 
> 
> -- 
> Regards,
> Eugene Prokopiev
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-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
> 
> 


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From miha at softnet.si  Sun Nov 11 14:38:07 2012
From: miha at softnet.si (Miha)
Date: Sun, 11 Nov 2012 13:38:07 +0200
Subject: [Freeswitch-users] Multiple profiles on a single domain
In-Reply-To: <CAKbxfG-yLHQ7bAnEHhzwHF01xssjn+fYEr3AL9TqPL=m2KPW_A@mail.gmail.com>
References: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>
	<FFD7A3F2-1F67-4C46-944F-024715BE7BF0@kavun.ch>
	<CAKbxfG-yLHQ7bAnEHhzwHF01xssjn+fYEr3AL9TqPL=m2KPW_A@mail.gmail.com>
Message-ID: <web-87816060@softnet.si>

Hi,

I was also testing some things with multple profiles and
also experiance the incoming calls were not working for
profile with port 5070. After i use presence for storing
reg user in db users were avalible from ouside.

If i read it right usinig presence thakes a lot of cpu
resurses, so is there any outer whey?

Thanks?

On Sat, 10 Nov 2012 22:00:59 -0600
 Anthony Minessale <anthony.minessale at gmail.com> wrote:
> You forgot to change the name of the profile too in the
> name attribute.
> You can't have 2 profiles with the same name....
> On Nov 10, 2012 4:53 PM, "Emrah" <lists at kavun.ch> wrote:
> 
> > Hi all,
> >
> > This is not urgent and I am now tunneling my traffic
> through a VPN.
> > However if it still something I am interested in and
> any insight about how
> > domains are aliased would be greatly appreciated.
> >
> > Cheers!
> > On Nov 7, 2012, at 7:46 PM, Emrah <lists at kavun.ch>
> wrote:
> >
> > > Hi Freeswitchers,
> > >
> > > I had to start an additional profile on port 5070
> because my Verizon
> > MiFi blocks 5060 completely.
> > > For now, 5070 allows me to have a call for 120
> seconds? Than the RTP
> > session drops.
> > >
> > > Here is how I started my new profile. Sounds too
> simple to be true!
> > > cp internal.xml internal-5070.xml
> > > Edit internal-5070.xml to change the port number
> > > Run reloadxml followed by sofia profile internal-5070
> start?
> > >
> > > I can register and make short calls, but I do not see
> my registration
> > when I issue a  sofia_contact username at domain and
> incoming calls do not
> > work.
> > >
> > > I did not edit anything else in internal-5070.xml and
> do not force a
> > registration domain either. Sofia status does not show
> me an additional
> > domain name being created.
> > >
> > > The documentation is a bit unclear to me on how FS
> finds it's domains? I
> > am sure it is something with the automatic parsing
> parameter at the top of
> > my sofia profiles, but it still doesn't click?
> > >
> > > any hint will be enormously appreciated.
> > >
> > > Thanks!
> > > E
> >
> >
> >
>
_________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> >
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >



From lconroy at insensate.co.uk  Sun Nov 11 16:09:16 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Sun, 11 Nov 2012 13:09:16 +0000
Subject: [Freeswitch-users] Domains and profiles
Message-ID: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>

Hi Folks,
 I've started a new thread as it's not quite the same issue, and domains & profiles have confused the heck out of me every time I have developed a new setup for fS.
I have sometimes had to hack/hard-doce the dialstring to make multiple domains in one profile work, had hours of fun with presence, db and force register settings, and have still had some odd gotchas that have required extensive meditation.
[... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract these elements ]

Coming at this from standards/specs and rolling my own SIP stacks, sofia/fS seems to use the term "domain" differently from sipdomain, and alias seems to be tied to the directory (and thus to the profile listed in a directory file), but I'm not sure. 
so ...
Before I capture to the sofia conf xml wiki page, I have a couple of questions on the sip-profile XML setup;

Q: Is there a particular reason why there's a parameter called alias and an (entirely different) setting also called alias?
 The sofia conf xml wiki's comment on the setting "alias" shows I'm not alone.
 I agree that's what it appears to be doing, but can we nail this down please (and what happens if an external client uses this connection to register and call)?

In the current sofia conf xml wiki page, the domain setting is not exactly well documented :).
The current internal.xml vanilla example from git (as of time of writing) has the following lines:
-------------------------
...
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
 <!--aliases are other names that will work as a valid profile name for this profile-->
 <aliases>
   <!--
        <alias name="default"/>
    -->
 </aliases>
 <!-- Outbound Registrations -->
 <gateways>
   <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
 </gateways>
 
 <domains>
   <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
   <!--<domain name="$${domain}" parse="true"/>-->
   <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
   <!--<domain name="all" alias="true" parse="true"/>-->
   <domain name="all" alias="true" parse="false"/>
 </domains>
...
-------------------------

This stuff is entirely missing from the sofia.conf.xml wiki page, and it IS really important.


Q: what's the default value for the alias parameter in the domain element? -- it is missing from the first example.
Q: if there is more than one profile, what's the impact of setting parse = true in one (or all) of the profiles' XML files?
   (or parse = false, or missing the parameter altogether)?
AFAICT, the gateways get pulled in via the pre-process directive just fine, regardless of the value of the parse parameter -- it works for me, at least.

Q: if there is more than one profile, what's the impact of putting domain name="all" into one (or all) of the profiles' XML files?

Ideally, having more than one sipdomain tied to one profile "would be good", but aliases doesn't do that -- as the git file says, these are aliases for the profile name.

Before I start scribbling, Answers on a postcard to this ML, please.

all the best,
  Lawrence

From abaci64 at gmail.com  Mon Nov 12 02:35:39 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 11 Nov 2012 18:35:39 -0500
Subject: [Freeswitch-users] general question about phone provisioning
Message-ID: <50A0364B.8050701@gmail.com>

This question is not specific to FreeSWITCH, just a general question 
that I would like to get feedback from other FreeSWITCH users.
I'm thinking of setting up phone provisioning via http, my question is 
how to make this setup secure. say my provisioning server will listen on 
https://myserver.com and a phone with the mac address 00-15-65-22-F4-23 
will try to pull the config as https://myserver.com /00156518425Dhow do 
I prevent hackers from trying to get config files using a brute force 
attack. is there any standard way of securing against these types of 
attacks?


From prasd.d.b at gmail.com  Mon Nov 12 03:21:17 2012
From: prasd.d.b at gmail.com (Prasd D)
Date: Sun, 11 Nov 2012 16:21:17 -0800
Subject: [Freeswitch-users] voicebox email notifications on Debian with
	exim4
In-Reply-To: <50633768.3060601@gmail.com>
References: <50604618.4010602@gmail.com>
	<CAKV-jFzQdV5biKpWagccJ0CjO5hXUaSMAD77sa_LZC1KRH948A@mail.gmail.com>
	<50633768.3060601@gmail.com>
Message-ID: <CAPZnxCRox9CWVnzf_Z2kO1yi-ee=kc-3a9c+EGC8c5Mt6vD-7A@mail.gmail.com>

I am unable to get it to work with eximcompat.sh as mentioned in the
Wiki. I tried with msmtp and mailx and same problem.
I don't see the email being sent (or received). Actually I don't even
see any attempt to send email. In console with debug level logging I
don't see any reference to email at all.

I changed these two lines accordingly in switch.conf.xml (in each
case, mailx example is shown below).
<param name="mailer-app" value="mailx"/>
    <!--<param name="mailer-app-args" value="-t"/>-->
   <param name="mailer-app-args" value=""/>

Its as though freeswitch doesn't attempt to use the email at all !
I can hear the play the voicemail from sip client fine

Any help is appreciated.

Thanks,
Prasd

On 9/26/12, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
> I already added to the wiki by the time you replied (also sent an update
> to the mailing list). Before that, msmtp was mentioned only for windows.
>
> You can find what I added at:
>    *
> http://wiki.freeswitch.org/wiki/Mod_voicemail#Using_MSMTP_for_Local_Relay_to_Exim4_on_Debian
>
>
>
> Plus a note at beginning of:
>    * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>
> Cheers,
> Daniel
>
> On 9/24/12 11:12 PM, Brian Foster wrote:
>>
>> Im pretty sure this is mentioned already on the wiki, but please
>> elaborate your process to fill in some gaps.
>>
>> -BDF
>>
>> On Sep 24, 2012 8:33 AM, "Daniel-Constantin Mierla" <miconda at gmail.com
>> <mailto:miconda at gmail.com>> wrote:
>>
>>     Hello,
>>
>>     I got into the crashing issue for freeswitch voicemail email
>>     notifications on Debian with exim4, reported at:
>>        * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>>
>>     My solution was to use msmtp alongside exim4 -- msmtp is used to
>> relay
>>     the emails to the local exim4 instance. All seems to work ok, I
>> wonder
>>     if anyone else is using same approach and got any issues or there are
>>     other solutions found meanwhile for debian+exim4 combination.
>>
>>     I plan to put more details there, once the registration process for
>> my
>>     account on the wiki is completed.
>>
>>     Cheers,
>>     Daniel
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -
> http://asipto.com/u/katu
>
>


-- 
Thanks,
Prasd


From jnvines at gmail.com  Mon Nov 12 04:03:49 2012
From: jnvines at gmail.com (Nick Vines)
Date: Sun, 11 Nov 2012 20:03:49 -0500
Subject: [Freeswitch-users] general question about phone provisioning
In-Reply-To: <50A0364B.8050701@gmail.com>
References: <50A0364B.8050701@gmail.com>
Message-ID: <CA+NjZ2vZveFbaVB9zPGZBMpJt+93W=adDgfKJLA21qdx1uWGQw@mail.gmail.com>

I used to use HTTP with no auth, but changed for a bit more security. When
I used basic HTTP, here are some of the things I did:

1. Only leave config files on server when something needs to be changed.
After device has synced, the file gets taken off public access.
2. Files are encrypted per device settings (I know Grandstream and Cisco
support some instance of this, I expect most do).
3. Random path and prefix that I already gave to device in pre-provisioning
before sent to customer.
4. Make sure you don't have indexes enabled on your webserver. For example,
see here <http://wiki.apache.org/httpd/DirectoryListings>. If you can type
in myserver.com/blah404_not_valid and see a list of the files and folders,
you need to change that.

But, if you want more, you could enable authentication for your devices and
have the certificate/username/password already loaded on the device (first
provision before you send it out). That will be more specific to your
device.

I'm sure other have more suggestions, but the above should help you stay
relatively secure. Keeping files off the server, with random paths, and
prefixes, should help prevent a brute force scan being successful.

Nick






On Sun, Nov 11, 2012 at 6:35 PM, Abaci <abaci64 at gmail.com> wrote:

> This question is not specific to FreeSWITCH, just a general question
> that I would like to get feedback from other FreeSWITCH users.
> I'm thinking of setting up phone provisioning via http, my question is
> how to make this setup secure. say my provisioning server will listen on
> https://myserver.com and a phone with the mac address 00-15-65-22-F4-23
> will try to pull the config as https://myserver.com /00156518425Dhow do
> I prevent hackers from trying to get config files using a brute force
> attack. is there any standard way of securing against these types of
> attacks?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nandy1925 at gmail.com  Mon Nov 12 04:33:58 2012
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Mon, 12 Nov 2012 09:33:58 +0800
Subject: [Freeswitch-users] general question about phone provisioning
In-Reply-To: <50A0364B.8050701@gmail.com>
References: <50A0364B.8050701@gmail.com>
Message-ID: <CAHKqMb68SLtEW+nFJqXM+=Gj2C7DLuoZXeReL0zOK=aZyEKXHA@mail.gmail.com>

Hi abaci,

IMHO, start with encrypted config files. Then, I also see some ATA with
password option which means login sessions can be setup with your
provisioning servers for more security.

/nandy

On Mon, Nov 12, 2012 at 7:35 AM, Abaci <abaci64 at gmail.com> wrote:

> This question is not specific to FreeSWITCH, just a general question
> that I would like to get feedback from other FreeSWITCH users.
> I'm thinking of setting up phone provisioning via http, my question is
> how to make this setup secure. say my provisioning server will listen on
> https://myserver.com and a phone with the mac address 00-15-65-22-F4-23
> will try to pull the config as https://myserver.com /00156518425Dhow do
> I prevent hackers from trying to get config files using a brute force
> attack. is there any standard way of securing against these types of
> attacks?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at kavun.ch  Mon Nov 12 06:09:28 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 11 Nov 2012 22:09:28 -0500
Subject: [Freeswitch-users] Domains and profiles
In-Reply-To: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
References: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
Message-ID: <1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>

Bless you!

Thanks for putting this together. You've beautifully summed up all my questions.
On Nov 11, 2012, at 8:09 AM, Lawrence Conroy <lconroy at insensate.co.uk> wrote:

> Hi Folks,
> I've started a new thread as it's not quite the same issue, and domains & profiles have confused the heck out of me every time I have developed a new setup for fS.
> I have sometimes had to hack/hard-doce the dialstring to make multiple domains in one profile work, had hours of fun with presence, db and force register settings, and have still had some odd gotchas that have required extensive meditation.
> [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract these elements ]
> 
> Coming at this from standards/specs and rolling my own SIP stacks, sofia/fS seems to use the term "domain" differently from sipdomain, and alias seems to be tied to the directory (and thus to the profile listed in a directory file), but I'm not sure. 
> so ...
> Before I capture to the sofia conf xml wiki page, I have a couple of questions on the sip-profile XML setup;
> 
> Q: Is there a particular reason why there's a parameter called alias and an (entirely different) setting also called alias?
> The sofia conf xml wiki's comment on the setting "alias" shows I'm not alone.
> I agree that's what it appears to be doing, but can we nail this down please (and what happens if an external client uses this connection to register and call)?
> 
> In the current sofia conf xml wiki page, the domain setting is not exactly well documented :).
> The current internal.xml vanilla example from git (as of time of writing) has the following lines:
> -------------------------
> ...
> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> <!--aliases are other names that will work as a valid profile name for this profile-->
> <aliases>
>   <!--
>        <alias name="default"/>
>    -->
> </aliases>
> <!-- Outbound Registrations -->
> <gateways>
>   <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
> </gateways>
> 
> <domains>
>   <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
>   <!--<domain name="$${domain}" parse="true"/>-->
>   <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
>   <!--<domain name="all" alias="true" parse="true"/>-->
>   <domain name="all" alias="true" parse="false"/>
> </domains>
> ...
> -------------------------
> 
> This stuff is entirely missing from the sofia.conf.xml wiki page, and it IS really important.
> 
> 
> Q: what's the default value for the alias parameter in the domain element? -- it is missing from the first example.
> Q: if there is more than one profile, what's the impact of setting parse = true in one (or all) of the profiles' XML files?
>   (or parse = false, or missing the parameter altogether)?
> AFAICT, the gateways get pulled in via the pre-process directive just fine, regardless of the value of the parse parameter -- it works for me, at least.
> 
> Q: if there is more than one profile, what's the impact of putting domain name="all" into one (or all) of the profiles' XML files?
> 
> Ideally, having more than one sipdomain tied to one profile "would be good", but aliases doesn't do that -- as the git file says, these are aliases for the profile name.
> 
> Before I start scribbling, Answers on a postcard to this ML, please.
> 
> all the best,
>  Lawrence
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lists at kavun.ch  Mon Nov 12 06:12:36 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 11 Nov 2012 22:12:36 -0500
Subject: [Freeswitch-users] Multiple profiles on a single domain
In-Reply-To: <CANBhyPC9nKZGQK4puX4Q5BUdozEh-yEq08_S8OyvvDHjKDPfPg@mail.gmail.com>
References: <386C5562-B6C9-4BB9-A074-D97304CA4C7A@kavun.ch>
	<FFD7A3F2-1F67-4C46-944F-024715BE7BF0@kavun.ch>
	<CAKbxfG-yLHQ7bAnEHhzwHF01xssjn+fYEr3AL9TqPL=m2KPW_A@mail.gmail.com>
	<F7FD48A0-C55A-4A09-B979-8CEB027FAC0F@kavun.ch>
	<CANBhyPC9nKZGQK4puX4Q5BUdozEh-yEq08_S8OyvvDHjKDPfPg@mail.gmail.com>
Message-ID: <491C3A59-92AE-454E-B81B-A3A09C1C7F12@kavun.ch>

That is not the question. I know how to address my calls and seem to have grasped 95% of the use of domain names in FS.
If you have 2 profiles on the same IP on port 5060 and 5070 and do not enforce a domain name, it's the first profile that comes up that will lock the registrations table for your domain.

Let's continue this in Lawrence's thread, thanks for your help.
On Nov 11, 2012, at 4:32 AM, Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:

> sofia_contact */user at domain
> 
> also check dial_string
> 
> 
> 2012/11/11 Emrah <lists at kavun.ch>
> You're good, I actually did change the name but forgot to mention it in the email.
> The profile is up but registrations do not list in sofia_contact.
> 
> Can you point me out to some documentation on how FS uses domains? I thoroughly checked the Wiki? I understand all that is there to understand. But yet I am not sure how it all connects together.
> I know how it works technically and can operate a multi domain set up, but how does a profile identify itself to a domain?
> I am not forcing a domain in the params and let it parse by itself. O at least I think it parses it?
> 
> Thanks!
> On Nov 10, 2012, at 11:00 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
> > You forgot to change the name of the profile too in the name attribute.  You can't have 2 profiles with the same name....
> >
> > On Nov 10, 2012 4:53 PM, "Emrah" <lists at kavun.ch> wrote:
> > Hi all,
> >
> > This is not urgent and I am now tunneling my traffic through a VPN. However if it still something I am interested in and any insight about how domains are aliased would be greatly appreciated.
> >
> > Cheers!
> > On Nov 7, 2012, at 7:46 PM, Emrah <lists at kavun.ch> wrote:
> >
> > > Hi Freeswitchers,
> > >
> > > I had to start an additional profile on port 5070 because my Verizon MiFi blocks 5060 completely.
> > > For now, 5070 allows me to have a call for 120 seconds? Than the RTP session drops.
> > >
> > > Here is how I started my new profile. Sounds too simple to be true!
> > > cp internal.xml internal-5070.xml
> > > Edit internal-5070.xml to change the port number
> > > Run reloadxml followed by sofia profile internal-5070 start?
> > >
> > > I can register and make short calls, but I do not see my registration when I issue a  sofia_contact username at domain and incoming calls do not work.
> > >
> > > I did not edit anything else in internal-5070.xml and do not force a registration domain either. Sofia status does not show me an additional domain name being created.
> > >
> > > The documentation is a bit unclear to me on how FS finds it's domains? I am sure it is something with the automatic parsing parameter at the top of my sofia profiles, but it still doesn't click?
> > >
> > > any hint will be enormously appreciated.
> > >
> > > Thanks!
> > > E
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From meditel at gmail.com  Mon Nov 12 03:18:59 2012
From: meditel at gmail.com (Meditel)
Date: Mon, 12 Nov 2012 01:18:59 +0100
Subject: [Freeswitch-users] Bridging an incoming call to outbound gateway
Message-ID: <CAHkDqS_=2u1XvbkwH3fRdt750tX833qJ50SUmpOs+77ChV_+zg@mail.gmail.com>

Hi,

I have actually 2 providers : p1 & p2
p1 is used as a DID only for incoming calls
p2 is used for making outgoing calls

I have 1 registered user : ext 1000

I have 2 GSM phones : GSM1 & GSM2

My dialplan is as :
<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(MY_P1_DID)$">
        <action application="transfer" data="1000 XML default"/>
    </condition>
  </extension>
</include>

For incoming calls "1000" is ringing and i can make a call without any
problem (GSM1 => P1_DID => ext 1000)
But whene trying to bridge the incoming call to p2 gateway (GSM1 => P1_DID
=> P2_GW => GSM2), i can see that GSM2 is ringing but whene i answer the
call i can't hear any audio ...
This is my dialplan for the second case:
<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(MY_P1_DID)$">
        <action application="bridge"
data="sofia/gateway/p2_gw/GSM2_NUMBER"/>
    </condition>
  </extension>
</include>

Any help are welcome
best regards
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From sdame at 207me.com  Mon Nov 12 14:27:28 2012
From: sdame at 207me.com (Stephen Dame)
Date: Mon, 12 Nov 2012 06:27:28 -0500
Subject: [Freeswitch-users] mod_opus codec name.
In-Reply-To: <CANBhyPDhHcc1_Gw+Q3BQS=mL0nUv0_-U2ACbg=WyfK+4_ymGKw@mail.gmail.com>
References: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com><CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com><003b0
	1cdbfa1$8e6a3f60$ab3ebe20$@207me.com>
	<CANBhyPDhHcc1_Gw+Q3BQS=mL0nUv0_-U2ACbg=WyfK+4_ymGKw@mail.gmail.com>
Message-ID: <015e01cdc0c8$b402ffa0$1c08fee0$@207me.com>

Anton,  I have reloaded and restarted FS. And made sure mod_opus is loaded.

 

I still see no documentation in forums or wiki, as to what name should be
used for opus in global codec preferences.

 

I get 488,  which looks like  freeswitch is never looking to compare it
against the preference list.

 

Thanks

Stephen

 

T 2012/11/12 06:20:38.776257 67.253.29.100:42778 -> 10.126.171.197:5080 [AP]

INVITE sip:5000 at 107.22.240.239 SIP/2.0.

Call-ID: f3a55676d1f742c782a8eb400049d626 at 0:0:0:0:0:0:0:0.

CSeq: 2 INVITE.

From: "Stephen Dame" <sip:1000 at 107.22.240.239>;tag=771781e0.

To: <sip:5000 at 107.22.240.239>.

Max-Forwards: 70.

Contact: "Stephen Dame"
<sip:1000 at 192.168.99.198:50441;transport=tcp;registering_acc=107_22_240_239>
.

User-Agent: Jitsi1.1.4310.10045Windows 7.

Content-Type: application/sdp.

Via: SIP/2.0/TCP
192.168.99.198:50441;branch=z9hG4bK-333236-9a42b9b6a906d0549253b69d9eb87431.

Proxy-Authorization: Digest
username="1000",realm="107.22.240.239",nonce="9afe2f18-8924-41be-9cab-82c371
bc7529",uri="sip:5000 at 107.22.240.239",response="f93d24497a79f51d81bee9a3a851
81bf",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001.

Content-Length: 668.

.

v=0.

o=1000 0 0 IN IP4 192.168.99.198.

s=-.

c=IN IP4 192.168.99.198.

t=0 0.

m=audio 5060 RTP/AVP 96.

a=rtpmap:96 opus/48000.

a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level.

a=zrtp-hash:1.10
fa977be972f6bf8a035e3519ba22b89dfbf191750ec9c3d027ce9cc73c7da199.

m=video 5062 RTP/AVP 97 99.

a=recvonly.

a=rtpmap:97 H264/90000.

a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1.

a=imageattr:97 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]].

a=rtpmap:99 H264/90000.

a=fmtp:99 profile-level-id=4DE01f.

a=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]].

a=zrtp-hash:1.10
711982f565cb5da0fe3added9c7c50e046df17a0d44b463371900ab295bc972f.

 

 

T 2012/11/12 06:20:38.776481 10.126.171.197:5080 -> 67.253.29.100:42778 [AP]

SIP/2.0 100 Trying.

Via: SIP/2.0/TCP
192.168.99.198:50441;branch=z9hG4bK-333236-9a42b9b6a906d0549253b69d9eb87431;
received=67.253.29.100;rport=42778.

From: "Stephen Dame" <sip:1000 at 107.22.240.239>;tag=771781e0.

To: <sip:5000 at 107.22.240.239>.

Call-ID: f3a55676d1f742c782a8eb400049d626 at 0:0:0:0:0:0:0:0.

CSeq: 2 INVITE.

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1ddd29b 2011-12-18 12-08-17
-0500.

Content-Length: 0.

.

 

 

T 2012/11/12 06:20:38.783929 10.126.171.197:5080 -> 67.253.29.100:42778 [AP]

SIP/2.0 488 Not Acceptable Here.

Via: SIP/2.0/TCP
192.168.99.198:50441;branch=z9hG4bK-333236-9a42b9b6a906d0549253b69d9eb87431;
received=67.253.29.100;rport=42778.

From: "Stephen Dame" <sip:1000 at 107.22.240.239>;tag=771781e0.

To: <sip:5000 at 107.22.240.239>;tag=pDHc0ma2Q58cr.

Call-ID: f3a55676d1f742c782a8eb400049d626 at 0:0:0:0:0:0:0:0.

CSeq: 2 INVITE.

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1ddd29b 2011-12-18 12-08-17
-0500.

Accept: application/sdp.

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.

Supported: timer, precondition, path, replaces.

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.

Content-Length: 0.

Remote-Party-ID: "5000"
<sip:5000 at 107.22.240.239>;party=calling;privacy=off;screen=no.

.

 

 

.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton
Kvashenkin
Sent: Sunday, November 11, 2012 1:26 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_opus codec name.

 

Don't forget to make reloadxml (F6). Can you paste full debug log with
<sofia profile internla siptrace on> or

 

ngrep -d <your iface> -qt -W byline port 5060 and host <which making call>

 

2012/11/11 Stephen Dame <sdame at 207me.com>

Anton,   thanks for info,  I get call not acceptable here. So no codec seems
to be matching.   is OPUS the right value,  it never shows when its
searching.

 

c=IN IP4 192.168.99.198

t=0 0

m=audio 5032 RTP/AVP 96

a=rtpmap:96 opus/48000

a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level

a=zrtp-hash:1.10
6e63145d4d42de5a83db9967b61e247360ccf093e5d3299eb3605b976901f71
e

m=video 5034 RTP/AVP 97 99

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1

a=rtpmap:99 H264/90000

a=fmtp:99 profile-level-id=4DE01f

a=recvonly

a=imageattr:97 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]

a=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]

a=zrtp-hash:1.10
14c0fd7b5b9c4386a10aa646894fdf8f03710de838f71e8241fca33b79a6592
9

 

2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:362
(sofia/intern
al/1000 at 107.22.240.239) Running State Change CS_NEW

2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:380
(sofia/intern
al/1000 at 107.22.240.239) State NEW

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G726-24:123:8000:20:24000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G729:18:8000:20:8000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[SPEEX:99:32000:20:44000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[SPEEX:99:16000:20:42200]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[SPEEX:99:8000:20:24600]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G7221:115:32000:20:48000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G7221:107:16000:20:32000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[G722:9:8000:20:64000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[PCMU:0:8000:20:64000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[PCMA:8:8000:20:64000]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[opus:9
6:48000:20:0]/[GSM:3:8000:20:13200]

2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4930 No 2833 in SDP.
Disable 28                                         33 dtmf and switch to
INFO

2012-11-10 18:56:32.159321 [DEBUG] switch_channel.c:2846
(sofia/internal/1000 at 10
7.22.240.239) Callstate Change DOWN -> HANGUP

2012-11-10 18:56:32.159321 [NOTICE] sofia.c:5743 Hangup
sofia/internal/1000 at 107.                                         22.240.239
[CS_NEW] [INCOMPATIBLE_DESTINATION]

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton
Kvashenkin
Sent: Saturday, November 10, 2012 3:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_opus codec name.

 

As we see in sdp - it's definitely works, but I'm not shure about jitsi.
When I was testing jitsi nightly build for OPUS support it was not working
at all.

 

2012/11/10 Stephen Dame <sdame at 207me.com>

I have mod_opus loaded.

 

Trying to find the documentation as to what the name of the CODEC is for
including it in codec preferences.

 

I tried OPUS,   but that doesn't seem to work.

 

<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h,
G7221 at 16000h,G722,PCMU,PCMA,GSM" />

<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=speex at 16000h@20i,PCMU,PCMA,GSM" />

 

Trying to get Jitsi to negotiate 

 

2012-11-10 12:54:22.039330 [DEBUG] sofia.c:5475 Remote SDP:

v=0

o=1000 0 0 IN IP4 192.168.99.198

s=-

c=IN IP4 192.168.99.198

t=0 0

m=audio 5020 RTP/AVP 96

a=rtpmap:96 opus/48000

a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level

a=zrtp-hash:1.10
29740c437bd6d867c08a8854b65368c5052c70ec0b463a06f748062657529239

 

 

thanks in advance,   I googled the list, and docs don't seem to find the
answer.

 

Regards,

Stephen

                                                                     


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




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


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




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

FreeSWITCH-users mailing 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 frank at carmickle.com  Mon Nov 12 18:29:06 2012
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 12 Nov 2012 10:29:06 -0500
Subject: [Freeswitch-users] mod_opus codec name.
In-Reply-To: <015e01cdc0c8$b402ffa0$1c08fee0$@207me.com>
References: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com><CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com><003b0
	1cdbfa1$8e6a3f60$ab3ebe20$@207me.com>
	<CANBhyPDhHcc1_Gw+Q3BQS=mL0nUv0_-U2ACbg=WyfK+4_ymGKw@mail.gmail.com>
	<015e01cdc0c8$b402ffa0$1c08fee0$@207me.com>
Message-ID: <87689103-2F3F-4527-90C9-0FC41F51EBF0@carmickle.com>


On Nov 12, 2012, at 6:27 AM, Stephen Dame <sdame at 207me.com> wrote:

> Anton,  I have reloaded and restarted FS. And made sure mod_opus is loaded.
>  
> I still see no documentation in forums or wiki, as to what name should be used for opus in global codec preferences.
>  
> I get 488,  which looks like  freeswitch is never looking to compare it against the preference list.

It used to be Opus-0.9.0.  I believe it is now Opus-1.0.1.

HTH
--FC



From a.villa at seletech.com  Mon Nov 12 19:39:24 2012
From: a.villa at seletech.com (alberto Villa)
Date: Mon, 12 Nov 2012 17:39:24 +0100
Subject: [Freeswitch-users] Call FORBIDDEN when setting
	max-registrations-per-extnsion parameter
Message-ID: <50A1263C.6050100@seletech.com>

Hello, I found that if I set the "max-registrations-per-extension"
parameter for an extension 2000 as follows

 <param name="max-registrations-per-extension" value="1"/>

in the <params> section of this account .xml file, then this phone
cannot execute any call as the server response is always "forbidden".

Is this a bug? if not why a call of an already registered user sholud
trigger on such parameter?

I'm using FreeSWITCH version: 1.2.0 (git-b3b2c37 2012-05-18 13-41-16 -0500)

Thanks in advance

Alberto

-- 
Dr. Villa Alberto
Sw Engineer

SeleTech srl
via Collodi, 8 20052 Monza (MI)
tel: +39 039 5962000
fax: +39 039 9716905
email: a.villa at seletech.com
web: www.seletech.com
     www.seletech.eu


From msc at freeswitch.org  Mon Nov 12 20:20:52 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 09:20:52 -0800
Subject: [Freeswitch-users] Can I use mod_shout in a macro?
In-Reply-To: <CAMRwkEC6dD9=Uy4tJqk+AcQwXs64bfjSbwNKcGsw-Or3c8+cUQ@mail.gmail.com>
References: <CAMRwkEC6dD9=Uy4tJqk+AcQwXs64bfjSbwNKcGsw-Or3c8+cUQ@mail.gmail.com>
Message-ID: <CAKzWOxU5b+Yqoc2SUUVEdGhB=_+VsVoXhqrpBQGS2H4a6fR1Jg@mail.gmail.com>

On Tue, Nov 6, 2012 at 2:27 PM, JP <jaykris at gmail.com> wrote:

> Is there a way that I can use mod_shout from a phrase macro?
>
> Thanks
> JP
>

Just curious - what are you trying to do?
-MC
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From jdiaz at coinfru.com  Mon Nov 12 20:56:43 2012
From: jdiaz at coinfru.com (Josue Diaz Cruz)
Date: Mon, 12 Nov 2012 18:56:43 +0100
Subject: [Freeswitch-users] About OPUS codec features
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAK2luhPmpPJCnBg1zt0rT1HCgAAAEAAAABJiGmkH4yFImmgThuLUbOwBAAAAAA==@coinfru.com>

I saw in the Features of this codec that Support for up to 255 channels. Is
this multiplexing? What this does mean and if this have any utility in
Freeswitch like trunking or something like this?
 

Josue Diaz Cruz

Departamento Tecnico y Soporte

 <mailto:jdiaz at coinfru.com> jdiaz at coinfru.com

 

C/ Balsicas 3

Alquerias | 30580 | Murcia

  <http://www.coinfru.com/> www.coinfru.com

 



 

 

Este e-mail contiene informaci?n confidencial, el contenido de la mismo se
encuentra protegido por Ley. Cualquier persona distinta a su destinataria
tiene prohibida su reproducci?n, uso, divulgaci?n o impresi?n total o
parcial. Si ha recibido este mensaje por error, notifiquelo de inmediato al
remitente borrando el mensaje original juntamente con sus ficheros anexos.
Gracias.

 

 

 

 
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From msc at freeswitch.org  Mon Nov 12 20:55:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 09:55:25 -0800
Subject: [Freeswitch-users] problem with receiving and sending faxes
In-Reply-To: <1352398794952-7584436.post@n2.nabble.com>
References: <1352398794952-7584436.post@n2.nabble.com>
Message-ID: <CAKzWOxUM5qcQOzs1WZvh9xNsaBtXQFsm7ETtaydqMvB_aNB4-g@mail.gmail.com>

Hello!

What is the device that is user 500? To use the dialplan like you have it
now you'll need to have your fax device be registered to your FreeSWITCH
server with user ID of "500". If you're using the example configs then this
command will show the list of registered users:

sofia status profile internal reg

Make sure there really is a user 500 and that it can answer calls. Also,
may I suggest that you avoid using loopback. In fact, you may wish to avoid
using a dialplan entry altogether. You can specify all the items right at
the command line:

originate {fax_enable_t38=true,execute_on_answer='t38_gateway
self'}user/500 &txfax(/tmp/txfax-sample.tiff)

BTW, you shouldn't need to set proxy_media. Give it a try and let us know
how it goes.

-MC

P.S. - I recommend that you put FS logs in pastebin.freeswitch.org and
select "FreeSWITCH Log" as the syntax highlighting - it makes it much
easier to read.

On Thu, Nov 8, 2012 at 10:19 AM, sterned <sterned at xakep.ru> wrote:

> Hello Please help me with the fax. I created a user 500 and 501. created a
> dialplan for 500 and am trying to send a fax from the server command:
> freeswitch @ internal> originate loopback/500 & txfax (/ tmp /
> txfax-sample.tiff)
> use: soft fax Venta Fax&Voice
> dialplan:
> <extension name="fax">
>   <condition>
>      <action application="set" data="fax_enable_t38=true"/>
>     <action application="set" data="execute_on_answer=t38_gateway self"/>
>     <action application="set" data="proxy_media=true" />
>       <action application="bridge" data="user/500 at 192.168.13.18"/>
>   </condition>
> </extension>
> log:
> freeswitch at internal> originate loopback/500 &txfax(/tmp/txfax-sample.tiff)
> -ERR NORMAL_CLEARING
>
> 2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:2005 Parsing
> global variables
> 2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:951 New Channel
> loopback/500-a [c2807b1f-acac-4e17-9271-6cf5c61000f2]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:154 loopback/500-a setup
> codec L16/8000/20
> 2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:949 Rename Channel
> loopback/500-a->loopback/500-a [c2807b1f-acac-4e17-9271-6cf5c61000f2]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:1069 (loopback/500-a)
> State Change CS_NEW -> CS_INIT
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-a [BREAK]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-a
> CHANNEL
> KILL
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-a) Running State Change CS_INIT
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
> (loopback/500-a) State INIT
> 2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:951 New Channel
> loopback/500-b [d3de53aa-7a3e-483b-bd3f-434eba87815b]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:154 loopback/500-b setup
> codec L16/8000/20
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:268 (loopback/500-b)
> State
> Change CS_NEW -> CS_INIT
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-b [BREAK]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-b
> CHANNEL
> KILL
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:324 (loopback/500-a)
> State
> Change CS_INIT -> CS_ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-a [BREAK]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-a
> CHANNEL
> KILL
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
> (loopback/500-a) State INIT going to sleep
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-a) Running State Change CS_ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1964 (loopback/500-a)
> Callstate Change DOWN -> RINGING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
> (loopback/500-a) State ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:356 loopback/500-a
> CHANNEL
> ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:67
> (loopback/500-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-a [BREAK]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-a
> CHANNEL
> KILL
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
> (loopback/500-a) State ROUTING going to sleep
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-a) Running State Change CS_CONSUME_MEDIA
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:465
> (loopback/500-a) State CONSUME_MEDIA
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:561 CHANNEL CONSUME_MEDIA
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:465
> (loopback/500-a) State CONSUME_MEDIA going to sleep
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-b) Running State Change CS_INIT
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
> (loopback/500-b) State INIT
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:324 (loopback/500-b)
> State
> Change CS_INIT -> CS_ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-b [BREAK]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-b
> CHANNEL
> KILL
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:437
> (loopback/500-b) State INIT going to sleep
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-b) Running State Change CS_ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1964 (loopback/500-b)
> Callstate Change DOWN -> RINGING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
> (loopback/500-b) State ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:356 loopback/500-b
> CHANNEL
> ROUTING
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:117
> loopback/500-b Standard ROUTING
> 2012-11-08 21:21:08.409369 [INFO] mod_dialplan_xml.c:485 Processing
> <0000000000>->500 in context default
> Dialplan: loopback/500-b parsing [default->unloop] continue=false
> Dialplan: loopback/500-b Regex (PASS) [unloop] ${unroll_loops}(true) =~
> /^true$/ break=on-false
> Dialplan: loopback/500-b Regex (FAIL) [unloop] ${sip_looped_call}() =~
> /^true$/ break=on-false
> Dialplan: loopback/500-b parsing [default->tod_example] continue=true
> Dialplan: loopback/500-b Date/TimeMatch (FAIL) [tod_example] break=on-false
> Dialplan: loopback/500-b parsing [default->holiday_example] continue=true
> Dialplan: loopback/500-b Date/TimeMatch (FAIL) [holiday_example]
> break=on-false
> Dialplan: loopback/500-b parsing [default->global-intercept] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [global-intercept]
> destination_number(500) =~ /^886$/ break=on-false
> Dialplan: loopback/500-b parsing [default->group-intercept] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [group-intercept]
> destination_number(500) =~ /^\*8$/ break=on-false
> Dialplan: loopback/500-b parsing [default->intercept-ext] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [intercept-ext]
> destination_number(500) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->redial] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [redial] destination_number(500) =~
> /^(redial|870)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->global] continue=true
> Dialplan: loopback/500-b Regex (FAIL) [global] ${call_debug}(false) =~
> /^true$/ break=never
> Dialplan: loopback/500-b Regex (FAIL) [global] ${sip_has_crypto}() =~
> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: loopback/500-b Absolute Condition [global]
> Dialplan: loopback/500-b Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: loopback/500-b Action
>
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: loopback/500-b Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: loopback/500-b Action export(RFC2822_DATE=${strftime(%a, %d %b %Y
> %T %z)})
> Dialplan: loopback/500-b parsing [default->snom-demo-2] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [snom-demo-2] destination_number(500)
> =~ /^9001$/ break=on-false
> Dialplan: loopback/500-b parsing [default->snom-demo-1] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [snom-demo-1] destination_number(500)
> =~ /^9000$/ break=on-false
> Dialplan: loopback/500-b parsing [default->eavesdrop] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [eavesdrop] destination_number(500)
> =~
> /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->eavesdrop] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [eavesdrop] destination_number(500)
> =~
> /^779$/ break=on-false
> Dialplan: loopback/500-b parsing [default->call_return] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [call_return] destination_number(500)
> =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: loopback/500-b parsing [default->del-group] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [del-group] destination_number(500)
> =~
> /^80(\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->add-group] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [add-group] destination_number(500)
> =~
> /^81(\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->call-group-simo] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [call-group-simo]
> destination_number(500) =~ /^82(\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->call-group-order] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [call-group-order]
> destination_number(500) =~ /^83(\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->extension-intercom]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [extension-intercom]
> destination_number(500) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: loopback/500-b parsing [default->Local_Extension] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [Local_Extension]
> destination_number(500) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: loopback/500-b parsing [default->Local_Extension_Skinny]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [Local_Extension_Skinny]
> destination_number(500) =~ /^(11[01][0-9])$/ break=on-false
> Dialplan: loopback/500-b parsing [default->group_dial_sales] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [group_dial_sales]
> destination_number(500) =~ /^2000$/ break=on-false
> Dialplan: loopback/500-b parsing [default->group_dial_support]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [group_dial_support]
> destination_number(500) =~ /^2001$/ break=on-false
> Dialplan: loopback/500-b parsing [default->group_dial_billing]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [group_dial_billing]
> destination_number(500) =~ /^2002$/ break=on-false
> Dialplan: loopback/500-b parsing [default->operator] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [operator] destination_number(500) =~
> /^(operator|0)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->vmain] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [vmain] destination_number(500) =~
> /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: loopback/500-b parsing [default->sip_uri] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [sip_uri] destination_number(500) =~
> /^sip:(.*)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->nb_conferences] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [nb_conferences]
> destination_number(500) =~ /^(30\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->wb_conferences] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [wb_conferences]
> destination_number(500) =~ /^(31\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->uwb_conferences] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [uwb_conferences]
> destination_number(500) =~ /^(32\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->cdquality_conferences]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [cdquality_conferences]
> destination_number(500) =~ /^(33\d{2})$/ break=on-false
> Dialplan: loopback/500-b parsing [default->freeswitch_public_conf_via_sip]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [freeswitch_public_conf_via_sip]
> destination_number(500) =~ /^9(888|8888|1616|3232)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->mad_boss_intercom]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [mad_boss_intercom]
> destination_number(500) =~ /^0911$/ break=on-false
> Dialplan: loopback/500-b parsing [default->mad_boss_intercom]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [mad_boss_intercom]
> destination_number(500) =~ /^0912$/ break=on-false
> Dialplan: loopback/500-b parsing [default->mad_boss] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [mad_boss] destination_number(500) =~
> /^0913$/ break=on-false
> Dialplan: loopback/500-b parsing [default->ivr_demo] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [ivr_demo] destination_number(500) =~
> /^5000$/ break=on-false
> Dialplan: loopback/500-b parsing [default->dynamic_conference]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [dynamic_conference]
> destination_number(500) =~ /^5001$/ break=on-false
> Dialplan: loopback/500-b parsing [default->rtp_multicast_page]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [rtp_multicast_page]
> destination_number(500) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: loopback/500-b parsing [default->park] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [park] destination_number(500) =~
> /^5900$/ break=on-false
> Dialplan: loopback/500-b parsing [default->unpark] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [unpark] destination_number(500) =~
> /^5901$/ break=on-false
> Dialplan: loopback/500-b parsing [default->valet_park] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [valet_park] destination_number(500)
> =~ /^(6000)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->valet_park] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [valet_park] destination_number(500)
> =~ /^(60\d[1-9])$/ break=on-false
> Dialplan: loopback/500-b parsing [default->park] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [park] source(mod_loopback) =~
> /mod_sofia/ break=on-false
> Dialplan: loopback/500-b parsing [default->unpark] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [unpark] source(mod_loopback) =~
> /mod_sofia/ break=on-false
> Dialplan: loopback/500-b parsing [default->park] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [park] source(mod_loopback) =~
> /mod_sofia/ break=on-false
> Dialplan: loopback/500-b parsing [default->unpark] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [unpark] source(mod_loopback) =~
> /mod_sofia/ break=on-false
> Dialplan: loopback/500-b parsing [default->wait] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [wait] destination_number(500) =~
> /^wait$/ break=on-false
> Dialplan: loopback/500-b parsing [default->fax_receive] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [fax_receive] destination_number(500)
> =~ /^9178$/ break=on-false
> Dialplan: loopback/500-b parsing [default->fax_transmit] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [fax_transmit]
> destination_number(500)
> =~ /^9179$/ break=on-false
> Dialplan: loopback/500-b parsing [default->ringback_180] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [ringback_180]
> destination_number(500)
> =~ /^9180$/ break=on-false
> Dialplan: loopback/500-b parsing [default->ringback_183_uk_ring]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [ringback_183_uk_ring]
> destination_number(500) =~ /^9181$/ break=on-false
> Dialplan: loopback/500-b parsing [default->ringback_183_music_ring]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [ringback_183_music_ring]
> destination_number(500) =~ /^9182$/ break=on-false
> Dialplan: loopback/500-b parsing [default->ringback_post_answer_uk_ring]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [ringback_post_answer_uk_ring]
> destination_number(500) =~ /^9183$/ break=on-false
> Dialplan: loopback/500-b parsing [default->ringback_post_answer_music]
> continue=false
> Dialplan: loopback/500-b Regex (FAIL) [ringback_post_answer_music]
> destination_number(500) =~ /^9184$/ break=on-false
> Dialplan: loopback/500-b parsing [default->ClueCon] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [ClueCon] destination_number(500) =~
> /^9191$/ break=on-false
> Dialplan: loopback/500-b parsing [default->show_info] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [show_info] destination_number(500)
> =~
> /^9192$/ break=on-false
> Dialplan: loopback/500-b parsing [default->video_record] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [video_record]
> destination_number(500)
> =~ /^9193$/ break=on-false
> Dialplan: loopback/500-b parsing [default->video_playback] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [video_playback]
> destination_number(500) =~ /^9194$/ break=on-false
> Dialplan: loopback/500-b parsing [default->delay_echo] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [delay_echo] destination_number(500)
> =~ /^9195$/ break=on-false
> Dialplan: loopback/500-b parsing [default->echo] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [echo] destination_number(500) =~
> /^9196$/ break=on-false
> Dialplan: loopback/500-b parsing [default->milliwatt] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [milliwatt] destination_number(500)
> =~
> /^9197$/ break=on-false
> Dialplan: loopback/500-b parsing [default->tone_stream] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [tone_stream] destination_number(500)
> =~ /^9198$/ break=on-false
> Dialplan: loopback/500-b parsing [default->zrtp_enrollement] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [zrtp_enrollement]
> destination_number(500) =~ /^9787$/ break=on-false
> Dialplan: loopback/500-b parsing [default->hold_music] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [hold_music] destination_number(500)
> =~ /^9664$/ break=on-false
> Dialplan: loopback/500-b parsing [default->laugh break] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [laugh break] destination_number(500)
> =~ /^9386$/ break=on-false
> Dialplan: loopback/500-b parsing [default->5555555] continue=false
> Dialplan: loopback/500-b Regex (FAIL) [5555555] destination_number(500) =~
> /^(1000)$/ break=on-false
> Dialplan: loopback/500-b parsing [default->fax] continue=false
> Dialplan: loopback/500-b Absolute Condition [fax]
> Dialplan: loopback/500-b Action set(fax_enable_t38=true)
> Dialplan: loopback/500-b Action set(execute_on_answer=t38_gateway self)
> Dialplan: loopback/500-b Action set(proxy_media=true)
> Dialplan: loopback/500-b Action bridge(user/500 at 192.168.13.18)
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:167
> (loopback/500-b) State Change CS_ROUTING -> CS_EXECUTE
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-b [BREAK]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:501 loopback/500-b
> CHANNEL
> KILL
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:446
> (loopback/500-b) State ROUTING going to sleep
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-b) Running State Change CS_EXECUTE
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:453
> (loopback/500-b) State EXECUTE
> 2012-11-08 21:21:08.409369 [DEBUG] mod_loopback.c:395 loopback/500-b
> CHANNEL
> EXECUTE
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_state_machine.c:209
> loopback/500-b Standard EXECUTE
> EXECUTE loopback/500-b
>
> hash(insert/192.168.13.18-spymap/0000000000/d3de53aa-7a3e-483b-bd3f-434eba87815b)
> EXECUTE loopback/500-b hash(insert/192.168.13.18-last_dial/0000000000/500)
> EXECUTE loopback/500-b
>
> hash(insert/192.168.13.18-last_dial/global/d3de53aa-7a3e-483b-bd3f-434eba87815b)
> EXECUTE loopback/500-b export(RFC2822_DATE=Thu, 08 Nov 2012 21:21:08 +0400)
> 2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1118 EXPORT
> (export_vars) [RFC2822_DATE]=[Thu, 08 Nov 2012 21:21:08 +0400]
> EXECUTE loopback/500-b set(fax_enable_t38=true)
> 2012-11-08 21:21:08.409369 [DEBUG] mod_dptools.c:1319 loopback/500-b SET
> [fax_enable_t38]=[true]
> EXECUTE loopback/500-b set(execute_on_answer=t38_gateway self)
> 2012-11-08 21:21:08.409369 [DEBUG] mod_dptools.c:1319 loopback/500-b SET
> [execute_on_answer]=[t38_gateway self]
> EXECUTE loopback/500-b set(proxy_media=true)
> 2012-11-08 21:21:08.409369 [DEBUG] mod_dptools.c:1319 loopback/500-b SET
> [proxy_media]=[true]
> EXECUTE loopback/500-b bridge(user/500 at 192.168.13.18)
> 2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1072 loopback/500-b
> EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Nov 2012 21:21:08 +0400] to
> event
> 2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:2005 Parsing
> global variables
> 2012-11-08 21:21:08.409369 [DEBUG] switch_channel.c:1072 loopback/500-b
> EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Nov 2012 21:21:08 +0400] to
> event
> 2012-11-08 21:21:08.409369 [DEBUG] switch_ivr_originate.c:2005 Parsing
> global variables
> 2012-11-08 21:21:08.409369 [DEBUG] switch_event.c:1569 Parsing variable
> [sip_invite_domain]=[192.168.13.18]
> 2012-11-08 21:21:08.409369 [DEBUG] switch_event.c:1569 Parsing variable
> [presence_id]=[500 at 192.168.13.18]
> 2012-11-08 21:21:08.409369 [NOTICE] switch_channel.c:951 New Channel
> sofia/internal/sip:500 at 192.168.13.233:5060
> [f617d5ed-1e77-4a22-85a1-d0db3da0fc4f]
> 2012-11-08 21:21:08.409369 [DEBUG] mod_sofia.c:4879
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_NEW ->
> CS_INIT
> 2012-11-08 21:21:08.409369 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_INIT
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:437
> (sofia/internal/sip:500 at 192.168.13.233:5060) State INIT
> 2012-11-08 21:21:08.429387 [DEBUG] mod_sofia.c:86
> sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA INIT
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:08.429387 [DEBUG] mod_sofia.c:126
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_INIT ->
> CS_ROUTING
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:437
> (sofia/internal/sip:500 at 192.168.13.233:5060) State INIT going to sleep
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_ROUTING
> 2012-11-08 21:21:08.429387 [DEBUG] switch_channel.c:1964
> (sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change DOWN ->
> RINGING
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING
> 2012-11-08 21:21:08.429387 [DEBUG] mod_sofia.c:149
> sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA ROUTING
> 2012-11-08 21:21:08.429387 [DEBUG] switch_ivr_originate.c:67
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING going to sleep
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_CONSUME_MEDIA
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:465
> (sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA
> 2012-11-08 21:21:08.429387 [DEBUG] switch_core_state_machine.c:465
> (sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA going to
> sleep
> 2012-11-08 21:21:08.429387 [DEBUG] sofia.c:6282 Channel
> sofia/internal/sip:500 at 192.168.13.233:5060 entering state [calling][0]
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:29.609239 [DEBUG] switch_channel.c:2950
> (sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change RINGING ->
> HANGUP
> 2012-11-08 21:21:29.609239 [NOTICE] sofia.c:711 Hangup
> sofia/internal/sip:500 at 192.168.13.233:5060 [CS_CONSUME_MEDIA]
> [NORMAL_CLEARING]
> 2012-11-08 21:21:29.609239 [DEBUG] switch_channel.c:2973 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [KILL]
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_HANGUP
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:638
> (sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP
> 2012-11-08 21:21:29.609239 [DEBUG] mod_sofia.c:483 Channel
> sofia/internal/sip:500 at 192.168.13.233:5060 hanging up, cause:
> NORMAL_CLEARING
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:48
> sofia/internal/sip:500 at 192.168.13.233:5060 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:638
> (sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP going to sleep
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:429
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_HANGUP ->
> CS_REPORTING
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_REPORTING
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:703
> (sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:92
> sofia/internal/sip:500 at 192.168.13.233:5060 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:703
> (sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING going to
> sleep
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_REPORTING ->
> CS_DESTROY
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:21:29.609239 [DEBUG] switch_core_session.c:1415 Session 35
> (sofia/internal/sip:500 at 192.168.13.233:5060) Locked, Waiting on external
> entities
> 2012-11-08 21:21:29.629589 [DEBUG] switch_ivr_originate.c:3508 Originate
> Resulted in Error Cause: 16 [NORMAL_CLEARING]
> 2012-11-08 21:21:29.629589 [NOTICE] switch_ivr_originate.c:2591 Cannot
> create outgoing channel of type [user] cause: [NORMAL_CLEARING]
> 2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1433 Session 35
> (sofia/internal/sip:500 at 192.168.13.233:5060) Ended
> 2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1437 Close
> Channel
> sofia/internal/sip:500 at 192.168.13.233:5060 [CS_DESTROY]
> 2012-11-08 21:21:29.629589 [DEBUG] switch_ivr_originate.c:3508 Originate
> Resulted in Error Cause: 16 [NORMAL_CLEARING]
> 2012-11-08 21:21:29.629589 [INFO] mod_dptools.c:3027 Originate Failed.
> Cause: NORMAL_CLEARING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2950 (loopback/500-b)
> Callstate Change RINGING -> HANGUP
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:527
> (sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change HANGUP ->
> DOWN
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:530
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
> (sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] mod_sofia.c:376
> sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:99
> sofia/internal/sip:500 at 192.168.13.233:5060 Standard DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
> (sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY going to sleep
> 2012-11-08 21:21:29.629589 [NOTICE] mod_dptools.c:3147 Hangup
> loopback/500-b
> [CS_EXECUTE] [NORMAL_CLEARING]
> 2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2973 Send signal
> loopback/500-b [KILL]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-b [BREAK]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:2553
> loopback/500-b
> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup
> already)
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:453
> (loopback/500-b) State EXECUTE going to sleep
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-b) Running State Change CS_HANGUP
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
> (loopback/500-b) State HANGUP
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:453 loopback/500-b
> CHANNEL
> HANGUP
> 2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2950 (loopback/500-a)
> Callstate Change RINGING -> HANGUP
> 2012-11-08 21:21:29.629589 [NOTICE] mod_loopback.c:464 Hangup
> loopback/500-a
> [CS_CONSUME_MEDIA] [NORMAL_CLEARING]
> 2012-11-08 21:21:29.629589 [DEBUG] switch_channel.c:2973 Send signal
> loopback/500-a [KILL]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-a) Running State Change CS_HANGUP
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
> (loopback/500-a) State HANGUP
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:453 loopback/500-a
> CHANNEL
> HANGUP
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:48
> loopback/500-a Standard HANGUP, cause: NORMAL_CLEARING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
> (loopback/500-a) State HANGUP going to sleep
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:429
> (loopback/500-a) State Change CS_HANGUP -> CS_REPORTING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-a [BREAK]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-a) Running State Change CS_REPORTING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
> (loopback/500-a) State REPORTING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:92
> loopback/500-a Standard REPORTING, cause: NORMAL_CLEARING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
> (loopback/500-a) State REPORTING going to sleep
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:423
> (loopback/500-a) State Change CS_REPORTING -> CS_DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-a [BREAK]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1415 Session 33
> (loopback/500-a) Locked, Waiting on external entities
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-a [BREAK]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-a
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:48
> loopback/500-b Standard HANGUP, cause: NORMAL_CLEARING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:638
> (loopback/500-b) State HANGUP going to sleep
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:429
> (loopback/500-b) State Change CS_HANGUP -> CS_REPORTING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-b [BREAK]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:398
> (loopback/500-b) Running State Change CS_REPORTING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_ivr_originate.c:3508 Originate
> Resulted in Error Cause: 16 [NORMAL_CLEARING]
> 2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1433 Session 33
> (loopback/500-a) Ended
> 2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1437 Close
> Channel
> loopback/500-a [CS_DESTROY]
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:527
> (loopback/500-a) Callstate Change HANGUP -> DOWN
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:530
> (loopback/500-a) Running State Change CS_DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
> (loopback/500-a) State DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:99
> loopback/500-a Standard DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
> (loopback/500-a) State DESTROY going to sleep
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
> (loopback/500-b) State REPORTING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:92
> loopback/500-b Standard REPORTING, cause: NORMAL_CLEARING
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:703
> (loopback/500-b) State REPORTING going to sleep
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:423
> (loopback/500-b) State Change CS_REPORTING -> CS_DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1210 Send signal
> loopback/500-b [BREAK]
> 2012-11-08 21:21:29.629589 [DEBUG] mod_loopback.c:501 loopback/500-b
> CHANNEL
> KILL
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_session.c:1415 Session 34
> (loopback/500-b) Locked, Waiting on external entities
> 2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1433 Session 34
> (loopback/500-b) Ended
> 2012-11-08 21:21:29.629589 [NOTICE] switch_core_session.c:1437 Close
> Channel
> loopback/500-b [CS_DESTROY]
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:527
> (loopback/500-b) Callstate Change HANGUP -> DOWN
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:530
> (loopback/500-b) Running State Change CS_DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
> (loopback/500-b) State DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:99
> loopback/500-b Standard DESTROY
> 2012-11-08 21:21:29.629589 [DEBUG] switch_core_state_machine.c:540
> (loopback/500-b) State DESTROY going to sleep
>
> so I'm trying to send a fax to the fax 501 -> 500
> log:
> 2012-11-08 21:18:21.109351 [NOTICE] switch_channel.c:951 New Channel
> sofia/internal/501 at 192.168.13.18 [ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d]
> 2012-11-08 21:18:21.109351 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.129757 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/501 at 192.168.13.18) Running State Change CS_NEW
> 2012-11-08 21:18:21.129757 [DEBUG] switch_core_state_machine.c:416
> (sofia/internal/501 at 192.168.13.18) State NEW
> 2012-11-08 21:18:21.129757 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.149399 [DEBUG] sofia.c:8412 IP 192.168.13.71 Rejected
> by
> acl "domains". Falling back to Digest auth.
> 2012-11-08 21:18:21.149399 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.149399 [DEBUG] sofia.c:1728 detaching session
> ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d
> 2012-11-08 21:18:21.149399 [DEBUG] sofia.c:1820 Re-attaching to session
> ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d
> 2012-11-08 21:18:21.149399 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.149399 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] sofia.c:8412 IP 192.168.13.71 Rejected
> by
> acl "domains". Falling back to Digest auth.
> 2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6282 Channel
> sofia/internal/501 at 192.168.13.18 entering state [received][100]
> 2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6293 Remote SDP:
> v=0
> o=- 1352395099 1 IN IP4 192.168.13.71
> s=VFT38M/v.25.08.2012
> c=IN IP4 192.168.13.71
> t=0 0
> m=audio 5018 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16,32,36
> a=maxptime:240
>
> 2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6506
> (sofia/internal/501 at 192.168.13.18) State Change CS_NEW -> CS_INIT
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/501 at 192.168.13.18) Running State Change CS_INIT
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
> (sofia/internal/501 at 192.168.13.18) State INIT
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:86
> sofia/internal/501 at 192.168.13.18 SOFIA INIT
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:126
> (sofia/internal/501 at 192.168.13.18) State Change CS_INIT -> CS_ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
> (sofia/internal/501 at 192.168.13.18) State INIT going to sleep
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/501 at 192.168.13.18) Running State Change CS_ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1964
> (sofia/internal/501 at 192.168.13.18) Callstate Change DOWN -> RINGING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/501 at 192.168.13.18) State ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:149
> sofia/internal/501 at 192.168.13.18 SOFIA ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:117
> sofia/internal/501 at 192.168.13.18 Standard ROUTING
> 2012-11-08 21:18:21.169346 [INFO] mod_dialplan_xml.c:485 Processing 501
> <501>->500 in context default
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unloop]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->tod_example]
> continue=true
> Dialplan: sofia/internal/501 at 192.168.13.18 Date/TimeMatch (FAIL)
> [tod_example] break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->holiday_example] continue=true
> Dialplan: sofia/internal/501 at 192.168.13.18 Date/TimeMatch (FAIL)
> [holiday_example] break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->global-intercept] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [global-intercept]
> destination_number(500) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->group-intercept] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [group-intercept]
> destination_number(500) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->intercept-ext]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [intercept-ext]
> destination_number(500) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->redial]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [redial]
> destination_number(500) =~ /^(redial|870)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->global]
> continue=true
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [global]
> ${sip_has_crypto}() =~
> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
> Dialplan: sofia/internal/501 at 192.168.13.18 Absolute Condition [global]
> Dialplan: sofia/internal/501 at 192.168.13.18 Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/501 at 192.168.13.18 Action
>
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/internal/501 at 192.168.13.18 Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/internal/501 at 192.168.13.18 Action
> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->snom-demo-2]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [snom-demo-2]
> destination_number(500) =~ /^9001$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->snom-demo-1]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [snom-demo-1]
> destination_number(500) =~ /^9000$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [eavesdrop]
> destination_number(500) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->eavesdrop]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [eavesdrop]
> destination_number(500) =~ /^779$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->call_return]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [call_return]
> destination_number(500) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->del-group]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [del-group]
> destination_number(500) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->add-group]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [add-group]
> destination_number(500) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->call-group-simo] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [call-group-simo]
> destination_number(500) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->call-group-order] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [call-group-order]
> destination_number(500) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->extension-intercom] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [extension-intercom]
> destination_number(500) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->Local_Extension] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [Local_Extension]
> destination_number(500) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->Local_Extension_Skinny] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [Local_Extension_Skinny] destination_number(500) =~ /^(11[01][0-9])$/
> break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->group_dial_sales] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [group_dial_sales]
> destination_number(500) =~ /^2000$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->group_dial_support] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [group_dial_support]
> destination_number(500) =~ /^2001$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->group_dial_billing] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [group_dial_billing]
> destination_number(500) =~ /^2002$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->operator]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [operator]
> destination_number(500) =~ /^(operator|0)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->vmain]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [vmain]
> destination_number(500) =~ /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->sip_uri]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [sip_uri]
> destination_number(500) =~ /^sip:(.*)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->nb_conferences]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [nb_conferences]
> destination_number(500) =~ /^(30\d{2})$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->wb_conferences]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [wb_conferences]
> destination_number(500) =~ /^(31\d{2})$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->uwb_conferences] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [uwb_conferences]
> destination_number(500) =~ /^(32\d{2})$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->cdquality_conferences] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [cdquality_conferences] destination_number(500) =~ /^(33\d{2})$/
> break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(500) =~
> /^9(888|8888|1616|3232)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->mad_boss_intercom] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [mad_boss_intercom]
> destination_number(500) =~ /^0911$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->mad_boss_intercom] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [mad_boss_intercom]
> destination_number(500) =~ /^0912$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->mad_boss]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [mad_boss]
> destination_number(500) =~ /^0913$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->ivr_demo]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [ivr_demo]
> destination_number(500) =~ /^5000$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->dynamic_conference] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [dynamic_conference]
> destination_number(500) =~ /^5001$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->rtp_multicast_page] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [rtp_multicast_page]
> destination_number(500) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [park]
> destination_number(500) =~ /^5900$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unpark]
> destination_number(500) =~ /^5901$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->valet_park]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [valet_park]
> destination_number(500) =~ /^(6000)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->valet_park]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [valet_park]
> destination_number(500) =~ /^(60\d[1-9])$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [park]
> destination_number(500) =~ /park\+(\d+)/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unpark]
> destination_number(500) =~ /^parking$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->park]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [park]
> destination_number(500) =~ /callpark/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->unpark]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [unpark]
> destination_number(500) =~ /pickup/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->wait]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [wait]
> destination_number(500) =~ /^wait$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->fax_receive]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [fax_receive]
> destination_number(500) =~ /^9178$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->fax_transmit]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [fax_transmit]
> destination_number(500) =~ /^9179$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->ringback_180]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [ringback_180]
> destination_number(500) =~ /^9180$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->ringback_183_uk_ring] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [ringback_183_uk_ring] destination_number(500) =~ /^9181$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->ringback_183_music_ring] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [ringback_183_music_ring] destination_number(500) =~ /^9182$/
> break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->ringback_post_answer_uk_ring] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(500) =~ /^9183$/
> break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->ringback_post_answer_music] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL)
> [ringback_post_answer_music] destination_number(500) =~ /^9184$/
> break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->ClueCon]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [ClueCon]
> destination_number(500) =~ /^9191$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->show_info]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [show_info]
> destination_number(500) =~ /^9192$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->video_record]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [video_record]
> destination_number(500) =~ /^9193$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->video_playback]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [video_playback]
> destination_number(500) =~ /^9194$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->delay_echo]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [delay_echo]
> destination_number(500) =~ /^9195$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->echo]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [echo]
> destination_number(500) =~ /^9196$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->milliwatt]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [milliwatt]
> destination_number(500) =~ /^9197$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->tone_stream]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [tone_stream]
> destination_number(500) =~ /^9198$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing
> [default->zrtp_enrollement] continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [zrtp_enrollement]
> destination_number(500) =~ /^9787$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->hold_music]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [hold_music]
> destination_number(500) =~ /^9664$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->laugh break]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [laugh break]
> destination_number(500) =~ /^9386$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->5555555]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Regex (FAIL) [5555555]
> destination_number(500) =~ /^(1000)$/ break=on-false
> Dialplan: sofia/internal/501 at 192.168.13.18 parsing [default->fax]
> continue=false
> Dialplan: sofia/internal/501 at 192.168.13.18 Absolute Condition [fax]
> Dialplan: sofia/internal/501 at 192.168.13.18 Action set(fax_enable_t38=true)
> Dialplan: sofia/internal/501 at 192.168.13.18 Action
> set(execute_on_answer=t38_gateway self)
> Dialplan: sofia/internal/501 at 192.168.13.18 Action set(proxy_media=true)
> Dialplan: sofia/internal/501 at 192.168.13.18 Action
> bridge(user/500 at 192.168.13.18)
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:167
> (sofia/internal/501 at 192.168.13.18) State Change CS_ROUTING -> CS_EXECUTE
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/501 at 192.168.13.18) State ROUTING going to sleep
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/501 at 192.168.13.18) Running State Change CS_EXECUTE
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:453
> (sofia/internal/501 at 192.168.13.18) State EXECUTE
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:242
> sofia/internal/501 at 192.168.13.18 SOFIA EXECUTE
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:209
> sofia/internal/501 at 192.168.13.18 Standard EXECUTE
> EXECUTE sofia/internal/501 at 192.168.13.18
> hash(insert/192.168.13.18-spymap/501/ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d)
> EXECUTE sofia/internal/501 at 192.168.13.18
> hash(insert/192.168.13.18-last_dial/501/500)
> EXECUTE sofia/internal/501 at 192.168.13.18
>
> hash(insert/192.168.13.18-last_dial/global/ce35cc96-6ce5-4cf7-a9f0-71f7d31e454d)
> EXECUTE sofia/internal/501 at 192.168.13.18 export(RFC2822_DATE=Thu, 08 Nov
> 2012 21:18:21 +0400)
> 2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1118 EXPORT
> (export_vars) [RFC2822_DATE]=[Thu, 08 Nov 2012 21:18:21 +0400]
> EXECUTE sofia/internal/501 at 192.168.13.18 set(fax_enable_t38=true)
> 2012-11-08 21:18:21.169346 [DEBUG] mod_dptools.c:1319
> sofia/internal/501 at 192.168.13.18 SET [fax_enable_t38]=[true]
> EXECUTE sofia/internal/501 at 192.168.13.18 set(execute_on_answer=t38_gateway
> self)
> 2012-11-08 21:18:21.169346 [DEBUG] mod_dptools.c:1319
> sofia/internal/501 at 192.168.13.18 SET [execute_on_answer]=[t38_gateway
> self]
> EXECUTE sofia/internal/501 at 192.168.13.18 set(proxy_media=true)
> 2012-11-08 21:18:21.169346 [DEBUG] mod_dptools.c:1319
> sofia/internal/501 at 192.168.13.18 SET [proxy_media]=[true]
> EXECUTE sofia/internal/501 at 192.168.13.18 bridge(user/500 at 192.168.13.18)
> 2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1072
> sofia/internal/501 at 192.168.13.18 EXPORTING[export_vars]
> [RFC2822_DATE]=[Thu,
> 08 Nov 2012 21:18:21 +0400] to event
> 2012-11-08 21:18:21.169346 [DEBUG] switch_ivr_originate.c:2005 Parsing
> global variables
> 2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1072
> sofia/internal/501 at 192.168.13.18 EXPORTING[export_vars]
> [RFC2822_DATE]=[Thu,
> 08 Nov 2012 21:18:21 +0400] to event
> 2012-11-08 21:18:21.169346 [DEBUG] switch_ivr_originate.c:2005 Parsing
> global variables
> 2012-11-08 21:18:21.169346 [DEBUG] switch_event.c:1569 Parsing variable
> [sip_invite_domain]=[192.168.13.18]
> 2012-11-08 21:18:21.169346 [DEBUG] switch_event.c:1569 Parsing variable
> [presence_id]=[500 at 192.168.13.18]
> 2012-11-08 21:18:21.169346 [NOTICE] switch_channel.c:951 New Channel
> sofia/internal/sip:500 at 192.168.13.233:5060
> [a318696b-d309-4724-abc6-1f0aca4f0dd3]
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:4879
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_NEW ->
> CS_INIT
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:4954 [zrtp_passthru] Setting
> a-leg inherit_codec=true
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:4957 [zrtp_passthru] Setting
> b-leg absolute_codec_string='PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b'
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change CS_INIT
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
> (sofia/internal/sip:500 at 192.168.13.233:5060) State INIT
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:86
> sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA INIT
> 2012-11-08 21:18:21.169346 [DEBUG] sofia_glue.c:1920
> sofia/internal/sip:500 at 192.168.13.233:5060 Patched SDP
> ---
> v=0
> o=- 1352395099 1 IN IP4 192.168.13.71
> s=VFT38M/v.25.08.2012
> c=IN IP4 192.168.13.71
> t=0 0
> m=audio 5018 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16,32,36
> a=maxptime:240
>
> +++
> v=0
> o=FreeSWITCH 3287184146 3287184147 IN IP4 192.168.13.18
> s=FreeSWITCH
> c=IN IP4 192.168.13.18
> t=0 0
> m=audio 16966 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16,32,36
> a=maxptime:240
>
> 2012-11-08 21:18:21.169346 [DEBUG] sofia_glue.c:2637 Local SDP:
> v=0
> o=FreeSWITCH 3287184146 3287184147 IN IP4 192.168.13.18
> s=FreeSWITCH
> c=IN IP4 192.168.13.18
> t=0 0
> m=audio 16966 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16,32,36
> a=maxptime:240
>
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:126
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_INIT ->
> CS_ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:437
> (sofia/internal/sip:500 at 192.168.13.233:5060) State INIT going to sleep
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_channel.c:1964
> (sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change DOWN ->
> RINGING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] mod_sofia.c:149
> sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA ROUTING
> 2012-11-08 21:18:21.169346 [DEBUG] switch_ivr_originate.c:67
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/sip:500 at 192.168.13.233:5060) State ROUTING going to sleep
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_CONSUME_MEDIA
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:465
> (sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA
> 2012-11-08 21:18:21.169346 [DEBUG] switch_core_state_machine.c:465
> (sofia/internal/sip:500 at 192.168.13.233:5060) State CONSUME_MEDIA going to
> sleep
> 2012-11-08 21:18:21.169346 [DEBUG] sofia.c:6282 Channel
> sofia/internal/sip:500 at 192.168.13.233:5060 entering state [calling][0]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:53.169624 [DEBUG] sofia.c:6282 Channel
> sofia/internal/sip:500 at 192.168.13.233:5060 entering state
> [terminated][408]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_channel.c:2950
> (sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change RINGING ->
> HANGUP
> 2012-11-08 21:18:53.169624 [NOTICE] sofia.c:7082 Hangup
> sofia/internal/sip:500 at 192.168.13.233:5060 [CS_CONSUME_MEDIA]
> [RECOVERY_ON_TIMER_EXPIRE]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_channel.c:2973 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [KILL]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_HANGUP
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:638
> (sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP
> 2012-11-08 21:18:53.169624 [DEBUG] mod_sofia.c:483 Channel
> sofia/internal/sip:500 at 192.168.13.233:5060 hanging up, cause:
> RECOVERY_ON_TIMER_EXPIRE
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:48
> sofia/internal/sip:500 at 192.168.13.233:5060 Standard HANGUP, cause:
> RECOVERY_ON_TIMER_EXPIRE
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:638
> (sofia/internal/sip:500 at 192.168.13.233:5060) State HANGUP going to sleep
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:429
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_HANGUP ->
> CS_REPORTING
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_REPORTING
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:703
> (sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:92
> sofia/internal/sip:500 at 192.168.13.233:5060 Standard REPORTING, cause:
> RECOVERY_ON_TIMER_EXPIRE
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:703
> (sofia/internal/sip:500 at 192.168.13.233:5060) State REPORTING going to
> sleep
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/sip:500 at 192.168.13.233:5060) State Change CS_REPORTING ->
> CS_DESTROY
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/sip:500 at 192.168.13.233:5060 [BREAK]
> 2012-11-08 21:18:53.169624 [DEBUG] switch_core_session.c:1415 Session 30
> (sofia/internal/sip:500 at 192.168.13.233:5060) Locked, Waiting on external
> entities
> 2012-11-08 21:18:53.190125 [DEBUG] switch_ivr_originate.c:3508 Originate
> Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
> 2012-11-08 21:18:53.190125 [NOTICE] switch_ivr_originate.c:2591 Cannot
> create outgoing channel of type [user] cause: [RECOVERY_ON_TIMER_EXPIRE]
> 2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1433 Session 30
> (sofia/internal/sip:500 at 192.168.13.233:5060) Ended
> 2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1437 Close
> Channel
> sofia/internal/sip:500 at 192.168.13.233:5060 [CS_DESTROY]
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:527
> (sofia/internal/sip:500 at 192.168.13.233:5060) Callstate Change HANGUP ->
> DOWN
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:530
> (sofia/internal/sip:500 at 192.168.13.233:5060) Running State Change
> CS_DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
> (sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:376
> sofia/internal/sip:500 at 192.168.13.233:5060 SOFIA DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:99
> sofia/internal/sip:500 at 192.168.13.233:5060 Standard DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
> (sofia/internal/sip:500 at 192.168.13.233:5060) State DESTROY going to sleep
> 2012-11-08 21:18:53.190125 [DEBUG] switch_ivr_originate.c:3508 Originate
> Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
> 2012-11-08 21:18:53.190125 [INFO] mod_dptools.c:3027 Originate Failed.
> Cause: RECOVERY_ON_TIMER_EXPIRE
> 2012-11-08 21:18:53.190125 [DEBUG] switch_channel.c:2950
> (sofia/internal/501 at 192.168.13.18) Callstate Change RINGING -> HANGUP
> 2012-11-08 21:18:53.190125 [NOTICE] mod_dptools.c:3147 Hangup
> sofia/internal/501 at 192.168.13.18 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE]
> 2012-11-08 21:18:53.190125 [DEBUG] switch_channel.c:2973 Send signal
> sofia/internal/501 at 192.168.13.18 [KILL]
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:2553
> sofia/internal/501 at 192.168.13.18 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:453
> (sofia/internal/501 at 192.168.13.18) State EXECUTE going to sleep
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/501 at 192.168.13.18) Running State Change CS_HANGUP
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:638
> (sofia/internal/501 at 192.168.13.18) State HANGUP
> 2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:477
> sofia/internal/501 at 192.168.13.18 Overriding SIP cause 504 with 408 from
> the
> other leg
> 2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:483 Channel
> sofia/internal/501 at 192.168.13.18 hanging up, cause:
> RECOVERY_ON_TIMER_EXPIRE
> 2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:613 Responding to INVITE
> with: 408
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:48
> sofia/internal/501 at 192.168.13.18 Standard HANGUP, cause:
> RECOVERY_ON_TIMER_EXPIRE
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:638
> (sofia/internal/501 at 192.168.13.18) State HANGUP going to sleep
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:429
> (sofia/internal/501 at 192.168.13.18) State Change CS_HANGUP -> CS_REPORTING
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:398
> (sofia/internal/501 at 192.168.13.18) Running State Change CS_REPORTING
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:703
> (sofia/internal/501 at 192.168.13.18) State REPORTING
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:92
> sofia/internal/501 at 192.168.13.18 Standard REPORTING, cause:
> RECOVERY_ON_TIMER_EXPIRE
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:703
> (sofia/internal/501 at 192.168.13.18) State REPORTING going to sleep
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:423
> (sofia/internal/501 at 192.168.13.18) State Change CS_REPORTING -> CS_DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1210 Send signal
> sofia/internal/501 at 192.168.13.18 [BREAK]
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_session.c:1415 Session 29
> (sofia/internal/501 at 192.168.13.18) Locked, Waiting on external entities
> 2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1433 Session 29
> (sofia/internal/501 at 192.168.13.18) Ended
> 2012-11-08 21:18:53.190125 [NOTICE] switch_core_session.c:1437 Close
> Channel
> sofia/internal/501 at 192.168.13.18 [CS_DESTROY]
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:527
> (sofia/internal/501 at 192.168.13.18) Callstate Change HANGUP -> DOWN
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:530
> (sofia/internal/501 at 192.168.13.18) Running State Change CS_DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
> (sofia/internal/501 at 192.168.13.18) State DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] mod_sofia.c:376
> sofia/internal/501 at 192.168.13.18 SOFIA DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:99
> sofia/internal/501 at 192.168.13.18 Standard DESTROY
> 2012-11-08 21:18:53.190125 [DEBUG] switch_core_state_machine.c:540
> (sofia/internal/501 at 192.168.13.18) State DESTROY going to sleep
>
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/problem-with-receiving-and-sending-faxes-tp7584436.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 12 21:11:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 10:11:30 -0800
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1352421083.47663.YahooMailNeo@web120002.mail.ne1.yahoo.com>
References: <1352421083.47663.YahooMailNeo@web120002.mail.ne1.yahoo.com>
Message-ID: <CAKzWOxXeFmU1n7Vk41i4EubYe-3aSfQL470RWt-6yD6AvMDBHg@mail.gmail.com>

what happens in the media stream prior to the call being answered?
-MC

On Thu, Nov 8, 2012 at 4:31 PM, Jim Daley <jim.daley1223 at yahoo.com> wrote:

> I hope this is the right place to ask this question.
>
> I am running freeswitch using LUA. I'm doing outbound calling for a load
> tester.
>
> I have a number that when you call it you need to enter 2654# before the
> call get's answered. So I need a way of dialing DTMF digits before the
> other end answers. I tried to just sleep and send_dtmf after the
> session.ready but that doesn't work.
>
> Is there a way to do this. Sample code below...thanks
>
> if (session:ready()) then
> session:answer();
> session:setVariable("CallCmp", "Script_dcetest1")
> session:setVariable("PromptFail", "Script_dcetest1") ;
> api = freeswitch.API();
> session:sleep(1000);
>  --status = session:getVariable("SilenceStatus");
>  session:execute("wait_for_silence", "200 30 20 20000"); --
>  -- Continue in english, press 2
> status = session:getVariable("SilenceStatus"); -- This will get past
> langauge and main menu prompts
> if((status == "unset")and(session:ready())) then
> session:execute("send_dtmf", "2 at 100"); --@ determines dtmf length, enter
> NPI
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jaykris at gmail.com  Mon Nov 12 21:15:48 2012
From: jaykris at gmail.com (JP)
Date: Mon, 12 Nov 2012 10:15:48 -0800
Subject: [Freeswitch-users] Can I use mod_shout in a macro?
In-Reply-To: <CAKzWOxU5b+Yqoc2SUUVEdGhB=_+VsVoXhqrpBQGS2H4a6fR1Jg@mail.gmail.com>
References: <CAMRwkEC6dD9=Uy4tJqk+AcQwXs64bfjSbwNKcGsw-Or3c8+cUQ@mail.gmail.com>
	<CAKzWOxU5b+Yqoc2SUUVEdGhB=_+VsVoXhqrpBQGS2H4a6fR1Jg@mail.gmail.com>
Message-ID: <CAMRwkEDd=9wTCUNo8RAwi_eYAGegBmCp+jQUnwiJwoe4dB25FA@mail.gmail.com>

I want to use mod_shout with google Translate for doing TTS. So I thought
of embedding this in a macro so that I can send the text I wanted
translated as an argument to the macro and reuse the same macro from
different points in my dialplan. Is there a better way do this? Thanks,

JP

On Mon, Nov 12, 2012 at 9:20 AM, Michael Collins <msc at freeswitch.org> wrote:

>
> On Tue, Nov 6, 2012 at 2:27 PM, JP <jaykris at gmail.com> wrote:
>
>> Is there a way that I can use mod_shout from a phrase macro?
>>
>> Thanks
>> JP
>>
>
> Just curious - what are you trying to do?
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Nov 12 21:20:47 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 10:20:47 -0800
Subject: [Freeswitch-users] mod_voicemail.c
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2328597@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2328597@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxUh+kzY08YQ7HOP3y1+Osh9bt9s7E1ZpGz8LPd+6nngEw@mail.gmail.com>

This information gets passed in to the function in which this query is
called. Look for calls to the function named "deliver_vm" to see what
arguments the calling process sends to it.

-MC

On Fri, Nov 9, 2012 at 4:49 AM, Archana Venugopan <a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> I have got a doubt. In mod_voicemail.c script I see the below query****
>
> ** **
>
> "insert into voicemail_msgs(created_epoch, read_epoch, username, domain,
> uuid, cid_name, "****
>
>                                                           "cid_number,
> in_folder, file_path, message_len, flags, read_flags, forwarded_by) "****
>
>
> "values(%ld,0,'%q','%q','%q','%q','%q','%q','%q','%u','','%q','%q')",
> (long) switch_epoch_time_now(NULL),****
>
>                                                           myid,
> domain_name, use_uuid, caller_id_name, caller_id_number,****
>
>                                                           myfolder,
> file_path, message_len, read_flags, switch_str_nil(forwarded_by));****
>
> ** **
>
> Want to know from where myid and domain_name are picked up. Initially I
> thought it was picking from voicemail_prefs but even if there are no
> entries for a caller in voicemail_prefs table it still insert the value.**
> **
>
> Please let me know on this. ****
>
> ** **
>
> Mod_voicemail.c code is the same as present online.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 12 21:25:26 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 10:25:26 -0800
Subject: [Freeswitch-users] No Media on bridging two external calls
In-Reply-To: <CANGqFgMEiGyXtAw4wBp7WW8pu5r8eMGVYp2eUOmEncMPCh3WBA@mail.gmail.com>
References: <CANGqFgMEiGyXtAw4wBp7WW8pu5r8eMGVYp2eUOmEncMPCh3WBA@mail.gmail.com>
Message-ID: <CAKzWOxWwazXRsOTCYcy=C_xL=OzZKWxFQwc7+LZQSJxW076pow@mail.gmail.com>

Recommend viewing the SIP trace on working vs. non-working calls. See if
you can figure out where the audio is attempting to go on the non-working
call as that will be your first clue on what to do next. Depending on your
preference you can turn on SIP trace right at the command line or use a
tool like tcpdump and then analyze in Wireshark. If you're relatively new
to SIP traces then I'd use tcpdump + WS since it's a bit easier to read.
More info available on our wiki:

http://wiki.freeswitch.org/wiki/Packet_Capture

-MC

On Fri, Nov 9, 2012 at 8:14 AM, Lynn Nielson <
lynn.nielson at greenseedtechnologies.com> wrote:

> Running FreeSwitch on Ubuntu 12.04 FreeSWITCH Version
> 1.3.0+git~20120823T225232Z~fbc83cb0ea
>
> I'm having a problem with no media/audio when bridging two external calls,
> however, if one of the legs is an internal extension the the audio works
> fine. Also, if I add bypass_media=true, the audio works for the two
> external call bridge as well.  I would really like to understand why this
> works this way and if there is a configuration option that would enable the
> internal and external bridging to work the same.  Here are two examples of
> the problem from the command line using sofia.
>
> This fails on audio but makes a connection (external to external):
> originate sofia/gateway/flowroute/1801xxxxxxx
> &bridge(sofia/gateway/flowroute/1801yyyyyyy)
>
> This works (external to external):
> originate {bypass_media=true}sofia/gateway/flowroute/1801xxxxxxx
> &bridge(sofia/gateway/flowroute/1801yyyyyyy)
>
> This works (internal to external):
> originate sofia/internal/1010%IPADDRESS
> &bridge(sofia/gateway/flowroute/1801yyyyyyy)
>
> This also works (internal to external):
> originate {bypass_media=true}sofia/internal/1010%IPADDRESS
> &bridge(sofia/gateway/flowroute/1801yyyyyyy)
>
>
> Thanks for any explanation,
>
> Lynn
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Mon Nov 12 21:26:27 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Nov 2012 12:26:27 -0600
Subject: [Freeswitch-users] mod_opus codec name.
In-Reply-To: <87689103-2F3F-4527-90C9-0FC41F51EBF0@carmickle.com>
References: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com>
	<CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com>
	<CANBhyPDhHcc1_Gw+Q3BQS=mL0nUv0_-U2ACbg=WyfK+4_ymGKw@mail.gmail.com>
	<015e01cdc0c8$b402ffa0$1c08fee0$@207me.com>
	<87689103-2F3F-4527-90C9-0FC41F51EBF0@carmickle.com>
Message-ID: <CAKbxfG82Caiuz2noBR87X+=ayNdq3iMOcjzv3PoXViqcGT-j_Q@mail.gmail.com>

Incorrect.  If you have a version of the module from before the codec was
released officially it would be opus-0.9.0.  That is now discontinued so
you would need to update to latest and use "opus"



On Mon, Nov 12, 2012 at 9:29 AM, Frank Carmickle <frank at carmickle.com>wrote:

>
> On Nov 12, 2012, at 6:27 AM, Stephen Dame <sdame at 207me.com> wrote:
>
> > Anton,  I have reloaded and restarted FS. And made sure mod_opus is
> loaded.
> >
> > I still see no documentation in forums or wiki, as to what name should
> be used for opus in global codec preferences.
> >
> > I get 488,  which looks like  freeswitch is never looking to compare it
> against the preference list.
>
> It used to be Opus-0.9.0.  I believe it is now Opus-1.0.1.
>
> HTH
> --FC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From msc at freeswitch.org  Mon Nov 12 21:40:47 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 10:40:47 -0800
Subject: [Freeswitch-users] Return code from ESL Message Sending
In-Reply-To: <CACyT-3mexr4O9D1jSqM94U50x4TacjDCHi37UNh2Y6sEmHaoCQ@mail.gmail.com>
References: <CACyT-3mexr4O9D1jSqM94U50x4TacjDCHi37UNh2Y6sEmHaoCQ@mail.gmail.com>
Message-ID: <CAKzWOxXZr_e6Qd5Di8=7x8XL6LBBB7mnJq1NSNuTVBWJ4ciqoQ@mail.gmail.com>

I'm afraid the mod_sms page is a little sparse on details. If anyone has a
working chat application that can handle various responses like a 415 then
we'd love to see it doc'd on the mod_sms page.

-MC

P.S - IIRC there won't be a reply to this event with the response from the
other server. You simply fire your event and that's it. I don't believe
there are even any FreeSWITCH event system events thrown when the SIP
response comes back. I'll defer to the expertise of those more experienced
than I, but if I understand correctly most (all?) of the SIP messaging
takes place down in the Sofia stack and never actually causes events to be
fired.


On Fri, Nov 9, 2012 at 8:42 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>wrote:

> Hello Freeswitch Users:
>
> We're currently trying to get the return code from a MESSAGE we send using
> ESL. The closest we've found is this jira:
> http://jira.freeswitch.org/browse/FS-4453 which seems to provide similar
> functionality for the chat command, but nothing for ESL.
>
> Here's a pastebin of our current code:
> http://pastebin.freeswitch.org/20201
>
> The server we are hitting is returning a "415 Unsupported Content Type"
> (which is correct) and we're trying to discover that in freeswitch, instead
> of assuming the message was received correctly. Right now, we get that the
> recvEventTimed is returning None. This is all done on the a pull of FS from
> yesterday.
>
> Any suggestions?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 12 21:51:54 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 10:51:54 -0800
Subject: [Freeswitch-users] Check user state in dialplan
In-Reply-To: <CANM4RuigJ6CRDgZSYBzH3nrMYUGAkO-fT67V6U48ng+Cg2W_6A@mail.gmail.com>
References: <CANM4RuigJ6CRDgZSYBzH3nrMYUGAkO-fT67V6U48ng+Cg2W_6A@mail.gmail.com>
Message-ID: <CAKzWOxUm0NgmC+9VD0qZy9i5T8xYk1ZQCDHEFgAg2egQZMMADw@mail.gmail.com>

Are you planning on using mod_xml_curl for this? Just curious.

In any event, the example configuration has conf/dialplan/default.xml with
"Local_Extension" entry that handles all of these except for the first
case. However, I have to ask - what is wrong with the default way that FS
handles unknown numbers?

-MC

On Sat, Nov 10, 2012 at 11:02 AM, Eugene Prokopiev <
eugene.prokopiev at gmail.com> wrote:

> Hi,
>
> I need to perform different actions in response to the following
> conditions:
>
> * The called number is not exists in the directory (check id and
> number-alias)
> * The called number is known, but not registered
> * The called number is busy
> * The called number is not answered due timeout
> * The called number is answered
>
> In the first case I need to send the call through a particular gateway, in
> the next two cases I need to play a file, in two remaining cases I need
> to call user and play a file only on timeout.
>
> What is the right to do it in the dialplan?
>
> --
> Regards,
> Eugene Prokopiev
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 12 21:58:53 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 10:58:53 -0800
Subject: [Freeswitch-users] Bridging an incoming call to outbound gateway
In-Reply-To: <CAHkDqS_=2u1XvbkwH3fRdt750tX833qJ50SUmpOs+77ChV_+zg@mail.gmail.com>
References: <CAHkDqS_=2u1XvbkwH3fRdt750tX833qJ50SUmpOs+77ChV_+zg@mail.gmail.com>
Message-ID: <CAKzWOxV70Dt9GjQF7SrS842CD_Qaow0BENJ7Qa0MR_A_vVfxTw@mail.gmail.com>

You'll want to get the SIP trace and console debug log and put on
pastebin.freeswitch.org with "FreeSWITCH Log" as the syntax highlighting.

-MC

On Sun, Nov 11, 2012 at 4:18 PM, Meditel <meditel at gmail.com> wrote:

> Hi,
>
> I have actually 2 providers : p1 & p2
> p1 is used as a DID only for incoming calls
> p2 is used for making outgoing calls
>
> I have 1 registered user : ext 1000
>
> I have 2 GSM phones : GSM1 & GSM2
>
> My dialplan is as :
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(MY_P1_DID)$">
>         <action application="transfer" data="1000 XML default"/>
>     </condition>
>   </extension>
> </include>
>
> For incoming calls "1000" is ringing and i can make a call without any
> problem (GSM1 => P1_DID => ext 1000)
> But whene trying to bridge the incoming call to p2 gateway (GSM1 => P1_DID
> => P2_GW => GSM2), i can see that GSM2 is ringing but whene i answer the
> call i can't hear any audio ...
> This is my dialplan for the second case:
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(MY_P1_DID)$">
>         <action application="bridge"
> data="sofia/gateway/p2_gw/GSM2_NUMBER"/>
>     </condition>
>   </extension>
> </include>
>
> Any help are welcome
> best regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 12 22:03:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 11:03:20 -0800
Subject: [Freeswitch-users] Call FORBIDDEN when setting
 max-registrations-per-extnsion parameter
In-Reply-To: <50A1263C.6050100@seletech.com>
References: <50A1263C.6050100@seletech.com>
Message-ID: <CAKzWOxWM2B-CdkkPpt6=ySTcXn43PyzvU-AkeOBRVQDYrUCXHQ@mail.gmail.com>

So the same phone that registers successfully cannot make outbound calls?
Just for the sake of testing, have you tried setting the max registrations
value to "2" or "3" and test again?

-MC

On Mon, Nov 12, 2012 at 8:39 AM, alberto Villa <a.villa at seletech.com> wrote:

> Hello, I found that if I set the "max-registrations-per-extension"
> parameter for an extension 2000 as follows
>
>  <param name="max-registrations-per-extension" value="1"/>
>
> in the <params> section of this account .xml file, then this phone
> cannot execute any call as the server response is always "forbidden".
>
> Is this a bug? if not why a call of an already registered user sholud
> trigger on such parameter?
>
> I'm using FreeSWITCH version: 1.2.0 (git-b3b2c37 2012-05-18 13-41-16 -0500)
>
> Thanks in advance
>
> Alberto
>
> --
> Dr. Villa Alberto
> Sw Engineer
>
> SeleTech srl
> via Collodi, 8 20052 Monza (MI)
> tel: +39 039 5962000
> fax: +39 039 9716905
> email: a.villa at seletech.com
> web: www.seletech.com
>      www.seletech.eu
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Mon Nov 12 22:16:50 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Nov 2012 13:16:50 -0600
Subject: [Freeswitch-users] Domains and profiles
In-Reply-To: <1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>
References: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
	<1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>
Message-ID: <CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>

The best thing to do is take a look at these things from a step back.

The domains inside the xml registry are completely different from the
domains on the internet and again completely different from domains in sip
packets.  The profiles are again entirely different from any of the above.
 Its up to you to align them if you so choose.


The default configuration distributed with FreeSWITCH sets up the scenario
most likely to load on any machine and work out of the box.  That is the
primary goal of that configuration, so, It sets the domain in both the
directory, the global default domain variable and the name of the internal
profile to be identical to the ip on the box that can reach the internet.
 Then it sets the sip to force everything to that value.  When you want to
detach from this behavior, you are probably on a venture to do some kind of
multi-home setup.


Aliases in the <aliases> tag are a list of keys you want to use to use that
lead to the current profile your are configuring.  Think of it as the
/etc/hosts file in unix only for profiles.   When you define aliases to
match all of the possible domains hosted on a particular profile, then when
you try to take a user at host.com notation and decide which profile it came
from, you can use the aliases to find it providing you have added <alias
name="host.com"/> to that profile.

The <domains> tag is an indicator telling the profile to open the xml
registry in FreeSWITCH and run through any domains defined therein.
The 2 key attributes are:

alias: [true/false]  (automatically create an alias for this domain as
mentioned above)
parse: [true/false] (scan the domain for gateway entries and include them
into this profile)
name: [<string>] (either the name of a specific domain or 'all' to denote
parsing every domain in the directory)

As you showed in your question the default config has

<domain name="all" alias="true" parse="false"/>

If you apply what you have learned above, it will scan for every domain
(there is only one by default) and add an alias for it and not parse it for
gateways.  The default directory uses global config vars to set the domain
to match the local ip on the box.  So now you will have a domain in your
config that is your ip, and the internal profile will attach to it and add
an alias so that value expands to match it.


This is explained in a comment at the top of directory/default.xml

    FreeSWITCH works off the concept of users and domains just like email.


    You have users that are in domains for example 1000 at domain.com.





    When freeswitch gets a register packet it looks for the user in the
directory

    based on the from or to domain in the packet depending on how your
sofia profile

    is configured.  Out of the box the default domain will be the IP
address of the

    machine running FreeSWITCH.  This IP can be found by typing "sofia
status" at the

    CLI.  You will register your phones to the IP and not the hostname by
default.

    If you wish to register using the domain please open vars.xml in the
root conf

    directory and set the default domain to the hostname you desire.  Then
you would

    use the domain name in the client instead of the IP address to register


    with FreeSWITCH.




So having more than one profile with the default of

<domain name="all" alias="true" parse="false"/>

is going to end up aliasing the same domains into all profiles who call it
and cause an overwrite in the lookup table and probably an error in your
logs somewhere.  If you had parse="true" on all of them, they would all try
and register to the gateways in all of your domains.


If you look at the stock config, external.xml is a good example of
a secondary profile, it has

<domain name="all" alias="false" parse="true"/>

so no aliases, and yes parse ... the exact opposite of the internal so that
all the gateways would register from external and internal would bind to
the local ip.

So, you probably want to use separate <domain name="my.com"/> per domain
per profile you want to bind it to in more complicated setups.



























On Sun, Nov 11, 2012 at 9:09 PM, Emrah <lists at kavun.ch> wrote:

> Bless you!
>
> Thanks for putting this together. You've beautifully summed up all my
> questions.
> On Nov 11, 2012, at 8:09 AM, Lawrence Conroy <lconroy at insensate.co.uk>
> wrote:
>
> > Hi Folks,
> > I've started a new thread as it's not quite the same issue, and domains
> & profiles have confused the heck out of me every time I have developed a
> new setup for fS.
> > I have sometimes had to hack/hard-doce the dialstring to make multiple
> domains in one profile work, had hours of fun with presence, db and force
> register settings, and have still had some odd gotchas that have required
> extensive meditation.
> > [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract
> these elements ]
> >
> > Coming at this from standards/specs and rolling my own SIP stacks,
> sofia/fS seems to use the term "domain" differently from sipdomain, and
> alias seems to be tied to the directory (and thus to the profile listed in
> a directory file), but I'm not sure.
> > so ...
> > Before I capture to the sofia conf xml wiki page, I have a couple of
> questions on the sip-profile XML setup;
> >
> > Q: Is there a particular reason why there's a parameter called alias and
> an (entirely different) setting also called alias?
> > The sofia conf xml wiki's comment on the setting "alias" shows I'm not
> alone.
> > I agree that's what it appears to be doing, but can we nail this down
> please (and what happens if an external client uses this connection to
> register and call)?
> >
> > In the current sofia conf xml wiki page, the domain setting is not
> exactly well documented :).
> > The current internal.xml vanilla example from git (as of time of
> writing) has the following lines:
> > -------------------------
> > ...
> > <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> > <!--aliases are other names that will work as a valid profile name for
> this profile-->
> > <aliases>
> >   <!--
> >        <alias name="default"/>
> >    -->
> > </aliases>
> > <!-- Outbound Registrations -->
> > <gateways>
> >   <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
> > </gateways>
> >
> > <domains>
> >   <!-- indicator to parse the directory for domains with parse="true" to
> get gateways-->
> >   <!--<domain name="$${domain}" parse="true"/>-->
> >   <!-- indicator to parse the directory for domains with parse="true" to
> get gateways and alias every domain to this profile -->
> >   <!--<domain name="all" alias="true" parse="true"/>-->
> >   <domain name="all" alias="true" parse="false"/>
> > </domains>
> > ...
> > -------------------------
> >
> > This stuff is entirely missing from the sofia.conf.xml wiki page, and it
> IS really important.
> >
> >
> > Q: what's the default value for the alias parameter in the domain
> element? -- it is missing from the first example.
> > Q: if there is more than one profile, what's the impact of setting parse
> = true in one (or all) of the profiles' XML files?
> >   (or parse = false, or missing the parameter altogether)?
> > AFAICT, the gateways get pulled in via the pre-process directive just
> fine, regardless of the value of the parse parameter -- it works for me, at
> least.
> >
> > Q: if there is more than one profile, what's the impact of putting
> domain name="all" into one (or all) of the profiles' XML files?
> >
> > Ideally, having more than one sipdomain tied to one profile "would be
> good", but aliases doesn't do that -- as the git file says, these are
> aliases for the profile name.
> >
> > Before I start scribbling, Answers on a postcard to this ML, please.
> >
> > all the best,
> >  Lawrence
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From kris at kriskinc.com  Tue Nov 13 00:30:11 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 12 Nov 2012 16:30:11 -0500
Subject: [Freeswitch-users] About OPUS codec features
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAK2luhPmpPJCnBg1zt0rT1HCgAAAEAAAABJiGmkH4yFImmgThuLUbOwBAAAAAA==@coinfru.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAK2luhPmpPJCnBg1zt0rT1HCgAAAEAAAABJiGmkH4yFImmgThuLUbOwBAAAAAA==@coinfru.com>
Message-ID: <CAKDfjgeQVYZC=jmO3V31ufhDASvNPrTPCPSCYBMQT9+LkRkEOA@mail.gmail.com>

Josue,

  Opus is a codec and as a codec it supports 255 channels of audio - think
of surround sound in home theater applications that may have five, seven,
or more discrete channels of audio.

  It has nothing to do with SIP trunking.


2012/11/12 Josue Diaz Cruz <jdiaz at coinfru.com>

> **
> I saw in the Features of this codec that Support for up to 255 channels.
> Is this multiplexing? What this does mean and if this have any utility in
> Freeswitch like trunking or something like this?
>
>
> *Josue Diaz Cruz*
>
> *Departamento Tecnico y Soporte*
>
> *jdiaz at coinfru.com* <jdiaz at coinfru.com>******
>
> *** ***
>
> *C/ Balsicas 3*
>
> *Alquerias | 30580 | Murcia*
>
> * **www.coinfru.com* <http://www.coinfru.com/>
>
>
>
>
>
>
>
> Este e-mail contiene informaci?n confidencial, el contenido de la mismo se
> encuentra protegido por Ley. Cualquier persona distinta a su destinataria
> tiene prohibida su reproducci?n, uso, divulgaci?n o impresi?n total o
> parcial. Si ha recibido este mensaje por error, notifiquelo de inmediato al
> remitente borrando el mensaje original juntamente con sus ficheros anexos.
> Gracias.
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Kristian Kielhofner
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From meditel at gmail.com  Tue Nov 13 00:47:04 2012
From: meditel at gmail.com (Meditel)
Date: Mon, 12 Nov 2012 22:47:04 +0100
Subject: [Freeswitch-users] Bridging an incoming call to outbound gateway
In-Reply-To: <CAKzWOxV70Dt9GjQF7SrS842CD_Qaow0BENJ7Qa0MR_A_vVfxTw@mail.gmail.com>
References: <CAHkDqS_=2u1XvbkwH3fRdt750tX833qJ50SUmpOs+77ChV_+zg@mail.gmail.com>
	<CAKzWOxV70Dt9GjQF7SrS842CD_Qaow0BENJ7Qa0MR_A_vVfxTw@mail.gmail.com>
Message-ID: <CAHkDqS8=mErwyTcytPYSzD5GQqARNoM+OSu4y0q5KBod=Dp3BQ@mail.gmail.com>

hi Michael,

sip traces : http://pastebin.freeswitch.org/20209

gsm is ringing and no audio in both ways..

Best regards

2012/11/12 Michael Collins <msc at freeswitch.org>

> You'll want to get the SIP trace and console debug log and put on
> pastebin.freeswitch.org with "FreeSWITCH Log" as the syntax highlighting.
>
> -MC
>
> On Sun, Nov 11, 2012 at 4:18 PM, Meditel <meditel at gmail.com> wrote:
>
>> Hi,
>>
>> I have actually 2 providers : p1 & p2
>> p1 is used as a DID only for incoming calls
>> p2 is used for making outgoing calls
>>
>> I have 1 registered user : ext 1000
>>
>> I have 2 GSM phones : GSM1 & GSM2
>>
>> My dialplan is as :
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^(MY_P1_DID)$">
>>         <action application="transfer" data="1000 XML default"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> For incoming calls "1000" is ringing and i can make a call without any
>> problem (GSM1 => P1_DID => ext 1000)
>> But whene trying to bridge the incoming call to p2 gateway (GSM1 =>
>> P1_DID => P2_GW => GSM2), i can see that GSM2 is ringing but whene i answer
>> the call i can't hear any audio ...
>> This is my dialplan for the second case:
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^(MY_P1_DID)$">
>>         <action application="bridge"
>> data="sofia/gateway/p2_gw/GSM2_NUMBER"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> Any help are welcome
>> best regards
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From meditel at gmail.com  Tue Nov 13 00:50:34 2012
From: meditel at gmail.com (Meditel)
Date: Mon, 12 Nov 2012 22:50:34 +0100
Subject: [Freeswitch-users]  Bridging an incoming call to outbound gateway
Message-ID: <CAHkDqS9s9HSYbu8hBTVRBVmp0WmbqRg80VO7F+a4XCSPCNLk_A@mail.gmail.com>

hi Michael,

sip traces : http://pastebin.freeswitch.org/20209

gsm is ringing and no audio in both ways..

Best regards
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From msc at freeswitch.org  Tue Nov 13 01:20:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 14:20:31 -0800
Subject: [Freeswitch-users] Domains and profiles
In-Reply-To: <CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>
References: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
	<1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>
	<CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>
Message-ID: <CAKzWOxVzM9moL99DWHx3ksPFE1ObnCF55-fne7Q0_-j+kpf7Xg@mail.gmail.com>

Any time Tony writes an email this long you can safely bet that it's worth
adding to the wiki. :) In this case I added a section to the main SIP
config page:

http://wiki.freeswitch.org/wiki/Sofia.conf.xml#The_Relationship_Between_SIP_Profiles_and_Domains

For those of you who know this stuff inside and out, feel free to add your
experience to the melting pot of knowledge that is our wiki.

Thanks,
MC

On Mon, Nov 12, 2012 at 11:16 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The best thing to do is take a look at these things from a step back.
>
> The domains inside the xml registry are completely different from the
> domains on the internet and again completely different from domains in sip
> packets.  The profiles are again entirely different from any of the above.
>  Its up to you to align them if you so choose.
>
>
> The default configuration distributed with FreeSWITCH sets up the scenario
> most likely to load on any machine and work out of the box.  That is the
> primary goal of that configuration, so, It sets the domain in both the
> directory, the global default domain variable and the name of the internal
> profile to be identical to the ip on the box that can reach the internet.
>  Then it sets the sip to force everything to that value.  When you want to
> detach from this behavior, you are probably on a venture to do some kind of
> multi-home setup.
>
>
> Aliases in the <aliases> tag are a list of keys you want to use to use
> that lead to the current profile your are configuring.  Think of it as the
> /etc/hosts file in unix only for profiles.   When you define aliases to
> match all of the possible domains hosted on a particular profile, then when
> you try to take a user at host.com notation and decide which profile it came
> from, you can use the aliases to find it providing you have added <alias
> name="host.com"/> to that profile.
>
> The <domains> tag is an indicator telling the profile to open the xml
> registry in FreeSWITCH and run through any domains defined therein.
> The 2 key attributes are:
>
> alias: [true/false]  (automatically create an alias for this domain as
> mentioned above)
> parse: [true/false] (scan the domain for gateway entries and include them
> into this profile)
> name: [<string>] (either the name of a specific domain or 'all' to denote
> parsing every domain in the directory)
>
> As you showed in your question the default config has
>
> <domain name="all" alias="true" parse="false"/>
>
> If you apply what you have learned above, it will scan for every domain
> (there is only one by default) and add an alias for it and not parse it for
> gateways.  The default directory uses global config vars to set the domain
> to match the local ip on the box.  So now you will have a domain in your
> config that is your ip, and the internal profile will attach to it and add
> an alias so that value expands to match it.
>
>
> This is explained in a comment at the top of directory/default.xml
>
>     FreeSWITCH works off the concept of users and domains just like email.
>
>
>     You have users that are in domains for example 1000 at domain.com.
>
>
>
>
>
>     When freeswitch gets a register packet it looks for the user in the
> directory
>
>     based on the from or to domain in the packet depending on how your
> sofia profile
>
>     is configured.  Out of the box the default domain will be the IP
> address of the
>
>     machine running FreeSWITCH.  This IP can be found by typing "sofia
> status" at the
>
>     CLI.  You will register your phones to the IP and not the hostname by
> default.
>
>     If you wish to register using the domain please open vars.xml in the
> root conf
>
>     directory and set the default domain to the hostname you desire.  Then
> you would
>
>     use the domain name in the client instead of the IP address to
> register
>
>     with FreeSWITCH.
>
>
>
>
> So having more than one profile with the default of
>
> <domain name="all" alias="true" parse="false"/>
>
> is going to end up aliasing the same domains into all profiles who call it
> and cause an overwrite in the lookup table and probably an error in your
> logs somewhere.  If you had parse="true" on all of them, they would all try
> and register to the gateways in all of your domains.
>
>
> If you look at the stock config, external.xml is a good example of
> a secondary profile, it has
>
> <domain name="all" alias="false" parse="true"/>
>
> so no aliases, and yes parse ... the exact opposite of the internal so
> that all the gateways would register from external and internal would bind
> to the local ip.
>
> So, you probably want to use separate <domain name="my.com"/> per domain
> per profile you want to bind it to in more complicated setups.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Nov 11, 2012 at 9:09 PM, Emrah <lists at kavun.ch> wrote:
>
>> Bless you!
>>
>> Thanks for putting this together. You've beautifully summed up all my
>> questions.
>> On Nov 11, 2012, at 8:09 AM, Lawrence Conroy <lconroy at insensate.co.uk>
>> wrote:
>>
>> > Hi Folks,
>> > I've started a new thread as it's not quite the same issue, and domains
>> & profiles have confused the heck out of me every time I have developed a
>> new setup for fS.
>> > I have sometimes had to hack/hard-doce the dialstring to make multiple
>> domains in one profile work, had hours of fun with presence, db and force
>> register settings, and have still had some odd gotchas that have required
>> extensive meditation.
>> > [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract
>> these elements ]
>> >
>> > Coming at this from standards/specs and rolling my own SIP stacks,
>> sofia/fS seems to use the term "domain" differently from sipdomain, and
>> alias seems to be tied to the directory (and thus to the profile listed in
>> a directory file), but I'm not sure.
>> > so ...
>> > Before I capture to the sofia conf xml wiki page, I have a couple of
>> questions on the sip-profile XML setup;
>> >
>> > Q: Is there a particular reason why there's a parameter called alias
>> and an (entirely different) setting also called alias?
>> > The sofia conf xml wiki's comment on the setting "alias" shows I'm not
>> alone.
>> > I agree that's what it appears to be doing, but can we nail this down
>> please (and what happens if an external client uses this connection to
>> register and call)?
>> >
>> > In the current sofia conf xml wiki page, the domain setting is not
>> exactly well documented :).
>> > The current internal.xml vanilla example from git (as of time of
>> writing) has the following lines:
>> > -------------------------
>> > ...
>> > <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>> > <!--aliases are other names that will work as a valid profile name for
>> this profile-->
>> > <aliases>
>> >   <!--
>> >        <alias name="default"/>
>> >    -->
>> > </aliases>
>> > <!-- Outbound Registrations -->
>> > <gateways>
>> >   <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
>> > </gateways>
>> >
>> > <domains>
>> >   <!-- indicator to parse the directory for domains with parse="true"
>> to get gateways-->
>> >   <!--<domain name="$${domain}" parse="true"/>-->
>> >   <!-- indicator to parse the directory for domains with parse="true"
>> to get gateways and alias every domain to this profile -->
>> >   <!--<domain name="all" alias="true" parse="true"/>-->
>> >   <domain name="all" alias="true" parse="false"/>
>> > </domains>
>> > ...
>> > -------------------------
>> >
>> > This stuff is entirely missing from the sofia.conf.xml wiki page, and
>> it IS really important.
>> >
>> >
>> > Q: what's the default value for the alias parameter in the domain
>> element? -- it is missing from the first example.
>> > Q: if there is more than one profile, what's the impact of setting
>> parse = true in one (or all) of the profiles' XML files?
>> >   (or parse = false, or missing the parameter altogether)?
>> > AFAICT, the gateways get pulled in via the pre-process directive just
>> fine, regardless of the value of the parse parameter -- it works for me, at
>> least.
>> >
>> > Q: if there is more than one profile, what's the impact of putting
>> domain name="all" into one (or all) of the profiles' XML files?
>> >
>> > Ideally, having more than one sipdomain tied to one profile "would be
>> good", but aliases doesn't do that -- as the git file says, these are
>> aliases for the profile name.
>> >
>> > Before I start scribbling, Answers on a postcard to this ML, please.
>> >
>> > all the best,
>> >  Lawrence
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Nov 13 01:29:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Nov 2012 14:29:17 -0800
Subject: [Freeswitch-users] Can I use mod_shout in a macro?
In-Reply-To: <CAMRwkEDd=9wTCUNo8RAwi_eYAGegBmCp+jQUnwiJwoe4dB25FA@mail.gmail.com>
References: <CAMRwkEC6dD9=Uy4tJqk+AcQwXs64bfjSbwNKcGsw-Or3c8+cUQ@mail.gmail.com>
	<CAKzWOxU5b+Yqoc2SUUVEdGhB=_+VsVoXhqrpBQGS2H4a6fR1Jg@mail.gmail.com>
	<CAMRwkEDd=9wTCUNo8RAwi_eYAGegBmCp+jQUnwiJwoe4dB25FA@mail.gmail.com>
Message-ID: <CAKzWOxWzMqQDEjTvj+aKJFy=3=Qg08TFsz=uR=cRf5bKpedpOA@mail.gmail.com>

As far as I can tell this should work if you have the correct URL for the
shout stream. Inside your macro it would just be a simple playback call:

<action function="execute" data="playback(shout://<URL_here>)"/>

Give it a try and let us know what happens.
-MC

On Mon, Nov 12, 2012 at 10:15 AM, JP <jaykris at gmail.com> wrote:

> I want to use mod_shout with google Translate for doing TTS. So I thought
> of embedding this in a macro so that I can send the text I wanted
> translated as an argument to the macro and reuse the same macro from
> different points in my dialplan. Is there a better way do this? Thanks,
>
> JP
>
> On Mon, Nov 12, 2012 at 9:20 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>> On Tue, Nov 6, 2012 at 2:27 PM, JP <jaykris at gmail.com> wrote:
>>
>>> Is there a way that I can use mod_shout from a phrase macro?
>>>
>>> Thanks
>>> JP
>>>
>>
>> Just curious - what are you trying to do?
>> -MC
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From victortho at gmail.com  Tue Nov 13 04:25:40 2012
From: victortho at gmail.com (Victor)
Date: Mon, 12 Nov 2012 17:25:40 -0800
Subject: [Freeswitch-users] E-mail voicemails - msmtp in Windows
Message-ID: <CAG0Fp+_kQf1pZyZpCDWuqAVMpYsCfA3p1gUyrSzWK24VNi9Yjg@mail.gmail.com>

My system is having trouble e-mailing voicemails. This happened when I
upgraded from 1.0.head (1/23/2012) to 1.2.3 or 1.3.2.

E-mails are being sent to the correct recipient, but they are blank (msmtp
logs indicate mailsize=0).

Nov 12 15:47:55 host=smtp.gmail.com tls=on auth=on user=noreply@*****.com
from=31 at 192.168.0.11 recipients=***@*****.com mailsize=0 smtpstatus=250
smtpmsg='250 2.0.0 OK 1352764075 os5sm4913355pbc.15' exitcode=EX_OK

I see in FreeSWITCH's console "The syntax of the command is incorrect"
whenever an e-mail is being sent out.

All the configuration files are the same, voicemail files are being written
to disk.  Only difference/change I know of is the upgrade to 1.2.3.
Reverting back to 1.0.x fixes the problem.

--Victor
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From cgoudie at callware.com  Tue Nov 13 03:55:29 2012
From: cgoudie at callware.com (Clint Goudie-Nice)
Date: Mon, 12 Nov 2012 17:55:29 -0700
Subject: [Freeswitch-users] Making SRTP mandatory
Message-ID: <B3426479DE2E6C4ABFC4C61B5C3DF790736FE4BE09@cwexch.callware.com>

Greetings all,

I have been configuring a FreeSWITCH / FusionPBX system to do TLS and SRTP.

When I configure it to do srtp in the outbound route, the SDP produced looks like:

o=FreeSWITCH 1352745516 1352745517 IN IP4 10.16.0.131
s=FreeSWITCH
c=IN IP4 10.16.0.131
t=0 0
m=audio 22222 RTP/SAVP 9 0 8 3 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:djfKObSYrk0J8pVI1w/OfQBjTYwNgTi80C68VgBB
a=ptime:20
m=audio 22222 RTP/AVP 9 0 8 3 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

This is great if I want to support both SRTP and non-SRTP calls; but I cannot figure out how to remove the AVP block so that SRTP is mandatory.

How do I require SRTP?

Thanks,

Clint

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From anton.vazir at gmail.com  Tue Nov 13 09:38:26 2012
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 13 Nov 2012 10:38:26 +0400
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
 Web Site Dev/Maint
In-Reply-To: <5038D10E.3060102@puzzled.xs4all.nl>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<fba5f967342.50377451@shaw.ca>
	<CAKzWOxXV39R6BbNvyW3u8q6u67sTjQpGdE0cBO62G08wSNsm-Q@mail.gmail.com>
	<5038D10E.3060102@puzzled.xs4all.nl>
Message-ID: <CAGX+Xzt+4EG7xSLFUZBDiatinZBe7zm=Qo4nzUb0+BtONy4cPg@mail.gmail.com>

Joomla has proven to be much less secure than Drupal or WP, surely not
a little depends to the admin, and in my experience of running small
web-hosting a while ago, there was uncountable security cases and
issues with customers installing/running Joomla based sites and almost
none for Drupal or WP. Would not like to start a religious war, I was
only hosting that stuff, but had to deal with security complaints.

2012/8/25 Patrick Lists <freeswitch-list at puzzled.xs4all.nl>:
> On 25-08-12 02:17, Michael Collins wrote:
>> Tchavar,
>>
>> I'm so glad you replied! You are the first person with any Wordpress
>> knowledge who has contacted us. We are investigating the possibility of
>> migrating from Drupal to Wordpress for FreeSWITCH.org.
>
> While Wordpress is an excellent solution I thought it lacked Community
> features. If Community/social platform features are required how about
> Joomla 2.5 with an extension like JomSocial or Community Builder?
>
> http://www.jomsocial.com/
> http://www.joomlapolis.com/
>
> Regards,
> Patrick (no affiliation)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From dar at helia.ca  Tue Nov 13 05:22:33 2012
From: dar at helia.ca (Dar Zuch)
Date: Mon, 12 Nov 2012 19:22:33 -0700
Subject: [Freeswitch-users] Asynchronous session:streamFile in Lua
Message-ID: <03442375-C4B9-4F7A-8973-9C0A5F342F52@helia.ca>

I have a Lua script that 1) answers a session 2) initiates a new session 3) plays two different wave files on each leg 4) bridges the sessions.

I'm using session:streamFile() to play the sound file but this is happening synchronously (i.e. file completes playing on one leg before the streamFile() starts on the other leg).

Is there a way to play sounds files on each leg at the same time?


Dar @ Helia





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From enp at itx.ru  Tue Nov 13 07:09:06 2012
From: enp at itx.ru (Eugene Prokopiev)
Date: Tue, 13 Nov 2012 07:09:06 +0300
Subject: [Freeswitch-users] Check user state in dialplan
In-Reply-To: <CAKzWOxUm0NgmC+9VD0qZy9i5T8xYk1ZQCDHEFgAg2egQZMMADw@mail.gmail.com>
References: <CANM4RuigJ6CRDgZSYBzH3nrMYUGAkO-fT67V6U48ng+Cg2W_6A@mail.gmail.com>
	<CAKzWOxUm0NgmC+9VD0qZy9i5T8xYk1ZQCDHEFgAg2egQZMMADw@mail.gmail.com>
Message-ID: <CANM4Ruj24EKsu3ij6g-D=iacwLGzK_w+AuNKNMqWD0cM4k0aXA@mail.gmail.com>

Hi,

With "Local_Extension" I can't distinguish all cases, I can distinguish
only successful or failed answer.

About first case: I have no sequential numbers for local users and can't
describe range as regex. So, the best way to identify if local user exists
is to search id (and number-alias) in directory. I know about mod_xml_curl
which can be used for filling dialplan and directory from some database,
but I tried to find more simple way.

-- 
Regards,
Eugene Prokopiev
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From a.villa at seletech.com  Tue Nov 13 10:52:43 2012
From: a.villa at seletech.com (alberto Villa)
Date: Tue, 13 Nov 2012 08:52:43 +0100
Subject: [Freeswitch-users] Call FORBIDDEN when setting
 max-registrations-per-extnsion parameter
In-Reply-To: <CAKzWOxWM2B-CdkkPpt6=ySTcXn43PyzvU-AkeOBRVQDYrUCXHQ@mail.gmail.com>
References: <50A1263C.6050100@seletech.com>
	<CAKzWOxWM2B-CdkkPpt6=ySTcXn43PyzvU-AkeOBRVQDYrUCXHQ@mail.gmail.com>
Message-ID: <50A1FC4B.7000207@seletech.com>

No, I haven't tried it yet, just give me some time and  I will try with
a value of 2 or 3 and will provide some more info from fs log. What I
wanted to achieve when I set this parameter was to forbid any
registration following the first for any enabled extension: if A
successfully registers with ID ID_A and then B try to register with ID_A
I wanted the registration of B to be rejected (as successfully is when I
set max-registrations-per-extension parameter to 1).

Alberto


Il 12/11/2012 20:03, Michael Collins ha scritto:
> So the same phone that registers successfully cannot make outbound
> calls? Just for the sake of testing, have you tried setting the max
> registrations value to "2" or "3" and test again?
>
> -MC
>
> On Mon, Nov 12, 2012 at 8:39 AM, alberto Villa <a.villa at seletech.com
> <mailto:a.villa at seletech.com>> wrote:
>
>     Hello, I found that if I set the "max-registrations-per-extension"
>     parameter for an extension 2000 as follows
>
>      <param name="max-registrations-per-extension" value="1"/>
>
>     in the <params> section of this account .xml file, then this phone
>     cannot execute any call as the server response is always "forbidden".
>
>     Is this a bug? if not why a call of an already registered user sholud
>     trigger on such parameter?
>
>     I'm using FreeSWITCH version: 1.2.0 (git-b3b2c37 2012-05-18
>     13-41-16 -0500)
>
>     Thanks in advance
>
>     Alberto
>
>     --
>     Dr. Villa Alberto
>     Sw Engineer
>
>     SeleTech srl
>     via Collodi, 8 20052 Monza (MI)
>     tel: +39 039 5962000
>     fax: +39 039 9716905
>     email: a.villa at seletech.com <mailto:a.villa at seletech.com>
>     web: www.seletech.com <http://www.seletech.com>
>          www.seletech.eu <http://www.seletech.eu>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Dr. Villa Alberto
Sw Engineer

SeleTech srl
via Collodi, 8 20052 Monza (MI)
tel: +39 039 5962000
fax: +39 039 9716905
email: a.villa at seletech.com
web: www.seletech.com
     www.seletech.eu 

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From avi at avimarcus.net  Tue Nov 13 10:55:54 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 13 Nov 2012 09:55:54 +0200
Subject: [Freeswitch-users] Asynchronous session:streamFile in Lua
In-Reply-To: <03442375-C4B9-4F7A-8973-9C0A5F342F52@helia.ca>
References: <03442375-C4B9-4F7A-8973-9C0A5F342F52@helia.ca>
Message-ID: <CALNZuwmELn3=igy20oBJv3TMM-xAUtQhUdSG-OaW7x-TciZ77g@mail.gmail.com>

Not 100% sure if this is what you need.. but
uuid_broadcast<http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast>is
async.

-Avi


On Tue, Nov 13, 2012 at 4:22 AM, Dar Zuch <dar at helia.ca> wrote:

> I have a Lua script that 1) answers a session 2) initiates a new session
> 3) plays two different wave files on each leg 4) bridges the sessions.
>
> I'm using session:streamFile() to play the sound file but this is
> happening synchronously (i.e. file completes playing on one leg before the
> streamFile() starts on the other leg).
>
> Is there a way to play sounds files on each leg at the same time?
>
>
>   Dar @ Helia
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From a.villa at seletech.com  Tue Nov 13 11:17:30 2012
From: a.villa at seletech.com (alberto Villa)
Date: Tue, 13 Nov 2012 09:17:30 +0100
Subject: [Freeswitch-users] Call FORBIDDEN when setting
 max-registrations-per-extnsion parameter
In-Reply-To: <50A1FC4B.7000207@seletech.com>
References: <50A1263C.6050100@seletech.com>
	<CAKzWOxWM2B-CdkkPpt6=ySTcXn43PyzvU-AkeOBRVQDYrUCXHQ@mail.gmail.com>
	<50A1FC4B.7000207@seletech.com>
Message-ID: <50A2021A.5090200@seletech.com>

Just tried, here are the results:

- if I set max-registrations-per-extension to 2 everything works fine
and the phone can succesfully call

- if I set max-registrations-per-extension to 1 and call extension 1000
from extension 2000 I get the "FORBIDDEN" response on phone 2000 and fs
print this on his log:

2012-11-13 09:15:05.368001 [DEBUG] sofia.c:7847 IP 192.168.2.246
Rejected by acl "domains". Falling back to Digest auth.
2012-11-13 09:15:05.368001 [WARNING] sofia_reg.c:1446 SIP auth challenge
(INVITE) on sofia profile 'internal' for [1000 at 192.168.2.40] from ip
192.168.2.246
2012-11-13 09:15:05.488001 [DEBUG] sofia.c:7847 IP 192.168.2.246
Rejected by acl "domains". Falling back to Digest auth.
2012-11-13 09:15:05.488001 [WARNING] sofia_reg.c:2575 SIP auth failure
(REGISTER) due to reaching max allowed registrations.  Count: 1
2012-11-13 09:15:05.488001 [WARNING] sofia_reg.c:1391 SIP auth failure
(INVITE) on sofia profile 'internal' for [1000 at 192.168.2.40] from ip
192.168.2.246



Il 13/11/2012 08:52, alberto Villa ha scritto:
> No, I haven't tried it yet, just give me some time and  I will try
> with a value of 2 or 3 and will provide some more info from fs log.
> What I wanted to achieve when I set this parameter was to forbid any
> registration following the first for any enabled extension: if A
> successfully registers with ID ID_A and then B try to register with
> ID_A I wanted the registration of B to be rejected (as successfully is
> when I set max-registrations-per-extension parameter to 1).
>
> Alberto
>
>
> Il 12/11/2012 20:03, Michael Collins ha scritto:
>> So the same phone that registers successfully cannot make outbound
>> calls? Just for the sake of testing, have you tried setting the max
>> registrations value to "2" or "3" and test again?
>>
>> -MC
>>
>> On Mon, Nov 12, 2012 at 8:39 AM, alberto Villa <a.villa at seletech.com
>> <mailto:a.villa at seletech.com>> wrote:
>>
>>     Hello, I found that if I set the "max-registrations-per-extension"
>>     parameter for an extension 2000 as follows
>>
>>      <param name="max-registrations-per-extension" value="1"/>
>>
>>     in the <params> section of this account .xml file, then this phone
>>     cannot execute any call as the server response is always "forbidden".
>>
>>     Is this a bug? if not why a call of an already registered user sholud
>>     trigger on such parameter?
>>
>>     I'm using FreeSWITCH version: 1.2.0 (git-b3b2c37 2012-05-18
>>     13-41-16 -0500)
>>
>>     Thanks in advance
>>
>>     Alberto
>>
>>     --
>>     Dr. Villa Alberto
>>     Sw Engineer
>>
>>     SeleTech srl
>>     via Collodi, 8 20052 Monza (MI)
>>     tel: +39 039 5962000
>>     fax: +39 039 9716905
>>     email: a.villa at seletech.com <mailto:a.villa at seletech.com>
>>     web: www.seletech.com <http://www.seletech.com>
>>          www.seletech.eu <http://www.seletech.eu>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-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
>>
>>
>>
>>
>> -- 
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> -- 
> Dr. Villa Alberto
> Sw Engineer
>
> SeleTech srl
> via Collodi, 8 20052 Monza (MI)
> tel: +39 039 5962000
> fax: +39 039 9716905
> email: a.villa at seletech.com
> web: www.seletech.com
>      www.seletech.eu 
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Dr. Villa Alberto
Sw Engineer

SeleTech srl
via Collodi, 8 20052 Monza (MI)
tel: +39 039 5962000
fax: +39 039 9716905
email: a.villa at seletech.com
web: www.seletech.com
     www.seletech.eu 

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From miha at softnet.si  Tue Nov 13 12:16:35 2012
From: miha at softnet.si (Miha)
Date: Tue, 13 Nov 2012 10:16:35 +0100
Subject: [Freeswitch-users] FS server
Message-ID: <50A20FF3.3070404@softnet.si>

Hi,

we are buying new server for FS. Does FS knows to work with more than 
one CPU?

BR,
Miha


From freeswitch-list at puzzled.xs4all.nl  Tue Nov 13 12:29:51 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 13 Nov 2012 10:29:51 +0100
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
 Web Site Dev/Maint
In-Reply-To: <CAGX+Xzt+4EG7xSLFUZBDiatinZBe7zm=Qo4nzUb0+BtONy4cPg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<fba5f967342.50377451@shaw.ca>
	<CAKzWOxXV39R6BbNvyW3u8q6u67sTjQpGdE0cBO62G08wSNsm-Q@mail.gmail.com>
	<5038D10E.3060102@puzzled.xs4all.nl>
	<CAGX+Xzt+4EG7xSLFUZBDiatinZBe7zm=Qo4nzUb0+BtONy4cPg@mail.gmail.com>
Message-ID: <50A2130F.5030008@puzzled.xs4all.nl>

On 11/13/2012 07:38 AM, Anton VG wrote:
> Joomla has proven to be much less secure than Drupal or WP, surely not
> a little depends to the admin, and in my experience of running small
> web-hosting a while ago, there was uncountable security cases and
> issues with customers installing/running Joomla based sites and almost
> none for Drupal or WP. Would not like to start a religious war, I was
> only hosting that stuff, but had to deal with security complaints.

I don't know much about Drupal but having used Wordpress and Joomla for 
many years now both seem equally susceptible to security flaws when they 
are loaded with one-man-band plugins that were once fired and then 
forgotten about. The Core of both projects indeed had their fair share 
of security issues in older versions. These days if you use either 
Joomla 2.5.x or Wordpress 3.4.x with well supported (commercial) plugins 
and follow the extensive security guidelines you should have a fairly 
secure CMS. Self-signed certificates with client certificate 
authentication go a long way making sure the admin area is protected. 
And there are even Yubikey plugins for both projects which add 
multifactor authentication with one-time passwords.

Regards,
Patrick



From peter.olsson at visionutveckling.se  Tue Nov 13 13:46:43 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 13 Nov 2012 10:46:43 +0000
Subject: [Freeswitch-users] FS server
Message-ID: <1FFF97C269757C458224B7C895F35F151BE3BE@cantor.std.visionutv.se>

Yes, the more CPU's/cores, the better...

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
Skickat: den 13 november 2012 10:17
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] FS server

Hi,

we are buying new server for FS. Does FS knows to work with more than one CPU?

BR,
Miha

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

 

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

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

!DSPAM:50a214aa32762023633067!



From a.venugopan at mundio.com  Tue Nov 13 14:21:41 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Tue, 13 Nov 2012 11:21:41 +0000
Subject: [Freeswitch-users] Lua script
Message-ID: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>

Hi,
I have lua script in freeswitch, I tried putting print statements there to get that in freeswitch logs. But its not appearing in logs. Would someone please suggest me how to print any string to freeswitch logs.

print("test")

Thanks.

Regards,
Archana

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From steveayre at gmail.com  Tue Nov 13 15:15:20 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 13 Nov 2012 12:15:20 +0000
Subject: [Freeswitch-users] FS server
In-Reply-To: <50A20FF3.3070404@softnet.si>
References: <50A20FF3.3070404@softnet.si>
Message-ID: <CAFiqYun2cCkKpM=aJoNgoK+G+SQdynVSSM8pCnOK_=TdUw_JbQ@mail.gmail.com>

Yes


On 13 November 2012 09:16, Miha <miha at softnet.si> wrote:

> Hi,
>
> we are buying new server for FS. Does FS knows to work with more than
> one CPU?
>
> BR,
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bfmtl at hotmail.com  Tue Nov 13 15:48:23 2012
From: bfmtl at hotmail.com (BF)
Date: Tue, 13 Nov 2012 07:48:23 -0500
Subject: [Freeswitch-users] Freeswitch and Centos 6
Message-ID: <BLU0-SMTP286E22BBD85851441DD3F04BE6C0@phx.gbl>

Hello,

I've read that Centos 6 had CPU spikes and other issues, including crashes with Freeswitch. Is it still the case with Centos 6.3 with latest Freeswitch release? I would like to upgrade from Centos 5.8. to 6.3.

Thank you

Bernard 

From pen.sokha at gmail.com  Tue Nov 13 12:18:28 2012
From: pen.sokha at gmail.com (sokha pen)
Date: Tue, 13 Nov 2012 16:18:28 +0700
Subject: [Freeswitch-users] failed to compile Mod_gsmopen on windows
Message-ID: <CAD33_DBGJSEoZFd_sU_YCwJPpFF527KmqvUwigwRArM+9D-3-g@mail.gmail.com>

Dear All,

I follow instruction in wiki

http://wiki.freeswitch.org/wiki/Mod_gsmopen

I am using VS2010.

however when i compiled mod_gsmopen still fail....


below is log:
1>------ Skipped Build: Project: gsmlib, Configuration: Release Win32 ------
1>Project not selected to build for this solution configuration
2>------ Build started: Project: mod_gsmopen, Configuration: Release Win32
------
2>     Creating library D:\FS_GIT\Win32\Release\mod\mod_gsmopen.lib and
object D:\FS_GIT\Win32\Release\mod\mod_gsmopen.exp
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_queue_indication at 8
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_perform_mark_ring_ready_value at 20
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_mutex_lock at 4
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_get_channel at 4
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_sleep at 8
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol "public:
int __thiscall ctb::SerialPort_x::Open(char const *,int,char const *,enum
ctb::SerialPort_x::FlowControl)" (?Open at SerialPort_x@ctb@
@QAEHPBDH0W4FlowControl at 12@@Z)
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol "public:
__thiscall ctb::SerialPort::SerialPort(void)" (??0SerialPort at ctb@@QAE at XZ)
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_rwunlock at 4
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_log_printf
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_perform_locate at 16
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_perform_hangup at 20
2>gsmopen_protocol.obj : error LNK2001: unresolved external symbol
__imp__switch_mutex_unlock at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_request_uuid at 20
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_xml_attr_soft at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_codec_destroy at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_fire_detailed at 20
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_xml_attr at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_set_caller_profile at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_time_now at 0
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__dtmf_rx_parms at 20
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__dtmf_rx at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_codec_init_with_bitrate at 40
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_caller_profile_new at 48
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_queue_dtmf at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_test_ready at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_separate_string at 16
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_file_write at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_is_number at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_timer_next at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_event_set_data at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_timer_sync at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_add_header_string at 16
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_thread_create at 20
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_set_write_codec at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_thread_launch at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_get_name at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_threadattr_stacksize_set at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_set_private at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_get_private at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_get_pool at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_add_body
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_xml_free at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__dtmf_rx_init at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_get_uuid at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_swap_linear at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_test_flag at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_threadattr_create at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_loadable_module_create_module_interface at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_snprintf
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_xml_child at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_reserve_subclass_detailed at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_loadable_module_create_interface at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_caller_profile_clone at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_thread_join at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_set_name at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_get_header_idx at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_direction at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_add_stream at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_xml_open_cfg at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_set_read_codec at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_default_dtmf_duration at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_chat_send at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_perform_mark_answered at 16
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_perform_mark_pre_answered at 16
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_timer_init at 20
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_destroy at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__dtmf_rx_get at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_copy_string at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_free_subclass_detailed at 8
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_get_body at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_timer_destroy at 4
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_add_header
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_event_create_subclass_detailed at 24
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_mutex_init at 12
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_channel_perform_set_state at 20
2>mod_gsmopen.obj : error LNK2001: unresolved external symbol
__imp__switch_core_session_perform_destroy at 16
2>D:\FS_GIT\Win32\Release\mod\mod_gsmopen.dll : fatal error LNK1120: 76
unresolved externals
========== Build: 0 succeeded, 1 failed, 0 up-to-date, 1 skipped ==========



Regards,
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From Chris.Martineau at semafone.com  Tue Nov 13 14:28:09 2012
From: Chris.Martineau at semafone.com (Chris Martineau)
Date: Tue, 13 Nov 2012 11:28:09 +0000
Subject: [Freeswitch-users] Event Channel Bridge not firing
Message-ID: <870204F45EE7D34E8D27CC0E602E11A1B4648D@EX01.semafone.local>

Hi,

I am writing a simple interface to read bridge and unbridge events via the event socket.

All was going well until I noticed that a lot of the events are missing. Out of 10 I will lose 30%.

Increased tcp buffers, increased read requests but with no effect.

Looked at wireshark and the events are not being sent?

Made a simple freeswitch module to read and report on events and get exactly the same in the module.

I attach the output from freeswitch  so you can see that even though the channel states trigger the bridge and sometimes the unbridge events do not (or at least they are not reaching my module or the esl module)?

Any idea why this may be the case?

Freeswitch version 1.3.0+git 20121023T030122Z~8589e031d0 (git 8589e03 2012-10-23 03:01:22Z)

(Ignore the ERR level, output from the module at this level so it was easy to see just these logs)

2012-11-13 10:50:03.564141 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:03.584210 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2049 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:03.644145 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.563188 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.623215 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:05.563322 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=3-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:06.563430 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=4-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:07.563046 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:07.583102 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=5-2049 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=5-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:08.563249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:08.603325 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=6-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=7-2049 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=7-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:10.563401 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:11.563509 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=9-2049 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=9-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:12.563113 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:50:12.583790 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=10-2049 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=10-2049 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:35.043882 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2061 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2061 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2062 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2062 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130 State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:53:01.383362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2063 at 192.168.86.130 State=CS_HIBERNATE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2063 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130 State=CS_DESTROY
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_DESTROY


NO BRIDGE OR UNBRIDGE EVENTS REPORTED FOR THIS CALL?
freeswitch at ubuntu> 2012-11-13 10:53:16.243477 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130 State=CS_ROUTING
2012-11-13 10:53:16.266281 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130 State=CS_EXECUTE
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130 State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_DESTROY
2012-11-13 10:53:16.303504 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130 State=CS_DESTROY

Many thanks

Chris


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From ssinyagin at yahoo.com  Tue Nov 13 16:19:30 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Tue, 13 Nov 2012 05:19:30 -0800 (PST)
Subject: [Freeswitch-users] FS server
In-Reply-To: <50A20FF3.3070404@softnet.si>
References: <50A20FF3.3070404@softnet.si>
Message-ID: <1352812770.6594.YahooMailNeo@web39304.mail.mud.yahoo.com>

Miha,
I would propose to invest in buying and reading the FreeSWITCH book, and in gaining some practical experience before buying the hardware.

answering your question, yes, FreeSWITCH is a highly scalable server which spreads the workload among as many cores as you have.







>________________________________
> From: Miha <miha at softnet.si>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Tuesday, November 13, 2012 10:16 AM
>Subject: [Freeswitch-users] FS server
> 
>Hi,
>
>we are buying new server for FS. Does FS knows to work with more than 
>one CPU?
>
>BR,
>Miha
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
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From shaheryarkh at googlemail.com  Tue Nov 13 17:24:02 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Tue, 13 Nov 2012 15:24:02 +0100
Subject: [Freeswitch-users] Lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
Message-ID: <CAFZQphwYdch+KVH=T9-vMYZyicBVZ=GiTXR+KeS8N+tOqENzeA@mail.gmail.com>

use freeswitch.consoleLog method, e.g.

freeswitch.consoleLog("ERR","Result of system call: " .. res .. "\n");

Thank you.


On Tue, Nov 13, 2012 at 12:21 PM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> I have lua script in freeswitch, I tried putting print statements there to
> get that in freeswitch logs. But its not appearing in logs. Would someone
> please suggest me how to print any string to freeswitch logs.****
>
> ** **
>
> print(?test?)****
>
> ** **
>
> Thanks.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From shaheryarkh at googlemail.com  Tue Nov 13 17:26:05 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Tue, 13 Nov 2012 15:26:05 +0100
Subject: [Freeswitch-users] FS server
In-Reply-To: <CAFiqYun2cCkKpM=aJoNgoK+G+SQdynVSSM8pCnOK_=TdUw_JbQ@mail.gmail.com>
References: <50A20FF3.3070404@softnet.si>
	<CAFiqYun2cCkKpM=aJoNgoK+G+SQdynVSSM8pCnOK_=TdUw_JbQ@mail.gmail.com>
Message-ID: <CAFZQphwv9gLbG_GeRooLW2okXUwtE+bEJYkmgh7JU7HUdEkeTw@mail.gmail.com>

One of my production server has 16 cores and FS works flawlessly. :-)

Thank you.


On Tue, Nov 13, 2012 at 1:15 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Yes
>
>
> On 13 November 2012 09:16, Miha <miha at softnet.si> wrote:
>
>> Hi,
>>
>> we are buying new server for FS. Does FS knows to work with more than
>> one CPU?
>>
>> BR,
>> Miha
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From evgeniy at bestnet.kharkov.ua  Tue Nov 13 17:27:13 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Tue, 13 Nov 2012 16:27:13 +0200
Subject: [Freeswitch-users] Lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
Message-ID: <50A258C1.8010003@bestnet.kharkov.ua>

freeswitch.consoleLog("INFO","Hello!\n");

13.11.2012 13:21, Archana Venugopan ?????:
> Hi,
> I have lua script in freeswitch, I tried putting print statements there to get that in freeswitch logs. But its not appearing in logs. Would someone please suggest me how to print any string to freeswitch logs.
>
> print("test")
>
> Thanks.
>
> Regards,
> Archana
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From anton.jugatsu at gmail.com  Tue Nov 13 17:47:25 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 13 Nov 2012 18:47:25 +0400
Subject: [Freeswitch-users] mod_opus codec name.
In-Reply-To: <CAKbxfG82Caiuz2noBR87X+=ayNdq3iMOcjzv3PoXViqcGT-j_Q@mail.gmail.com>
References: <001801cdbf6d$1e24ef90$5a6eceb0$@207me.com>
	<CANBhyPBCXaw8Q=WTKwToMPXQLo6PNCMZfkR10_Fh_V6w1hC4ow@mail.gmail.com>
	<CANBhyPDhHcc1_Gw+Q3BQS=mL0nUv0_-U2ACbg=WyfK+4_ymGKw@mail.gmail.com>
	<015e01cdc0c8$b402ffa0$1c08fee0$@207me.com>
	<87689103-2F3F-4527-90C9-0FC41F51EBF0@carmickle.com>
	<CAKbxfG82Caiuz2noBR87X+=ayNdq3iMOcjzv3PoXViqcGT-j_Q@mail.gmail.com>
Message-ID: <CANBhyPBkOCWTpvpxGWJEhRvpaDb7pARHEqLC_cx_U1=qJfXaDA@mail.gmail.com>

Can you paste output from

sofia status profile external


2012/11/12 Anthony Minessale <anthony.minessale at gmail.com>

> Incorrect.  If you have a version of the module from before the codec was
> released officially it would be opus-0.9.0.  That is now discontinued so
> you would need to update to latest and use "opus"
>
>
>
> On Mon, Nov 12, 2012 at 9:29 AM, Frank Carmickle <frank at carmickle.com>wrote:
>
>>
>> On Nov 12, 2012, at 6:27 AM, Stephen Dame <sdame at 207me.com> wrote:
>>
>> > Anton,  I have reloaded and restarted FS. And made sure mod_opus is
>> loaded.
>> >
>> > I still see no documentation in forums or wiki, as to what name should
>> be used for opus in global codec preferences.
>> >
>> > I get 488,  which looks like  freeswitch is never looking to compare it
>> against the preference list.
>>
>> It used to be Opus-0.9.0.  I believe it is now Opus-1.0.1.
>>
>> HTH
>> --FC
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Tue Nov 13 17:53:24 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 13 Nov 2012 06:53:24 -0800 (PST)
Subject: [Freeswitch-users] failed to compile Mod_gsmopen on windows
In-Reply-To: <CAD33_DBGJSEoZFd_sU_YCwJPpFF527KmqvUwigwRArM+9D-3-g@mail.gmail.com>
References: <CAD33_DBGJSEoZFd_sU_YCwJPpFF527KmqvUwigwRArM+9D-3-g@mail.gmail.com>
Message-ID: <1352818404721-7584542.post@n2.nabble.com>

Update to git head, there was a missing reference.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/failed-to-compile-Mod-gsmopen-on-windows-tp7584539p7584542.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mike at jerris.com  Tue Nov 13 17:56:26 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Nov 2012 09:56:26 -0500
Subject: [Freeswitch-users] Freeswitch and Centos 6
In-Reply-To: <BLU0-SMTP286E22BBD85851441DD3F04BE6C0@phx.gbl>
References: <BLU0-SMTP286E22BBD85851441DD3F04BE6C0@phx.gbl>
Message-ID: <01F4E2AD-570F-46D7-8BC6-20DD5F5FE875@jerris.com>

We think these issues are all resolved in the latest centos 6.

On Nov 13, 2012, at 7:48 AM, BF <bfmtl at hotmail.com> wrote:

> Hello,
> 
> I've read that Centos 6 had CPU spikes and other issues, including crashes with Freeswitch. Is it still the case with Centos 6.3 with latest Freeswitch release? I would like to upgrade from Centos 5.8. to 6.3.
> 
> Thank you
> 
> Bernard 



From peter.olsson at visionutveckling.se  Tue Nov 13 18:10:14 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 13 Nov 2012 15:10:14 +0000
Subject: [Freeswitch-users] Event Channel Bridge not firing
Message-ID: <1FFF97C269757C458224B7C895F35F151BEAD2@cantor.std.visionutv.se>

It's impossible to see anything without the full FS logs. There is nothing in these logs that even say that you were trying to bridge a call or not.

Please supply the full debug logs, and pastebin them.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau
Skickat: den 13 november 2012 12:28
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Event Channel Bridge not firing

Hi,

I am writing a simple interface to read bridge and unbridge events via the event socket.

All was going well until I noticed that a lot of the events are missing. Out of 10 I will lose 30%.

Increased tcp buffers, increased read requests but with no effect.

Looked at wireshark and the events are not being sent?

Made a simple freeswitch module to read and report on events and get exactly the same in the module.

I attach the output from freeswitch  so you can see that even though the channel states trigger the bridge and sometimes the unbridge events do not (or at least they are not reaching my module or the esl module)?

Any idea why this may be the case?

Freeswitch version 1.3.0+git 20121023T030122Z~8589e031d0 (git 8589e03 2012-10-23 03:01:22Z)

(Ignore the ERR level, output from the module at this level so it was easy to see just these logs)

2012-11-13 10:50:03.564141 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:03.584210 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:03.644145 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.563188 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.623215 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:05.563322 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:06.563430 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:07.563046 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:07.583102 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:08.563249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:08.603325 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:10.563401 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:11.563509 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:12.563113 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:12.583790 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:35.043882 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:53:01.383362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_DESTROY


NO BRIDGE OR UNBRIDGE EVENTS REPORTED FOR THIS CALL?
freeswitch at ubuntu> 2012-11-13 10:53:16.243477 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:53:16.266281 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_DESTROY
2012-11-13 10:53:16.303504 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_DESTROY

Many thanks

Chris


!DSPAM:50a259a132764491816697!
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From jeff at jefflenk.com  Tue Nov 13 17:55:21 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 13 Nov 2012 06:55:21 -0800 (PST)
Subject: [Freeswitch-users] E-mail voicemails - msmtp in Windows
In-Reply-To: <CAG0Fp+_kQf1pZyZpCDWuqAVMpYsCfA3p1gUyrSzWK24VNi9Yjg@mail.gmail.com>
References: <CAG0Fp+_kQf1pZyZpCDWuqAVMpYsCfA3p1gUyrSzWK24VNi9Yjg@mail.gmail.com>
Message-ID: <1352818521085-7584543.post@n2.nabble.com>

Please report this to Jira at jira.freeswitch.org



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/E-mail-voicemails-msmtp-in-Windows-tp7584525p7584543.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue Nov 13 18:57:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Nov 2012 07:57:27 -0800
Subject: [Freeswitch-users] Check user state in dialplan
In-Reply-To: <CANM4Ruj24EKsu3ij6g-D=iacwLGzK_w+AuNKNMqWD0cM4k0aXA@mail.gmail.com>
References: <CANM4RuigJ6CRDgZSYBzH3nrMYUGAkO-fT67V6U48ng+Cg2W_6A@mail.gmail.com>
	<CAKzWOxUm0NgmC+9VD0qZy9i5T8xYk1ZQCDHEFgAg2egQZMMADw@mail.gmail.com>
	<CANM4Ruj24EKsu3ij6g-D=iacwLGzK_w+AuNKNMqWD0cM4k0aXA@mail.gmail.com>
Message-ID: <CAKzWOxVTBBDeCSn3vDxOsCriBwYXQGihuXrq6r1B03k2cjZD0w@mail.gmail.com>

Which case(s) are not handled by Local_Extension, other than user not in
the directory?

Also, if you use the "user/xxxx" dialstring format you'll get different
errors depending on whether or not the user exists in the database:

freeswitch at default> originate user/1000 9664
-ERR USER_NOT_REGISTERED

freeswitch at default> originate user/1900 9664
-ERR SUBSCRIBER_ABSENT

User 1000 exists in the database but is not registered. User 1900 does not
exist at all. In the case of a bridge app you can use chan var
continue_on_fail=true and then check bridge_hangup_cause to determine what
happened on the bridge attempt. I'm pretty sure you can then handle all of
your scenarios.

-MC

On Mon, Nov 12, 2012 at 8:09 PM, Eugene Prokopiev <enp at itx.ru> wrote:

> Hi,
>
> With "Local_Extension" I can't distinguish all cases, I can distinguish
> only successful or failed answer.
>
> About first case: I have no sequential numbers for local users and can't
> describe range as regex. So, the best way to identify if local user exists
> is to search id (and number-alias) in directory. I know about
> mod_xml_curl which can be used for filling dialplan and directory from some
> database, but I tried to find more simple way.
>
> --
> Regards,
> Eugene Prokopiev
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From chad at apartmentlines.com  Tue Nov 13 19:15:26 2012
From: chad at apartmentlines.com (Chad Phillips)
Date: Tue, 13 Nov 2012 08:15:26 -0800
Subject: [Freeswitch-users] Lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
Message-ID: <21133B9558044308A2F7A6B21A8E2168@gmail.com>

i use something like this:  

function debug_print(info)
    freeswitch.consoleLog("info", info .. "\n")
end



On Tuesday, November 13, 2012 at 3:21 AM, Archana Venugopan wrote:

> Hi,
> I have lua script in freeswitch, I tried putting print statements there to get that in freeswitch logs. But its not appearing in logs. Would someone please suggest me how to print any string to freeswitch logs.
>   
> print(?test?)
>   
> Thanks.
>   
> Regards,
> Archana
>   
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-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
>  
>  


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From a.venugopan at mundio.com  Tue Nov 13 20:21:31 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Tue, 13 Nov 2012 17:21:31 +0000
Subject: [Freeswitch-users] Lua script
In-Reply-To: <CAFZQphwYdch+KVH=T9-vMYZyicBVZ=GiTXR+KeS8N+tOqENzeA@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
	<CAFZQphwYdch+KVH=T9-vMYZyicBVZ=GiTXR+KeS8N+tOqENzeA@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2329621@Mail-Kilo.squay.com>

Thanks. But I want to put print statements within lua script, just to make sure if the script passes that line. In that case too consolelog we need to use?

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad
Sent: 13 November 2012 14:24
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua script

use freeswitch.consoleLog method, e.g.

freeswitch.consoleLog("ERR","Result of system call: " .. res .. "\n");

Thank you.

On Tue, Nov 13, 2012 at 12:21 PM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
I have lua script in freeswitch, I tried putting print statements there to get that in freeswitch logs. But its not appearing in logs. Would someone please suggest me how to print any string to freeswitch logs.

print(?test?)

Thanks.

Regards,
Archana


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




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

FreeSWITCH-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



--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
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From babak.freeswitch at gmail.com  Tue Nov 13 20:23:51 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Tue, 13 Nov 2012 20:53:51 +0330
Subject: [Freeswitch-users] tx fax: call drops prematurely
In-Reply-To: <CAFVUaV-jFrzNHrDK824F4bVK6QnE6CQUTBOq1HBTd-k6jN-eqw@mail.gmail.com>
References: <CAEVfcaZcYHV+e0jJ8cKHWyvG41-b7F9_3M4Ho=zSFwTg-0jxaA@mail.gmail.com>
	<CALn233pFO_pvKzJ+=yi=mhPTS2BdRhjAPBOOycSKCkLcf5XyZQ@mail.gmail.com>
	<CAFVUaV-jFrzNHrDK824F4bVK6QnE6CQUTBOq1HBTd-k6jN-eqw@mail.gmail.com>
Message-ID: <CAEVfcaYJ5x7daR3XZJ1zyTW63qiTbaaDtSeE-yXiu8s3QjCXBQ@mail.gmail.com>

Hi Yiftach
can you share your code please?

On Mon, Oct 29, 2012 at 8:44 PM, Yiftach Golan <yiftah at choochee.com> wrote:

> Hi Babak,
> we have a lot of experience with Faxes and FreeSWITCH
> We had to add some changes to the both the T.30 and the T.38 code in order
> for the fax to be stable
> I can share those changes with the community, most of the changes were
> making larger timeouts and transitions between states mostly around the DCN
> )
> However, we have a pretty old version of FreeSWITCH I know that the fax
> code has changed a lot since then
>
> Thanks,
> Yiftach.
>
> On Sun, Oct 28, 2012 at 1:05 PM, Mitch Capper <mitch.capper at gmail.com>wrote:
>
>> Hi Babak,
>> Check the wiki for the suggested methods to try for faxing also
>> provider can make a big difference.  Let us know what you have tried
>> /are doing.
>> ~mitch
>>
>> On Sun, Oct 28, 2012 at 11:45 AM, Babak Yakhchali
>> <babak.freeswitch at gmail.com> wrote:
>> > Hi
>> > Sending faxes results in dropped calls: "The call dropped prematurely"
>> > http://pastebin.freeswitch.org/20133
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Tue Nov 13 21:04:43 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 13 Nov 2012 10:04:43 -0800
Subject: [Freeswitch-users] FS server
In-Reply-To: <1352812770.6594.YahooMailNeo@web39304.mail.mud.yahoo.com>
References: <50A20FF3.3070404@softnet.si>
	<1352812770.6594.YahooMailNeo@web39304.mail.mud.yahoo.com>
Message-ID: <CAMWiMy9O5nahQMwjqF_QPkaM87h+2HxZ6Hc572buEUFkcZ0iSA@mail.gmail.com>

Quick answer to that: FreeSWITCH is highly multithreaded ;)

On Tue, Nov 13, 2012 at 5:19 AM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> Miha,
> I would propose to invest in buying and reading the FreeSWITCH book, and in
> gaining some practical experience before buying the hardware.
>
> answering your question, yes, FreeSWITCH is a highly scalable server which
> spreads the workload among as many cores as you have.
>
>
>
>
> ________________________________
> From: Miha <miha at softnet.si>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, November 13, 2012 10:16 AM
> Subject: [Freeswitch-users] FS server
>
> Hi,
>
> we are buying new server for FS. Does FS knows to work with more than
> one CPU?
>
> BR,
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From tgraziano at myitdepartment.net  Tue Nov 13 21:36:17 2012
From: tgraziano at myitdepartment.net (myITdepartment)
Date: 13 Nov 2012 12:36:17 -0600
Subject: [Freeswitch-users] Ticket created:  FS server (ticket #556115)
Message-ID: <WWW39rf3yXvEbjkyqU7000f97ad@smtp.estreamdesk.com>

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From lists at kavun.ch  Tue Nov 13 22:23:05 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 13 Nov 2012 20:23:05 +0100
Subject: [Freeswitch-users] Domains and profiles
In-Reply-To: <CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>
References: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
	<1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>
	<CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>
Message-ID: <3FC39EED-0A60-4A87-866D-0AAA629F9560@kavun.ch>

This is precious. I had figured out how the domain portion affects FS, I just didn't know how to declare my domains to my SIP profiles. Which I believe I now know and will experiment a bit.

Thanks!
On Nov 12, 2012, at 8:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> The best thing to do is take a look at these things from a step back.
> 
> The domains inside the xml registry are completely different from the domains on the internet and again completely different from domains in sip packets.  The profiles are again entirely different from any of the above.  Its up to you to align them if you so choose.
> 
> 
> The default configuration distributed with FreeSWITCH sets up the scenario most likely to load on any machine and work out of the box.  That is the primary goal of that configuration, so, It sets the domain in both the directory, the global default domain variable and the name of the internal profile to be identical to the ip on the box that can reach the internet.  Then it sets the sip to force everything to that value.  When you want to detach from this behavior, you are probably on a venture to do some kind of multi-home setup.
> 
> 
> Aliases in the <aliases> tag are a list of keys you want to use to use that lead to the current profile your are configuring.  Think of it as the /etc/hosts file in unix only for profiles.   When you define aliases to match all of the possible domains hosted on a particular profile, then when you try to take a user at host.com notation and decide which profile it came from, you can use the aliases to find it providing you have added <alias name="host.com"/> to that profile.
> 
> The <domains> tag is an indicator telling the profile to open the xml registry in FreeSWITCH and run through any domains defined therein.
> The 2 key attributes are:
> 
> alias: [true/false]  (automatically create an alias for this domain as mentioned above)
> parse: [true/false] (scan the domain for gateway entries and include them into this profile)
> name: [<string>] (either the name of a specific domain or 'all' to denote parsing every domain in the directory)
> 
> As you showed in your question the default config has
> 
> <domain name="all" alias="true" parse="false"/>
> 
> If you apply what you have learned above, it will scan for every domain (there is only one by default) and add an alias for it and not parse it for gateways.  The default directory uses global config vars to set the domain to match the local ip on the box.  So now you will have a domain in your config that is your ip, and the internal profile will attach to it and add an alias so that value expands to match it.
> 
> 
> This is explained in a comment at the top of directory/default.xml
> 
>     FreeSWITCH works off the concept of users and domains just like email.                                                                                      
>     You have users that are in domains for example 1000 at domain.com.                                                                                             
>                                                                                                                                                                 
>     When freeswitch gets a register packet it looks for the user in the directory                                                                               
>     based on the from or to domain in the packet depending on how your sofia profile                                                                            
>     is configured.  Out of the box the default domain will be the IP address of the                                                                             
>     machine running FreeSWITCH.  This IP can be found by typing "sofia status" at the                                                                           
>     CLI.  You will register your phones to the IP and not the hostname by default.                                                                              
>     If you wish to register using the domain please open vars.xml in the root conf                                                                              
>     directory and set the default domain to the hostname you desire.  Then you would                                                                            
>     use the domain name in the client instead of the IP address to register                                                                                     
>     with FreeSWITCH.                                                                    
> 
> 
> 
> So having more than one profile with the default of 
> 
> <domain name="all" alias="true" parse="false"/>
> 
> is going to end up aliasing the same domains into all profiles who call it and cause an overwrite in the lookup table and probably an error in your logs somewhere.  If you had parse="true" on all of them, they would all try and register to the gateways in all of your domains.
> 
> 
> If you look at the stock config, external.xml is a good example of a secondary profile, it has
> 
> <domain name="all" alias="false" parse="true"/>
> 
> so no aliases, and yes parse ... the exact opposite of the internal so that all the gateways would register from external and internal would bind to the local ip.
> 
> So, you probably want to use separate <domain name="my.com"/> per domain per profile you want to bind it to in more complicated setups.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>  
> 
> 
> 
> 
> 
> 
> On Sun, Nov 11, 2012 at 9:09 PM, Emrah <lists at kavun.ch> wrote:
> Bless you!
> 
> Thanks for putting this together. You've beautifully summed up all my questions.
> On Nov 11, 2012, at 8:09 AM, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
> 
> > Hi Folks,
> > I've started a new thread as it's not quite the same issue, and domains & profiles have confused the heck out of me every time I have developed a new setup for fS.
> > I have sometimes had to hack/hard-doce the dialstring to make multiple domains in one profile work, had hours of fun with presence, db and force register settings, and have still had some odd gotchas that have required extensive meditation.
> > [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract these elements ]
> >
> > Coming at this from standards/specs and rolling my own SIP stacks, sofia/fS seems to use the term "domain" differently from sipdomain, and alias seems to be tied to the directory (and thus to the profile listed in a directory file), but I'm not sure.
> > so ...
> > Before I capture to the sofia conf xml wiki page, I have a couple of questions on the sip-profile XML setup;
> >
> > Q: Is there a particular reason why there's a parameter called alias and an (entirely different) setting also called alias?
> > The sofia conf xml wiki's comment on the setting "alias" shows I'm not alone.
> > I agree that's what it appears to be doing, but can we nail this down please (and what happens if an external client uses this connection to register and call)?
> >
> > In the current sofia conf xml wiki page, the domain setting is not exactly well documented :).
> > The current internal.xml vanilla example from git (as of time of writing) has the following lines:
> > -------------------------
> > ...
> > <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> > <!--aliases are other names that will work as a valid profile name for this profile-->
> > <aliases>
> >   <!--
> >        <alias name="default"/>
> >    -->
> > </aliases>
> > <!-- Outbound Registrations -->
> > <gateways>
> >   <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
> > </gateways>
> >
> > <domains>
> >   <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
> >   <!--<domain name="$${domain}" parse="true"/>-->
> >   <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
> >   <!--<domain name="all" alias="true" parse="true"/>-->
> >   <domain name="all" alias="true" parse="false"/>
> > </domains>
> > ...
> > -------------------------
> >
> > This stuff is entirely missing from the sofia.conf.xml wiki page, and it IS really important.
> >
> >
> > Q: what's the default value for the alias parameter in the domain element? -- it is missing from the first example.
> > Q: if there is more than one profile, what's the impact of setting parse = true in one (or all) of the profiles' XML files?
> >   (or parse = false, or missing the parameter altogether)?
> > AFAICT, the gateways get pulled in via the pre-process directive just fine, regardless of the value of the parse parameter -- it works for me, at least.
> >
> > Q: if there is more than one profile, what's the impact of putting domain name="all" into one (or all) of the profiles' XML files?
> >
> > Ideally, having more than one sipdomain tied to one profile "would be good", but aliases doesn't do that -- as the git file says, these are aliases for the profile name.
> >
> > Before I start scribbling, Answers on a postcard to this ML, please.
> >
> > all the best,
> >  Lawrence
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Hector.Geraldino at ipsoft.com  Tue Nov 13 23:30:03 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Tue, 13 Nov 2012 20:30:03 +0000
Subject: [Freeswitch-users] Event Channel Bridge not firing
In-Reply-To: <1FFF97C269757C458224B7C895F35F151BEAD2@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151BEAD2@cantor.std.visionutv.se>
Message-ID: <ADF32A4A0F985A428C85B277B8E1C9D80F47987E@NY1-EXMB-02.ip-soft.net>

Also it's always a good idea to subscribe only to the events you need. Instead of sending an "event plain all" command you should register only to the events you're interested in, like "event plain CHANNEL_BRIDGE CHANNEL_UNBRIDGE".

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: Tuesday, November 13, 2012 10:10 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Event Channel Bridge not firing

It's impossible to see anything without the full FS logs. There is nothing in these logs that even say that you were trying to bridge a call or not.

Please supply the full debug logs, and pastebin them.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau
Skickat: den 13 november 2012 12:28
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Event Channel Bridge not firing

Hi,

I am writing a simple interface to read bridge and unbridge events via the event socket.

All was going well until I noticed that a lot of the events are missing. Out of 10 I will lose 30%.

Increased tcp buffers, increased read requests but with no effect.

Looked at wireshark and the events are not being sent?

Made a simple freeswitch module to read and report on events and get exactly the same in the module.

I attach the output from freeswitch  so you can see that even though the channel states trigger the bridge and sometimes the unbridge events do not (or at least they are not reaching my module or the esl module)?

Any idea why this may be the case?

Freeswitch version 1.3.0+git 20121023T030122Z~8589e031d0 (git 8589e03 2012-10-23 03:01:22Z)

(Ignore the ERR level, output from the module at this level so it was easy to see just these logs)

2012-11-13 10:50:03.564141 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:03.584210 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:03.644145 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.563188 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.623215 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:05.563322 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:06.563430 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:07.563046 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:07.583102 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:08.563249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:08.603325 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:10.563401 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:11.563509 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:12.563113 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:12.583790 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:35.043882 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:53:01.383362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_DESTROY


NO BRIDGE OR UNBRIDGE EVENTS REPORTED FOR THIS CALL?
freeswitch at ubuntu> 2012-11-13 10:53:16.243477 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:53:16.266281 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_DESTROY
2012-11-13 10:53:16.303504 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_DESTROY

Many thanks

Chris


!DSPAM:50a259a132764491816697!
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From chad at apartmentlines.com  Wed Nov 14 01:30:54 2012
From: chad at apartmentlines.com (Chad Phillips)
Date: Tue, 13 Nov 2012 14:30:54 -0800
Subject: [Freeswitch-users] Lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2329621@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
	<CAFZQphwYdch+KVH=T9-vMYZyicBVZ=GiTXR+KeS8N+tOqENzeA@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2329621@Mail-Kilo.squay.com>
Message-ID: <4F13DF04E5FB460CB59F5651D4B54082@gmail.com>

When you're invoking Lua via FreeSWITCH, there's nowhere to 'print' other than to the console -- so, yes, consoleLog() is the easy way to go.  

You could also log to a file if you preferred, Lua documentation can help you there.

Chad  


On Tuesday, November 13, 2012 at 9:21 AM, Archana Venugopan wrote:

> Thanks. But I want to put print statements within lua script, just to make sure if the script passes that line. In that case too consolelog we need to use?
>   
> Regards,
> Archana
>   
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad
> Sent: 13 November 2012 14:24
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Lua script  
>   
> use freeswitch.consoleLog method, e.g.
>   
>  
> freeswitch.consoleLog("ERR","Result of system call: " .. res .. "\n");
>  
>   
>  
> Thank you.
>  
>   
> On Tue, Nov 13, 2012 at 12:21 PM, Archana Venugopan <a.venugopan at mundio.com (mailto:a.venugopan at mundio.com)> wrote:
> Hi,
> I have lua script in freeswitch, I tried putting print statements there to get that in freeswitch logs. But its not appearing in logs. Would someone please suggest me how to print any string to freeswitch logs.
>   
> print(?test?)
>   
> Thanks.
>   
> Regards,
> Archana
>   
>  
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-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  
>  
>  
>    
> --  
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com (mailto:shari_786pk at hotmail.com)
> Email: shaheryarkh at googlemail.com (mailto:shaheryarkh at googlemail.com)  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-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
>  
>  


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From josego85 at gmail.com  Tue Nov 13 21:18:28 2012
From: josego85 at gmail.com (Jose Alberto Gonzalez von Schmeling)
Date: Tue, 13 Nov 2012 15:18:28 -0300
Subject: [Freeswitch-users] raise the call to another extension
Message-ID: <CAAKtwi=siQ-A+us5wuq_x1yAV0VyKzOf0ix8mLdj6i6P7ZM+Sg@mail.gmail.com>

Hello:
I have an outside call. What I want is to raise the call to another
extension. How is that done?


Thanks, Jose

-- 
Podes encontrarme o comunicarte conmigo en:

   - *Mi blog*: http://proyectosbeta.net/<http://proyectosbeta.blogspot.com/>
   - *Facebook*:
   http://www.facebook.com/pages/Proyectos-Beta/113277785412256<http://www.facebook.com/jose.gonzalezvonschmeling>
   - *Twitter*: @proyectosbeta <http://twitter.com/proyectosbeta>
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From msc at freeswitch.org  Wed Nov 14 01:37:51 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Nov 2012 14:37:51 -0800
Subject: [Freeswitch-users] Lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2329621@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
	<CAFZQphwYdch+KVH=T9-vMYZyicBVZ=GiTXR+KeS8N+tOqENzeA@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2329621@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxVgeHWqhQBYHo+skJDYsRsRupj0RiiEMhYh_f-VqHXk-g@mail.gmail.com>

Yes, use consoleLog for that as well and just watch the FreeSWITCH CLI when
the script is running. I use WARNING and ERR debug levels to make the print
stand out in purple and red lettering, respectively.

-MC

On Tue, Nov 13, 2012 at 9:21 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Thanks. But I want to put print statements within lua script, just to
> make sure if the script passes that line. In that case too consolelog we
> need to use?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
> Shahzad
> *Sent:* 13 November 2012 14:24
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Lua script****
>
> ** **
>
> use freeswitch.consoleLog method, e.g.****
>
> ** **
>
> freeswitch.consoleLog("ERR","Result of system call: " .. res .. "\n");****
>
> ** **
>
> Thank you.****
>
> ** **
>
> On Tue, Nov 13, 2012 at 12:21 PM, Archana Venugopan <
> a.venugopan at mundio.com> wrote:****
>
> Hi,****
>
> I have lua script in freeswitch, I tried putting print statements there to
> get that in freeswitch logs. But its not appearing in logs. Would someone
> please suggest me how to print any string to freeswitch logs.****
>
>  ****
>
> print(?test?)****
>
>  ****
>
> Thanks.****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From dujinfang at gmail.com  Wed Nov 14 01:41:04 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 14 Nov 2012 06:41:04 +0800
Subject: [Freeswitch-users] Lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF23295A0@Mail-Kilo.squay.com>
Message-ID: <6EFFA56507314D1094B6A5DE9A71BA91@gmail.com>

try

session:execute("log", "ERR blahblah\n")  

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


On Tuesday, November 13, 2012 at 7:21 PM, Archana Venugopan wrote:

> Hi,
> I have lua script in freeswitch, I tried putting print statements there to get that in freeswitch logs. But its not appearing in logs. Would someone please suggest me how to print any string to freeswitch logs.
>   
> print(?test?)
>   
> Thanks.
>   
> Regards,
> Archana
>   
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-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
>  
>  


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From msc at freeswitch.org  Wed Nov 14 01:44:11 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Nov 2012 14:44:11 -0800
Subject: [Freeswitch-users] Making SRTP mandatory
In-Reply-To: <B3426479DE2E6C4ABFC4C61B5C3DF790736FE4BE09@cwexch.callware.com>
References: <B3426479DE2E6C4ABFC4C61B5C3DF790736FE4BE09@cwexch.callware.com>
Message-ID: <CAKzWOxWry9SgCRf6hZuwLzhgvetwar8KT7fn+93b+EfqL3ZMfg@mail.gmail.com>

FYI,

For posterity's sake we've researched and documented the answer to this
question:
http://wiki.freeswitch.org/wiki/SRTP#Forcing_SAVP_in_the_SDP

Enjoy!

-MC

On Mon, Nov 12, 2012 at 4:55 PM, Clint Goudie-Nice <cgoudie at callware.com>wrote:

>  Greetings all,****
>
> ** **
>
> I have been configuring a FreeSWITCH / FusionPBX system to do TLS and SRTP.
> ****
>
> ** **
>
> When I configure it to do srtp in the outbound route, the SDP produced
> looks like:****
>
> ** **
>
> o=FreeSWITCH 1352745516 1352745517 IN IP4 10.16.0.131****
>
> s=FreeSWITCH****
>
> c=IN IP4 10.16.0.131****
>
> t=0 0****
>
> m=audio 22222 RTP/SAVP 9 0 8 3 101 13****
>
> a=rtpmap:101 telephone-event/8000****
>
> a=fmtp:101 0-16****
>
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:djfKObSYrk0J8pVI1w/OfQBjTYwNgTi80C68VgBB****
>
> a=ptime:20****
>
> m=audio 22222 RTP/AVP 9 0 8 3 101 13****
>
> a=rtpmap:101 telephone-event/8000****
>
> a=fmtp:101 0-16****
>
> a=ptime:20****
>
> ** **
>
> This is great if I want to support both SRTP and non-SRTP calls; but I
> cannot figure out how to remove the AVP block so that SRTP is mandatory.**
> **
>
> ** **
>
> How do I require SRTP?****
>
> ** **
>
> Thanks,****
>
> ** **
>
> Clint****
>
> ------------------------------
> CONFIDENTIALITY NOTICE: This e-mail and any attachments are for the
> exclusive and confidential use of the intended recipient. If you received
> this in error, please do not read, distribute, or take action in reliance
> upon this message. Instead, please notify us immediately by return email
> and promptly delete this message and its attachments from your computer
> system.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From Jacob_Hodges at data3.com.au  Wed Nov 14 07:00:55 2012
From: Jacob_Hodges at data3.com.au (Jacob Hodges)
Date: Wed, 14 Nov 2012 04:00:55 +0000
Subject: [Freeswitch-users] Freeswitch to Lync with TLS
Message-ID: <2F018F4D35BC204EB5043281571CAFA93520B01E@D3TOOEXM01.data3.com.au>

Hi all,
Does anyone have any experience configuration Freeswitch to work with Lync using TLS?

Jacob.

---------------------------------------------------------------------------
Details of Data#3's transmission content disclaimer can be accessed here - 
http://www.data3.com.au/transdisc
---------------------------------------------------------------------------
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From 8f27e956 at gmail.com  Wed Nov 14 07:01:20 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Tue, 13 Nov 2012 23:01:20 -0500
Subject: [Freeswitch-users] Freeswitch and Centos 6
In-Reply-To: <01F4E2AD-570F-46D7-8BC6-20DD5F5FE875@jerris.com>
References: <BLU0-SMTP286E22BBD85851441DD3F04BE6C0@phx.gbl>
	<01F4E2AD-570F-46D7-8BC6-20DD5F5FE875@jerris.com>
Message-ID: <-2321702856529270929@unknownmsgid>

Been running FS 1.2.3 for over a month with zero issues observed on
CentOS 6.3_X86-64_MINIMAL_IOS (WITH its distro's latest O/S YUM
updates applied, plus yum'ed in the freeSWITCH dependencies).  Kernel
clocksource0 = hpet (override acpi_pm default).  FS was git obtained
and built with ./configure --enable-64 --enable-optimization
--enable-zrtp.  Using stock FS sqlite3 and lua scripting.  No
local-machine SQL server.  No p-language.

First built and ran in Windows 7 hosted Virtualbox for two weeks (test
and proof of concept as we migrated off asterisk) and then/now on an
ASUS ATOM D525 Dual-core 1.8 GHz hyper-threading enabled mobo.


?????
iThing:  Big thumbs & little keys. Please excuse typo, spelling and
grammar errors ? Good planets are hard to find ? think before you
print ? My desire to be well-informed is currently at odds with my
desire to remain sane. ? Last night I played a blank CD at full blast.
The Mime next door went nuts.


On 2012-11-13, at 12:19, Michael Jerris <mike at jerris.com> wrote:

> We think these issues are all resolved in the latest centos 6.
>
> On Nov 13, 2012, at 7:48 AM, BF <bfmtl at hotmail.com> wrote:
>
>> Hello,
>>
>> I've read that Centos 6 had CPU spikes and other issues, including crashes with Freeswitch. Is it still the case with Centos 6.3 with latest Freeswitch release? I would like to upgrade from Centos 5.8. to 6.3.
>>
>> Thank you
>>
>> Bernard
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From miha at softnet.si  Wed Nov 14 10:03:00 2012
From: miha at softnet.si (Miha)
Date: Wed, 14 Nov 2012 08:03:00 +0100
Subject: [Freeswitch-users] FS server
In-Reply-To: <CAMWiMy9O5nahQMwjqF_QPkaM87h+2HxZ6Hc572buEUFkcZ0iSA@mail.gmail.com>
References: <50A20FF3.3070404@softnet.si>
	<1352812770.6594.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAMWiMy9O5nahQMwjqF_QPkaM87h+2HxZ6Hc572buEUFkcZ0iSA@mail.gmail.com>
Message-ID: <50A34224.1010101@softnet.si>

Hi,

thanks for replays:)

BR,
Miha

On 11/13/2012 7:04 PM, curriegrad2004 wrote:
> Quick answer to that: FreeSWITCH is highly multithreaded ;)
>
> On Tue, Nov 13, 2012 at 5:19 AM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>> Miha,
>> I would propose to invest in buying and reading the FreeSWITCH book, and in
>> gaining some practical experience before buying the hardware.
>>
>> answering your question, yes, FreeSWITCH is a highly scalable server which
>> spreads the workload among as many cores as you have.
>>
>>
>>
>>
>> ________________________________
>> From: Miha <miha at softnet.si>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, November 13, 2012 10:16 AM
>> Subject: [Freeswitch-users] FS server
>>
>> Hi,
>>
>> we are buying new server for FS. Does FS knows to work with more than
>> one CPU?
>>
>> BR,
>> Miha
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From peter.olsson at visionutveckling.se  Wed Nov 14 11:07:58 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 14 Nov 2012 08:07:58 +0000
Subject: [Freeswitch-users] Freeswitch to Lync with TLS
Message-ID: <1FFF97C269757C458224B7C895F35F151BF805@cantor.std.visionutv.se>

I haven't tried, but it should work. I have one trunk using TCP up though, and that works without any problems.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jacob Hodges
Skickat: den 14 november 2012 05:01
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Freeswitch to Lync with TLS

Hi all,
Does anyone have any experience configuration Freeswitch to work with Lync using TLS?

Jacob.
________________________________

Details of Data#3's transmission content disclaimer can be accessed here - www.data3.com.au<http://www.data3.com.au/transdisc>

________________________________
!DSPAM:50a31c8b32768856899745!
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From 8f27e956 at gmail.com  Wed Nov 14 12:10:57 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Wed, 14 Nov 2012 04:10:57 -0500
Subject: [Freeswitch-users] FS server
In-Reply-To: <50A34224.1010101@softnet.si>
References: <50A20FF3.3070404@softnet.si>
	<1352812770.6594.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAMWiMy9O5nahQMwjqF_QPkaM87h+2HxZ6Hc572buEUFkcZ0iSA@mail.gmail.com>
	<50A34224.1010101@softnet.si>
Message-ID: <-4084970387553399957@unknownmsgid>

Intel's XEON E3-12xxL V2 series, emphasis on the V2 series, offer a
lot of punch at a surprisingly good price points.  The "L" is low
total power dissipation series and my or may not be important to you.

Cheers,

?????
iThing:  Big thumbs & little keys. Please excuse typo, spelling and
grammar errors ? Good planets are hard to find ? think before you
print ? Last night I played a blank CD at full blast. The Mime next
door went nuts.


On 2012-11-14, at 3:01, Miha <miha at softnet.si> wrote:

> Hi,
>
> thanks for replays:)
>
> BR,
> Miha
>
> On 11/13/2012 7:04 PM, curriegrad2004 wrote:
>> Quick answer to that: FreeSWITCH is highly multithreaded ;)
>>
>> On Tue, Nov 13, 2012 at 5:19 AM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>> Miha,
>>> I would propose to invest in buying and reading the FreeSWITCH book, and in
>>> gaining some practical experience before buying the hardware.
>>>
>>> answering your question, yes, FreeSWITCH is a highly scalable server which
>>> spreads the workload among as many cores as you have.
>>>
>>>
>>>
>>>
>>> ________________________________
>>> From: Miha <miha at softnet.si>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, November 13, 2012 10:16 AM
>>> Subject: [Freeswitch-users] FS server
>>>
>>> Hi,
>>>
>>> we are buying new server for FS. Does FS knows to work with more than
>>> one CPU?
>>>
>>> BR,
>>> Miha
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jan.t.ohlsen at gmail.com  Wed Nov 14 12:36:07 2012
From: jan.t.ohlsen at gmail.com (Jan Ohlsen)
Date: Wed, 14 Nov 2012 10:36:07 +0100
Subject: [Freeswitch-users] FS server
In-Reply-To: <50A34224.1010101@softnet.si>
References: <50A20FF3.3070404@softnet.si>
	<1352812770.6594.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAMWiMy9O5nahQMwjqF_QPkaM87h+2HxZ6Hc572buEUFkcZ0iSA@mail.gmail.com>
	<50A34224.1010101@softnet.si>
Message-ID: <CAMDWk-mQ3FHnC3m50vJBpUjCsKUi2_tW=x_XnqyQt8vGbFORAA@mail.gmail.com>

http://lwn.net/Articles/486858/

Could FS be made even more multi-CPU / NUMA (vs. cores / SMP) scalable
through numa_tbind(), e.g. placing conference channels / related threads
together in a NUMA group, with static file playback data cached
independently for each group?


On Wed, Nov 14, 2012 at 8:03 AM, Miha <miha at softnet.si> wrote:

> Hi,
>
> thanks for replays:)
>
> BR,
> Miha
>
> On 11/13/2012 7:04 PM, curriegrad2004 wrote:
> > Quick answer to that: FreeSWITCH is highly multithreaded ;)
> >
> > On Tue, Nov 13, 2012 at 5:19 AM, Stanislav Sinyagin <ssinyagin at yahoo.com>
> wrote:
> >> Miha,
> >> I would propose to invest in buying and reading the FreeSWITCH book,
> and in
> >> gaining some practical experience before buying the hardware.
> >>
> >> answering your question, yes, FreeSWITCH is a highly scalable server
> which
> >> spreads the workload among as many cores as you have.
> >>
> >>
> >>
> >>
> >> ________________________________
> >> From: Miha <miha at softnet.si>
> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> Sent: Tuesday, November 13, 2012 10:16 AM
> >> Subject: [Freeswitch-users] FS server
> >>
> >> Hi,
> >>
> >> we are buying new server for FS. Does FS knows to work with more than
> >> one CPU?
> >>
> >> BR,
> >> Miha
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Nov 14 17:16:50 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 14 Nov 2012 09:16:50 -0500
Subject: [Freeswitch-users] FS server
In-Reply-To: <CAMDWk-mQ3FHnC3m50vJBpUjCsKUi2_tW=x_XnqyQt8vGbFORAA@mail.gmail.com>
References: <50A20FF3.3070404@softnet.si>
	<1352812770.6594.YahooMailNeo@web39304.mail.mud.yahoo.com>
	<CAMWiMy9O5nahQMwjqF_QPkaM87h+2HxZ6Hc572buEUFkcZ0iSA@mail.gmail.com>
	<50A34224.1010101@softnet.si>
	<CAMDWk-mQ3FHnC3m50vJBpUjCsKUi2_tW=x_XnqyQt8vGbFORAA@mail.gmail.com>
Message-ID: <919555A9-BBE3-4AED-B678-EDFCE82D7A92@jerris.com>

With the current architecture, it would probably be very difficult.  There is some talk in the future about de-coupling some of the media handling from signaling handling a bit, but even that is a long way from an architecture that would lend well to this I think.  The real question is, what exactly problem are you trying to solve.  In recent code the scalability of the software is vastly beyond what any sane person would want to implement.  There is really no value to doing 10,000 calls at a time on one box because hardware and virtual hardware is so cheap.  You could get much much more flexibility, scalability, and stability with smaller freeswitch instances and some combination of HA and virtualization technologies.

On Nov 14, 2012, at 4:36 AM, Jan Ohlsen <jan.t.ohlsen at gmail.com> wrote:

> 
> http://lwn.net/Articles/486858/
> 
> Could FS be made even more multi-CPU / NUMA (vs. cores / SMP) scalable through numa_tbind(), e.g. placing conference channels / related threads together in a NUMA group, with static file playback data cached independently for each group?
> 
> 
> On Wed, Nov 14, 2012 at 8:03 AM, Miha <miha at softnet.si> wrote:
> Hi,
> 
> thanks for replays:)
> 
> BR,
> Miha
> 
> On 11/13/2012 7:04 PM, curriegrad2004 wrote:
> > Quick answer to that: FreeSWITCH is highly multithreaded ;)
> >
> > On Tue, Nov 13, 2012 at 5:19 AM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> >> Miha,
> >> I would propose to invest in buying and reading the FreeSWITCH book, and in
> >> gaining some practical experience before buying the hardware.
> >>
> >> answering your question, yes, FreeSWITCH is a highly scalable server which
> >> spreads the workload among as many cores as you have.
> >>
> >>
> >>
> >>
> >> ________________________________
> >> From: Miha <miha at softnet.si>
> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> Sent: Tuesday, November 13, 2012 10:16 AM
> >> Subject: [Freeswitch-users] FS server
> >>
> >> Hi,
> >>
> >> we are buying new server for FS. Does FS knows to work with more than
> >> one CPU?
> >>
> >> BR,
> >> Miha
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From a.venugopan at mundio.com  Wed Nov 14 17:21:52 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Wed, 14 Nov 2012 14:21:52 +0000
Subject: [Freeswitch-users] (no subject)
Message-ID: <592A9CF93E12394E8472A6CC66E66BF23361E6@Mail-Kilo.squay.com>

Hi,

I have an issue, currently incoming and outgoing calls are looking at a single lua script (directory.lua) and find the corresponding numbers from a table and route the calls. This is what I thought from the logs.
Now the requirement is to check for the numbers for incoming call from different table. If I change (directory.lua) script, then outgoing call display is also getting changed.

As I am new to freeswitch and lua scripts I could not understand from where I should proceed. From logs (<07867429523>->447574745125), this I am calling from offnet(Vodafone,O2) to internal number(which will be present in my Database).
Ideally I should change the script in such a way that 447574745125 should be looked up from did table and direct the call. But currently its looking up in dir_users table from num_lookup.lua(which is stored as variables) and using user_id its picking up the msisdn values from dir_users(directory.lua) and transferring the call.

Should I need to write a new lua so that outgoing call and incoming call looks up different lua scripts?
Or Can I change any of these existing lua?

I have no clue on this, Can anyone please guide me. Many thanks.

Regards,
Archana

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From freeswitch-list at puzzled.xs4all.nl  Wed Nov 14 17:51:51 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 14 Nov 2012 15:51:51 +0100
Subject: [Freeswitch-users] Freeswitch and Centos 6
In-Reply-To: <-2321702856529270929@unknownmsgid>
References: <BLU0-SMTP286E22BBD85851441DD3F04BE6C0@phx.gbl>
	<01F4E2AD-570F-46D7-8BC6-20DD5F5FE875@jerris.com>
	<-2321702856529270929@unknownmsgid>
Message-ID: <50A3B007.9090001@puzzled.xs4all.nl>

On 11/14/2012 05:01 AM, S. Scott wrote:
> Been running FS 1.2.3 for over a month with zero issues observed on
> CentOS 6.3_X86-64_MINIMAL_IOS (WITH its distro's latest O/S YUM
> updates applied, plus yum'ed in the freeSWITCH dependencies).  Kernel
> clocksource0 = hpet (override acpi_pm default).

For those who don't know what S. Scott's remark about the kernel 
clocksource means:

Available clocksources which can be found with:

[patrick at vps ~]$ cat 
/sys/devices/system/clocksource/clocksource0/available_clocksource
kvm-clock tsc hpet acpi_pm

The current clocksource can be found with:

[patrick at vps ~]$ cat 
/sys/devices/system/clocksource/clocksource0/current_clocksource
kvm-clock

If you want to force the clocksource to be hpet then add hpet=force to 
the "kernel ...." line in /boot/grub/grub.conf and reboot. This does not 
work on a KVM Virtual Machine as the clocksource is always set to kvm-clock.

Regards,
Patrick



From msc at freeswitch.org  Wed Nov 14 19:33:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Nov 2012 08:33:32 -0800
Subject: [Freeswitch-users] FreeSWITCH Conf Call Today
Message-ID: <CAKzWOxV3MyaH-FYE0UKPKQ-PraieBWDE4Kayt1RkR0rKy5DN=w@mail.gmail.com>

Hello All!

Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_14

Ken Rice and a few of our FS long time users will be presenting some tips
and tricks that you may not be aware of or may have forgotten about. Talk
to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Nov 14 19:50:24 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Nov 2012 08:50:24 -0800
Subject: [Freeswitch-users] raise the call to another extension
In-Reply-To: <CAAKtwi=siQ-A+us5wuq_x1yAV0VyKzOf0ix8mLdj6i6P7ZM+Sg@mail.gmail.com>
References: <CAAKtwi=siQ-A+us5wuq_x1yAV0VyKzOf0ix8mLdj6i6P7ZM+Sg@mail.gmail.com>
Message-ID: <CAKzWOxVsWDr-_TzqcZxSFDSTOq-QRWKdcvxgG_+FcE4ANpK0iA@mail.gmail.com>

What do you mean by "raise" the call to another extension? Do you mean
"transfer" it? Also, the word "extension" can have several meanings as
well. It could mean "dialplan extension" or "physical telephone extension."

Please clarify your question and we'll be happy to assist.
-MC

On Tue, Nov 13, 2012 at 10:18 AM, Jose Alberto Gonzalez von Schmeling <
josego85 at gmail.com> wrote:

> Hello:
> I have an outside call. What I want is to raise the call to another
> extension. How is that done?
>
>
> Thanks, Jose
>
> --
> Podes encontrarme o comunicarte conmigo en:
>
>    - *Mi blog*: http://proyectosbeta.net/<http://proyectosbeta.blogspot.com/>
>    - *Facebook*:
>    http://www.facebook.com/pages/Proyectos-Beta/113277785412256<http://www.facebook.com/jose.gonzalezvonschmeling>
>    - *Twitter*: @proyectosbeta <http://twitter.com/proyectosbeta>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Nov 14 19:55:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Nov 2012 08:55:13 -0800
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23361E6@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF23361E6@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxX4S_c2gkzoAero7U1P3OHJXmXZP8aQOK9ZxeU1jmiL4g@mail.gmail.com>

I'm not an expert in this area but I have a feeling that possibly mod_lcr
can do what you want to do. Have you looked at that module by any chance?
http://wiki.freeswitch.org/wiki/Mod_lcr

-MC

On Wed, Nov 14, 2012 at 6:21 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> I have an issue, currently incoming and outgoing calls are looking at a
> single lua script (directory.lua) and find the corresponding numbers from a
> table and route the calls. This is what I thought from the logs.****
>
> Now the requirement is to check for the numbers for incoming call from
> different table. If I change (directory.lua) script, then outgoing call
> display is also getting changed.****
>
> ** **
>
> As I am new to freeswitch and lua scripts I could not understand from
> where I should proceed. From logs (<07867429523>->447574745125), this I am
> calling from offnet(Vodafone,O2) to internal number(which will be present
> in my Database).****
>
> Ideally I should change the script in such a way that 447574745125 should
> be looked up from did table and direct the call. But currently its looking
> up in dir_users table from num_lookup.lua(which is stored as variables) and
> using user_id its picking up the msisdn values from
> dir_users(directory.lua) and transferring the call.****
>
> ** **
>
> Should I need to write a new lua so that outgoing call and incoming call
> looks up different lua scripts?****
>
> Or Can I change any of these existing lua?****
>
> ** **
>
> I have no clue on this, Can anyone please guide me. Many thanks.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From josego85 at gmail.com  Wed Nov 14 20:40:54 2012
From: josego85 at gmail.com (Jose Alberto Gonzalez von Schmeling)
Date: Wed, 14 Nov 2012 14:40:54 -0300
Subject: [Freeswitch-users] raise the call to another extension
In-Reply-To: <CAKzWOxVsWDr-_TzqcZxSFDSTOq-QRWKdcvxgG_+FcE4ANpK0iA@mail.gmail.com>
References: <CAAKtwi=siQ-A+us5wuq_x1yAV0VyKzOf0ix8mLdj6i6P7ZM+Sg@mail.gmail.com>
	<CAKzWOxVsWDr-_TzqcZxSFDSTOq-QRWKdcvxgG_+FcE4ANpK0iA@mail.gmail.com>
Message-ID: <CAAKtwikgmvqUy9FBtVUMwNvgFSA__Ov6hwNXD4sGsqoktp3ndg@mail.gmail.com>

i mean transfer. I mean dialplan extension..
Thanks, Jose

On Wed, Nov 14, 2012 at 1:50 PM, Michael Collins <msc at freeswitch.org> wrote:

> What do you mean by "raise" the call to another extension? Do you mean
> "transfer" it? Also, the word "extension" can have several meanings as
> well. It could mean "dialplan extension" or "physical telephone extension."
>
> Please clarify your question and we'll be happy to assist.
> -MC
>
> On Tue, Nov 13, 2012 at 10:18 AM, Jose Alberto Gonzalez von Schmeling <
> josego85 at gmail.com> wrote:
>
>> Hello:
>> I have an outside call. What I want is to raise the call to another
>> extension. How is that done?
>>
>>
>> Thanks, Jose
>>
>> --
>> Podes encontrarme o comunicarte conmigo en:
>>
>>    - *Mi blog*: http://proyectosbeta.net/<http://proyectosbeta.blogspot.com/>
>>    - *Facebook*:
>>    http://www.facebook.com/pages/Proyectos-Beta/113277785412256<http://www.facebook.com/jose.gonzalezvonschmeling>
>>    - *Twitter*: @proyectosbeta <http://twitter.com/proyectosbeta>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Podes encontrarme o comunicarte conmigo en:

   - *Mi blog*: http://proyectosbeta.net/<http://proyectosbeta.blogspot.com/>
   - *Facebook*:
   http://www.facebook.com/pages/Proyectos-Beta/113277785412256<http://www.facebook.com/jose.gonzalezvonschmeling>
   - *Twitter*: @proyectosbeta <http://twitter.com/proyectosbeta>
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From vbvbrj at gmail.com  Wed Nov 14 22:08:39 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 14 Nov 2012 21:08:39 +0200
Subject: [Freeswitch-users] Compile with PostgreSQL,
	postgresql build with prefix.
Message-ID: <50A3EC37.1080909@gmail.com>

Hello.

I've build PostreSQL with --prefix /opt/postresql

Now, I want to use --enable-core-pgsql-support
For odbc there is --with-odbc-lib=dir and --with-odbc-include=dir
What is equivalent for PostgreSQL?

I use export PATH=$PATH:/opt/postgresql/bin, but still get:
checking for pg_config... /opt/postgresql/bin/pg_config
checking for PostgreSQL libraries... checking for PQgetvalue in -lpq... no
configure: error: no usable libpq; please install PostgreSQL devel 
package or equivalent

What is difference between --enable-core-odbc-support and 
--with-odbc=PREFIX ?

-- 
Mimiko desu.


From gabe at gundy.org  Thu Nov 15 11:34:11 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 15 Nov 2012 01:34:11 -0700
Subject: [Freeswitch-users] raise the call to another extension
In-Reply-To: <CAKzWOxVsWDr-_TzqcZxSFDSTOq-QRWKdcvxgG_+FcE4ANpK0iA@mail.gmail.com>
References: <CAAKtwi=siQ-A+us5wuq_x1yAV0VyKzOf0ix8mLdj6i6P7ZM+Sg@mail.gmail.com>
	<CAKzWOxVsWDr-_TzqcZxSFDSTOq-QRWKdcvxgG_+FcE4ANpK0iA@mail.gmail.com>
Message-ID: <CAMwZ-tLY4xmOnaE4Dwr=cSzDqkLWvKeGHXdKukW+71-KQ2ddbg@mail.gmail.com>

On Wed, Nov 14, 2012 at 9:50 AM, Michael Collins <msc at freeswitch.org> wrote:
> What do you mean by "raise" the call to another extension? Do you mean
> "transfer" it?

I'm guessing he's looking at bridging to a registered endpoint. If
that's right, than here we go (see the 2nd example):

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#From_the_Dialplan


Best,
Gabe


From anton.jugatsu at gmail.com  Thu Nov 15 12:02:49 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 15 Nov 2012 13:02:49 +0400
Subject: [Freeswitch-users] Compile with PostgreSQL,
	postgresql build with prefix.
In-Reply-To: <50A3EC37.1080909@gmail.com>
References: <50A3EC37.1080909@gmail.com>
Message-ID: <CANBhyPC639cUXruisi=KXNk9HkZXLCanWCcNeM9wdT8_3Rm+yw@mail.gmail.com>

You should install libpq-dev.


2012/11/14 Mimiko <vbvbrj at gmail.com>

> Hello.
>
> I've build PostreSQL with --prefix /opt/postresql
>
> Now, I want to use --enable-core-pgsql-support
> For odbc there is --with-odbc-lib=dir and --with-odbc-include=dir
> What is equivalent for PostgreSQL?
>
> I use export PATH=$PATH:/opt/postgresql/bin, but still get:
> checking for pg_config... /opt/postgresql/bin/pg_config
> checking for PostgreSQL libraries... checking for PQgetvalue in -lpq... no
> configure: error: no usable libpq; please install PostgreSQL devel
> package or equivalent
>
> What is difference between --enable-core-odbc-support and
> --with-odbc=PREFIX ?
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ntomer at newgen.co.in  Thu Nov 15 12:59:53 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 15 Nov 2012 15:29:53 +0530
Subject: [Freeswitch-users] mod_db help needed
Message-ID: <00a001cdc317$f58c2900$e0a47b00$@co.in>

Hi,

 

I am trying to enter some data (entered by caller, received by Play and Get
Digits). But I am not able to get much information about it from mod_db
wiki. I want to enter data in an Oracle database. Please tell me what all I
need to do, in order to do this.

 

Regards

 

Nitin


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From a.venugopan at mundio.com  Thu Nov 15 13:40:21 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 15 Nov 2012 10:40:21 +0000
Subject: [Freeswitch-users] profile
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2336255@Mail-Kilo.squay.com>

Hi,
Can anyone please tell me how to get the profile name

local sipprofile = params:getHeader("sip_profile");
print ("sip profile is:" .. sipprofile);

When I give this I am getting a Nil value error instead of 'external' or 'internal'. I want the profile to return either external or internal value.

Regards,
Archana

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From avi at avimarcus.net  Thu Nov 15 14:42:37 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 15 Nov 2012 13:42:37 +0200
Subject: [Freeswitch-users] profile
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2336255@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2336255@Mail-Kilo.squay.com>
Message-ID: <CALNZuw=cyUFNvmmY1xWo9i3OB5xYRd-e3gRZCQimnSmm7NtHmQ@mail.gmail.com>

It's in "sofia_profile_name" but I don't know if that's available during
the call.
-Avi


On Thu, Nov 15, 2012 at 12:40 PM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> Can anyone please tell me how to get the profile name****
>
> ** **
>
> local sipprofile = params:getHeader("sip_profile");  ****
>
> print ("sip profile is:" .. sipprofile);  ****
>
> ** **
>
> When I give this I am getting a Nil value error instead of ?external? or
> ?internal?. I want the profile to return either external or internal value.
> ****
>
> ** **
>
> Regards,****
>
> Archana****
>
> **
>
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From ntomer at newgen.co.in  Thu Nov 15 15:51:09 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 15 Nov 2012 18:21:09 +0530
Subject: [Freeswitch-users] mod_db help needed
In-Reply-To: <00a001cdc317$f58c2900$e0a47b00$@co.in>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>
Message-ID: <00eb01cdc32f$e284b6c0$a78e2440$@co.in>

I understand that I will have to make changes in db.conf.xml -

 

<configuration name="db.conf" description="LIMIT DB Configuration">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

  </settings>

</configuration>

 

But the wiki also says this - "ODBC must be configured to use ODBC resources
(configure with --enable-core-odbc-support)." What does this mean? I
couldn't find any more references to it.

 

Wiki says - "Realm and key are arbitrary strings. Consider realm as a
container for keys." Does this mean that I can use a realm for a particular
type of data e.g. "callerdata", use caller number as key and set the entered
data against that key. But how would I be able to access this data from
outside FreeSwitch?

 

Is the realm mapped to a table in the database or something similar?

 

Thanks in advance

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Thursday, November 15, 2012 3:30 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_db help needed

 

Hi,

 

I am trying to enter some data (entered by caller, received by Play and Get
Digits). But I am not able to get much information about it from mod_db
wiki. I want to enter data in an Oracle database. Please tell me what all I
need to do, in order to do this.

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From benkokakao at gmail.com  Thu Nov 15 15:56:22 2012
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 15 Nov 2012 13:56:22 +0100
Subject: [Freeswitch-users] Wiki Search broken
Message-ID: <CAAMQ8bQA_zTDSFjDaD9wYBUU2=tw_UO20279Z+fsY6QQ=bBH5A@mail.gmail.com>

Hi!

Am i the only person who is trying to use the wiki-search at
http://wiki.freeswitch.org/ ?

The search has been broken since a few months now, with a
503-error-message from Google(I assume the mediawiki-search has been
switched to google-search at some point and the mechanism failed
since).

I've opened a bug-report at http://jira.freeswitch.org/browse/FSWIKI-13.

I'm only posting this mail to get some feedback from the community
because i'm puzzled that i've not found any post mentioning it on this
list or a bug report on Jira.

Well, maybe i'm just to clumsy to use both the wiki-search AND google
for "site:lists.freeswitch.org wiki search" to see if anyone else
mentioned it before. So please confirm! Otherwise i must believe
Google plays an evil prank on me and it works for everyone else...

Regards,
Christian


From tshepo.maphutha at gmail.com  Thu Nov 15 12:34:09 2012
From: tshepo.maphutha at gmail.com (Tshepo Maphutha)
Date: Thu, 15 Nov 2012 11:34:09 +0200
Subject: [Freeswitch-users] How to add dial plans
Message-ID: <CAC_Ms107gcHY5FCJkc0BNaorpfK+KuS0Km+53O65gNd-6FXteA@mail.gmail.com>

Hello,

I am new to FreeSwitch and currently just installed it and want to work
around with it a bit.

I have default settings and can make calls to extensions that are from 1000
-1019 .

How do i make calls to PSTN numbers or GSM numbers.

Please indicate what will be needed

Warm Regards,
-- 
TG Maphutha
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From alessio at asistar.it  Thu Nov 15 13:22:31 2012
From: alessio at asistar.it (Alessio)
Date: Thu, 15 Nov 2012 11:22:31 +0100
Subject: [Freeswitch-users] Create a simple dialplan
Message-ID: <50A4C267.2020103@asistar.it>

Hi all,
Can I create a dialplan for incoming calls that do the following?

<condition field="destination_number" expression=" ^0123456789$">
     <action application="export" data="originate_timeout=5"/>
     <action application="transfer" data="100 XML default"/>

     If 100 did not respond:

     <action application="export" data="originate_timeout=15"/>
     <action application="transfer" data="101 XML default"/>

     If 103 did not respond:

     <action application="export" data="originate_timeout=99"/>
     <action application="transfer" data="102 XML default"/>

</condition>

Can anyone help me to understand how I have to set dialplan for do this?

Alessio.


From peter.olsson at visionutveckling.se  Thu Nov 15 16:38:26 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 15 Nov 2012 13:38:26 +0000
Subject: [Freeswitch-users] Wiki Search broken
Message-ID: <1FFF97C269757C458224B7C895F35F151C174B@cantor.std.visionutv.se>

As far as I can see it works fine here...

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Christian Benke
Skickat: den 15 november 2012 13:56
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Wiki Search broken

Hi!

Am i the only person who is trying to use the wiki-search at http://wiki.freeswitch.org/ ?

The search has been broken since a few months now, with a 503-error-message from Google(I assume the mediawiki-search has been switched to google-search at some point and the mechanism failed since).

I've opened a bug-report at http://jira.freeswitch.org/browse/FSWIKI-13.

I'm only posting this mail to get some feedback from the community because i'm puzzled that i've not found any post mentioning it on this list or a bug report on Jira.

Well, maybe i'm just to clumsy to use both the wiki-search AND google for "site:lists.freeswitch.org wiki search" to see if anyone else mentioned it before. So please confirm! Otherwise i must believe Google plays an evil prank on me and it works for everyone else...

Regards,
Christian

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

 

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

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

!DSPAM:50a4eb8432766282415789!



From freeswitch-list at puzzled.xs4all.nl  Thu Nov 15 16:42:00 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 15 Nov 2012 14:42:00 +0100
Subject: [Freeswitch-users] Wiki Search broken
In-Reply-To: <CAAMQ8bQA_zTDSFjDaD9wYBUU2=tw_UO20279Z+fsY6QQ=bBH5A@mail.gmail.com>
References: <CAAMQ8bQA_zTDSFjDaD9wYBUU2=tw_UO20279Z+fsY6QQ=bBH5A@mail.gmail.com>
Message-ID: <50A4F128.1090205@puzzled.xs4all.nl>

On 11/15/2012 01:56 PM, Christian Benke wrote:
> Hi!
>
> Am i the only person who is trying to use the wiki-search at
> http://wiki.freeswitch.org/ ?
>
> The search has been broken since a few months now, with a
> 503-error-message from Google(I assume the mediawiki-search has been
> switched to google-search at some point and the mechanism failed
> since).
>
> I've opened a bug-report at http://jira.freeswitch.org/browse/FSWIKI-13.
>
> I'm only posting this mail to get some feedback from the community
> because i'm puzzled that i've not found any post mentioning it on this
> list or a bug report on Jira.
>
> Well, maybe i'm just to clumsy to use both the wiki-search AND google
> for "site:lists.freeswitch.org wiki search" to see if anyone else
> mentioned it before. So please confirm! Otherwise i must believe
> Google plays an evil prank on me and it works for everyone else...

I just entered "TLS" in the search bar labeled "Search" in the top left 
of the wiki front page and clicking on "Go" works fine as does clicking 
on "Search". Maybe clear your cache, delete cookies etc and see if the 
problem goes away. If all else fails maybe try a different browser.

Regards,
Patrick



From shaheryarkh at googlemail.com  Thu Nov 15 16:48:42 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Thu, 15 Nov 2012 14:48:42 +0100
Subject: [Freeswitch-users] Is there any documentation for Voicemail DB
	schema?
Message-ID: <CAFZQphz9o5JxD4BGqJka5jzcQP-nC10LCaVsT6XUZkPb2vxD6Q@mail.gmail.com>

Hi,

I can find db schema for voicemail module at below url,

http://wiki.freeswitch.org/wiki/Mod_voicemail#Database_Schema

But unfortunately it does not give any explanation of db fields, for
example i would like to know what are the types and possible values for
flags and read_flags column in voicemail_msgs table and so on.

Does anyone has any idea or documentation link for this?

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From bret at ticm.com  Thu Nov 15 16:48:54 2012
From: bret at ticm.com (Bret Watson)
Date: Thu, 15 Nov 2012 21:48:54 +0800
Subject: [Freeswitch-users] pennytel registration issues - aka 200 OK has
	fatal errors
Message-ID: <CAN4s1rigCeYWCLZvmU8tc9mR2Rmtsyfo6u75KmELERUnu6sNJg@mail.gmail.com>

Hi All,
running into pennytel registration errors - registration log is here
http://pastebin.com/7EFspd47

gateway config:
--
<include>
 <gateway name="Pennytel">
  <param name="username" value="61894360596"/>
  <param name="password" value="b01eca29"/>
  <param name="realm" value="sip.pennytel.com"/>
  <param name="proxy" value="sip.pennytel.com"/>
  <param name="register" value="true"/>
</gateway>
</include>
--

any thoughts?

Thanks

Bret
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From bret at ticm.com  Thu Nov 15 16:50:16 2012
From: bret at ticm.com (Bret Watson)
Date: Thu, 15 Nov 2012 21:50:16 +0800
Subject: [Freeswitch-users] pennytel registration issues etc..
Message-ID: <CAN4s1rgRXG6-8i2z+Ji7X+CdDa7ReZdVibwoJJDnSRGi-R7Cyw@mail.gmail.com>

Forgot to add... works fine when using x-lite

Cheers

Bret
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From josego85 at gmail.com  Thu Nov 15 17:09:41 2012
From: josego85 at gmail.com (Jose Alberto Gonzalez von Schmeling)
Date: Thu, 15 Nov 2012 11:09:41 -0300
Subject: [Freeswitch-users] doesn't work ivr when calling with linksys
Message-ID: <CAAKtwin5xjVNU78Nj7BR-gttFKabX+T_ZvC97DKxXhpUuLdyWA@mail.gmail.com>

Hello
Internally calling to 5000 (ivr) it works, but when calling with
linksys to 5000
(ivr) does not work. Why?

Thanks, Jose
-- 
Podes encontrarme o comunicarte conmigo en:

   - *Mi blog*: http://proyectosbeta.net/<http://proyectosbeta.blogspot.com/>
   - *Facebook*:
   http://www.facebook.com/pages/Proyectos-Beta/113277785412256<http://www.facebook.com/jose.gonzalezvonschmeling>
   - *Twitter*: @proyectosbeta <http://twitter.com/proyectosbeta>
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From juanito1982 at gmail.com  Thu Nov 15 17:10:42 2012
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 15 Nov 2012 15:10:42 +0100
Subject: [Freeswitch-users] mod_xml_curl and SIP profiles
Message-ID: <CAHLYDCvQgu8EGT=k5p347oLgfXYXkcTKfTBdYv4O_iPTeqYA7g@mail.gmail.com>

Hello boys,

   I am doing some tests with mod_xml_curl to server the directory
configuration. I was be able to make it work but users must register in
external profile instead the internal one. Calls are also processed in
public dialplan. Is there any way to get the users be registered in
internal profile and the calls processed in default dialplan?

Directory XML response from webserver is:

http://pastebin.com/hpjivASc

Regards
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From moshe3t at gmail.com  Thu Nov 15 18:00:01 2012
From: moshe3t at gmail.com (moshe3)
Date: Thu, 15 Nov 2012 10:00:01 -0500
Subject: [Freeswitch-users] Wiki Search broken
In-Reply-To: <CAAMQ8bQA_zTDSFjDaD9wYBUU2=tw_UO20279Z+fsY6QQ=bBH5A@mail.gmail.com>
References: <CAAMQ8bQA_zTDSFjDaD9wYBUU2=tw_UO20279Z+fsY6QQ=bBH5A@mail.gmail.com>
Message-ID: <170201cdc341$e3da2f70$ab8e8e50$@com>

Seems like you said "Google plays an evil prank on me and it works for
everyone else..."

Working great for me 


Following is the url to a search of api and attached is the screenshot 

http://wiki.freeswitch.org/wiki/Special:Search?search=api&go=Go 




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Christian Benke
Sent: Thursday, November 15, 2012 7:56 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Wiki Search broken

Hi!

Am i the only person who is trying to use the wiki-search at
http://wiki.freeswitch.org/ ?

The search has been broken since a few months now, with a 503-error-message
from Google(I assume the mediawiki-search has been switched to google-search
at some point and the mechanism failed since).

I've opened a bug-report at http://jira.freeswitch.org/browse/FSWIKI-13.

I'm only posting this mail to get some feedback from the community because
i'm puzzled that i've not found any post mentioning it on this list or a bug
report on Jira.

Well, maybe i'm just to clumsy to use both the wiki-search AND google for
"site:lists.freeswitch.org wiki search" to see if anyone else mentioned it
before. So please confirm! Otherwise i must believe Google plays an evil
prank on me and it works for everyone else...

Regards,
Christian

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From jnvines at gmail.com  Thu Nov 15 19:07:23 2012
From: jnvines at gmail.com (Nick Vines)
Date: Thu, 15 Nov 2012 11:07:23 -0500
Subject: [Freeswitch-users] Create a simple dialplan
In-Reply-To: <50A4C267.2020103@asistar.it>
References: <50A4C267.2020103@asistar.it>
Message-ID: <CA+NjZ2uM2MnwocfdZey33vqO7qt=qhcQQJDUW=+ndkmMOpx_FA@mail.gmail.com>

I think this will do what you want. It will try extension 100 for 5
seconds, then 101 for 15 seconds, then 102 for 99 seconds. You could also
use groups in addition to users.

<action application="bridge" data="[call_timeout=5]user/100 at domain|
[call_timeout=15]user/101 at domain|[call_timeout=99]user/102 at domain"/>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#From_the_Dialplan


Nick


On Thu, Nov 15, 2012 at 5:22 AM, Alessio <alessio at asistar.it> wrote:

> Hi all,
> Can I create a dialplan for incoming calls that do the following?
>
> <condition field="destination_number" expression=" ^0123456789$">
>      <action application="export" data="originate_timeout=5"/>
>      <action application="transfer" data="100 XML default"/>
>
>      If 100 did not respond:
>
>      <action application="export" data="originate_timeout=15"/>
>      <action application="transfer" data="101 XML default"/>
>
>      If 103 did not respond:
>
>      <action application="export" data="originate_timeout=99"/>
>      <action application="transfer" data="102 XML default"/>
>
> </condition>
>
> Can anyone help me to understand how I have to set dialplan for do this?
>
> Alessio.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vbvbrj at gmail.com  Thu Nov 15 19:31:22 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 15 Nov 2012 18:31:22 +0200
Subject: [Freeswitch-users] Compile with PostgreSQL,
 postgresql build with prefix.
In-Reply-To: <CANBhyPC639cUXruisi=KXNk9HkZXLCanWCcNeM9wdT8_3Rm+yw@mail.gmail.com>
References: <50A3EC37.1080909@gmail.com>
	<CANBhyPC639cUXruisi=KXNk9HkZXLCanWCcNeM9wdT8_3Rm+yw@mail.gmail.com>
Message-ID: <50A518DA.6040403@gmail.com>

On 15.11.2012 11:02, Anton Kvashenkin wrote:
> You should install libpq-dev.
>

Yes, I know, but I written, I've compiled PostgreSQL from sources. 
Anyway, I compiled PostgreSQL with --disable-shared option, so shared 
libs were not created. For now ./configure is running well with:

export PATH=$PATH:/opt/postgresql/bin && \
./configure --enable-static --with-openssl --enable-core-odbc-support 
--enable-portable-binary \
--enable-zrtp --prefix=/opt/freeswitch --enable-core-pgsql-support 
--enable-static --enable-shared \
LDFLAGS="-Wl,-R,\\\$\${ORIGIN}/../lib:\\\$\${ORIGIN}/../lib/x86_64-linux-gnu 
-L/lib -L/usr/lib -L/opt/postgresql/lib" \
CPPFLAGS="-I/usr/include -I/opt/postgresql/include"

But I get errors on make:

libtool: link: ranlib .libs/libcurl.a
libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s 
"../libcurl.la" "libcurl.la" )
make[3]: Leaving directory `/home/mimiko/src/freeswitch/libs/curl/lib'
make[2]: Leaving directory `/home/mimiko/src/freeswitch/libs/curl/lib'
Making all in src
make[2]: Entering directory `/home/mimiko/src/freeswitch/libs/curl/src'
make  all-am
make[3]: Entering directory `/home/mimiko/src/freeswitch/libs/curl/src'
make[3]: Leaving directory `/home/mimiko/src/freeswitch/libs/curl/src'
make[2]: Leaving directory `/home/mimiko/src/freeswitch/libs/curl/src'
make[2]: Entering directory `/home/mimiko/src/freeswitch/libs/curl'
make[2]: Nothing to be done for `all-am'.
make[2]: Leaving directory `/home/mimiko/src/freeswitch/libs/curl'
make[1]: Leaving directory `/home/mimiko/src/freeswitch/libs/curl'
cat /home/mimiko/src/freeswitch/src/include/switch_cpp.h | perl 
/home/mimiko/src/freeswitch/build/strip.pl > 
/home/mimiko/src/freeswitch/src/include/switch_swigable_cpp.h
make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" 
/home/mimiko/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | 
uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo 
$i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; 
then tmp_mods="$(grep -v "#" /home/mimiko/src/freeswitch/modules.conf | 
sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for 
i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" 
"OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" 
/home/mimiko/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | 
uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo 
$i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z 
"" ; then tmp_mods="$(grep -v "#" 
/home/mimiko/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | 
uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo 
$i-uninstall ; done)"; echo $mods )" 
"OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" 
/home/mimiko/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all 
; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep 
"#" /home/mimiko/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo 
$i-clean ; done )"; echo $mods )" 
"OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" 
/home/mimiko/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo 
$i-install ; done)"; echo $mods )" 
"OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" 
/home/mimiko/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo 
$i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
cc1: warnings being treated as errors
src/switch.c: In function ???main???:
src/switch.c:667: error: implicit declaration of function ???rpl_malloc???
src/switch.c:667: error: cast to pointer from integer of different size
src/switch.c:683: error: cast to pointer from integer of different size
src/switch.c:698: error: cast to pointer from integer of different size
src/switch.c:715: error: cast to pointer from integer of different size
src/switch.c:731: error: cast to pointer from integer of different size
src/switch.c:747: error: cast to pointer from integer of different size
src/switch.c:762: error: cast to pointer from integer of different size
src/switch.c:777: error: cast to pointer from integer of different size
src/switch.c:792: error: cast to pointer from integer of different size
src/switch.c:807: error: cast to pointer from integer of different size
src/switch.c:822: error: cast to pointer from integer of different size
src/switch.c:837: error: cast to pointer from integer of different size
src/switch.c:852: error: cast to pointer from integer of different size
src/switch.c:869: error: cast to pointer from integer of different size
make[1]: *** [freeswitch-switch.o] Error 1
make: *** [all] Error 2


Why is this?

-- 
Mimiko desu.


From a.venugopan at mundio.com  Thu Nov 15 19:59:20 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 15 Nov 2012 16:59:20 +0000
Subject: [Freeswitch-users] profile
In-Reply-To: <CALNZuw=cyUFNvmmY1xWo9i3OB5xYRd-e3gRZCQimnSmm7NtHmQ@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2336255@Mail-Kilo.squay.com>
	<CALNZuw=cyUFNvmmY1xWo9i3OB5xYRd-e3gRZCQimnSmm7NtHmQ@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF23362F9@Mail-Kilo.squay.com>

Hi,
Can you please let me know in which file this parameter would be defined.

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: 15 November 2012 11:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] profile

It's in "sofia_profile_name" but I don't know if that's available during the call.
-Avi

On Thu, Nov 15, 2012 at 12:40 PM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Can anyone please tell me how to get the profile name

local sipprofile = params:getHeader("sip_profile");
print ("sip profile is:" .. sipprofile);

When I give this I am getting a Nil value error instead of 'external' or 'internal'. I want the profile to return either external or internal value.

Regards,
Archana

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From abaci64 at gmail.com  Thu Nov 15 20:45:10 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 15 Nov 2012 12:45:10 -0500
Subject: [Freeswitch-users] Is there any documentation for Voicemail DB
 schema?
In-Reply-To: <CAFZQphz9o5JxD4BGqJka5jzcQP-nC10LCaVsT6XUZkPb2vxD6Q@mail.gmail.com>
References: <CAFZQphz9o5JxD4BGqJka5jzcQP-nC10LCaVsT6XUZkPb2vxD6Q@mail.gmail.com>
Message-ID: <50A52A26.8010306@gmail.com>

flags will be either '' for regular messages, 'save' for saved messages, 
or 'delete' for messages marked for deletion.
read_flags will be either 'B_NORMAL' for normal messages, or 'A_URGENT' 
for urgent messages.
you can update of the wiki to help the next person.

On 11/15/2012 8:48 AM, Muhammad Shahzad wrote:
> Hi,
>
> I can find db schema for voicemail module at below url,
>
> http://wiki.freeswitch.org/wiki/Mod_voicemail#Database_Schema
>
> But unfortunately it does not give any explanation of db fields, for 
> example i would like to know what are the types and possible values 
> for flags and read_flags column in voicemail_msgs table and so on.
>
> Does anyone has any idea or documentation link for this?
>
> Thank you.
>
>
> -- 
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com <mailto:shari_786pk at hotmail.com>
> Email: shaheryarkh at googlemail.com <mailto:shaheryarkh at googlemail.com>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From freeswitch-list at puzzled.xs4all.nl  Thu Nov 15 20:58:48 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 15 Nov 2012 18:58:48 +0100
Subject: [Freeswitch-users] pennytel registration issues - aka 200 OK
 has fatal errors
In-Reply-To: <CAN4s1rigCeYWCLZvmU8tc9mR2Rmtsyfo6u75KmELERUnu6sNJg@mail.gmail.com>
References: <CAN4s1rigCeYWCLZvmU8tc9mR2Rmtsyfo6u75KmELERUnu6sNJg@mail.gmail.com>
Message-ID: <50A52D58.8050600@puzzled.xs4all.nl>

On 11/15/2012 02:48 PM, Bret Watson wrote:
> Hi All,
> running into pennytel registration errors - registration log is here
> http://pastebin.com/7EFspd47
>
> gateway config:
> --
> <include>
>   <gateway name="Pennytel">
>    <param name="username" value="61894360596"/>
>    <param name="password" value="b01eca29"/>

That is a fake password right? If not then you may want to change it asap.

Regards,
Patrick



From benkokakao at gmail.com  Thu Nov 15 21:01:55 2012
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 15 Nov 2012 19:01:55 +0100
Subject: [Freeswitch-users] Wiki Search broken
In-Reply-To: <170201cdc341$e3da2f70$ab8e8e50$@com>
References: <CAAMQ8bQA_zTDSFjDaD9wYBUU2=tw_UO20279Z+fsY6QQ=bBH5A@mail.gmail.com>
	<170201cdc341$e3da2f70$ab8e8e50$@com>
Message-ID: <CAAMQ8bRcxfMMU-MB0Gux417nAD31xSwhGvOnAwjUTdEXL8+nZg@mail.gmail.com>

> Working great for me
>
>
> Following is the url to a search of api and attached is the screenshot
>
> http://wiki.freeswitch.org/wiki/Special:Search?search=api&go=Go

LOL WTF? See my screenshot attached... (Chrome 21.0.1180.89 on Linux)

It never occured to me, but i followed Patrick's advice and tried with
Firefox - and there it works just fine.

Edit: Well, i figured out what the cause was - it was the
chrome-extension "Disconnect"(http://bit.ly/WxRt8z) which blocked the
Google-Search, after disabling it it works just fine.

Sorry for the noise, that was one fun day :-)

Cheers,
Christian
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From a.venugopan at mundio.com  Thu Nov 15 21:17:01 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 15 Nov 2012 18:17:01 +0000
Subject: [Freeswitch-users] param configurations
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233632D@Mail-Kilo.squay.com>

Hi,
Can anyone please tell me in which file and path exactly the below params would be configured

Event-Name: REQUEST_PARAMS
Core-UUID: <uuid>
FreeSWITCH-Hostname: hostname
FreeSWITCH-IPv4: 192.168.1.11
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2010-08-06%2014%3A04%3A40
Event-Date-GMT: Fri,%2006%20Aug%202010%2018%3A04%3A40%20GMT
Event-Date-Timestamp: 1281117880813532
Event-Calling-File: sofia.c
Event-Calling-Function: config_sofia
Event-Calling-Line-Number: 3481
purpose: gateways
profile: internal

and what I need to do if i need to put a new "profile" say 'external'.

Regards,
Archana

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From vbvbrj at gmail.com  Thu Nov 15 21:29:53 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 15 Nov 2012 20:29:53 +0200
Subject: [Freeswitch-users] Compile with PostgreSQL,
 postgresql build with prefix.
In-Reply-To: <CANBhyPC639cUXruisi=KXNk9HkZXLCanWCcNeM9wdT8_3Rm+yw@mail.gmail.com>
References: <50A3EC37.1080909@gmail.com>
	<CANBhyPC639cUXruisi=KXNk9HkZXLCanWCcNeM9wdT8_3Rm+yw@mail.gmail.com>
Message-ID: <50A534A1.6030700@gmail.com>

So in order to "make" I had to do the following:

export PATH=$PATH:/opt/postgresql/bin && export 
LD_LIBRARY_PATH=${LD_LIBRARY_PATH}:/opt/postgresql/lib && \
./configure --with-openssl --enable-core-odbc-support 
--enable-portable-binary \
--enable-zrtp --prefix=/opt/freeswitch --enable-core-pgsql-support 
--enable-static --enable-shared \
LDFLAGS="-L/opt/postgresql/lib 
-Wl,-R,\\\$\${ORIGIN}/../lib:\\\$\${ORIGIN}/../lib/x86_64-linux-gnu"


It will be great to have a way to specify lib and bin path for 
PostgreSQL to the ./configure with an option the same way is done for odbc.

-- 
Mimiko desu.


From Chris.Martineau at semafone.com  Thu Nov 15 19:02:39 2012
From: Chris.Martineau at semafone.com (Chris Martineau)
Date: Thu, 15 Nov 2012 16:02:39 +0000
Subject: [Freeswitch-users] Event Channel Bridge not firing
In-Reply-To: <1FFF97C269757C458224B7C895F35F151BEAD2@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151BEAD2@cantor.std.visionutv.se>
Message-ID: <870204F45EE7D34E8D27CC0E602E11A1B4681A@EX01.semafone.local>

Hi Peter,

Sorry already found the issue, Sipp scenario firing too fast. I was using a canned scenario that didn't have any pause between setup and teardown which caused a bit of a race condition. Needed a minimum of 100ms between setup and teardown to report the states correctly.

Thanks for your time.

Regards

Chris


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 13 November 2012 15:10
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Event Channel Bridge not firing

It's impossible to see anything without the full FS logs. There is nothing in these logs that even say that you were trying to bridge a call or not.

Please supply the full debug logs, and pastebin them.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> F?r Chris Martineau
Skickat: den 13 november 2012 12:28
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: [Freeswitch-users] Event Channel Bridge not firing

Hi,

I am writing a simple interface to read bridge and unbridge events via the event socket.

All was going well until I noticed that a lot of the events are missing. Out of 10 I will lose 30%.

Increased tcp buffers, increased read requests but with no effect.

Looked at wireshark and the events are not being sent?

Made a simple freeswitch module to read and report on events and get exactly the same in the module.

I attach the output from freeswitch  so you can see that even though the channel states trigger the bridge and sometimes the unbridge events do not (or at least they are not reaching my module or the esl module)?

Any idea why this may be the case?

Freeswitch version 1.3.0+git 20121023T030122Z~8589e031d0 (git 8589e03 2012-10-23 03:01:22Z)

(Ignore the ERR level, output from the module at this level so it was easy to see just these logs)

2012-11-13 10:50:03.564141 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:03.584210 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:03.604108 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:03.624137 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2049 at 192.168.86.130<mailto:UUID=1-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:03.644145 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1236064-2d7f-11e2-9d1c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.563188 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:04.584337 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:04.603206 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e1bc097c-2d7f-11e2-9d26-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:04.623215 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=2-2049 at 192.168.86.130<mailto:UUID=2-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:05.563322 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:05.583317 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:05.603407 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2545330-2d7f-11e2-9d30-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:05.623330 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3-2049 at 192.168.86.130<mailto:UUID=3-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:06.563430 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:06.583602 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:06.603523 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e2ed6ade-2d7f-11e2-9d3a-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:06.623468 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4-2049 at 192.168.86.130<mailto:UUID=4-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:07.563046 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:07.583102 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:07.603059 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e385b514-2d7f-11e2-9d44-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:07.623083 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=5-2049 at 192.168.86.130<mailto:UUID=5-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:08.563249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:08.585648 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:08.603325 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e41ee432-2d7f-11e2-9d4e-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:08.623173 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=6-2049 at 192.168.86.130<mailto:UUID=6-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:09.587627 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:09.606362 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e4b6cb94-2d7f-11e2-9d58-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:09.623249 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=7-2049 at 192.168.86.130<mailto:UUID=7-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:10.563401 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:10.585417 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:10.603595 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e54f9694-2d7f-11e2-9d62-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:10.623350 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=8-2049 at 192.168.86.130<mailto:UUID=8-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:11.563509 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:11.583537 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:11.604376 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e5e7e322-2d7f-11e2-9d6c-158386780fa6 State=CS_DESTROY
2012-11-13 10:50:11.623112 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=9-2049 at 192.168.86.130<mailto:UUID=9-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:12.563113 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:50:12.583790 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_ROUTING
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:12.603153 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_HANGUP
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=10-2049 at 192.168.86.130<mailto:UUID=10-2049 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:50:12.627134 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=e6806ff2-2d7f-11e2-9d76-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:52:35.023536 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:35.043882 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:35.066203 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2061 at 192.168.86.130<mailto:UUID=1-2061 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:52:35.083087 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=3b69185c-2d80-11e2-9d80-158386780fa6 State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_ROUTING
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:52:50.003125 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_HANGUP
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4455dab8-2d80-11e2-9d8a-158386780fa6 State=CS_DESTROY
2012-11-13 10:52:50.032743 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2062 at 192.168.86.130<mailto:UUID=1-2062 at 192.168.86.130> State=CS_DESTROY

freeswitch at ubuntu> 2012-11-13 10:53:01.383362 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:01.403381 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_BRIDGE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HIBERNATE
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_UNBRIDGE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2063 at 192.168.86.130<mailto:UUID=1-2063 at 192.168.86.130> State=CS_DESTROY
2012-11-13 10:53:01.430829 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=4b1ff590-2d80-11e2-9d94-158386780fa6 State=CS_DESTROY


NO BRIDGE OR UNBRIDGE EVENTS REPORTED FOR THIS CALL?
freeswitch at ubuntu> 2012-11-13 10:53:16.243477 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_ROUTING
2012-11-13 10:53:16.266281 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_ROUTING
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_CONSUME_MEDIA
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_EXECUTE
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_HANGUP
2012-11-13 10:53:16.283823 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=53fc1842-2d80-11e2-9d9e-158386780fa6 State=CS_DESTROY
2012-11-13 10:53:16.303504 [ERR] mod_ccm.c:311 CHANNEL_CALLSTATE: UUID=1-2064 at 192.168.86.130<mailto:UUID=1-2064 at 192.168.86.130> State=CS_DESTROY

Many thanks

Chris


!DSPAM:50a259a132764491816697!
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From Chris.Martineau at semafone.com  Thu Nov 15 20:08:43 2012
From: Chris.Martineau at semafone.com (Chris Martineau)
Date: Thu, 15 Nov 2012 17:08:43 +0000
Subject: [Freeswitch-users] uuid_media leaves behind zombie call
Message-ID: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>

Hi,

I am trying to use uuid_media to switch in and out of the rtp stream and in essence it seems to work however I get some strange things happening.

The first thing is that when the re-invites go out, the a and the b legs get the codec list in a different order i.e. 0 8 3 to the a leg and 8 0 3 to the b leg!
If all the codecs are supported this forces a transcoding scenario as the a leg neg's pcmu and the b leg pcma! As this is done via the event socket and not the dialplan I cannot see what I am supposed to set to correct this behaviour ( the calls all start in media bypass mode so no codec neg is done at call start).
Any ideas why this should do this and how I can get round it? Setting the no transcoding variable has no effect on this.

The second thing is that if you go into the media path and then going out of it again and then clear down it seems to leave the a leg channel hungup in the system. Show channels lists the channel just sitting in CS_HANGUP state. In the log it says Locked, waiting on external entities?
Everything externally has cleared and wireshark shows a clean dialog with no resends of the BYE messages. Cleardown is from the a leg and FS has passed the BYE on to the b leg.
If you try to kill it with uuid_kill it says that the uuid does not exist? The only way I can clear it is to restart FS which on closing it hangs around with a sofia error saying waiting for 1 session.

If I just go into the media path and cleardown without coming out again the bridge process seems to be broken as any BYE message does not get passed through, requiring both ends to be cleared separately. Also no UNBRIDGE event is fired in this scenario which I suppose you would expect if it wasn't bridged, however rtp is flowing through as is signalling?

The same channel however is still left hanging.

I may be using this incorrectly or not preparing the environment however the info around this command is pretty basic and looks simple to use.

It is pretty critical for the application I am trying to build that this process works smoothly so any help you could offer would be greatly appreciated.

I have attached some debug logs.

Many thanks


Chris


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From shaheryarkh at googlemail.com  Thu Nov 15 23:18:43 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Thu, 15 Nov 2012 21:18:43 +0100
Subject: [Freeswitch-users] Is there any documentation for Voicemail DB
	schema?
In-Reply-To: <50A52A26.8010306@gmail.com>
References: <CAFZQphz9o5JxD4BGqJka5jzcQP-nC10LCaVsT6XUZkPb2vxD6Q@mail.gmail.com>
	<50A52A26.8010306@gmail.com>
Message-ID: <CAFZQphw-7Ynkr3g-r-NMucCjhKLn5zRmzg+w=N1dOYXfsP4CWg@mail.gmail.com>

I don't have account on FS Wiki. However, i sent request for it. Lets see
if it approves then i will update voicemail docs anytime soon.

Thank you.


On Thu, Nov 15, 2012 at 6:45 PM, Abaci <abaci64 at gmail.com> wrote:

>  flags will be either '' for regular messages, 'save' for saved messages,
> or 'delete' for messages marked for deletion.
> read_flags will be either 'B_NORMAL' for normal messages, or 'A_URGENT'
> for urgent messages.
> you can update of the wiki to help the next person.
>
>
> On 11/15/2012 8:48 AM, Muhammad Shahzad wrote:
>
> Hi,
>
>  I can find db schema for voicemail module at below url,
>
>  http://wiki.freeswitch.org/wiki/Mod_voicemail#Database_Schema
>
>  But unfortunately it does not give any explanation of db fields, for
> example i would like to know what are the types and possible values for
> flags and read_flags column in voicemail_msgs table and so on.
>
>  Does anyone has any idea or documentation link for this?
>
>  Thank you.
>
>
>  --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From msc at freeswitch.org  Thu Nov 15 23:31:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Nov 2012 12:31:25 -0800
Subject: [Freeswitch-users] profile
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23362F9@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2336255@Mail-Kilo.squay.com>
	<CALNZuw=cyUFNvmmY1xWo9i3OB5xYRd-e3gRZCQimnSmm7NtHmQ@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF23362F9@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxVt6RPyBkNRrZ4CA+85JUL2_-Soyxc2N0zSF6omdjGKtA@mail.gmail.com>

This is found in conf/sip_profiles/internal.xml or external.xml or whatever
SIP profile you have created. It's right at the top in the "name" attribute:

<profile name="internal">

One of the easiest ways to see what's available is to send a call through
the "info" app. Also, if you're looking at specific events then it's not a
bad idea to dump all the events to the screen or a file and then sift
through them. It's a lot of information at first but once you get
accustomed to looking for specific things it becomes much easier.

-MC

On Thu, Nov 15, 2012 at 8:59 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> Can you please let me know in which file this parameter would be defined.*
> ***
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* 15 November 2012 11:43
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] profile****
>
> ** **
>
> It's in "sofia_profile_name" but I don't know if that's available during
> the call.
> ****
>
> -Avi****
>
> ** **
>
> On Thu, Nov 15, 2012 at 12:40 PM, Archana Venugopan <
> a.venugopan at mundio.com> wrote:****
>
> Hi,****
>
> Can anyone please tell me how to get the profile name****
>
>  ****
>
> local sipprofile = params:getHeader("sip_profile");  ****
>
> print ("sip profile is:" .. sipprofile);  ****
>
>  ****
>
> When I give this I am getting a Nil value error instead of ?external? or
> ?internal?. I want the profile to return either external or internal value.
> ****
>
>  ****
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Nov 15 23:43:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Nov 2012 12:43:32 -0800
Subject: [Freeswitch-users] How to add dial plans
In-Reply-To: <CAC_Ms107gcHY5FCJkc0BNaorpfK+KuS0Km+53O65gNd-6FXteA@mail.gmail.com>
References: <CAC_Ms107gcHY5FCJkc0BNaorpfK+KuS0Km+53O65gNd-6FXteA@mail.gmail.com>
Message-ID: <CAKzWOxXazRN6zdGyrNVGQVinCJ2XvS+aV46XO9P-f_sASh1JBA@mail.gmail.com>

Hi Tshepo,

Welcome to FreeSWITCH! Hold on tight - telephony and VoIP are a wild ride.
:)

To make calls out to the PSTN you'll need some sort of connectivity,
usually a SIP trunk. Do you have that? If so we have a number of pages on
our wiki that are dedicated to assisting with specific providers:

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

Also, it looks like you are using the example config. If you open
conf/vars.xml you can set the default_provider variable and it will use
whichever provider you create. We have default dialing for NANPA numbers.
Look in conf/dialplan/default/01_example.xml for those. Modify as needed.
Lastly, drop by the #freeswitch channel on irc.freenode.net if you want
some real-time help.

-MC

On Thu, Nov 15, 2012 at 1:34 AM, Tshepo Maphutha
<tshepo.maphutha at gmail.com>wrote:

> Hello,
>
> I am new to FreeSwitch and currently just installed it and want to work
> around with it a bit.
>
> I have default settings and can make calls to extensions that are from
> 1000 -1019 .
>
> How do i make calls to PSTN numbers or GSM numbers.
>
> Please indicate what will be needed
>
> Warm Regards,
> --
> TG Maphutha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Nov 15 23:50:54 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Nov 2012 12:50:54 -0800
Subject: [Freeswitch-users] mod_xml_curl and SIP profiles
In-Reply-To: <CAHLYDCvQgu8EGT=k5p347oLgfXYXkcTKfTBdYv4O_iPTeqYA7g@mail.gmail.com>
References: <CAHLYDCvQgu8EGT=k5p347oLgfXYXkcTKfTBdYv4O_iPTeqYA7g@mail.gmail.com>
Message-ID: <CAKzWOxUDzJ-D8YVu=zMNRog-QZKTL90QUuuDwZ0Q4E9pZCxmqA@mail.gmail.com>

Why must your users register on the external profile?
-MC

On Thu, Nov 15, 2012 at 6:10 AM, Juan Antonio Iba?ez Santorum <
juanito1982 at gmail.com> wrote:

> Hello boys,
>
>    I am doing some tests with mod_xml_curl to server the directory
> configuration. I was be able to make it work but users must register in
> external profile instead the internal one. Calls are also processed in
> public dialplan. Is there any way to get the users be registered in
> internal profile and the calls processed in default dialplan?
>
> Directory XML response from webserver is:
>
> http://pastebin.com/hpjivASc
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From yiftah at choochee.com  Fri Nov 16 00:32:54 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Thu, 15 Nov 2012 13:32:54 -0800
Subject: [Freeswitch-users] mod_db help needed
In-Reply-To: <00eb01cdc32f$e284b6c0$a78e2440$@co.in>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>
	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>
Message-ID: <CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>

Hi Nitin,
Just install odbc on you system, I used this one (
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html)
but you can use FreeSWITCH's explanations as well (
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core)
Once you are done the configure should already know that you have odbc
support, just hit regular configure or as FreeSWITCH docs says :
Compile FreeSWITCH with ODBC support

UnixODBC support will be autodetected by ./configure, and if found, will be
compiled into FreeSWITCH.
As for the keys it is pretty easy just go to (
http://wiki.freeswitch.org/wiki/Mod_db) the query should be something like
:

<action application="set" data="var=${db(select/realm/key)}"/>

The documentation is actually pretty good so try to read it again it will
make sense
Thanks,
Yiftach.

On Thu, Nov 15, 2012 at 4:51 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

>  I understand that I will have to make changes in db.conf.xml -****
>
> ** **
>
> <configuration name="db.conf" description="LIMIT DB Configuration">****
>
>   <settings>****
>
>     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->****
>
>   </settings>****
>
> </configuration>****
>
> ** **
>
> But the wiki also says this ? ?ODBC must be configured to use ODBC
> resources (configure with --enable-core-odbc-support).? What does this
> mean? I couldn?t find any more references to it.****
>
> ** **
>
> Wiki says ? ?Realm and key are arbitrary strings. Consider realm as a
> container for keys.? Does this mean that I can use a realm for a particular
> type of data e.g. ?callerdata?, use caller number as key and set the
> entered data against that key. But how would I be able to access this data
> from outside FreeSwitch?****
>
> ** **
>
> Is the realm mapped to a table in the database or something similar?****
>
> ** **
>
> Thanks in advance****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nitin Tomer
> *Sent:* Thursday, November 15, 2012 3:30 PM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* [Freeswitch-users] mod_db help needed****
>
> ** **
>
> Hi,****
>
> ** **
>
> I am trying to enter some data (entered by caller, received by Play and
> Get Digits). But I am not able to get much information about it from mod_db
> wiki. I want to enter data in an Oracle database. Please tell me what all I
> need to do, in order to do this.****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> ** **
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
> ** **
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
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> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From lists at kavun.ch  Fri Nov 16 00:46:35 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 15 Nov 2012 22:46:35 +0100
Subject: [Freeswitch-users] Domains and profiles
In-Reply-To: <3FC39EED-0A60-4A87-866D-0AAA629F9560@kavun.ch>
References: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
	<1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>
	<CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>
	<3FC39EED-0A60-4A87-866D-0AAA629F9560@kavun.ch>
Message-ID: <8ED4CC3C-441A-4AA0-8DDB-1B57CE646A3C@kavun.ch>

I read and reread Anthony's explanation and am still not able to have multiple profile allow registering on the same domain.
If I have 2 identical profiles, it looks like the first one that is up will take the ownership of the domain name. Anything registering on the secondary one will not be visible to a sofia_contact. 

If sip.example.com:5070 is up before sip.example.com:5060, only phones registered to sip.example.com:5070 will be visible to FS.

How do I fix this? I am not enforcing the domain in my configs.

Thanks,
Emrah

On Nov 13, 2012, at 8:23 PM, Emrah <lists at kavun.ch> wrote:

> This is precious. I had figured out how the domain portion affects FS, I just didn't know how to declare my domains to my SIP profiles. Which I believe I now know and will experiment a bit.
> 
> Thanks!
> On Nov 12, 2012, at 8:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> The best thing to do is take a look at these things from a step back.
>> 
>> The domains inside the xml registry are completely different from the domains on the internet and again completely different from domains in sip packets.  The profiles are again entirely different from any of the above.  Its up to you to align them if you so choose.
>> 
>> 
>> The default configuration distributed with FreeSWITCH sets up the scenario most likely to load on any machine and work out of the box.  That is the primary goal of that configuration, so, It sets the domain in both the directory, the global default domain variable and the name of the internal profile to be identical to the ip on the box that can reach the internet.  Then it sets the sip to force everything to that value.  When you want to detach from this behavior, you are probably on a venture to do some kind of multi-home setup.
>> 
>> 
>> Aliases in the <aliases> tag are a list of keys you want to use to use that lead to the current profile your are configuring.  Think of it as the /etc/hosts file in unix only for profiles.   When you define aliases to match all of the possible domains hosted on a particular profile, then when you try to take a user at host.com notation and decide which profile it came from, you can use the aliases to find it providing you have added <alias name="host.com"/> to that profile.
>> 
>> The <domains> tag is an indicator telling the profile to open the xml registry in FreeSWITCH and run through any domains defined therein.
>> The 2 key attributes are:
>> 
>> alias: [true/false]  (automatically create an alias for this domain as mentioned above)
>> parse: [true/false] (scan the domain for gateway entries and include them into this profile)
>> name: [<string>] (either the name of a specific domain or 'all' to denote parsing every domain in the directory)
>> 
>> As you showed in your question the default config has
>> 
>> <domain name="all" alias="true" parse="false"/>
>> 
>> If you apply what you have learned above, it will scan for every domain (there is only one by default) and add an alias for it and not parse it for gateways.  The default directory uses global config vars to set the domain to match the local ip on the box.  So now you will have a domain in your config that is your ip, and the internal profile will attach to it and add an alias so that value expands to match it.
>> 
>> 
>> This is explained in a comment at the top of directory/default.xml
>> 
>>    FreeSWITCH works off the concept of users and domains just like email.                                                                                      
>>    You have users that are in domains for example 1000 at domain.com.                                                                                             
>> 
>>    When freeswitch gets a register packet it looks for the user in the directory                                                                               
>>    based on the from or to domain in the packet depending on how your sofia profile                                                                            
>>    is configured.  Out of the box the default domain will be the IP address of the                                                                             
>>    machine running FreeSWITCH.  This IP can be found by typing "sofia status" at the                                                                           
>>    CLI.  You will register your phones to the IP and not the hostname by default.                                                                              
>>    If you wish to register using the domain please open vars.xml in the root conf                                                                              
>>    directory and set the default domain to the hostname you desire.  Then you would                                                                            
>>    use the domain name in the client instead of the IP address to register                                                                                     
>>    with FreeSWITCH.                                                                    
>> 
>> 
>> 
>> So having more than one profile with the default of 
>> 
>> <domain name="all" alias="true" parse="false"/>
>> 
>> is going to end up aliasing the same domains into all profiles who call it and cause an overwrite in the lookup table and probably an error in your logs somewhere.  If you had parse="true" on all of them, they would all try and register to the gateways in all of your domains.
>> 
>> 
>> If you look at the stock config, external.xml is a good example of a secondary profile, it has
>> 
>> <domain name="all" alias="false" parse="true"/>
>> 
>> so no aliases, and yes parse ... the exact opposite of the internal so that all the gateways would register from external and internal would bind to the local ip.
>> 
>> So, you probably want to use separate <domain name="my.com"/> per domain per profile you want to bind it to in more complicated setups.
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> On Sun, Nov 11, 2012 at 9:09 PM, Emrah <lists at kavun.ch> wrote:
>> Bless you!
>> 
>> Thanks for putting this together. You've beautifully summed up all my questions.
>> On Nov 11, 2012, at 8:09 AM, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>> 
>>> Hi Folks,
>>> I've started a new thread as it's not quite the same issue, and domains & profiles have confused the heck out of me every time I have developed a new setup for fS.
>>> I have sometimes had to hack/hard-doce the dialstring to make multiple domains in one profile work, had hours of fun with presence, db and force register settings, and have still had some odd gotchas that have required extensive meditation.
>>> [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract these elements ]
>>> 
>>> Coming at this from standards/specs and rolling my own SIP stacks, sofia/fS seems to use the term "domain" differently from sipdomain, and alias seems to be tied to the directory (and thus to the profile listed in a directory file), but I'm not sure.
>>> so ...
>>> Before I capture to the sofia conf xml wiki page, I have a couple of questions on the sip-profile XML setup;
>>> 
>>> Q: Is there a particular reason why there's a parameter called alias and an (entirely different) setting also called alias?
>>> The sofia conf xml wiki's comment on the setting "alias" shows I'm not alone.
>>> I agree that's what it appears to be doing, but can we nail this down please (and what happens if an external client uses this connection to register and call)?
>>> 
>>> In the current sofia conf xml wiki page, the domain setting is not exactly well documented :).
>>> The current internal.xml vanilla example from git (as of time of writing) has the following lines:
>>> -------------------------
>>> ...
>>> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>>> <!--aliases are other names that will work as a valid profile name for this profile-->
>>> <aliases>
>>>  <!--
>>>       <alias name="default"/>
>>>   -->
>>> </aliases>
>>> <!-- Outbound Registrations -->
>>> <gateways>
>>>  <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
>>> </gateways>
>>> 
>>> <domains>
>>>  <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
>>>  <!--<domain name="$${domain}" parse="true"/>-->
>>>  <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
>>>  <!--<domain name="all" alias="true" parse="true"/>-->
>>>  <domain name="all" alias="true" parse="false"/>
>>> </domains>
>>> ...
>>> -------------------------
>>> 
>>> This stuff is entirely missing from the sofia.conf.xml wiki page, and it IS really important.
>>> 
>>> 
>>> Q: what's the default value for the alias parameter in the domain element? -- it is missing from the first example.
>>> Q: if there is more than one profile, what's the impact of setting parse = true in one (or all) of the profiles' XML files?
>>>  (or parse = false, or missing the parameter altogether)?
>>> AFAICT, the gateways get pulled in via the pre-process directive just fine, regardless of the value of the parse parameter -- it works for me, at least.
>>> 
>>> Q: if there is more than one profile, what's the impact of putting domain name="all" into one (or all) of the profiles' XML files?
>>> 
>>> Ideally, having more than one sipdomain tied to one profile "would be good", but aliases doesn't do that -- as the git file says, these are aliases for the profile name.
>>> 
>>> Before I start scribbling, Answers on a postcard to this ML, please.
>>> 
>>> all the best,
>>> Lawrence
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 tomasz.szuster at gmail.com  Fri Nov 16 01:12:27 2012
From: tomasz.szuster at gmail.com (Tomasz Szuster)
Date: Thu, 15 Nov 2012 23:12:27 +0100
Subject: [Freeswitch-users] How to add dial plans
In-Reply-To: <CAC_Ms107gcHY5FCJkc0BNaorpfK+KuS0Km+53O65gNd-6FXteA@mail.gmail.com>
References: <CAC_Ms107gcHY5FCJkc0BNaorpfK+KuS0Km+53O65gNd-6FXteA@mail.gmail.com>
Message-ID: <CAMAX9-2To+PcyGT1PJ_YonXMBqL3ms4P27c3t42=XBMW8gJyNg@mail.gmail.com>

Hi Tshepo,

You have to search for mod_gsmopen - gsm network, and dahdi for pstn.
Both informations are in freeswitch wiki.

Regards.

On Thu, Nov 15, 2012 at 10:34 AM, Tshepo Maphutha <tshepo.maphutha at gmail.com
> wrote:

> Hello,
>
> I am new to FreeSwitch and currently just installed it and want to work
> around with it a bit.
>
> I have default settings and can make calls to extensions that are from
> 1000 -1019 .
>
> How do i make calls to PSTN numbers or GSM numbers.
>
> Please indicate what will be needed
>
> Warm Regards,
> --
> TG Maphutha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Pozdrawiam
Tomasz
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From bret at ticm.com  Fri Nov 16 02:07:04 2012
From: bret at ticm.com (Bret Watson)
Date: Fri, 16 Nov 2012 07:07:04 +0800
Subject: [Freeswitch-users] pennytel registration issues - aka 200 OK
 has fatal errors
In-Reply-To: <50A52D58.8050600@puzzled.xs4all.nl>
References: <CAN4s1rigCeYWCLZvmU8tc9mR2Rmtsyfo6u75KmELERUnu6sNJg@mail.gmail.com>
	<50A52D58.8050600@puzzled.xs4all.nl>
Message-ID: <CAN4s1rhf4eo9sVFJPGngxFeFWLnRkY-0RzjXnq2jrSSO2-zczA@mail.gmail.com>

D'oh... and I was being careful removing stuff out of the log as well...
thanks!


On 16 November 2012 01:58, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl>wrote:

> On 11/15/2012 02:48 PM, Bret Watson wrote:
> > Hi All,
> > running into pennytel registration errors - registration log is here
> > http://pastebin.com/7EFspd47
> >
> > gateway config:
> > --
> > <include>
> >   <gateway name="Pennytel">
> >    <param name="username" value="61894360596"/>
> >    <param name="password" value="b01eca29"/>
>
> That is a fake password right? If not then you may want to change it asap.
>
> Regards,
> Patrick
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From k4kaleem at gmail.com  Fri Nov 16 02:22:35 2012
From: k4kaleem at gmail.com (kaleem rehman)
Date: Thu, 15 Nov 2012 23:22:35 +0000
Subject: [Freeswitch-users] call disconnects after 32 seconds
Message-ID: <CAFWsKqzS13Q7iXZm6x_jNR0E-KAGugSNSpRkHLwq-Stbwt7M4Q@mail.gmail.com>

Hi All,

my inbound calls are fine with no issues, my outbound calls get
disconnected after 32 seconds and its on all calls. i tried 2 different
suppliers and its same result.
please find the attached log file with sofia in debug mode. - caller was
extension 1234 and desination was 01908321682

your help will be greately appreciated.

regards,
Kaleem
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From msc at freeswitch.org  Fri Nov 16 03:01:11 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Nov 2012 16:01:11 -0800
Subject: [Freeswitch-users] Is there any documentation for Voicemail DB
	schema?
In-Reply-To: <CAFZQphw-7Ynkr3g-r-NMucCjhKLn5zRmzg+w=N1dOYXfsP4CWg@mail.gmail.com>
References: <CAFZQphz9o5JxD4BGqJka5jzcQP-nC10LCaVsT6XUZkPb2vxD6Q@mail.gmail.com>
	<50A52A26.8010306@gmail.com>
	<CAFZQphw-7Ynkr3g-r-NMucCjhKLn5zRmzg+w=N1dOYXfsP4CWg@mail.gmail.com>
Message-ID: <CAKzWOxWVjtDk8ab_8qU_WKCfR89AJMnRhehq15ag0xnHfZPGmg@mail.gmail.com>

Enabled! Go forth and document.
-MC

On Thu, Nov 15, 2012 at 12:18 PM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> I don't have account on FS Wiki. However, i sent request for it. Lets see
> if it approves then i will update voicemail docs anytime soon.
>
> Thank you.
>
>
> On Thu, Nov 15, 2012 at 6:45 PM, Abaci <abaci64 at gmail.com> wrote:
>
>>  flags will be either '' for regular messages, 'save' for saved
>> messages, or 'delete' for messages marked for deletion.
>> read_flags will be either 'B_NORMAL' for normal messages, or 'A_URGENT'
>> for urgent messages.
>> you can update of the wiki to help the next person.
>>
>>
>> On 11/15/2012 8:48 AM, Muhammad Shahzad wrote:
>>
>> Hi,
>>
>>  I can find db schema for voicemail module at below url,
>>
>>  http://wiki.freeswitch.org/wiki/Mod_voicemail#Database_Schema
>>
>>  But unfortunately it does not give any explanation of db fields, for
>> example i would like to know what are the types and possible values for
>> flags and read_flags column in voicemail_msgs table and so on.
>>
>>  Does anyone has any idea or documentation link for this?
>>
>>  Thank you.
>>
>>
>>  --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> 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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Fri Nov 16 03:26:52 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Nov 2012 18:26:52 -0600
Subject: [Freeswitch-users] uuid_media leaves behind zombie call
In-Reply-To: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>
References: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>
Message-ID: <CAKbxfG-YU0Sz2wd0mZ9+rv3MON13f6AYbJcwxPa1GdmsuGaRoQ@mail.gmail.com>

Issues should be reported on http://jira.freeswitch.org

You should be reproducing on git HEAD and including full logs as
attachments.
Also you need to enable the sofia sip traces.

console loglevel debug
sofia loglevel all 9
sofia global siptrace on

I'd like to figure out the steps you took to send this email so we can see
were have continued to fail to make the policy clear,



On Thu, Nov 15, 2012 at 11:08 AM, Chris Martineau <
Chris.Martineau at semafone.com> wrote:

>  Hi,****
>
> ** **
>
> I am trying to use uuid_media to switch in and out of the rtp stream and
> in essence it seems to work however I get some strange things happening.**
> **
>
> ** **
>
> The first thing is that when the re-invites go out, the a and the b legs
> get the codec list in a different order i.e. 0 8 3 to the a leg and 8 0 3
> to the b leg!****
>
> If all the codecs are supported this forces a transcoding scenario as the
> a leg neg?s pcmu and the b leg pcma! As this is done via the event socket
> and not the dialplan I cannot see what I am supposed to set to correct this
> behaviour ( the calls all start in media bypass mode so no codec neg is
> done at call start).****
>
> Any ideas why this should do this and how I can get round it? Setting the
> no transcoding variable has no effect on this.****
>
> ** **
>
> The second thing is that if you go into the media path and then going out
> of it again and then clear down it seems to leave the a leg channel hungup
> in the system. Show channels lists the channel just sitting in CS_HANGUP
> state. In the log it says Locked, waiting on external entities?****
>
> Everything externally has cleared and wireshark shows a clean dialog with
> no resends of the BYE messages. Cleardown is from the a leg and FS has
> passed the BYE on to the b leg.****
>
> If you try to kill it with uuid_kill it says that the uuid does not exist?
> The only way I can clear it is to restart FS which on closing it hangs
> around with a sofia error saying waiting for 1 session.****
>
> ** **
>
> If I just go into the media path and cleardown without coming out again
> the bridge process seems to be broken as any BYE message does not get
> passed through, requiring both ends to be cleared separately. Also no
> UNBRIDGE event is fired in this scenario which I suppose you would expect
> if it wasn?t bridged, however rtp is flowing through as is signalling?****
>
> ** **
>
> The same channel however is still left hanging.****
>
> ** **
>
> I may be using this incorrectly or not preparing the environment however
> the info around this command is pretty basic and looks simple to use.****
>
> ** **
>
> It is pretty critical for the application I am trying to build that this
> process works smoothly so any help you could offer would be greatly
> appreciated.****
>
> ** **
>
> I have attached some debug logs.****
>
> ** **
>
> Many thanks****
>
> ** **
>
> ** **
>
> Chris****
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From msc at freeswitch.org  Fri Nov 16 03:31:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Nov 2012 16:31:09 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFWsKqzS13Q7iXZm6x_jNR0E-KAGugSNSpRkHLwq-Stbwt7M4Q@mail.gmail.com>
References: <CAFWsKqzS13Q7iXZm6x_jNR0E-KAGugSNSpRkHLwq-Stbwt7M4Q@mail.gmail.com>
Message-ID: <CAKzWOxVyGpSkE1tSBYMK9hD6jd0jgqgYQQ9wSQWoEJUhTAF-_A@mail.gmail.com>

Get a SIP capture of this call as well. It might yield a clue.
-MC

On Thu, Nov 15, 2012 at 3:22 PM, kaleem rehman <k4kaleem at gmail.com> wrote:

> Hi All,
>
> my inbound calls are fine with no issues, my outbound calls get
> disconnected after 32 seconds and its on all calls. i tried 2 different
> suppliers and its same result.
> please find the attached log file with sofia in debug mode. - caller was
> extension 1234 and desination was 01908321682
>
> your help will be greately appreciated.
>
> regards,
> Kaleem
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Fri Nov 16 03:34:14 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 16 Nov 2012 00:34:14 +0000
Subject: [Freeswitch-users] How to add dial plans
In-Reply-To: <CAMAX9-2To+PcyGT1PJ_YonXMBqL3ms4P27c3t42=XBMW8gJyNg@mail.gmail.com>
References: <CAC_Ms107gcHY5FCJkc0BNaorpfK+KuS0Km+53O65gNd-6FXteA@mail.gmail.com>
	<CAMAX9-2To+PcyGT1PJ_YonXMBqL3ms4P27c3t42=XBMW8gJyNg@mail.gmail.com>
Message-ID: <C27D7369-6131-4E9D-B8B0-34425625A5C5@gmail.com>

Mod_dahdi is one TDM module. The main one is mod_freetdm though, which works with a lot more cards.  Both would need you to have a ISDN line (PRI/BRI) and a TDM card to interface with it.

Mod_gsmopen uses a USB mobile dongle to make calls via the gsm network.

If you don't have any of the above hardware available, the usual way most people send calls to the PSTN is via a SIP provider as Mike said. There are plenty around. All will need you to create an account they will bill you through. You would probably set them up as a Sofia gateway.

Whether calling to a landline or mobile number, the methods would be the same. Although if you have multiple routes it may affect the route you use (eg gsmopen may be cheaper than freetdm or sip providers to call mobiles, and visa versa for landlines).

Calling via any if the above would involve creating a dial plan extension that matches the number(s) you want to call and sends them via one of the above using the bridge app. The dial string determines the module used, the route used and number dialed. You can use $1 etc to generate the disk string from the matched number. There's plenty of info on the wiki.

Steve on iPhone



On 15 Nov 2012, at 22:12, Tomasz Szuster <tomasz.szuster at gmail.com> wrote:

> Hi Tshepo,
> 
> You have to search for mod_gsmopen - gsm network, and dahdi for pstn.
> Both informations are in freeswitch wiki.
> 
> Regards.
> 
> On Thu, Nov 15, 2012 at 10:34 AM, Tshepo Maphutha <tshepo.maphutha at gmail.com> wrote:
> Hello,
> 
> I am new to FreeSwitch and currently just installed it and want to work around with it a bit.
> 
> I have default settings and can make calls to extensions that are from 1000 -1019 .
> 
> How do i make calls to PSTN numbers or GSM numbers.
> 
> Please indicate what will be needed 
> 
> Warm Regards,
> -- 
> TG Maphutha
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Pozdrawiam
> Tomasz
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Fri Nov 16 03:36:19 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 16 Nov 2012 00:36:19 +0000
Subject: [Freeswitch-users] mod_db help needed
In-Reply-To: <00eb01cdc32f$e284b6c0$a78e2440$@co.in>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>
	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>
Message-ID: <D98A7905-5B7F-4BAB-A553-512D10DF978A@gmail.com>

--enable-core-odbc-support is an option to the configure script that compiles in ODBC support. It shouldn't be needed any more, it should be enabled by default (if available?)

You'll want unixodbc and it's dev package installed.

Steve on iPhone



On 15 Nov 2012, at 12:51, "Nitin Tomer" <ntomer at newgen.co.in> wrote:

> I understand that I will have to make changes in db.conf.xml -
>  
> <configuration name="db.conf" description="LIMIT DB Configuration">
>   <settings>
>     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
>   </settings>
> </configuration>
>  
> But the wiki also says this ? ?ODBC must be configured to use ODBC resources (configure with --enable-core-odbc-support).? What does this mean? I couldn?t find any more references to it.
>  
> Wiki says ? ?Realm and key are arbitrary strings. Consider realm as a container for keys.? Does this mean that I can use a realm for a particular type of data e.g. ?callerdata?, use caller number as key and set the entered data against that key. But how would I be able to access this data from outside FreeSwitch?
>  
> Is the realm mapped to a table in the database or something similar?
>  
> Thanks in advance
>  
> Nitin
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin Tomer
> Sent: Thursday, November 15, 2012 3:30 PM
> To: 'FreeSWITCH Users Help'
> Subject: [Freeswitch-users] mod_db help needed
>  
> Hi,
>  
> I am trying to enter some data (entered by caller, received by Play and Get Digits). But I am not able to get much information about it from mod_db wiki. I want to enter data in an Oracle database. Please tell me what all I need to do, in order to do this.
>  
> Regards
>  
> Nitin
>  
> Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL) accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
> 
>  
> 
> Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From steveayre at gmail.com  Fri Nov 16 03:49:03 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 16 Nov 2012 00:49:03 +0000
Subject: [Freeswitch-users] Create a simple dialplan
In-Reply-To: <50A4C267.2020103@asistar.it>
References: <50A4C267.2020103@asistar.it>
Message-ID: <1F90FE12-623A-4F32-941A-53033561E566@gmail.com>

Transfer is to move the incoming call to another point in the dialplan (and change the destination_number field). 

If you want an outgoing call you need the bridge app instead, to make the call and join it to the incoming channel.

As for extensions, there's a difference between dial plan extensions and user extensions.

For users bridge to user/username.

If you really need to try another dial plan extension with a timeout (and you probably don't) try bridging with the loopback endpoint.

Steve on iPhone



On 15 Nov 2012, at 10:22, Alessio <alessio at asistar.it> wrote:

> Hi all,
> Can I create a dialplan for incoming calls that do the following?
> 
> <condition field="destination_number" expression=" ^0123456789$">
>     <action application="export" data="originate_timeout=5"/>
>     <action application="transfer" data="100 XML default"/>
> 
>     If 100 did not respond:
> 
>     <action application="export" data="originate_timeout=15"/>
>     <action application="transfer" data="101 XML default"/>
> 
>     If 103 did not respond:
> 
>     <action application="export" data="originate_timeout=99"/>
>     <action application="transfer" data="102 XML default"/>
> 
> </condition>
> 
> Can anyone help me to understand how I have to set dialplan for do this?
> 
> Alessio.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From vetali100 at gmail.com  Fri Nov 16 04:43:47 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Thu, 15 Nov 2012 17:43:47 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFWsKqzS13Q7iXZm6x_jNR0E-KAGugSNSpRkHLwq-Stbwt7M4Q@mail.gmail.com>
References: <CAFWsKqzS13Q7iXZm6x_jNR0E-KAGugSNSpRkHLwq-Stbwt7M4Q@mail.gmail.com>
Message-ID: <CACc3D0X3FonmA0abxp7rRD4032bG+H-cGbCaGOEDiXMk0gnxUw@mail.gmail.com>

I saw this happened earlier when the remote party does not send SIP ACK
after receiving SIP OK, so the call is being disconnected after exactly 32
seconds.
Not sure if this is exact same scenario here, but just something to
consider...

Regards.
Vitalie


2012/11/15 kaleem rehman <k4kaleem at gmail.com>

> Hi All,
>
> my inbound calls are fine with no issues, my outbound calls get
> disconnected after 32 seconds and its on all calls. i tried 2 different
> suppliers and its same result.
> please find the attached log file with sofia in debug mode. - caller was
> extension 1234 and desination was 01908321682
>
> your help will be greately appreciated.
>
> regards,
> Kaleem
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Fri Nov 16 05:35:44 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 15 Nov 2012 20:35:44 -0600
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CACc3D0X3FonmA0abxp7rRD4032bG+H-cGbCaGOEDiXMk0gnxUw@mail.gmail.com>
Message-ID: <CCCB02A0.60B4A%krice@freeswitch.org>

This is probably the same scenario as this is exactly what to expect... Call
gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup the
call....

Quite common on networks with NAT issues or broken endpoints


On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com> wrote:

> I saw this happened earlier when the remote party does not send SIP ACK after
> receiving SIP OK, so the call is being disconnected after exactly 32 seconds.
> Not sure if this is exact same scenario here, but just something to
> consider...
> 
> Regards.
> Vitalie
> 
> 
> 2012/11/15 kaleem rehman <k4kaleem at gmail.com>
>> Hi All,
>> ?
>> my inbound calls are fine with no issues, my outbound calls get disconnected
>> after 32 seconds and its on all calls. i tried 2 different suppliers and its
>> same result.
>> please find the attached log file with sofia in debug mode. - caller was
>> extension 1234 and desination was 01908321682
>> ?
>> your help will be greately appreciated.
>> ?
>> regards,
>> Kaleem
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From packetandy at gmail.com  Fri Nov 16 06:58:56 2012
From: packetandy at gmail.com (andy)
Date: Thu, 15 Nov 2012 19:58:56 -0800
Subject: [Freeswitch-users] changing vm default announcement
Message-ID: <50A5BA00.5050209@gmail.com>

Hi all,

I am trying to change the default greeting that plays to the caller from 
the standard voicemail/callie when the call is not answered.. I have not 
been able to find where the script is for the ivr that answers the call. 
I tried overriding the default in the dialplan by using :

             <action application="set" data="voicemail_greeting_number=1"/>
             <action application="set" data="skip_instructions=true"/>

but this did not change anything - the default greeting still plays. I 
have tried to copy a different message into each user's directory as 
greeting_1.wav, but some flag is not been set so callie still plays

  Any suggestions?

drew


From juanito1982 at gmail.com  Fri Nov 16 09:40:31 2012
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 16 Nov 2012 07:40:31 +0100
Subject: [Freeswitch-users] mod_xml_curl and SIP profiles
In-Reply-To: <CAKzWOxUDzJ-D8YVu=zMNRog-QZKTL90QUuuDwZ0Q4E9pZCxmqA@mail.gmail.com>
References: <CAHLYDCvQgu8EGT=k5p347oLgfXYXkcTKfTBdYv4O_iPTeqYA7g@mail.gmail.com>
	<CAKzWOxUDzJ-D8YVu=zMNRog-QZKTL90QUuuDwZ0Q4E9pZCxmqA@mail.gmail.com>
Message-ID: <CAHLYDCvB=QRz8g2YezNn-nJ4gG_5ESU=CRSid5EAjeKrJ+TvwQ@mail.gmail.com>

Sorry, it was a mistake. The problem was the dialplan only and I think I
can mend it using user_context variable.

Regards

2012/11/15 Michael Collins <msc at freeswitch.org>

> Why must your users register on the external profile?
> -MC
>
> On Thu, Nov 15, 2012 at 6:10 AM, Juan Antonio Iba?ez Santorum <
> juanito1982 at gmail.com> wrote:
>
>> Hello boys,
>>
>>    I am doing some tests with mod_xml_curl to server the directory
>> configuration. I was be able to make it work but users must register in
>> external profile instead the internal one. Calls are also processed in
>> public dialplan. Is there any way to get the users be registered in
>> internal profile and the calls processed in default dialplan?
>>
>> Directory XML response from webserver is:
>>
>> http://pastebin.com/hpjivASc
>>
>> Regards
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Chris.Martineau at semafone.com  Fri Nov 16 12:24:23 2012
From: Chris.Martineau at semafone.com (Chris Martineau)
Date: Fri, 16 Nov 2012 09:24:23 +0000
Subject: [Freeswitch-users] uuid_media leaves behind zombie call
In-Reply-To: <CAKbxfG-YU0Sz2wd0mZ9+rv3MON13f6AYbJcwxPa1GdmsuGaRoQ@mail.gmail.com>
References: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>
	<CAKbxfG-YU0Sz2wd0mZ9+rv3MON13f6AYbJcwxPa1GdmsuGaRoQ@mail.gmail.com>
Message-ID: <870204F45EE7D34E8D27CC0E602E11A1B46979@EX01.semafone.local>

Hi,

Sorry,

Don't post that often and wasn't sure if it was a configuration or known issue that somebody was already aware of and could have been a "you haven't done this or set that type answer"?

I will produce the logs as requested and report as an issue on jira.


Regards

Chris


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: 16 November 2012 00:27
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] uuid_media leaves behind zombie call

Issues should be reported on http://jira.freeswitch.org

You should be reproducing on git HEAD and including full logs as attachments.
Also you need to enable the sofia sip traces.

console loglevel debug
sofia loglevel all 9
sofia global siptrace on

I'd like to figure out the steps you took to send this email so we can see were have continued to fail to make the policy clear,


On Thu, Nov 15, 2012 at 11:08 AM, Chris Martineau <Chris.Martineau at semafone.com<mailto:Chris.Martineau at semafone.com>> wrote:
Hi,

I am trying to use uuid_media to switch in and out of the rtp stream and in essence it seems to work however I get some strange things happening.

The first thing is that when the re-invites go out, the a and the b legs get the codec list in a different order i.e. 0 8 3 to the a leg and 8 0 3 to the b leg!
If all the codecs are supported this forces a transcoding scenario as the a leg neg's pcmu and the b leg pcma! As this is done via the event socket and not the dialplan I cannot see what I am supposed to set to correct this behaviour ( the calls all start in media bypass mode so no codec neg is done at call start).
Any ideas why this should do this and how I can get round it? Setting the no transcoding variable has no effect on this.

The second thing is that if you go into the media path and then going out of it again and then clear down it seems to leave the a leg channel hungup in the system. Show channels lists the channel just sitting in CS_HANGUP state. In the log it says Locked, waiting on external entities?
Everything externally has cleared and wireshark shows a clean dialog with no resends of the BYE messages. Cleardown is from the a leg and FS has passed the BYE on to the b leg.
If you try to kill it with uuid_kill it says that the uuid does not exist? The only way I can clear it is to restart FS which on closing it hangs around with a sofia error saying waiting for 1 session.

If I just go into the media path and cleardown without coming out again the bridge process seems to be broken as any BYE message does not get passed through, requiring both ends to be cleared separately. Also no UNBRIDGE event is fired in this scenario which I suppose you would expect if it wasn't bridged, however rtp is flowing through as is signalling?

The same channel however is still left hanging.

I may be using this incorrectly or not preparing the environment however the info around this command is pretty basic and looks simple to use.

It is pretty critical for the application I am trying to build that this process works smoothly so any help you could offer would be greatly appreciated.

I have attached some debug logs.

Many thanks


Chris



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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From gavin.henry at gmail.com  Fri Nov 16 13:55:32 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 16 Nov 2012 10:55:32 +0000
Subject: [Freeswitch-users] Outcall for FreeSWITCH
Message-ID: <CAA8_NKDcYnPSO9COUnuGKQ6m4CKMwgTKE4Bg6vpwAZZ7ow-nng@mail.gmail.com>

Hi all,

Some customers are currently using SIPTAPI, but wondered if any one is
using something like OutCall that is for Asterisk.

Ideas?

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk
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From jnvines at gmail.com  Fri Nov 16 16:01:22 2012
From: jnvines at gmail.com (Nick Vines)
Date: Fri, 16 Nov 2012 08:01:22 -0500
Subject: [Freeswitch-users] changing vm default announcement
In-Reply-To: <50A5BA00.5050209@gmail.com>
References: <50A5BA00.5050209@gmail.com>
Message-ID: <CA+NjZ2t2RqA0a-AgWjSg3vgKeXzGqvAZ=o3m434Br7T3QPJf7w@mail.gmail.com>

Which directory did you put the sound file?

1) conf/domain/user/greeting_1.wav
or
2) $${base_dir}/storage/voicemail/default/$${domain}/<user>/greeting_1.wav

"2" is listed on the wiki.
http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_greeting_number

Also, I think you will find the voicemail configurations in
conf/lang/en/vm/sounds.xml. If you have the 1.0.6 cookbook, you can read
what was current at that time on page 116.

Nick


On Thu, Nov 15, 2012 at 10:58 PM, andy <packetandy at gmail.com> wrote:

> Hi all,
>
> I am trying to change the default greeting that plays to the caller from
> the standard voicemail/callie when the call is not answered.. I have not
> been able to find where the script is for the ivr that answers the call.
> I tried overriding the default in the dialplan by using :
>
>              <action application="set" data="voicemail_greeting_number=1"/>
>              <action application="set" data="skip_instructions=true"/>
>
> but this did not change anything - the default greeting still plays. I
> have tried to copy a different message into each user's directory as
> greeting_1.wav, but some flag is not been set so callie still plays
>
>   Any suggestions?
>
> drew
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shaheryarkh at googlemail.com  Fri Nov 16 16:31:04 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 16 Nov 2012 14:31:04 +0100
Subject: [Freeswitch-users] Is there any documentation for Voicemail DB
	schema?
In-Reply-To: <CAKzWOxWVjtDk8ab_8qU_WKCfR89AJMnRhehq15ag0xnHfZPGmg@mail.gmail.com>
References: <CAFZQphz9o5JxD4BGqJka5jzcQP-nC10LCaVsT6XUZkPb2vxD6Q@mail.gmail.com>
	<50A52A26.8010306@gmail.com>
	<CAFZQphw-7Ynkr3g-r-NMucCjhKLn5zRmzg+w=N1dOYXfsP4CWg@mail.gmail.com>
	<CAKzWOxWVjtDk8ab_8qU_WKCfR89AJMnRhehq15ag0xnHfZPGmg@mail.gmail.com>
Message-ID: <CAFZQphxfYyaDVg_KF3aiHJjcsvPk7XkDmiNRZRR_tsRMDZ+Oag@mail.gmail.com>

Just updated voicemail module documentation. Hope this helps other.

Thanks to Abaci & MC for your help.

Thank you.


On Fri, Nov 16, 2012 at 1:01 AM, Michael Collins <msc at freeswitch.org> wrote:

> Enabled! Go forth and document.
> -MC
>
>
> On Thu, Nov 15, 2012 at 12:18 PM, Muhammad Shahzad <
> shaheryarkh at googlemail.com> wrote:
>
>> I don't have account on FS Wiki. However, i sent request for it. Lets see
>> if it approves then i will update voicemail docs anytime soon.
>>
>> Thank you.
>>
>>
>> On Thu, Nov 15, 2012 at 6:45 PM, Abaci <abaci64 at gmail.com> wrote:
>>
>>>  flags will be either '' for regular messages, 'save' for saved
>>> messages, or 'delete' for messages marked for deletion.
>>> read_flags will be either 'B_NORMAL' for normal messages, or 'A_URGENT'
>>> for urgent messages.
>>> you can update of the wiki to help the next person.
>>>
>>>
>>> On 11/15/2012 8:48 AM, Muhammad Shahzad wrote:
>>>
>>> Hi,
>>>
>>>  I can find db schema for voicemail module at below url,
>>>
>>>  http://wiki.freeswitch.org/wiki/Mod_voicemail#Database_Schema
>>>
>>>  But unfortunately it does not give any explanation of db fields, for
>>> example i would like to know what are the types and possible values for
>>> flags and read_flags column in voicemail_msgs table and so on.
>>>
>>>  Does anyone has any idea or documentation link for this?
>>>
>>>  Thank you.
>>>
>>>
>>>  --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +49 176 99 83 10 85
>>> MSN: shari_786pk at hotmail.com
>>> Email: shaheryarkh at googlemail.com
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> 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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From Chris.Martineau at semafone.com  Fri Nov 16 17:19:09 2012
From: Chris.Martineau at semafone.com (Chris Martineau)
Date: Fri, 16 Nov 2012 14:19:09 +0000
Subject: [Freeswitch-users] git down?
Message-ID: <870204F45EE7D34E8D27CC0E602E11A1B46A01@EX01.semafone.local>

Hi,

Cannot seem to get the latest git.

Get the following...
Fatal: unable to connect to git.freeswitch.org:
Git.freeswitch.org[0: 198.22.64.222]: errno=connection refused

In case you were not aware?

Regards

Chris
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From krice at freeswitch.org  Fri Nov 16 18:29:23 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Nov 2012 09:29:23 -0600
Subject: [Freeswitch-users] git down?
In-Reply-To: <870204F45EE7D34E8D27CC0E602E11A1B46A01@EX01.semafone.local>
Message-ID: <CCCBB7F3.60B78%krice@freeswitch.org>

It was having some issues earlier... Its been resolved now tho...


On 11/16/12 8:19 AM, "Chris Martineau" <Chris.Martineau at semafone.com> wrote:

> Hi,
>  
> Cannot seem to get the latest git.
>  
> Get the following?
> Fatal: unable to connect to git.freeswitch.org:
> Git.freeswitch.org[0: 198.22.64.222]: errno=connection refused
>  
> In case you were not aware?
>  
> Regards
>  
> Chris
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From sterned at xakep.ru  Fri Nov 16 18:46:11 2012
From: sterned at xakep.ru (sterned)
Date: Fri, 16 Nov 2012 07:46:11 -0800 (PST)
Subject: [Freeswitch-users] problem with receiving and sending faxes
In-Reply-To: <1352398794952-7584436.post@n2.nabble.com>
References: <1352398794952-7584436.post@n2.nabble.com>
Message-ID: <1353080771990-7584627.post@n2.nabble.com>

Hello. So I went to add the person 500 "Local Extension". Checked the caller
goes both ways. Then from the console fs _tseli run your command, still not
accepted.
By the way I use the fax software Kapanga with the default settings, just
enable the option "Detect CNG" Logs are attached.
http://pastebin.freeswitch.org/20227




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problem-with-receiving-and-sending-faxes-tp7584436p7584627.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Nov 16 19:14:01 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Nov 2012 10:14:01 -0600
Subject: [Freeswitch-users] uuid_media leaves behind zombie call
In-Reply-To: <870204F45EE7D34E8D27CC0E602E11A1B46979@EX01.semafone.local>
References: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>
	<CAKbxfG-YU0Sz2wd0mZ9+rv3MON13f6AYbJcwxPa1GdmsuGaRoQ@mail.gmail.com>
	<870204F45EE7D34E8D27CC0E602E11A1B46979@EX01.semafone.local>
Message-ID: <CAKbxfG8q5qn5XPeyNL2nhOmGzUjvbJMJ0qMZTEVnhwfYb5NwHw@mail.gmail.com>

No problem, I just want to try and trace peoples path into the community
and make sure we have all the proper information at first glance.

Even in a case of uncertainty, Jira is better because its a database of
"issues" and we always have a "not a bug" resolution.
Even in cases of mistaken configuration, it can be considered a valid issue
and we can change defaults etc.

Jiras are not expensive to create and easy to close, just a lot more
organized.



On Fri, Nov 16, 2012 at 3:24 AM, Chris Martineau <
Chris.Martineau at semafone.com> wrote:

>  Hi, ****
>
> ** **
>
> Sorry,****
>
> ** **
>
> Don?t post that often and wasn?t sure if it was a configuration or known
> issue that somebody was already aware of and could have been a ?you haven?t
> done this or set that type answer??****
>
> ** **
>
> I will produce the logs as requested and report as an issue on jira.****
>
> ** **
>
> ** **
>
> Regards****
>
> ** **
>
> Chris****
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* 16 November 2012 00:27
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] uuid_media leaves behind zombie call****
>
> ** **
>
> Issues should be reported on http://jira.freeswitch.org****
>
> ** **
>
> You should be reproducing on git HEAD and including full logs as
> attachments.****
>
> Also you need to enable the sofia sip traces.****
>
> ** **
>
> console loglevel debug****
>
> sofia loglevel all 9****
>
> sofia global siptrace on****
>
> ** **
>
> I'd like to figure out the steps you took to send this email so we can see
> were have continued to fail to make the policy clear,****
>
> ** **
>
> ** **
>
> On Thu, Nov 15, 2012 at 11:08 AM, Chris Martineau <
> Chris.Martineau at semafone.com> wrote:****
>
> Hi,****
>
>  ****
>
> I am trying to use uuid_media to switch in and out of the rtp stream and
> in essence it seems to work however I get some strange things happening.**
> **
>
>  ****
>
> The first thing is that when the re-invites go out, the a and the b legs
> get the codec list in a different order i.e. 0 8 3 to the a leg and 8 0 3
> to the b leg!****
>
> If all the codecs are supported this forces a transcoding scenario as the
> a leg neg?s pcmu and the b leg pcma! As this is done via the event socket
> and not the dialplan I cannot see what I am supposed to set to correct this
> behaviour ( the calls all start in media bypass mode so no codec neg is
> done at call start).****
>
> Any ideas why this should do this and how I can get round it? Setting the
> no transcoding variable has no effect on this.****
>
>  ****
>
> The second thing is that if you go into the media path and then going out
> of it again and then clear down it seems to leave the a leg channel hungup
> in the system. Show channels lists the channel just sitting in CS_HANGUP
> state. In the log it says Locked, waiting on external entities?****
>
> Everything externally has cleared and wireshark shows a clean dialog with
> no resends of the BYE messages. Cleardown is from the a leg and FS has
> passed the BYE on to the b leg.****
>
> If you try to kill it with uuid_kill it says that the uuid does not exist?
> The only way I can clear it is to restart FS which on closing it hangs
> around with a sofia error saying waiting for 1 session.****
>
>  ****
>
> If I just go into the media path and cleardown without coming out again
> the bridge process seems to be broken as any BYE message does not get
> passed through, requiring both ends to be cleared separately. Also no
> UNBRIDGE event is fired in this scenario which I suppose you would expect
> if it wasn?t bridged, however rtp is flowing through as is signalling?****
>
>  ****
>
> The same channel however is still left hanging.****
>
>  ****
>
> I may be using this incorrectly or not preparing the environment however
> the info around this command is pretty basic and looks simple to use.****
>
>  ****
>
> It is pretty critical for the application I am trying to build that this
> process works smoothly so any help you could offer would be greatly
> appreciated.****
>
>  ****
>
> I have attached some debug logs.****
>
>  ****
>
> Many thanks****
>
>  ****
>
>  ****
>
> Chris****
>
>  ****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From james.bravo at redmatter.com  Fri Nov 16 20:08:36 2012
From: james.bravo at redmatter.com (James Bravo)
Date: Fri, 16 Nov 2012 17:08:36 +0000
Subject: [Freeswitch-users] Response time of ESL interface has increased a
 great deal from version 1.2 to 1.3.2
Message-ID: <50A67314.1080803@redmatter.com>

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From yiftah at choochee.com  Fri Nov 16 20:45:32 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 16 Nov 2012 09:45:32 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CCCB02A0.60B4A%krice@freeswitch.org>
References: <CACc3D0X3FonmA0abxp7rRD4032bG+H-cGbCaGOEDiXMk0gnxUw@mail.gmail.com>
	<CCCB02A0.60B4A%krice@freeswitch.org>
Message-ID: <CAFVUaV875kE+rXMQEnFcUf20FwDH46bntUrGEhVQB=bwBTDsPQ@mail.gmail.com>

I know that it is kind out of the what RFC3261 instructs, but did anyone
think on giving the option in configuration not to hang up calls in case of
an ACK does not arrive?
I know that it has the risk of open sessions but there some other ways to
handle those cases

On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org> wrote:

>  This is probably the same scenario as this is exactly what to expect...
> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
> the call....
>
> Quite common on networks with NAT issues or broken endpoints
>
>
> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com> wrote:
>
> I saw this happened earlier when the remote party does not send SIP ACK
> after receiving SIP OK, so the call is being disconnected after exactly 32
> seconds.
> Not sure if this is exact same scenario here, but just something to
> consider...
>
> Regards.
> Vitalie
>
>
> 2012/11/15 kaleem rehman <k4kaleem at gmail.com>
>
> Hi All,
>
> my inbound calls are fine with no issues, my outbound calls get
> disconnected after 32 seconds and its on all calls. i tried 2 different
> suppliers and its same result.
> please find the attached log file with sofia in debug mode. - caller was
> extension 1234 and desination was 01908321682
>
> your help will be greately appreciated.
>
> regards,
> Kaleem
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Fri Nov 16 21:27:46 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Nov 2012 12:27:46 -0600
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV875kE+rXMQEnFcUf20FwDH46bntUrGEhVQB=bwBTDsPQ@mail.gmail.com>
Message-ID: <CCCBE1C2.60B86%krice@freeswitch.org>

That leaves to big a risk of open sessions and only masks the true issue
which is a problem with FS getting the ACK back...

Theres a reason FS is not getting the ACK, and FS will make several attempts
to get an ack by retransmitting the 200 OK several times before that timeout
occurs.

The real fix here is to fix the underlying cause, not masking it....


On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com> wrote:

> I know that it is kind out of the what RFC3261 instructs, but did anyone think
> on giving the option in configuration not to hang up calls in case of an ACK
> does not arrive?
> I know that it has the risk of open sessions but there some other ways to
> handle those cases
> ? 
> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org> wrote:
>> This is probably the same scenario as this is exactly what to expect... Call
>> gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup the
>> call....
>> 
>> Quite common on networks with NAT issues or broken endpoints
>> 
>> 
>> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com
>> <http://vetali100 at gmail.com> > wrote:
>> 
>>> I saw this happened earlier when the remote party does not send SIP ACK
>>> after receiving SIP OK, so the call is being disconnected after exactly 32
>>> seconds.
>>> Not sure if this is exact same scenario here, but just something to
>>> consider...
>>> 
>>> Regards.
>>> Vitalie
>>> 
>>> 
>>> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com> >
>>>> Hi All,
>>>> ?
>>>> my inbound calls are fine with no issues, my outbound calls get
>>>> disconnected after 32 seconds and its on all calls. i tried 2 different
>>>> suppliers and its same result.
>>>> please find the attached log file with sofia in debug mode. - caller was
>>>> extension 1234 and desination was 01908321682
>>>> ?
>>>> your help will be greately appreciated.
>>>> ?
>>>> regards,
>>>> Kaleem
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org

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

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From msc at freeswitch.org  Fri Nov 16 22:24:56 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Nov 2012 11:24:56 -0800
Subject: [Freeswitch-users] problem with receiving and sending faxes
In-Reply-To: <1353080771990-7584627.post@n2.nabble.com>
References: <1352398794952-7584436.post@n2.nabble.com>
	<1353080771990-7584627.post@n2.nabble.com>
Message-ID: <CAKzWOxVvGmNznXcuSw+BEJmR0ZwvU7zijo5ynLMO2HUa5X2VGQ@mail.gmail.com>

Well, it says that the call dropped prematurely. I'd get the SIP trace of
this call and see if there's any more information in there.
-MC

On Fri, Nov 16, 2012 at 7:46 AM, sterned <sterned at xakep.ru> wrote:

> Hello. So I went to add the person 500 "Local Extension". Checked the
> caller
> goes both ways. Then from the console fs _tseli run your command, still not
> accepted.
> By the way I use the fax software Kapanga with the default settings, just
> enable the option "Detect CNG" Logs are attached.
> http://pastebin.freeswitch.org/20227
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/problem-with-receiving-and-sending-faxes-tp7584436p7584627.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From yiftah at choochee.com  Fri Nov 16 23:19:02 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 16 Nov 2012 12:19:02 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CCCBE1C2.60B86%krice@freeswitch.org>
References: <CAFVUaV875kE+rXMQEnFcUf20FwDH46bntUrGEhVQB=bwBTDsPQ@mail.gmail.com>
	<CCCBE1C2.60B86%krice@freeswitch.org>
Message-ID: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>

While I agree on the details I disagree on the solution
Sometimes masking the problems can be a good solution but I guess it is a
philosophical debate


On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:

>  That leaves to big a risk of open sessions and only masks the true issue
> which is a problem with FS getting the ACK back...
>
> Theres a reason FS is not getting the ACK, and FS will make several
> attempts to get an ack by retransmitting the 200 OK several times before
> that timeout occurs.
>
> The real fix here is to fix the underlying cause, not masking it....
>
>
> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>
> I know that it is kind out of the what RFC3261 instructs, but did anyone
> think on giving the option in configuration not to hang up calls in case of
> an ACK does not arrive?
> I know that it has the risk of open sessions but there some other ways to
> handle those cases
>
> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> This is probably the same scenario as this is exactly what to expect...
> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
> the call....
>
> Quite common on networks with NAT issues or broken endpoints
>
>
> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
> http://vetali100 at gmail.com> > wrote:
>
> I saw this happened earlier when the remote party does not send SIP ACK
> after receiving SIP OK, so the call is being disconnected after exactly 32
> seconds.
> Not sure if this is exact same scenario here, but just something to
> consider...
>
> Regards.
> Vitalie
>
>
> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com> >
>
> Hi All,
>
> my inbound calls are fine with no issues, my outbound calls get
> disconnected after 32 seconds and its on all calls. i tried 2 different
> suppliers and its same result.
> please find the attached log file with sofia in debug mode. - caller was
> extension 1234 and desination was 01908321682
>
> your help will be greately appreciated.
>
> regards,
> Kaleem
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Fri Nov 16 23:29:26 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 16 Nov 2012 20:29:26 +0000
Subject: [Freeswitch-users] Response time of ESL interface has increased
 a great deal from version 1.2 to 1.3.2
Message-ID: <ojj9vdo4djm3vh609lkxx9x8.1353097761179@email.android.com>

Did you read my response on the jira.freeswitch.org issue?


/Peter

James Bravo <james.bravo at redmatter.com> skrev:
Hi

Just wondering if anyone else has experienced the following issue with later Freeswitch.

Using a simple php script (see below) to issue 20 ESL requests and display the response
times shows that the response time has increased from an average of 0.002 secs per
request for version 1.2 to 0.06 for version 1.3.2 .

I've tried issuing requests to localhost, 127.0.0.1 and from other hosts.
The results are always much slowed for version 1.3.2 .

The test php script is as follows:

<?php
require_once 'ESL.php';

$host = "localhost";
if ($argc >= 2)
        $host = $argv[1];

$sockFSServer = new ESLconnection($host, '8021', 'ClueCon');

function getTime() {
        list($usec, $sec) = explode(" ", microtime());
        return ((float)$usec + (float)$sec);
}

for($n=0; $n < 20; $n++) {
        $startTime = getTime();
        $res = $sockFSServer->sendRecv("api global_getvar local_ip_v4");
        if (!$res)
                echo "request failed\n";
        else
                echo "Got reply '" . $res->getBody() . "'. Time taken " . (getTime() - $startTime) . "\n";
}



cheers James Bravo

!DSPAM:50a6775532767648055020!

From krice at freeswitch.org  Fri Nov 16 23:58:13 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Nov 2012 14:58:13 -0600
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
Message-ID: <CCCC0505.60B94%krice@freeswitch.org>

In this case masking the issue can lead to massive bills... Imaging paying
by the minute... All of a sudden you are now leaving 2 minute calls up for
hours on end... And continuing to get billed for them... Or you are not
continuing to bill a customer for them... And now you have unexpected HUGE
bills coming in... Masking it is far worse then just fixing it...

If we mask this one issue, we might as well mask memory leaks, or passwords
that don?t work, etc... Sure sometimes we might have to mask an issue for
production to work in the short term, but that is never the correct answer
fix a problem


On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:

> While I agree on the details I disagree on the solution
> Sometimes masking the problems can be a good solution but I guess it is a
> philosophical debate
> ?? 
> 
> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>> That leaves to big a risk of open sessions and only masks the true issue
>> which is a problem with FS getting the ACK back...
>> 
>> Theres a reason FS is not getting the ACK, and FS will make several attempts
>> to get an ack by retransmitting the 200 OK several times before that timeout
>> occurs.
>> 
>> The real fix here is to fix the underlying cause, not masking it....
>> 
>> 
>> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com
>> <http://yiftah at choochee.com> > wrote:
>> 
>>> I know that it is kind out of the what RFC3261 instructs, but did anyone
>>> think on giving the option in configuration not to hang up calls in case of
>>> an ACK does not arrive?
>>> I know that it has the risk of open sessions but there some other ways to
>>> handle those cases
>>> ? 
>>> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org
>>> <http://krice at freeswitch.org> > wrote:
>>>> This is probably the same scenario as this is exactly what to expect...
>>>> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
>>>> the call....
>>>> 
>>>> Quite common on networks with NAT issues or broken endpoints
>>>> 
>>>> 
>>>> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com
>>>> <http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>>> 
>>>>> I saw this happened earlier when the remote party does not send SIP ACK
>>>>> after receiving SIP OK, so the call is being disconnected after exactly 32
>>>>> seconds.
>>>>> Not sure if this is exact same scenario here, but just something to
>>>>> consider...
>>>>> 
>>>>> Regards.
>>>>> Vitalie
>>>>> 
>>>>> 
>>>>> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
>>>>> <http://k4kaleem at gmail.com> >
>>>>>> Hi All,
>>>>>> ?
>>>>>> my inbound calls are fine with no issues, my outbound calls get
>>>>>> disconnected after 32 seconds and its on all calls. i tried 2 different
>>>>>> suppliers and its same result.
>>>>>> please find the attached log file with sofia in debug mode. - caller was
>>>>>> extension 1234 and desination was 01908321682
>>>>>> ?
>>>>>> your help will be greately appreciated.
>>>>>> ?
>>>>>> regards,
>>>>>> Kaleem
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>>>> <http://consulting at freeswitch.org>
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>>> <http://consulting at freeswitch.org>
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org

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

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From msc at freeswitch.org  Sat Nov 17 00:21:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Nov 2012 13:21:09 -0800
Subject: [Freeswitch-users] Friday Free-for-all!
Message-ID: <CAKzWOxVepyZ-puUBM7ikZBCEjvQFpTXMUEWaUoH7-URTrykuyA@mail.gmail.com>

Come join us! We're having a nice chat in the main conf room

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From yiftah at choochee.com  Sat Nov 17 00:56:58 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 16 Nov 2012 13:56:58 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CCCC0505.60B94%krice@freeswitch.org>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
Message-ID: <CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>

This is where I am getting a bit confused, if the 200OK arrived to the
other side and we checked that the RTP exists (with mod_sofia option) we
will not get to hours of calls (unless the other side did not hanged up)
In any case there is a good chance that the BYE is getting lost, so the
danger exist even without the ACK

I am guessing that the designers of the SIP protocol came up with the ACK
because there is a potential for open dialog that is not bound with time
and they therefore wanted to know that the other side actually ACKs the
request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
tied with RTP (at least in most normal SIP implementations) I'm not sure
that this ACK is that needed
but again as I said it is more of philosophical debate, maybe a potential
request in the new RFC

On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:

>  In this case masking the issue can lead to massive bills... Imaging
> paying by the minute... All of a sudden you are now leaving 2 minute calls
> up for hours on end... And continuing to get billed for them... Or you are
> not continuing to bill a customer for them... And now you have unexpected
> HUGE bills coming in... Masking it is far worse then just fixing it...
>
> If we mask this one issue, we might as well mask memory leaks, or
> passwords that don?t work, etc... Sure sometimes we might have to mask an
> issue for production to work in the short term, but that is never the
> correct answer fix a problem
>
>
> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>
> While I agree on the details I disagree on the solution
> Sometimes masking the problems can be a good solution but I guess it is a
> philosophical debate
>
>
> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> That leaves to big a risk of open sessions and only masks the true issue
> which is a problem with FS getting the ACK back...
>
> Theres a reason FS is not getting the ACK, and FS will make several
> attempts to get an ack by retransmitting the 200 OK several times before
> that timeout occurs.
>
> The real fix here is to fix the underlying cause, not masking it....
>
>
> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
> http://yiftah at choochee.com> > wrote:
>
> I know that it is kind out of the what RFC3261 instructs, but did anyone
> think on giving the option in configuration not to hang up calls in case of
> an ACK does not arrive?
> I know that it has the risk of open sessions but there some other ways to
> handle those cases
>
> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
> http://krice at freeswitch.org> > wrote:
>
> This is probably the same scenario as this is exactly what to expect...
> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
> the call....
>
> Quite common on networks with NAT issues or broken endpoints
>
>
> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>
> I saw this happened earlier when the remote party does not send SIP ACK
> after receiving SIP OK, so the call is being disconnected after exactly 32
> seconds.
> Not sure if this is exact same scenario here, but just something to
> consider...
>
> Regards.
> Vitalie
>
>
> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
>  <http://k4kaleem at gmail.com> >
>
> Hi All,
>
> my inbound calls are fine with no issues, my outbound calls get
> disconnected after 32 seconds and its on all calls. i tried 2 different
> suppliers and its same result.
> please find the attached log file with sofia in debug mode. - caller was
> extension 1234 and desination was 01908321682
>
> your help will be greately appreciated.
>
> regards,
> Kaleem
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>  <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>  <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Sat Nov 17 01:31:03 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 16 Nov 2012 17:31:03 -0500
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
Message-ID: <1546109.RqNAKPZJSR@sos>

 Don't mix signaling (SIP) and media (RTP). Signaling and media could run 
different ways. Why do you think FS will always be in media path?

On Friday 16 November 2012 13:56:58 Yiftach Golan wrote:
> This is where I am getting a bit confused, if the 200OK arrived to the
> other side and we checked that the RTP exists (with mod_sofia option) we
> will not get to hours of calls (unless the other side did not hanged up)
> In any case there is a good chance that the BYE is getting lost, so the
> danger exist even without the ACK
> 
> I am guessing that the designers of the SIP protocol came up with the ACK
> because there is a potential for open dialog that is not bound with time
> and they therefore wanted to know that the other side actually ACKs the
> request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
> tied with RTP (at least in most normal SIP implementations) I'm not sure
> that this ACK is that needed
> but again as I said it is more of philosophical debate, maybe a potential
> request in the new RFC
> 
> On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
> >  In this case masking the issue can lead to massive bills... Imaging
> > 
> > paying by the minute... All of a sudden you are now leaving 2 minute calls
> > up for hours on end... And continuing to get billed for them... Or you are
> > not continuing to bill a customer for them... And now you have unexpected
> > HUGE bills coming in... Masking it is far worse then just fixing it...
> > 
> > If we mask this one issue, we might as well mask memory leaks, or
> > passwords that don?t work, etc... Sure sometimes we might have to mask an
> > issue for production to work in the short term, but that is never the
> > correct answer fix a problem
> > 
> > 
> > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
> > 
> > While I agree on the details I disagree on the solution
> > Sometimes masking the problems can be a good solution but I guess it is a
> > philosophical debate
> > 
> > 
> > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
> > 
> > That leaves to big a risk of open sessions and only masks the true issue
> > which is a problem with FS getting the ACK back...
> > 
> > Theres a reason FS is not getting the ACK, and FS will make several
> > attempts to get an ack by retransmitting the 200 OK several times before
> > that timeout occurs.
> > 
> > The real fix here is to fix the underlying cause, not masking it....
> > 
> > 
> > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
> > http://yiftah at choochee.com> > wrote:
> > 
> > I know that it is kind out of the what RFC3261 instructs, but did anyone
> > think on giving the option in configuration not to hang up calls in case
> > of
> > an ACK does not arrive?
> > I know that it has the risk of open sessions but there some other ways to
> > handle those cases
> > 
> > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
> > http://krice at freeswitch.org> > wrote:
> > 
> > This is probably the same scenario as this is exactly what to expect...
> > Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
> > hangsup
> > the call....
> > 
> > Quite common on networks with NAT issues or broken endpoints
> > 
> > 
> > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
> > http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
> > 
> > I saw this happened earlier when the remote party does not send SIP ACK
> > after receiving SIP OK, so the call is being disconnected after exactly 32
> > seconds.
> > Not sure if this is exact same scenario here, but just something to
> > consider...
> > 
> > Regards.
> > Vitalie
> > 
> > 
> > 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
> > 
> >  <http://k4kaleem at gmail.com> >
> > 
> > Hi All,
> > 
> > my inbound calls are fine with no issues, my outbound calls get
> > disconnected after 32 seconds and its on all calls. i tried 2 different
> > suppliers and its same result.
> > please find the attached log file with sofia in debug mode. - caller was
> > extension 1234 and desination was 01908321682
> > 
> > your help will be greately appreciated.
> > 
> > regards,
> > Kaleem
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> > http://consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <
> > http://FreeSWITCH-users at lists.freeswitch.org>  <
> > http://FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> > 
> > ------------------------------
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> > http://consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <
> > http://FreeSWITCH-users at lists.freeswitch.org>  <
> > http://FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> > --
> > Ken
> > *http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> > *irc.freenode.net #freeswitch
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From vetali100 at gmail.com  Sat Nov 17 02:18:23 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Fri, 16 Nov 2012 15:18:23 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
Message-ID: <CACc3D0UZ1E-taSeBpftYPRNm_uDGVzgfZEuGiTsF8FYJAbwmhQ@mail.gmail.com>

What if RTP does not flow via FreeSWITCH but directly between the
endpoints. Then ACK is the only way to send a signal that endpoint started
to send RTP.
Not sure if it sends ACK before starting sending RTP or after... or it is
really not related to RTP, if so - why indeed ACK is needed?


2012/11/16 Yiftach Golan <yiftah at choochee.com>

> This is where I am getting a bit confused, if the 200OK arrived to the
> other side and we checked that the RTP exists (with mod_sofia option) we
> will not get to hours of calls (unless the other side did not hanged up)
> In any case there is a good chance that the BYE is getting lost, so the
> danger exist even without the ACK
>
> I am guessing that the designers of the SIP protocol came up with the ACK
> because there is a potential for open dialog that is not bound with time
> and they therefore wanted to know that the other side actually ACKs the
> request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
> tied with RTP (at least in most normal SIP implementations) I'm not sure
> that this ACK is that needed
> but again as I said it is more of philosophical debate, maybe a potential
> request in the new RFC
>
> On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  In this case masking the issue can lead to massive bills... Imaging
>> paying by the minute... All of a sudden you are now leaving 2 minute calls
>> up for hours on end... And continuing to get billed for them... Or you are
>> not continuing to bill a customer for them... And now you have unexpected
>> HUGE bills coming in... Masking it is far worse then just fixing it...
>>
>> If we mask this one issue, we might as well mask memory leaks, or
>> passwords that don?t work, etc... Sure sometimes we might have to mask an
>> issue for production to work in the short term, but that is never the
>> correct answer fix a problem
>>
>>
>> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>
>> While I agree on the details I disagree on the solution
>> Sometimes masking the problems can be a good solution but I guess it is a
>> philosophical debate
>>
>>
>> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> That leaves to big a risk of open sessions and only masks the true issue
>> which is a problem with FS getting the ACK back...
>>
>> Theres a reason FS is not getting the ACK, and FS will make several
>> attempts to get an ack by retransmitting the 200 OK several times before
>> that timeout occurs.
>>
>> The real fix here is to fix the underlying cause, not masking it....
>>
>>
>> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>> http://yiftah at choochee.com> > wrote:
>>
>> I know that it is kind out of the what RFC3261 instructs, but did anyone
>> think on giving the option in configuration not to hang up calls in case of
>> an ACK does not arrive?
>> I know that it has the risk of open sessions but there some other ways to
>> handle those cases
>>
>> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>> http://krice at freeswitch.org> > wrote:
>>
>> This is probably the same scenario as this is exactly what to expect...
>> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
>> the call....
>>
>> Quite common on networks with NAT issues or broken endpoints
>>
>>
>> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>
>> I saw this happened earlier when the remote party does not send SIP ACK
>> after receiving SIP OK, so the call is being disconnected after exactly 32
>> seconds.
>> Not sure if this is exact same scenario here, but just something to
>> consider...
>>
>> Regards.
>> Vitalie
>>
>>
>> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
>>  <http://k4kaleem at gmail.com> >
>>
>> Hi All,
>>
>> my inbound calls are fine with no issues, my outbound calls get
>> disconnected after 32 seconds and its on all calls. i tried 2 different
>> suppliers and its same result.
>> please find the attached log file with sofia in debug mode. - caller was
>> extension 1234 and desination was 01908321682
>>
>> your help will be greately appreciated.
>>
>> regards,
>> Kaleem
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Ken
>> *http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> *irc.freenode.net #freeswitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lconroy at insensate.co.uk  Sat Nov 17 02:23:37 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Fri, 16 Nov 2012 23:23:37 +0000
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
Message-ID: <D775C81D-ABCC-4D35-9102-9AC34FFEDCD5@insensate.co.uk>

Hi There,
 um ... the folk who thought up SIP used the Z-pattern for session initiation
to ensure synchronisation between the parties. That's also the reason the path
for the ACK is party-to-party direct, unlike the INVITE/200.

There's a real good reason for that -- it's a call, so both *ends* need to know
what state they're in -- as everyone here has been telling you.

Trust me -- I was there, and the discussions were intense/painful.
What you're proposing is exactly what was ditched back in 1997 -- RTP-based
call completion. This is brittle -- it's being carried over UDP.

No-RTP call abort is useful (check out 3GPP's discussion on this and how IMS
works -- in particular, the discussion on call handling when one end goes into
a tunnel), but it is only a fallback, and is nasty when call & media go
different routes.

Might I respectfully suggest that, if you think anyone is going to go over
that again to produce a new RFC, you really should cut down on the crack.

Ignoring failure to receive ACK (after a whole series of attempts) is ignoring
your call control channel being, to use the technical term, b*gg*r*d. Just say no.

all the best,
  Lawrence

On 16 Nov 2012, at 21:56, Yiftach Golan wrote:

> This is where I am getting a bit confused, if the 200OK arrived to the
> other side and we checked that the RTP exists (with mod_sofia option) we
> will not get to hours of calls (unless the other side did not hanged up)
> In any case there is a good chance that the BYE is getting lost, so the
> danger exist even without the ACK
> 
> I am guessing that the designers of the SIP protocol came up with the ACK
> because there is a potential for open dialog that is not bound with time
> and they therefore wanted to know that the other side actually ACKs the
> request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
> tied with RTP (at least in most normal SIP implementations) I'm not sure
> that this ACK is that needed
> but again as I said it is more of philosophical debate, maybe a potential
> request in the new RFC
> 
> On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
> 
>> In this case masking the issue can lead to massive bills... Imaging
>> paying by the minute... All of a sudden you are now leaving 2 minute calls
>> up for hours on end... And continuing to get billed for them... Or you are
>> not continuing to bill a customer for them... And now you have unexpected
>> HUGE bills coming in... Masking it is far worse then just fixing it...
>> 
>> If we mask this one issue, we might as well mask memory leaks, or
>> passwords that don?t work, etc... Sure sometimes we might have to mask an
>> issue for production to work in the short term, but that is never the
>> correct answer fix a problem
>> 
>> 
>> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>> 
>> While I agree on the details I disagree on the solution
>> Sometimes masking the problems can be a good solution but I guess it is a
>> philosophical debate
>> 
>> 
>> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>> 
>> That leaves to big a risk of open sessions and only masks the true issue
>> which is a problem with FS getting the ACK back...
>> 
>> Theres a reason FS is not getting the ACK, and FS will make several
>> attempts to get an ack by retransmitting the 200 OK several times before
>> that timeout occurs.
>> 
>> The real fix here is to fix the underlying cause, not masking it....
>> 
>> 
>> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>> http://yiftah at choochee.com> > wrote:
>> 
>> I know that it is kind out of the what RFC3261 instructs, but did anyone
>> think on giving the option in configuration not to hang up calls in case of
>> an ACK does not arrive?
>> I know that it has the risk of open sessions but there some other ways to
>> handle those cases
>> 
>> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>> http://krice at freeswitch.org> > wrote:
>> 
>> This is probably the same scenario as this is exactly what to expect...
>> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
>> the call....
>> 
>> Quite common on networks with NAT issues or broken endpoints
>> 
>> 
>> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>> 
>> I saw this happened earlier when the remote party does not send SIP ACK
>> after receiving SIP OK, so the call is being disconnected after exactly 32
>> seconds.
>> Not sure if this is exact same scenario here, but just something to
>> consider...
>> 
>> Regards.
>> Vitalie
>> 
>> 
>> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
>> <http://k4kaleem at gmail.com> >
>> 
>> Hi All,
>> 
>> my inbound calls are fine with no issues, my outbound calls get
>> disconnected after 32 seconds and its on all calls. i tried 2 different
>> suppliers and its same result.
>> please find the attached log file with sofia in debug mode. - caller was
>> extension 1234 and desination was 01908321682
>> 
>> your help will be greately appreciated.
>> 
>> regards,
>> Kaleem
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> --
>> Ken
>> *http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> *irc.freenode.net #freeswitch
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From yiftah at choochee.com  Sat Nov 17 02:50:19 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 16 Nov 2012 15:50:19 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <1546109.RqNAKPZJSR@sos>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<1546109.RqNAKPZJSR@sos>
Message-ID: <CAFVUaV9VnVwVeNn_KFHZwV3Q4N5u8rrcMv5RbPzfh143j29KTA@mail.gmail.com>

Yes I you are right if the RTP is not tied with SIP this theory is not
really valid
But this is again true about BYE that does not reach to the destination so
the risk of not closing the dialog still exist
In most of the voice implementation that I saw there are three options :
1. Softswitch (FreeSWITCH, Asterisk, etc)
2. MGW (Avaya, Cisco, etc)
3. Media release but only between two phones
You can tie your RTP to the SIP with option 1 and option 2
Option 3 is the problematic one but you will never release your media for
billing purpose so usually your risk will be extension to extension open
dialog which is less riskier
Again as I said it is pretty philosophical debate but from my long
experience in SIP I do not think that there is a practical use for it but
maybe I am missing something

On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

>  Don't mix signaling (SIP) and media (RTP). Signaling and media could run
> different ways. Why do you think FS will always be in media path?
>
> On Friday 16 November 2012 14:56:58 Yiftach Golan wrote:
> > This is where I am getting a bit confused, if the 200OK arrived to the
> > other side and we checked that the RTP exists (with mod_sofia option) we
> > will not get to hours of calls (unless the other side did not hanged up)
> > In any case there is a good chance that the BYE is getting lost, so the
> > danger exist even without the ACK
> >
> > I am guessing that the designers of the SIP protocol came up with the ACK
> > because there is a potential for open dialog that is not bound with time
> > and they therefore wanted to know that the other side actually ACKs the
> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
> > tied with RTP (at least in most normal SIP implementations) I'm not sure
> > that this ACK is that needed
> > but again as I said it is more of philosophical debate, maybe a potential
> > request in the new RFC
> >
> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
> > >  In this case masking the issue can lead to massive bills... Imaging
> > >
> > > paying by the minute... All of a sudden you are now leaving 2 minute
> calls
> > > up for hours on end... And continuing to get billed for them... Or you
> are
> > > not continuing to bill a customer for them... And now you have
> unexpected
> > > HUGE bills coming in... Masking it is far worse then just fixing it...
> > >
> > > If we mask this one issue, we might as well mask memory leaks, or
> > > passwords that don?t work, etc... Sure sometimes we might have to mask
> an
> > > issue for production to work in the short term, but that is never the
> > > correct answer fix a problem
> > >
> > >
> > > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
> > >
> > > While I agree on the details I disagree on the solution
> > > Sometimes masking the problems can be a good solution but I guess it
> is a
> > > philosophical debate
> > >
> > >
> > > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
> wrote:
> > >
> > > That leaves to big a risk of open sessions and only masks the true
> issue
> > > which is a problem with FS getting the ACK back...
> > >
> > > Theres a reason FS is not getting the ACK, and FS will make several
> > > attempts to get an ack by retransmitting the 200 OK several times
> before
> > > that timeout occurs.
> > >
> > > The real fix here is to fix the underlying cause, not masking it....
> > >
> > >
> > > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
> > > http://yiftah at choochee.com> > wrote:
> > >
> > > I know that it is kind out of the what RFC3261 instructs, but did
> anyone
> > > think on giving the option in configuration not to hang up calls in
> case
> > > of
> > > an ACK does not arrive?
> > > I know that it has the risk of open sessions but there some other ways
> to
> > > handle those cases
> > >
> > > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
> > > http://krice at freeswitch.org> > wrote:
> > >
> > > This is probably the same scenario as this is exactly what to expect...
> > > Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
> > > hangsup
> > > the call....
> > >
> > > Quite common on networks with NAT issues or broken endpoints
> > >
> > >
> > > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
> > > http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
> > >
> > > I saw this happened earlier when the remote party does not send SIP ACK
> > > after receiving SIP OK, so the call is being disconnected after
> exactly 32
> > > seconds.
> > > Not sure if this is exact same scenario here, but just something to
> > > consider...
> > >
> > > Regards.
> > > Vitalie
> > >
> > >
> > > 2012/11/15 kaleem rehman <k4kaleem at gmail.com <
> http://k4kaleem at gmail.com>
> > >
> > >  <http://k4kaleem at gmail.com> >
> > >
> > > Hi All,
> > >
> > > my inbound calls are fine with no issues, my outbound calls get
> > > disconnected after 32 seconds and its on all calls. i tried 2 different
> > > suppliers and its same result.
> > > please find the attached log file with sofia in debug mode. - caller
> was
> > > extension 1234 and desination was 01908321682
> > >
> > > your help will be greately appreciated.
> > >
> > > regards,
> > > Kaleem
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> > > http://consulting at freeswitch.org>
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org <
> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
> > > http://FreeSWITCH-users at lists.freeswitch.org>
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > ------------------------------
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> > > http://consulting at freeswitch.org>
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org <
> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
> > > http://FreeSWITCH-users at lists.freeswitch.org>
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > > --
> > > Ken
> > > *http://www.FreeSWITCH.org
> > > http://www.ClueCon.com
> > > http://www.OSTAG.org
> > > *irc.freenode.net #freeswitch
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yiftah at choochee.com  Sat Nov 17 03:07:13 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 16 Nov 2012 16:07:13 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV9VnVwVeNn_KFHZwV3Q4N5u8rrcMv5RbPzfh143j29KTA@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<1546109.RqNAKPZJSR@sos>
	<CAFVUaV9VnVwVeNn_KFHZwV3Q4N5u8rrcMv5RbPzfh143j29KTA@mail.gmail.com>
Message-ID: <CAFVUaV-hPaPUZLQtp+J_VLwHfYNnMQE_TzcUymPqurmbm6AmAQ@mail.gmail.com>

Here is the reason from the RFC3261 :
   "...The reason for this separation is rooted in the importance of
      delivering all 200 (OK) responses to an INVITE to the UAC.  To
      deliver them all to the UAC, the UAS alone takes responsibility
      for retransmitting them (see Section 13.3.1.4), and the UAC alone
      takes responsibility for acknowledging them with ACK (see Section
      13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
      effectively considered its own transaction..."

On Fri, Nov 16, 2012 at 3:50 PM, Yiftach Golan <yiftah at choochee.com> wrote:

> Yes I you are right if the RTP is not tied with SIP this theory is not
> really valid
> But this is again true about BYE that does not reach to the destination so
> the risk of not closing the dialog still exist
> In most of the voice implementation that I saw there are three options :
> 1. Softswitch (FreeSWITCH, Asterisk, etc)
> 2. MGW (Avaya, Cisco, etc)
> 3. Media release but only between two phones
> You can tie your RTP to the SIP with option 1 and option 2
> Option 3 is the problematic one but you will never release your media for
> billing purpose so usually your risk will be extension to extension open
> dialog which is less riskier
> Again as I said it is pretty philosophical debate but from my long
> experience in SIP I do not think that there is a practical use for it but
> maybe I am missing something
>
> On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>>  Don't mix signaling (SIP) and media (RTP). Signaling and media could run
>> different ways. Why do you think FS will always be in media path?
>>
>> On Friday 16 November 2012 14:56:58 Yiftach Golan wrote:
>> > This is where I am getting a bit confused, if the 200OK arrived to the
>> > other side and we checked that the RTP exists (with mod_sofia option) we
>> > will not get to hours of calls (unless the other side did not hanged up)
>> > In any case there is a good chance that the BYE is getting lost, so the
>> > danger exist even without the ACK
>> >
>> > I am guessing that the designers of the SIP protocol came up with the
>> ACK
>> > because there is a potential for open dialog that is not bound with time
>> > and they therefore wanted to know that the other side actually ACKs the
>> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
>> > tied with RTP (at least in most normal SIP implementations) I'm not sure
>> > that this ACK is that needed
>> > but again as I said it is more of philosophical debate, maybe a
>> potential
>> > request in the new RFC
>> >
>> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org>
>> wrote:
>> > >  In this case masking the issue can lead to massive bills... Imaging
>> > >
>> > > paying by the minute... All of a sudden you are now leaving 2 minute
>> calls
>> > > up for hours on end... And continuing to get billed for them... Or
>> you are
>> > > not continuing to bill a customer for them... And now you have
>> unexpected
>> > > HUGE bills coming in... Masking it is far worse then just fixing it...
>> > >
>> > > If we mask this one issue, we might as well mask memory leaks, or
>> > > passwords that don?t work, etc... Sure sometimes we might have to
>> mask an
>> > > issue for production to work in the short term, but that is never the
>> > > correct answer fix a problem
>> > >
>> > >
>> > > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>> > >
>> > > While I agree on the details I disagree on the solution
>> > > Sometimes masking the problems can be a good solution but I guess it
>> is a
>> > > philosophical debate
>> > >
>> > >
>> > > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
>> wrote:
>> > >
>> > > That leaves to big a risk of open sessions and only masks the true
>> issue
>> > > which is a problem with FS getting the ACK back...
>> > >
>> > > Theres a reason FS is not getting the ACK, and FS will make several
>> > > attempts to get an ack by retransmitting the 200 OK several times
>> before
>> > > that timeout occurs.
>> > >
>> > > The real fix here is to fix the underlying cause, not masking it....
>> > >
>> > >
>> > > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>> > > http://yiftah at choochee.com> > wrote:
>> > >
>> > > I know that it is kind out of the what RFC3261 instructs, but did
>> anyone
>> > > think on giving the option in configuration not to hang up calls in
>> case
>> > > of
>> > > an ACK does not arrive?
>> > > I know that it has the risk of open sessions but there some other
>> ways to
>> > > handle those cases
>> > >
>> > > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>> > > http://krice at freeswitch.org> > wrote:
>> > >
>> > > This is probably the same scenario as this is exactly what to
>> expect...
>> > > Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
>> > > hangsup
>> > > the call....
>> > >
>> > > Quite common on networks with NAT issues or broken endpoints
>> > >
>> > >
>> > > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>> > > http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>> > >
>> > > I saw this happened earlier when the remote party does not send SIP
>> ACK
>> > > after receiving SIP OK, so the call is being disconnected after
>> exactly 32
>> > > seconds.
>> > > Not sure if this is exact same scenario here, but just something to
>> > > consider...
>> > >
>> > > Regards.
>> > > Vitalie
>> > >
>> > >
>> > > 2012/11/15 kaleem rehman <k4kaleem at gmail.com <
>> http://k4kaleem at gmail.com>
>> > >
>> > >  <http://k4kaleem at gmail.com> >
>> > >
>> > > Hi All,
>> > >
>> > > my inbound calls are fine with no issues, my outbound calls get
>> > > disconnected after 32 seconds and its on all calls. i tried 2
>> different
>> > > suppliers and its same result.
>> > > please find the attached log file with sofia in debug mode. - caller
>> was
>> > > extension 1234 and desination was 01908321682
>> > >
>> > > your help will be greately appreciated.
>> > >
>> > > regards,
>> > > Kaleem
>> > >
>> > >
>> _________________________________________________________________________
>> > > Professional FreeSWITCH Consulting Services:
>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> > > http://consulting at freeswitch.org>
>> > > http://www.freeswitchsolutions.com
>> > >
>> > > 
>> > > 
>> > >
>> > > Official FreeSWITCH Sites
>> > > http://www.freeswitch.org
>> > > http://wiki.freeswitch.org
>> > > http://www.cluecon.com
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org <
>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> > >
>> > > ------------------------------
>> > >
>> _________________________________________________________________________
>> > > Professional FreeSWITCH Consulting Services:
>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> > > http://consulting at freeswitch.org>
>> > > http://www.freeswitchsolutions.com
>> > >
>> > > 
>> > > 
>> > >
>> > > Official FreeSWITCH Sites
>> > > http://www.freeswitch.org
>> > > http://wiki.freeswitch.org
>> > > http://www.cluecon.com
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org <
>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> > > --
>> > > Ken
>> > > *http://www.FreeSWITCH.org
>> > > http://www.ClueCon.com
>> > > http://www.OSTAG.org
>> > > *irc.freenode.net #freeswitch
>> > >
>> > >
>> _________________________________________________________________________
>> > > Professional FreeSWITCH Consulting Services:
>> > > consulting at freeswitch.org
>> > > http://www.freeswitchsolutions.com
>> > >
>> > > 
>> > > 
>> > >
>> > > Official FreeSWITCH Sites
>> > > http://www.freeswitch.org
>> > > http://wiki.freeswitch.org
>> > > http://www.cluecon.com
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From yiftah at choochee.com  Sat Nov 17 03:40:10 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Fri, 16 Nov 2012 16:40:10 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <D775C81D-ABCC-4D35-9102-9AC34FFEDCD5@insensate.co.uk>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<D775C81D-ABCC-4D35-9102-9AC34FFEDCD5@insensate.co.uk>
Message-ID: <CAFVUaV_Ba8zBdq3aEsdh7ChKPsLrzvnCzjbu_SDo40mVdBr6gw@mail.gmail.com>

Lawrence it still does not give an answer what do you do when the BYE does
not arrive
In this case only one end knows when the call ended
The 30 sec disconnecting is very annoying to the users and it creates many
problems with your customers
When you disconnect the call it is much more problematic then a call that
did not established, further more most of the HA systems are working well
before the call is established, once it is established the HA no longer
available
As I said there is a lot of philosophy behind it, but I feel that they make
a mistake in this point

On Fri, Nov 16, 2012 at 3:23 PM, Lawrence Conroy <lconroy at insensate.co.uk>wrote:

> Hi There,
>  um ... the folk who thought up SIP used the Z-pattern for session
> initiation
> to ensure synchronisation between the parties. That's also the reason the
> path
> for the ACK is party-to-party direct, unlike the INVITE/200.
>
> There's a real good reason for that -- it's a call, so both *ends* need to
> know
> what state they're in -- as everyone here has been telling you.
>
> Trust me -- I was there, and the discussions were intense/painful.
> What you're proposing is exactly what was ditched back in 1997 -- RTP-based
> call completion. This is brittle -- it's being carried over UDP.
>
> No-RTP call abort is useful (check out 3GPP's discussion on this and how
> IMS
> works -- in particular, the discussion on call handling when one end goes
> into
> a tunnel), but it is only a fallback, and is nasty when call & media go
> different routes.
>
> Might I respectfully suggest that, if you think anyone is going to go over
> that again to produce a new RFC, you really should cut down on the crack.
>
> Ignoring failure to receive ACK (after a whole series of attempts) is
> ignoring
> your call control channel being, to use the technical term, b*gg*r*d. Just
> say no.
>
> all the best,
>   Lawrence
>
> On 16 Nov 2012, at 21:56, Yiftach Golan wrote:
>
> > This is where I am getting a bit confused, if the 200OK arrived to the
> > other side and we checked that the RTP exists (with mod_sofia option) we
> > will not get to hours of calls (unless the other side did not hanged up)
> > In any case there is a good chance that the BYE is getting lost, so the
> > danger exist even without the ACK
> >
> > I am guessing that the designers of the SIP protocol came up with the ACK
> > because there is a potential for open dialog that is not bound with time
> > and they therefore wanted to know that the other side actually ACKs the
> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
> > tied with RTP (at least in most normal SIP implementations) I'm not sure
> > that this ACK is that needed
> > but again as I said it is more of philosophical debate, maybe a potential
> > request in the new RFC
> >
> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
> >
> >> In this case masking the issue can lead to massive bills... Imaging
> >> paying by the minute... All of a sudden you are now leaving 2 minute
> calls
> >> up for hours on end... And continuing to get billed for them... Or you
> are
> >> not continuing to bill a customer for them... And now you have
> unexpected
> >> HUGE bills coming in... Masking it is far worse then just fixing it...
> >>
> >> If we mask this one issue, we might as well mask memory leaks, or
> >> passwords that don?t work, etc... Sure sometimes we might have to mask
> an
> >> issue for production to work in the short term, but that is never the
> >> correct answer fix a problem
> >>
> >>
> >> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
> >>
> >> While I agree on the details I disagree on the solution
> >> Sometimes masking the problems can be a good solution but I guess it is
> a
> >> philosophical debate
> >>
> >>
> >> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
> wrote:
> >>
> >> That leaves to big a risk of open sessions and only masks the true issue
> >> which is a problem with FS getting the ACK back...
> >>
> >> Theres a reason FS is not getting the ACK, and FS will make several
> >> attempts to get an ack by retransmitting the 200 OK several times before
> >> that timeout occurs.
> >>
> >> The real fix here is to fix the underlying cause, not masking it....
> >>
> >>
> >> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
> >> http://yiftah at choochee.com> > wrote:
> >>
> >> I know that it is kind out of the what RFC3261 instructs, but did anyone
> >> think on giving the option in configuration not to hang up calls in
> case of
> >> an ACK does not arrive?
> >> I know that it has the risk of open sessions but there some other ways
> to
> >> handle those cases
> >>
> >> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
> >> http://krice at freeswitch.org> > wrote:
> >>
> >> This is probably the same scenario as this is exactly what to expect...
> >> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
> hangsup
> >> the call....
> >>
> >> Quite common on networks with NAT issues or broken endpoints
> >>
> >>
> >> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
> >> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
> >>
> >> I saw this happened earlier when the remote party does not send SIP ACK
> >> after receiving SIP OK, so the call is being disconnected after exactly
> 32
> >> seconds.
> >> Not sure if this is exact same scenario here, but just something to
> >> consider...
> >>
> >> Regards.
> >> Vitalie
> >>
> >>
> >> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com
> >
> >> <http://k4kaleem at gmail.com> >
> >>
> >> Hi All,
> >>
> >> my inbound calls are fine with no issues, my outbound calls get
> >> disconnected after 32 seconds and its on all calls. i tried 2 different
> >> suppliers and its same result.
> >> please find the attached log file with sofia in debug mode. - caller was
> >> extension 1234 and desination was 01908321682
> >>
> >> your help will be greately appreciated.
> >>
> >> regards,
> >> Kaleem
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> >> http://consulting at freeswitch.org>
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org <
> >> http://FreeSWITCH-users at lists.freeswitch.org>  <
> >> http://FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> ------------------------------
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
> >> http://consulting at freeswitch.org>
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org <
> >> http://FreeSWITCH-users at lists.freeswitch.org>  <
> >> http://FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> --
> >> Ken
> >> *http://www.FreeSWITCH.org
> >> http://www.ClueCon.com
> >> http://www.OSTAG.org
> >> *irc.freenode.net #freeswitch
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Sat Nov 17 05:44:17 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Nov 2012 20:44:17 -0600
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
Message-ID: <CAKbxfG9v3FP1Q5_CMiG8bd-g_S=Dt2a=GXM+cLv3rPJ1Djh2xg@mail.gmail.com>

Why do you keep hijacking peoples threads and starting arguments?  An
attempt to promote your site maybe?

This is not even slightly philosophical... No endpoint will tolerate not
geting the ACK.  The whole point of it is to make sure 2-way communication
has been established.  The ACK contains vial information to set up the call
properly and until you have it, you have not completed the call setup.....
We will absolutely not try to hack our sip stack.

If you are here to participate in the community, try to help rather than
force comments on people and start arguments.  Maybe try listening to the
advice people are giving you......





On Fri, Nov 16, 2012 at 3:56 PM, Yiftach Golan <yiftah at choochee.com> wrote:

> This is where I am getting a bit confused, if the 200OK arrived to the
> other side and we checked that the RTP exists (with mod_sofia option) we
> will not get to hours of calls (unless the other side did not hanged up)
> In any case there is a good chance that the BYE is getting lost, so the
> danger exist even without the ACK
>
> I am guessing that the designers of the SIP protocol came up with the ACK
> because there is a potential for open dialog that is not bound with time
> and they therefore wanted to know that the other side actually ACKs the
> request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
> tied with RTP (at least in most normal SIP implementations) I'm not sure
> that this ACK is that needed
> but again as I said it is more of philosophical debate, maybe a potential
> request in the new RFC
>
> On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  In this case masking the issue can lead to massive bills... Imaging
>> paying by the minute... All of a sudden you are now leaving 2 minute calls
>> up for hours on end... And continuing to get billed for them... Or you are
>> not continuing to bill a customer for them... And now you have unexpected
>> HUGE bills coming in... Masking it is far worse then just fixing it...
>>
>> If we mask this one issue, we might as well mask memory leaks, or
>> passwords that don?t work, etc... Sure sometimes we might have to mask an
>> issue for production to work in the short term, but that is never the
>> correct answer fix a problem
>>
>>
>> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>
>> While I agree on the details I disagree on the solution
>> Sometimes masking the problems can be a good solution but I guess it is a
>> philosophical debate
>>
>>
>> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> That leaves to big a risk of open sessions and only masks the true issue
>> which is a problem with FS getting the ACK back...
>>
>> Theres a reason FS is not getting the ACK, and FS will make several
>> attempts to get an ack by retransmitting the 200 OK several times before
>> that timeout occurs.
>>
>> The real fix here is to fix the underlying cause, not masking it....
>>
>>
>> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>> http://yiftah at choochee.com> > wrote:
>>
>> I know that it is kind out of the what RFC3261 instructs, but did anyone
>> think on giving the option in configuration not to hang up calls in case of
>> an ACK does not arrive?
>> I know that it has the risk of open sessions but there some other ways to
>> handle those cases
>>
>> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>> http://krice at freeswitch.org> > wrote:
>>
>> This is probably the same scenario as this is exactly what to expect...
>> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
>> the call....
>>
>> Quite common on networks with NAT issues or broken endpoints
>>
>>
>> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>
>> I saw this happened earlier when the remote party does not send SIP ACK
>> after receiving SIP OK, so the call is being disconnected after exactly 32
>> seconds.
>> Not sure if this is exact same scenario here, but just something to
>> consider...
>>
>> Regards.
>> Vitalie
>>
>>
>> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
>>  <http://k4kaleem at gmail.com> >
>>
>> Hi All,
>>
>> my inbound calls are fine with no issues, my outbound calls get
>> disconnected after 32 seconds and its on all calls. i tried 2 different
>> suppliers and its same result.
>> please find the attached log file with sofia in debug mode. - caller was
>> extension 1234 and desination was 01908321682
>>
>> your help will be greately appreciated.
>>
>> regards,
>> Kaleem
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Ken
>> *http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> *irc.freenode.net #freeswitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From kaushalshriyan at gmail.com  Sat Nov 17 06:05:01 2012
From: kaushalshriyan at gmail.com (Kaushal Shriyan)
Date: Sat, 17 Nov 2012 08:35:01 +0530
Subject: [Freeswitch-users] Pager Duty Service on FreeSWITCH
Message-ID: <CAD7Ssm_iS83R2Hm9e52nxaNZjN9DwfB0KsWyZRaXzywZP5iWdQ@mail.gmail.com>

Hi,

Does FreeSwitch has pager duty feature and write ups or How To's to setup?

Regards,

Kaushal
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From 8f27e956 at gmail.com  Sat Nov 17 06:07:31 2012
From: 8f27e956 at gmail.com (Scott)
Date: Fri, 16 Nov 2012 22:07:31 -0500
Subject: [Freeswitch-users] [drm:drm_kms_helper_poll_enable] *ERROR* delayed
	enqueue failed -125
Message-ID: <CA+Y7Lf5j7FWUPsMncgMnLvRMH9O1x1gieqmfDwq=0WRsrKofJQ@mail.gmail.com>

Periodically, the following pops up on the running console.  No discernible
interval but at least once after a restart or start, only after a period of
time and comparative inactivity (almost like it's a housekeeping thread or
the like, but dunno). It pops as shown (no leading timestamp or [loglevel]
that is typical for most console messages.

[drm:drm_kms_helper_poll_enable] *ERROR* delayed enqueue failed -125

Not aware of any problem.  Too new to FS to know what part of the code this
is and if I RELY on it, or if it's a mod that's loaded that I don't really
use but is in the default mix.

FS 1.2.3+ 20121029... O/S centOS 6.3-X86-64

Question:

1. Should I care?

2. Should you, devel's, care?

Let me know if/what more info wanted/needed.

Thanks/Cheers,
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From 8f27e956 at gmail.com  Sat Nov 17 06:15:50 2012
From: 8f27e956 at gmail.com (Scott)
Date: Fri, 16 Nov 2012 22:15:50 -0500
Subject: [Freeswitch-users] extension name="Your_Are_Here"
Message-ID: <CA+Y7Lf7cfqigqjnK+t8XpNnR5Gx54MsGxSKv9Ofm8m1tHfqcLg@mail.gmail.com>

In the dialplan xml, for the <extension name="You_Are_Here" > clause, how
does one reference the value of the name= ?

e.g. <action application="log" data="INFO this message emitted from [${name}]"
/>

It's in the debug output so it "in there somewhere."

:-)

With thanks,
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From krice at freeswitch.org  Sat Nov 17 06:31:24 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Nov 2012 21:31:24 -0600
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
Message-ID: <B7EB28B2-F8F0-4648-A5AF-481C2482C396@freeswitch.org>

dont forget that sip and rtp do not nessecarily go through the same hosts on the same call... how can you monitor rtp on a host if the rtp never goes there

Ken
Sent from my iPad

On Nov 16, 2012, at 3:56 PM, Yiftach Golan <yiftah at choochee.com> wrote:

> This is where I am getting a bit confused, if the 200OK arrived to the other side and we checked that the RTP exists (with mod_sofia option) we will not get to hours of calls (unless the other side did not hanged up)
> In any case there is a good chance that the BYE is getting lost, so the danger exist even without the ACK 
> 
> I am guessing that the designers of the SIP protocol came up with the ACK because there is a potential for open dialog that is not bound with time and they therefore wanted to know that the other side actually ACKs the request, but since ACK is tied to INVITE only (AFAIK) and INVITE always tied with RTP (at least in most normal SIP implementations) I'm not sure that this ACK is that needed    
> but again as I said it is more of philosophical debate, maybe a potential request in the new RFC 
> 
> On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
> In this case masking the issue can lead to massive bills... Imaging paying by the minute... All of a sudden you are now leaving 2 minute calls up for hours on end... And continuing to get billed for them... Or you are not continuing to bill a customer for them... And now you have unexpected HUGE bills coming in... Masking it is far worse then just fixing it... 
> 
> If we mask this one issue, we might as well mask memory leaks, or passwords that don?t work, etc... Sure sometimes we might have to mask an issue for production to work in the short term, but that is never the correct answer fix a problem
> 
> 
> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
> 
> While I agree on the details I disagree on the solution
> Sometimes masking the problems can be a good solution but I guess it is a philosophical debate 
>    
> 
> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
> That leaves to big a risk of open sessions and only masks the true issue which is a problem with FS getting the ACK back...
> 
> Theres a reason FS is not getting the ACK, and FS will make several attempts to get an ack by retransmitting the 200 OK several times before that timeout occurs.
> 
> The real fix here is to fix the underlying cause, not masking it....
> 
> 
> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <http://yiftah at choochee.com> > wrote:
> 
> I know that it is kind out of the what RFC3261 instructs, but did anyone think on giving the option in configuration not to hang up calls in case of an ACK does not arrive?
> I know that it has the risk of open sessions but there some other ways to handle those cases
>   
> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <http://krice at freeswitch.org> > wrote:
> This is probably the same scenario as this is exactly what to expect... Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup the call....
> 
> Quite common on networks with NAT issues or broken endpoints
> 
> 
> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
> 
> I saw this happened earlier when the remote party does not send SIP ACK after receiving SIP OK, so the call is being disconnected after exactly 32 seconds.
> Not sure if this is exact same scenario here, but just something to consider...
> 
> Regards.
> Vitalie
> 
> 
> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>  <http://k4kaleem at gmail.com> >
> Hi All,
>  
> my inbound calls are fine with no issues, my outbound calls get disconnected after 32 seconds and its on all calls. i tried 2 different suppliers and its same result.
> please find the attached log file with sofia in debug mode. - caller was extension 1234 and desination was 01908321682
>  
> your help will be greately appreciated.
>  
> regards,
> Kaleem
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>  <http://consulting at freeswitch.org> 
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <http://FreeSWITCH-users at lists.freeswitch.org>  <http://FreeSWITCH-users at lists.freeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>  <http://consulting at freeswitch.org> 
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <http://FreeSWITCH-users at lists.freeswitch.org>  <http://FreeSWITCH-users at lists.freeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From packetandy at gmail.com  Sat Nov 17 07:20:50 2012
From: packetandy at gmail.com (andy)
Date: Fri, 16 Nov 2012 20:20:50 -0800
Subject: [Freeswitch-users]  changing vm default announcement
References: 50A5BA00.5050209@gmail.com
Message-ID: <50A710A2.90606@gmail.com>

hi Nick, thanks for the reply.

Greetings are in 
$${base_dir}/storage/voicemail/default/$${domain}/<user>/greeting_1.wav 
as specified in the wiki.

trying to  force freeswitch to use this custom greeting by using/<action application="set" data="voicemail_greeting_number=1"/>
/does not work/./Default greeting from Callie still plays/.

/If one records a greeting and then overwrites the recorded greeting with another file in the same directory - that works.

Guys, is this a bug, or do I not understand what the 'voicemail_greeting_number variable is supposed to do?

drew

BTW - I am on FS 1.2.1

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From jason.holden at start.ca  Sat Nov 17 09:33:04 2012
From: jason.holden at start.ca (Jason Holden)
Date: Sat, 17 Nov 2012 01:33:04 -0500
Subject: [Freeswitch-users] retreiving voicemail dropping after 30 seconds
Message-ID: <6626166B66164AB4B5C0E344762D7E4A@bob>

Hi.

When accessing voicemail to listen to messages I am finding that it is
dropping at 30 seconds each time with a message of 100 sleep timer.

Does anyone have any recommendations?

I am using a Sipura 3000 connected to my freeswitch server.

 

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From anthony.minessale at gmail.com  Sat Nov 17 18:21:22 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Nov 2012 09:21:22 -0600
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV-hPaPUZLQtp+J_VLwHfYNnMQE_TzcUymPqurmbm6AmAQ@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<1546109.RqNAKPZJSR@sos>
	<CAFVUaV9VnVwVeNn_KFHZwV3Q4N5u8rrcMv5RbPzfh143j29KTA@mail.gmail.com>
	<CAFVUaV-hPaPUZLQtp+J_VLwHfYNnMQE_TzcUymPqurmbm6AmAQ@mail.gmail.com>
Message-ID: <CAKbxfG-dWuNCgykuNWHV=QxTkwHkFhjKDwC0wK6ezTFVPhhcWg@mail.gmail.com>

The only reason that there would be a failure to get the ACK would be if
the 2 endpoints have lost communication.
If you can't get an ACK to the host, you can't get a BYE to them either.
 Even if you hacked it to tolerate no ACK, the session timers would fail in
another 30 seconds.

You are just focused on your business needs and just disregarding logic.
 You ask if you are missing something and several individuals are telling
you YES.  This is the 4th time I am telling you that you should focus
your enthusiasm on learning rather than affirming a m00t point because we
are not going to change the behavior.

I have not seen a seen a single case of missing ACK that did not point out
a real problem that can be simply fixed by either using some NAT related
configuration and proper formed Contact hosts.

If this argument continues it will be ended with moderation.....


On Fri, Nov 16, 2012 at 6:07 PM, Yiftach Golan <yiftah at choochee.com> wrote:

> Here is the reason from the RFC3261 :
>    "...The reason for this separation is rooted in the importance of
>       delivering all 200 (OK) responses to an INVITE to the UAC.  To
>       deliver them all to the UAC, the UAS alone takes responsibility
>       for retransmitting them (see Section 13.3.1.4), and the UAC alone
>       takes responsibility for acknowledging them with ACK (see Section
>       13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
>       effectively considered its own transaction..."
>
>
> On Fri, Nov 16, 2012 at 3:50 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>
>> Yes I you are right if the RTP is not tied with SIP this theory is not
>> really valid
>> But this is again true about BYE that does not reach to the destination
>> so the risk of not closing the dialog still exist
>> In most of the voice implementation that I saw there are three options :
>> 1. Softswitch (FreeSWITCH, Asterisk, etc)
>> 2. MGW (Avaya, Cisco, etc)
>> 3. Media release but only between two phones
>> You can tie your RTP to the SIP with option 1 and option 2
>> Option 3 is the problematic one but you will never release your media for
>> billing purpose so usually your risk will be extension to extension open
>> dialog which is less riskier
>> Again as I said it is pretty philosophical debate but from my long
>> experience in SIP I do not think that there is a practical use for it but
>> maybe I am missing something
>>
>> On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin <
>> sos at sokhapkin.dyndns.org> wrote:
>>
>>>  Don't mix signaling (SIP) and media (RTP). Signaling and media could run
>>> different ways. Why do you think FS will always be in media path?
>>>
>>> On Friday 16 November 2012 14:56:58 Yiftach Golan wrote:
>>> > This is where I am getting a bit confused, if the 200OK arrived to the
>>> > other side and we checked that the RTP exists (with mod_sofia option)
>>> we
>>> > will not get to hours of calls (unless the other side did not hanged
>>> up)
>>> > In any case there is a good chance that the BYE is getting lost, so the
>>> > danger exist even without the ACK
>>> >
>>> > I am guessing that the designers of the SIP protocol came up with the
>>> ACK
>>> > because there is a potential for open dialog that is not bound with
>>> time
>>> > and they therefore wanted to know that the other side actually ACKs the
>>> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
>>> > tied with RTP (at least in most normal SIP implementations) I'm not
>>> sure
>>> > that this ACK is that needed
>>> > but again as I said it is more of philosophical debate, maybe a
>>> potential
>>> > request in the new RFC
>>> >
>>> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org>
>>> wrote:
>>> > >  In this case masking the issue can lead to massive bills... Imaging
>>> > >
>>> > > paying by the minute... All of a sudden you are now leaving 2 minute
>>> calls
>>> > > up for hours on end... And continuing to get billed for them... Or
>>> you are
>>> > > not continuing to bill a customer for them... And now you have
>>> unexpected
>>> > > HUGE bills coming in... Masking it is far worse then just fixing
>>> it...
>>> > >
>>> > > If we mask this one issue, we might as well mask memory leaks, or
>>> > > passwords that don?t work, etc... Sure sometimes we might have to
>>> mask an
>>> > > issue for production to work in the short term, but that is never the
>>> > > correct answer fix a problem
>>> > >
>>> > >
>>> > > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>> > >
>>> > > While I agree on the details I disagree on the solution
>>> > > Sometimes masking the problems can be a good solution but I guess it
>>> is a
>>> > > philosophical debate
>>> > >
>>> > >
>>> > > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
>>> wrote:
>>> > >
>>> > > That leaves to big a risk of open sessions and only masks the true
>>> issue
>>> > > which is a problem with FS getting the ACK back...
>>> > >
>>> > > Theres a reason FS is not getting the ACK, and FS will make several
>>> > > attempts to get an ack by retransmitting the 200 OK several times
>>> before
>>> > > that timeout occurs.
>>> > >
>>> > > The real fix here is to fix the underlying cause, not masking it....
>>> > >
>>> > >
>>> > > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>>> > > http://yiftah at choochee.com> > wrote:
>>> > >
>>> > > I know that it is kind out of the what RFC3261 instructs, but did
>>> anyone
>>> > > think on giving the option in configuration not to hang up calls in
>>> case
>>> > > of
>>> > > an ACK does not arrive?
>>> > > I know that it has the risk of open sessions but there some other
>>> ways to
>>> > > handle those cases
>>> > >
>>> > > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>>> > > http://krice at freeswitch.org> > wrote:
>>> > >
>>> > > This is probably the same scenario as this is exactly what to
>>> expect...
>>> > > Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
>>> > > hangsup
>>> > > the call....
>>> > >
>>> > > Quite common on networks with NAT issues or broken endpoints
>>> > >
>>> > >
>>> > > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>>> > > http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>> > >
>>> > > I saw this happened earlier when the remote party does not send SIP
>>> ACK
>>> > > after receiving SIP OK, so the call is being disconnected after
>>> exactly 32
>>> > > seconds.
>>> > > Not sure if this is exact same scenario here, but just something to
>>> > > consider...
>>> > >
>>> > > Regards.
>>> > > Vitalie
>>> > >
>>> > >
>>> > > 2012/11/15 kaleem rehman <k4kaleem at gmail.com <
>>> http://k4kaleem at gmail.com>
>>> > >
>>> > >  <http://k4kaleem at gmail.com> >
>>> > >
>>> > > Hi All,
>>> > >
>>> > > my inbound calls are fine with no issues, my outbound calls get
>>> > > disconnected after 32 seconds and its on all calls. i tried 2
>>> different
>>> > > suppliers and its same result.
>>> > > please find the attached log file with sofia in debug mode. - caller
>>> was
>>> > > extension 1234 and desination was 01908321682
>>> > >
>>> > > your help will be greately appreciated.
>>> > >
>>> > > regards,
>>> > > Kaleem
>>> > >
>>> > >
>>> _________________________________________________________________________
>>> > > Professional FreeSWITCH Consulting Services:
>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>> > > http://consulting at freeswitch.org>
>>> > > http://www.freeswitchsolutions.com
>>> > >
>>> > > 
>>> > > 
>>> > >
>>> > > Official FreeSWITCH Sites
>>> > > http://www.freeswitch.org
>>> > > http://wiki.freeswitch.org
>>> > > http://www.cluecon.com
>>> > >
>>> > > FreeSWITCH-users mailing list
>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > > http://www.freeswitch.org
>>> > >
>>> > >
>>> > >
>>> > > ------------------------------
>>> > >
>>> _________________________________________________________________________
>>> > > Professional FreeSWITCH Consulting Services:
>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>> > > http://consulting at freeswitch.org>
>>> > > http://www.freeswitchsolutions.com
>>> > >
>>> > > 
>>> > > 
>>> > >
>>> > > Official FreeSWITCH Sites
>>> > > http://www.freeswitch.org
>>> > > http://wiki.freeswitch.org
>>> > > http://www.cluecon.com
>>> > >
>>> > > FreeSWITCH-users mailing list
>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > > http://www.freeswitch.org
>>> > >
>>> > >
>>> > > --
>>> > > Ken
>>> > > *http://www.FreeSWITCH.org
>>> > > http://www.ClueCon.com
>>> > > http://www.OSTAG.org
>>> > > *irc.freenode.net #freeswitch
>>> > >
>>> > >
>>> _________________________________________________________________________
>>> > > Professional FreeSWITCH Consulting Services:
>>> > > consulting at freeswitch.org
>>> > > http://www.freeswitchsolutions.com
>>> > >
>>> > > 
>>> > > 
>>> > >
>>> > > Official FreeSWITCH Sites
>>> > > http://www.freeswitch.org
>>> > > http://wiki.freeswitch.org
>>> > > http://www.cluecon.com
>>> > >
>>> > > FreeSWITCH-users mailing list
>>> > > FreeSWITCH-users at lists.freeswitch.org
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From anthony.minessale at gmail.com  Sat Nov 17 18:25:18 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Nov 2012 09:25:18 -0600
Subject: [Freeswitch-users] [drm:drm_kms_helper_poll_enable] *ERROR*
 delayed enqueue failed -125
In-Reply-To: <CA+Y7Lf5j7FWUPsMncgMnLvRMH9O1x1gieqmfDwq=0WRsrKofJQ@mail.gmail.com>
References: <CA+Y7Lf5j7FWUPsMncgMnLvRMH9O1x1gieqmfDwq=0WRsrKofJQ@mail.gmail.com>
Message-ID: <CAKbxfG_vVh6CW1-Zb0Xfc_-sDAZxCRajDZQ-YEadWynfap1R5g@mail.gmail.com>

I think that is coming from linux.  It's not from FS.  Like a driver err or
something...



On Fri, Nov 16, 2012 at 9:07 PM, Scott <8f27e956 at gmail.com> wrote:

> Periodically, the following pops up on the running console.  No
> discernible interval but at least once after a restart or start, only after
> a period of time and comparative inactivity (almost like it's a
> housekeeping thread or the like, but dunno). It pops as shown (no leading
> timestamp or [loglevel] that is typical for most console messages.
>
> [drm:drm_kms_helper_poll_enable] *ERROR* delayed enqueue failed -125
>
> Not aware of any problem.  Too new to FS to know what part of the code
> this is and if I RELY on it, or if it's a mod that's loaded that I don't
> really use but is in the default mix.
>
> FS 1.2.3+ 20121029... O/S centOS 6.3-X86-64
>
> Question:
>
> 1. Should I care?
>
> 2. Should you, devel's, care?
>
> Let me know if/what more info wanted/needed.
>
> Thanks/Cheers,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From Tim.Meade at Millicorp.com  Sat Nov 17 18:34:11 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Sat, 17 Nov 2012 15:34:11 +0000
Subject: [Freeswitch-users] G729 File version of $${us-ring}
Message-ID: <804D48104511D4468F0D60DF9D3100350AD99E66@MAIL.millicorp.com>

I was wondering if anyone had a G729 file of the $${us-ring}

We want to use it for playing the ringback with G729 pass through.

<action application="set"  data="ringback=SomeRingBackFile.G729"/>


Thanks

Tim
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From oseslija at gmail.com  Sat Nov 17 18:57:52 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 17 Nov 2012 16:57:52 +0100
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV_Ba8zBdq3aEsdh7ChKPsLrzvnCzjbu_SDo40mVdBr6gw@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<D775C81D-ABCC-4D35-9102-9AC34FFEDCD5@insensate.co.uk>
	<CAFVUaV_Ba8zBdq3aEsdh7ChKPsLrzvnCzjbu_SDo40mVdBr6gw@mail.gmail.com>
Message-ID: <CAM21H-PPaH6dUTWK5DjODSM1tfqW8GCv52CG9hdMgaHFfhna8A@mail.gmail.com>

Check SIP session timers which are fully supported in FreeSWITCH.

On Sat, Nov 17, 2012 at 1:40 AM, Yiftach Golan <yiftah at choochee.com> wrote:

> Lawrence it still does not give an answer what do you do when the BYE does
> not arrive
> In this case only one end knows when the call ended
> The 30 sec disconnecting is very annoying to the users and it creates many
> problems with your customers
> When you disconnect the call it is much more problematic then a call that
> did not established, further more most of the HA systems are working well
> before the call is established, once it is established the HA no longer
> available
> As I said there is a lot of philosophy behind it, but I feel that they
> make a mistake in this point
>
> On Fri, Nov 16, 2012 at 3:23 PM, Lawrence Conroy <lconroy at insensate.co.uk>wrote:
>
>> Hi There,
>>  um ... the folk who thought up SIP used the Z-pattern for session
>> initiation
>> to ensure synchronisation between the parties. That's also the reason the
>> path
>> for the ACK is party-to-party direct, unlike the INVITE/200.
>>
>> There's a real good reason for that -- it's a call, so both *ends* need
>> to know
>> what state they're in -- as everyone here has been telling you.
>>
>> Trust me -- I was there, and the discussions were intense/painful.
>> What you're proposing is exactly what was ditched back in 1997 --
>> RTP-based
>> call completion. This is brittle -- it's being carried over UDP.
>>
>> No-RTP call abort is useful (check out 3GPP's discussion on this and how
>> IMS
>> works -- in particular, the discussion on call handling when one end goes
>> into
>> a tunnel), but it is only a fallback, and is nasty when call & media go
>> different routes.
>>
>> Might I respectfully suggest that, if you think anyone is going to go over
>> that again to produce a new RFC, you really should cut down on the crack.
>>
>> Ignoring failure to receive ACK (after a whole series of attempts) is
>> ignoring
>> your call control channel being, to use the technical term, b*gg*r*d.
>> Just say no.
>>
>> all the best,
>>   Lawrence
>>
>> On 16 Nov 2012, at 21:56, Yiftach Golan wrote:
>>
>> > This is where I am getting a bit confused, if the 200OK arrived to the
>> > other side and we checked that the RTP exists (with mod_sofia option) we
>> > will not get to hours of calls (unless the other side did not hanged up)
>> > In any case there is a good chance that the BYE is getting lost, so the
>> > danger exist even without the ACK
>> >
>> > I am guessing that the designers of the SIP protocol came up with the
>> ACK
>> > because there is a potential for open dialog that is not bound with time
>> > and they therefore wanted to know that the other side actually ACKs the
>> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
>> > tied with RTP (at least in most normal SIP implementations) I'm not sure
>> > that this ACK is that needed
>> > but again as I said it is more of philosophical debate, maybe a
>> potential
>> > request in the new RFC
>> >
>> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org>
>> wrote:
>> >
>> >> In this case masking the issue can lead to massive bills... Imaging
>> >> paying by the minute... All of a sudden you are now leaving 2 minute
>> calls
>> >> up for hours on end... And continuing to get billed for them... Or you
>> are
>> >> not continuing to bill a customer for them... And now you have
>> unexpected
>> >> HUGE bills coming in... Masking it is far worse then just fixing it...
>> >>
>> >> If we mask this one issue, we might as well mask memory leaks, or
>> >> passwords that don?t work, etc... Sure sometimes we might have to mask
>> an
>> >> issue for production to work in the short term, but that is never the
>> >> correct answer fix a problem
>> >>
>> >>
>> >> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>> >>
>> >> While I agree on the details I disagree on the solution
>> >> Sometimes masking the problems can be a good solution but I guess it
>> is a
>> >> philosophical debate
>> >>
>> >>
>> >> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
>> wrote:
>> >>
>> >> That leaves to big a risk of open sessions and only masks the true
>> issue
>> >> which is a problem with FS getting the ACK back...
>> >>
>> >> Theres a reason FS is not getting the ACK, and FS will make several
>> >> attempts to get an ack by retransmitting the 200 OK several times
>> before
>> >> that timeout occurs.
>> >>
>> >> The real fix here is to fix the underlying cause, not masking it....
>> >>
>> >>
>> >> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>> >> http://yiftah at choochee.com> > wrote:
>> >>
>> >> I know that it is kind out of the what RFC3261 instructs, but did
>> anyone
>> >> think on giving the option in configuration not to hang up calls in
>> case of
>> >> an ACK does not arrive?
>> >> I know that it has the risk of open sessions but there some other ways
>> to
>> >> handle those cases
>> >>
>> >> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>> >> http://krice at freeswitch.org> > wrote:
>> >>
>> >> This is probably the same scenario as this is exactly what to expect...
>> >> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
>> hangsup
>> >> the call....
>> >>
>> >> Quite common on networks with NAT issues or broken endpoints
>> >>
>> >>
>> >> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>> >> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>> >>
>> >> I saw this happened earlier when the remote party does not send SIP ACK
>> >> after receiving SIP OK, so the call is being disconnected after
>> exactly 32
>> >> seconds.
>> >> Not sure if this is exact same scenario here, but just something to
>> >> consider...
>> >>
>> >> Regards.
>> >> Vitalie
>> >>
>> >>
>> >> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <
>> http://k4kaleem at gmail.com>
>> >> <http://k4kaleem at gmail.com> >
>> >>
>> >> Hi All,
>> >>
>> >> my inbound calls are fine with no issues, my outbound calls get
>> >> disconnected after 32 seconds and its on all calls. i tried 2 different
>> >> suppliers and its same result.
>> >> please find the attached log file with sofia in debug mode. - caller
>> was
>> >> extension 1234 and desination was 01908321682
>> >>
>> >> your help will be greately appreciated.
>> >>
>> >> regards,
>> >> Kaleem
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> >> http://consulting at freeswitch.org>
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org <
>> >> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> >> http://FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> ------------------------------
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> >> http://consulting at freeswitch.org>
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org <
>> >> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> >> http://FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> --
>> >> Ken
>> >> *http://www.FreeSWITCH.org
>> >> http://www.ClueCon.com
>> >> http://www.OSTAG.org
>> >> *irc.freenode.net #freeswitch
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From bote_radio at botecomm.com  Sat Nov 17 19:13:01 2012
From: bote_radio at botecomm.com (Bote Man)
Date: Sat, 17 Nov 2012 11:13:01 -0500
Subject: [Freeswitch-users] uuid_media leaves behind zombie call
In-Reply-To: <CAKbxfG8q5qn5XPeyNL2nhOmGzUjvbJMJ0qMZTEVnhwfYb5NwHw@mail.gmail.com>
References: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>
	<CAKbxfG-YU0Sz2wd0mZ9+rv3MON13f6AYbJcwxPa1GdmsuGaRoQ@mail.gmail.com>
	<870204F45EE7D34E8D27CC0E602E11A1B46979@EX01.semafone.local>
	<CAKbxfG8q5qn5XPeyNL2nhOmGzUjvbJMJ0qMZTEVnhwfYb5NwHw@mail.gmail.com>
Message-ID: <027901cdc4de$6e305ee0$4a911ca0$@com>

Well, if you want the additional workload, fine.

 

One might reasonably wonder why this user mailing list exists in the
first place, if not to ask questions and learn about FreeSWITCH?

 

Bote

 

 

From: Anthony Minessale
Sent: Friday, 16 November, 2012 11:14



No problem, I just want to try and trace peoples path into the community
and make sure we have all the proper information at first glance.

 

Even in a case of uncertainty, Jira is better because its a database of
"issues" and we always have a "not a bug" resolution.

Even in cases of mistaken configuration, it can be considered a valid
issue and we can change defaults etc.

 

Jiras are not expensive to create and easy to close, just a lot more
organized.

 

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From Tim.Meade at Millicorp.com  Sat Nov 17 19:25:56 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Sat, 17 Nov 2012 16:25:56 +0000
Subject: [Freeswitch-users] G729 File version of $${us-ring}
Message-ID: <804D48104511D4468F0D60DF9D3100350AD9A4EC@MAIL.millicorp.com>


I was wondering if anyone had a G729 file of the $${us-ring}

We want to use it for playing the ringback with G729 pass through.

<action application="set"  data="ringback=SomeRingBackFile.G729"/>


Thanks

Tim
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From k4kaleem at gmail.com  Sat Nov 17 20:14:09 2012
From: k4kaleem at gmail.com (kaleem rehman)
Date: Sat, 17 Nov 2012 17:14:09 +0000
Subject: [Freeswitch-users] call disconnects after 32 seconds issue -
 FreeSWITCH-users Digest, Vol 77, Issue 117
Message-ID: <CAFWsKqzkeURVdTvmTKvVxBFhtKHpLRE0LcKB9VATjhsJiNKUyg@mail.gmail.com>

Hi All,

Its kaleem, i reported the issue with call disconnecting after 32 seconds,
i have managed to fix the issue. please find the detailed fix below;

i had to get the external Interface (internet facing) to pass 5060 TCP/UDP
& 5080 TCP/UDP to freeswitch server.

if you are experiencing similar issues then please follow following steps.

run fswitch console using command "fs_cli"

once in run command "sofia status" this should show all gateways and
profiles.

check if external profile has your external IP to it. (to confirm go to
www.whatsmyip.org)
this will confirm freeswitch is using NAT.

now turn loggin on using command  "sofia loglevel all 9" and "sofia profile
external siptrace on" where external is name of profile which has your
external IP addr.
please note this will give you a lot of information and you might have
digestion issues.

i used putty and changed putty settings to show 3000 lines (right click on
putty title bar, CHANGE SETTINGS and click on WINDOWS and change SCROLLBACK
lines from default 200 to 3000) also you can enable log in putty so it
saves all data in putty to a file for you.

now make the call and when it disconencts go to log windows and type
"/quit" to quit freeswitch terminal and no more logs are shown.

now either copy and paste the data or open the log file and search for "50"
and try to see what ports are being used.

now log on to your router and change all port fowarding (please refer to
documentation for your router) forward port you can see in log file(in my
case 5060 & 5080) to freeswitch server.

i restarted my server just to be on safe side and everything went fine
after that.

also i had a bad experience with BT SIP, British telecoms business sip,
they dont respond with SRV & DNS queries so FreeSwitch doesnt register with
them and gateway status goes to fail_wait.

i had to play around and manage to fix it, i will do a proper guide so no
one else has to suffer like me where it took me 4 days and nights to sort
that out.

i dont want to post messy documentation so will clean up a bit and
hopefully FreeSwitch admins will import to WIKI


hope this helps.

regards,
K


On Sat, Nov 17, 2012 at 3:21 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>         freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>         freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>         freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>    1.  changing vm default announcement (andy)
>    2. retreiving voicemail dropping after 30 seconds (Jason Holden)
>    3. Re: call disconnects after 32 seconds (Anthony Minessale)
>
>
> ---------- Forwarded message ----------
> From: andy <packetandy at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Fri, 16 Nov 2012 20:20:50 -0800
> Subject: [Freeswitch-users] changing vm default announcement
>  hi Nick, thanks for the reply.
>
> Greetings are in
> $${base_dir}/storage/voicemail/default/$${domain}/<user>/greeting_1.wav as
> specified in the wiki.
>
> trying to  force freeswitch to use this custom greeting by using *<action application="set" data="voicemail_greeting_number=1"/> *does not work*. *Default greeting from Callie still plays*.
> *If one records a greeting and then overwrites the recorded greeting with another file in the same directory - that works.
>
> Guys, is this a bug, or do I not understand what the 'voicemail_greeting_number variable is supposed to do?
>
> drew
>
> BTW - I am on FS 1.2.1
>
>
>
> ---------- Forwarded message ----------
> From: "Jason Holden" <jason.holden at start.ca>
> To: <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 17 Nov 2012 01:33:04 -0500
> Subject: [Freeswitch-users] retreiving voicemail dropping after 30 seconds
>
> Hi.****
>
> When accessing voicemail to listen to messages I am finding that it is
> dropping at 30 seconds each time with a message of 100 sleep timer.****
>
> Does anyone have any recommendations?****
>
> I am using a Sipura 3000 connected to my freeswitch server.****
>
> ** **
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 17 Nov 2012 09:21:22 -0600
> Subject: Re: [Freeswitch-users] call disconnects after 32 seconds
> The only reason that there would be a failure to get the ACK would be if
> the 2 endpoints have lost communication.
> If you can't get an ACK to the host, you can't get a BYE to them either.
>  Even if you hacked it to tolerate no ACK, the session timers would fail in
> another 30 seconds.
>
> You are just focused on your business needs and just disregarding logic.
>  You ask if you are missing something and several individuals are telling
> you YES.  This is the 4th time I am telling you that you should focus
> your enthusiasm on learning rather than affirming a m00t point because we
> are not going to change the behavior.
>
> I have not seen a seen a single case of missing ACK that did not point out
> a real problem that can be simply fixed by either using some NAT related
> configuration and proper formed Contact hosts.
>
> If this argument continues it will be ended with moderation.....
>
>
> On Fri, Nov 16, 2012 at 6:07 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>
>> Here is the reason from the RFC3261 :
>>    "...The reason for this separation is rooted in the importance of
>>       delivering all 200 (OK) responses to an INVITE to the UAC.  To
>>       deliver them all to the UAC, the UAS alone takes responsibility
>>       for retransmitting them (see Section 13.3.1.4), and the UAC alone
>>       takes responsibility for acknowledging them with ACK (see Section
>>       13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
>>       effectively considered its own transaction..."
>>
>>
>> On Fri, Nov 16, 2012 at 3:50 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>>
>>> Yes I you are right if the RTP is not tied with SIP this theory is not
>>> really valid
>>> But this is again true about BYE that does not reach to the destination
>>> so the risk of not closing the dialog still exist
>>> In most of the voice implementation that I saw there are three options :
>>> 1. Softswitch (FreeSWITCH, Asterisk, etc)
>>> 2. MGW (Avaya, Cisco, etc)
>>> 3. Media release but only between two phones
>>> You can tie your RTP to the SIP with option 1 and option 2
>>> Option 3 is the problematic one but you will never release your media
>>> for billing purpose so usually your risk will be extension to extension
>>> open dialog which is less riskier
>>> Again as I said it is pretty philosophical debate but from my long
>>> experience in SIP I do not think that there is a practical use for it but
>>> maybe I am missing something
>>>
>>> On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin <
>>> sos at sokhapkin.dyndns.org> wrote:
>>>
>>>>  Don't mix signaling (SIP) and media (RTP). Signaling and media could
>>>> run
>>>> different ways. Why do you think FS will always be in media path?
>>>>
>>>> On Friday 16 November 2012 14:56:58 Yiftach Golan wrote:
>>>> > This is where I am getting a bit confused, if the 200OK arrived to the
>>>> > other side and we checked that the RTP exists (with mod_sofia option)
>>>> we
>>>> > will not get to hours of calls (unless the other side did not hanged
>>>> up)
>>>> > In any case there is a good chance that the BYE is getting lost, so
>>>> the
>>>> > danger exist even without the ACK
>>>> >
>>>> > I am guessing that the designers of the SIP protocol came up with the
>>>> ACK
>>>> > because there is a potential for open dialog that is not bound with
>>>> time
>>>> > and they therefore wanted to know that the other side actually ACKs
>>>> the
>>>> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE
>>>> always
>>>> > tied with RTP (at least in most normal SIP implementations) I'm not
>>>> sure
>>>> > that this ACK is that needed
>>>> > but again as I said it is more of philosophical debate, maybe a
>>>> potential
>>>> > request in the new RFC
>>>> >
>>>> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org>
>>>> wrote:
>>>> > >  In this case masking the issue can lead to massive bills... Imaging
>>>> > >
>>>> > > paying by the minute... All of a sudden you are now leaving 2
>>>> minute calls
>>>> > > up for hours on end... And continuing to get billed for them... Or
>>>> you are
>>>> > > not continuing to bill a customer for them... And now you have
>>>> unexpected
>>>> > > HUGE bills coming in... Masking it is far worse then just fixing
>>>> it...
>>>> > >
>>>> > > If we mask this one issue, we might as well mask memory leaks, or
>>>> > > passwords that don?t work, etc... Sure sometimes we might have to
>>>> mask an
>>>> > > issue for production to work in the short term, but that is never
>>>> the
>>>> > > correct answer fix a problem
>>>> > >
>>>> > >
>>>> > > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>>> > >
>>>> > > While I agree on the details I disagree on the solution
>>>> > > Sometimes masking the problems can be a good solution but I guess
>>>> it is a
>>>> > > philosophical debate
>>>> > >
>>>> > >
>>>> > > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
>>>> wrote:
>>>> > >
>>>> > > That leaves to big a risk of open sessions and only masks the true
>>>> issue
>>>> > > which is a problem with FS getting the ACK back...
>>>> > >
>>>> > > Theres a reason FS is not getting the ACK, and FS will make several
>>>> > > attempts to get an ack by retransmitting the 200 OK several times
>>>> before
>>>> > > that timeout occurs.
>>>> > >
>>>> > > The real fix here is to fix the underlying cause, not masking it....
>>>> > >
>>>> > >
>>>> > > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>>>> > > http://yiftah at choochee.com> > wrote:
>>>> > >
>>>> > > I know that it is kind out of the what RFC3261 instructs, but did
>>>> anyone
>>>> > > think on giving the option in configuration not to hang up calls in
>>>> case
>>>> > > of
>>>> > > an ACK does not arrive?
>>>> > > I know that it has the risk of open sessions but there some other
>>>> ways to
>>>> > > handle those cases
>>>> > >
>>>> > > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>>>> > > http://krice at freeswitch.org> > wrote:
>>>> > >
>>>> > > This is probably the same scenario as this is exactly what to
>>>> expect...
>>>> > > Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
>>>> > > hangsup
>>>> > > the call....
>>>> > >
>>>> > > Quite common on networks with NAT issues or broken endpoints
>>>> > >
>>>> > >
>>>> > > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>>>> > > http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>>> > >
>>>> > > I saw this happened earlier when the remote party does not send SIP
>>>> ACK
>>>> > > after receiving SIP OK, so the call is being disconnected after
>>>> exactly 32
>>>> > > seconds.
>>>> > > Not sure if this is exact same scenario here, but just something to
>>>> > > consider...
>>>> > >
>>>> > > Regards.
>>>> > > Vitalie
>>>> > >
>>>> > >
>>>> > > 2012/11/15 kaleem rehman <k4kaleem at gmail.com <
>>>> http://k4kaleem at gmail.com>
>>>> > >
>>>> > >  <http://k4kaleem at gmail.com> >
>>>> > >
>>>> > > Hi All,
>>>> > >
>>>> > > my inbound calls are fine with no issues, my outbound calls get
>>>> > > disconnected after 32 seconds and its on all calls. i tried 2
>>>> different
>>>> > > suppliers and its same result.
>>>> > > please find the attached log file with sofia in debug mode. -
>>>> caller was
>>>> > > extension 1234 and desination was 01908321682
>>>> > >
>>>> > > your help will be greately appreciated.
>>>> > >
>>>> > > regards,
>>>> > > Kaleem
>>>> > >
>>>> > >
>>>> _________________________________________________________________________
>>>> > > Professional FreeSWITCH Consulting Services:
>>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>>> > > http://consulting at freeswitch.org>
>>>> > > http://www.freeswitchsolutions.com
>>>> > >
>>>> > > 
>>>> > > 
>>>> > >
>>>> > > Official FreeSWITCH Sites
>>>> > > http://www.freeswitch.org
>>>> > > http://wiki.freeswitch.org
>>>> > > http://www.cluecon.com
>>>> > >
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > > http://www.freeswitch.org
>>>> > >
>>>> > >
>>>> > >
>>>> > > ------------------------------
>>>> > >
>>>> _________________________________________________________________________
>>>> > > Professional FreeSWITCH Consulting Services:
>>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>>> > > http://consulting at freeswitch.org>
>>>> > > http://www.freeswitchsolutions.com
>>>> > >
>>>> > > 
>>>> > > 
>>>> > >
>>>> > > Official FreeSWITCH Sites
>>>> > > http://www.freeswitch.org
>>>> > > http://wiki.freeswitch.org
>>>> > > http://www.cluecon.com
>>>> > >
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > > http://www.freeswitch.org
>>>> > >
>>>> > >
>>>> > > --
>>>> > > Ken
>>>> > > *http://www.FreeSWITCH.org
>>>> > > http://www.ClueCon.com
>>>> > > http://www.OSTAG.org
>>>> > > *irc.freenode.net #freeswitch
>>>> > >
>>>> > >
>>>> _________________________________________________________________________
>>>> > > Professional FreeSWITCH Consulting Services:
>>>> > > consulting at freeswitch.org
>>>> > > http://www.freeswitchsolutions.com
>>>> > >
>>>> > > 
>>>> > > 
>>>> > >
>>>> > > Official FreeSWITCH Sites
>>>> > > http://www.freeswitch.org
>>>> > > http://wiki.freeswitch.org
>>>> > > http://www.cluecon.com
>>>> > >
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > > http://www.freeswitch.org
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yiftah at choochee.com  Sat Nov 17 21:03:15 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sat, 17 Nov 2012 10:03:15 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAKbxfG9v3FP1Q5_CMiG8bd-g_S=Dt2a=GXM+cLv3rPJ1Djh2xg@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<CAKbxfG9v3FP1Q5_CMiG8bd-g_S=Dt2a=GXM+cLv3rPJ1Djh2xg@mail.gmail.com>
Message-ID: <CAFVUaV8at9=aSwzHPN5PvmA7_TkoxL-sk=N-=6s9Rz6_RnzS1g@mail.gmail.com>

I am not trying to promote any site and our Start Up does not even sell
directly to the customer we just develop the technology
We actually trying to promote FreeSWITCH and we have been paying a lot of
money to its consultant
I wanted to have an open minded discussion and question every aspect of
technology apparently it is not welcomed in here and people get offended by
questioning and raising doubts then I will stop

Thanks,
Yiftach.


On Fri, Nov 16, 2012 at 6:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Why do you keep hijacking peoples threads and starting arguments?  An
> attempt to promote your site maybe?
>
> This is not even slightly philosophical... No endpoint will tolerate not
> geting the ACK.  The whole point of it is to make sure 2-way communication
> has been established.  The ACK contains vial information to set up the call
> properly and until you have it, you have not completed the call setup.....
> We will absolutely not try to hack our sip stack.
>
> If you are here to participate in the community, try to help rather than
> force comments on people and start arguments.  Maybe try listening to the
> advice people are giving you......
>
>
>
>
>
> On Fri, Nov 16, 2012 at 3:56 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>
>> This is where I am getting a bit confused, if the 200OK arrived to the
>> other side and we checked that the RTP exists (with mod_sofia option) we
>> will not get to hours of calls (unless the other side did not hanged up)
>> In any case there is a good chance that the BYE is getting lost, so the
>> danger exist even without the ACK
>>
>> I am guessing that the designers of the SIP protocol came up with the ACK
>> because there is a potential for open dialog that is not bound with time
>> and they therefore wanted to know that the other side actually ACKs the
>> request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
>> tied with RTP (at least in most normal SIP implementations) I'm not sure
>> that this ACK is that needed
>> but again as I said it is more of philosophical debate, maybe a potential
>> request in the new RFC
>>
>> On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  In this case masking the issue can lead to massive bills... Imaging
>>> paying by the minute... All of a sudden you are now leaving 2 minute calls
>>> up for hours on end... And continuing to get billed for them... Or you are
>>> not continuing to bill a customer for them... And now you have unexpected
>>> HUGE bills coming in... Masking it is far worse then just fixing it...
>>>
>>> If we mask this one issue, we might as well mask memory leaks, or
>>> passwords that don?t work, etc... Sure sometimes we might have to mask an
>>> issue for production to work in the short term, but that is never the
>>> correct answer fix a problem
>>>
>>>
>>> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>>
>>> While I agree on the details I disagree on the solution
>>> Sometimes masking the problems can be a good solution but I guess it is
>>> a philosophical debate
>>>
>>>
>>> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>> That leaves to big a risk of open sessions and only masks the true issue
>>> which is a problem with FS getting the ACK back...
>>>
>>> Theres a reason FS is not getting the ACK, and FS will make several
>>> attempts to get an ack by retransmitting the 200 OK several times before
>>> that timeout occurs.
>>>
>>> The real fix here is to fix the underlying cause, not masking it....
>>>
>>>
>>> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>>> http://yiftah at choochee.com> > wrote:
>>>
>>> I know that it is kind out of the what RFC3261 instructs, but did anyone
>>> think on giving the option in configuration not to hang up calls in case of
>>> an ACK does not arrive?
>>> I know that it has the risk of open sessions but there some other ways
>>> to handle those cases
>>>
>>> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>>> http://krice at freeswitch.org> > wrote:
>>>
>>> This is probably the same scenario as this is exactly what to expect...
>>> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
>>> the call....
>>>
>>> Quite common on networks with NAT issues or broken endpoints
>>>
>>>
>>> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>>> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>>
>>> I saw this happened earlier when the remote party does not send SIP ACK
>>> after receiving SIP OK, so the call is being disconnected after exactly 32
>>> seconds.
>>> Not sure if this is exact same scenario here, but just something to
>>> consider...
>>>
>>> Regards.
>>> Vitalie
>>>
>>>
>>> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
>>>  <http://k4kaleem at gmail.com> >
>>>
>>> Hi All,
>>>
>>> my inbound calls are fine with no issues, my outbound calls get
>>> disconnected after 32 seconds and its on all calls. i tried 2 different
>>> suppliers and its same result.
>>> please find the attached log file with sofia in debug mode. - caller was
>>> extension 1234 and desination was 01908321682
>>>
>>> your help will be greately appreciated.
>>>
>>> regards,
>>> Kaleem
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>> http://consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <
>>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>>> http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>> http://consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <
>>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>>> http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Ken
>>> *http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>> *irc.freenode.net #freeswitch
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yiftah at choochee.com  Sat Nov 17 21:04:53 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sat, 17 Nov 2012 10:04:53 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <B7EB28B2-F8F0-4648-A5AF-481C2482C396@freeswitch.org>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<B7EB28B2-F8F0-4648-A5AF-481C2482C396@freeswitch.org>
Message-ID: <CAFVUaV9OHUfUwcisPrgVZFDw19J9351U2aGU58_cD2wxq4gKsQ@mail.gmail.com>

I do not want to continue this discussion because apparently it hurts
people when you have different opinion if you would like to have a fact
base discussion with no personal offenses we can continue talking about it
through are private mails

On Fri, Nov 16, 2012 at 7:31 PM, Ken Rice <krice at freeswitch.org> wrote:

> dont forget that sip and rtp do not nessecarily go through the same hosts
> on the same call... how can you monitor rtp on a host if the rtp never goes
> there
>
> Ken
> Sent from my iPad
>
> On Nov 16, 2012, at 3:56 PM, Yiftach Golan <yiftah at choochee.com> wrote:
>
> This is where I am getting a bit confused, if the 200OK arrived to the
> other side and we checked that the RTP exists (with mod_sofia option) we
> will not get to hours of calls (unless the other side did not hanged up)
> In any case there is a good chance that the BYE is getting lost, so the
> danger exist even without the ACK
>
> I am guessing that the designers of the SIP protocol came up with the ACK
> because there is a potential for open dialog that is not bound with time
> and they therefore wanted to know that the other side actually ACKs the
> request, but since ACK is tied to INVITE only (AFAIK) and INVITE always
> tied with RTP (at least in most normal SIP implementations) I'm not sure
> that this ACK is that needed
> but again as I said it is more of philosophical debate, maybe a potential
> request in the new RFC
>
> On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  In this case masking the issue can lead to massive bills... Imaging
>> paying by the minute... All of a sudden you are now leaving 2 minute calls
>> up for hours on end... And continuing to get billed for them... Or you are
>> not continuing to bill a customer for them... And now you have unexpected
>> HUGE bills coming in... Masking it is far worse then just fixing it...
>>
>> If we mask this one issue, we might as well mask memory leaks, or
>> passwords that don?t work, etc... Sure sometimes we might have to mask an
>> issue for production to work in the short term, but that is never the
>> correct answer fix a problem
>>
>>
>> On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>
>> While I agree on the details I disagree on the solution
>> Sometimes masking the problems can be a good solution but I guess it is a
>> philosophical debate
>>
>>
>> On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> That leaves to big a risk of open sessions and only masks the true issue
>> which is a problem with FS getting the ACK back...
>>
>> Theres a reason FS is not getting the ACK, and FS will make several
>> attempts to get an ack by retransmitting the 200 OK several times before
>> that timeout occurs.
>>
>> The real fix here is to fix the underlying cause, not masking it....
>>
>>
>> On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>> http://yiftah at choochee.com> > wrote:
>>
>> I know that it is kind out of the what RFC3261 instructs, but did anyone
>> think on giving the option in configuration not to hang up calls in case of
>> an ACK does not arrive?
>> I know that it has the risk of open sessions but there some other ways to
>> handle those cases
>>
>> On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>> http://krice at freeswitch.org> > wrote:
>>
>> This is probably the same scenario as this is exactly what to expect...
>> Call gets answered far end doesn?t ACK FS sending them a 200OK , fs hangsup
>> the call....
>>
>> Quite common on networks with NAT issues or broken endpoints
>>
>>
>> On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>> http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>
>> I saw this happened earlier when the remote party does not send SIP ACK
>> after receiving SIP OK, so the call is being disconnected after exactly 32
>> seconds.
>> Not sure if this is exact same scenario here, but just something to
>> consider...
>>
>> Regards.
>> Vitalie
>>
>>
>> 2012/11/15 kaleem rehman <k4kaleem at gmail.com <http://k4kaleem at gmail.com>
>>  <http://k4kaleem at gmail.com> >
>>
>> Hi All,
>>
>> my inbound calls are fine with no issues, my outbound calls get
>> disconnected after 32 seconds and its on all calls. i tried 2 different
>> suppliers and its same result.
>> please find the attached log file with sofia in debug mode. - caller was
>> extension 1234 and desination was 01908321682
>>
>> your help will be greately appreciated.
>>
>> regards,
>> Kaleem
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>> http://consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <
>> http://FreeSWITCH-users at lists.freeswitch.org>  <
>> http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Ken
>> *http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> *irc.freenode.net #freeswitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From william.suffill at gmail.com  Sat Nov 17 21:13:04 2012
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 17 Nov 2012 13:13:04 -0500
Subject: [Freeswitch-users] Pager Duty Service on FreeSWITCH
In-Reply-To: <CAD7Ssm_iS83R2Hm9e52nxaNZjN9DwfB0KsWyZRaXzywZP5iWdQ@mail.gmail.com>
References: <CAD7Ssm_iS83R2Hm9e52nxaNZjN9DwfB0KsWyZRaXzywZP5iWdQ@mail.gmail.com>
Message-ID: <CAF-ZooSBy1z_mwS1F8a1JNwjqBtq4_8ow2LHNyXmDe2UQtbhtg@mail.gmail.com>

Not really sure what you are after. If you could clarify what you are
trying to do someone might be able to give you better insight on how to get
what you are after.


On Fri, Nov 16, 2012 at 10:05 PM, Kaushal Shriyan
<kaushalshriyan at gmail.com>wrote:

> Hi,
>
> Does FreeSwitch has pager duty feature and write ups or How To's to setup?
>
> Regards,
>
> Kaushal
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yiftah at choochee.com  Sat Nov 17 21:17:33 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sat, 17 Nov 2012 10:17:33 -0800
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAKbxfG-dWuNCgykuNWHV=QxTkwHkFhjKDwC0wK6ezTFVPhhcWg@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<1546109.RqNAKPZJSR@sos>
	<CAFVUaV9VnVwVeNn_KFHZwV3Q4N5u8rrcMv5RbPzfh143j29KTA@mail.gmail.com>
	<CAFVUaV-hPaPUZLQtp+J_VLwHfYNnMQE_TzcUymPqurmbm6AmAQ@mail.gmail.com>
	<CAKbxfG-dWuNCgykuNWHV=QxTkwHkFhjKDwC0wK6ezTFVPhhcWg@mail.gmail.com>
Message-ID: <CAFVUaV-8HUU++2m=6sah4ovQz6sde4-hL4Hv2qDKdzF00WXFwA@mail.gmail.com>

Again Anthony I am not trying to push you to change anything just wanted to
have a discussion if I want to change anything I would change in the code
by myself
You are looking for an ulterior motive when there is none
I get it that you want this forum to be more questions and answers about
FreeSWITCH configurations and as I stated  before I will not correct people
or give my thoughts on what I think even if they are missing some facts in
their assumptions (including in this mail) which they consider as premise

On Sat, Nov 17, 2012 at 7:21 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The only reason that there would be a failure to get the ACK would be if
> the 2 endpoints have lost communication.
> If you can't get an ACK to the host, you can't get a BYE to them either.
>  Even if you hacked it to tolerate no ACK, the session timers would fail in
> another 30 seconds.
>
> You are just focused on your business needs and just disregarding logic.
>  You ask if you are missing something and several individuals are telling
> you YES.  This is the 4th time I am telling you that you should focus
> your enthusiasm on learning rather than affirming a m00t point because we
> are not going to change the behavior.
>
> I have not seen a seen a single case of missing ACK that did not point out
> a real problem that can be simply fixed by either using some NAT related
> configuration and proper formed Contact hosts.
>
> If this argument continues it will be ended with moderation.....
>
>
> On Fri, Nov 16, 2012 at 6:07 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>
>> Here is the reason from the RFC3261 :
>>    "...The reason for this separation is rooted in the importance of
>>       delivering all 200 (OK) responses to an INVITE to the UAC.  To
>>       deliver them all to the UAC, the UAS alone takes responsibility
>>       for retransmitting them (see Section 13.3.1.4), and the UAC alone
>>       takes responsibility for acknowledging them with ACK (see Section
>>       13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
>>       effectively considered its own transaction..."
>>
>>
>> On Fri, Nov 16, 2012 at 3:50 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>>
>>> Yes I you are right if the RTP is not tied with SIP this theory is not
>>> really valid
>>> But this is again true about BYE that does not reach to the destination
>>> so the risk of not closing the dialog still exist
>>> In most of the voice implementation that I saw there are three options :
>>> 1. Softswitch (FreeSWITCH, Asterisk, etc)
>>> 2. MGW (Avaya, Cisco, etc)
>>> 3. Media release but only between two phones
>>> You can tie your RTP to the SIP with option 1 and option 2
>>> Option 3 is the problematic one but you will never release your media
>>> for billing purpose so usually your risk will be extension to extension
>>> open dialog which is less riskier
>>> Again as I said it is pretty philosophical debate but from my long
>>> experience in SIP I do not think that there is a practical use for it but
>>> maybe I am missing something
>>>
>>> On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin <
>>> sos at sokhapkin.dyndns.org> wrote:
>>>
>>>>  Don't mix signaling (SIP) and media (RTP). Signaling and media could
>>>> run
>>>> different ways. Why do you think FS will always be in media path?
>>>>
>>>> On Friday 16 November 2012 14:56:58 Yiftach Golan wrote:
>>>> > This is where I am getting a bit confused, if the 200OK arrived to the
>>>> > other side and we checked that the RTP exists (with mod_sofia option)
>>>> we
>>>> > will not get to hours of calls (unless the other side did not hanged
>>>> up)
>>>> > In any case there is a good chance that the BYE is getting lost, so
>>>> the
>>>> > danger exist even without the ACK
>>>> >
>>>> > I am guessing that the designers of the SIP protocol came up with the
>>>> ACK
>>>> > because there is a potential for open dialog that is not bound with
>>>> time
>>>> > and they therefore wanted to know that the other side actually ACKs
>>>> the
>>>> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE
>>>> always
>>>> > tied with RTP (at least in most normal SIP implementations) I'm not
>>>> sure
>>>> > that this ACK is that needed
>>>> > but again as I said it is more of philosophical debate, maybe a
>>>> potential
>>>> > request in the new RFC
>>>> >
>>>> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org>
>>>> wrote:
>>>> > >  In this case masking the issue can lead to massive bills... Imaging
>>>> > >
>>>> > > paying by the minute... All of a sudden you are now leaving 2
>>>> minute calls
>>>> > > up for hours on end... And continuing to get billed for them... Or
>>>> you are
>>>> > > not continuing to bill a customer for them... And now you have
>>>> unexpected
>>>> > > HUGE bills coming in... Masking it is far worse then just fixing
>>>> it...
>>>> > >
>>>> > > If we mask this one issue, we might as well mask memory leaks, or
>>>> > > passwords that don?t work, etc... Sure sometimes we might have to
>>>> mask an
>>>> > > issue for production to work in the short term, but that is never
>>>> the
>>>> > > correct answer fix a problem
>>>> > >
>>>> > >
>>>> > > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>>> > >
>>>> > > While I agree on the details I disagree on the solution
>>>> > > Sometimes masking the problems can be a good solution but I guess
>>>> it is a
>>>> > > philosophical debate
>>>> > >
>>>> > >
>>>> > > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
>>>> wrote:
>>>> > >
>>>> > > That leaves to big a risk of open sessions and only masks the true
>>>> issue
>>>> > > which is a problem with FS getting the ACK back...
>>>> > >
>>>> > > Theres a reason FS is not getting the ACK, and FS will make several
>>>> > > attempts to get an ack by retransmitting the 200 OK several times
>>>> before
>>>> > > that timeout occurs.
>>>> > >
>>>> > > The real fix here is to fix the underlying cause, not masking it....
>>>> > >
>>>> > >
>>>> > > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>>>> > > http://yiftah at choochee.com> > wrote:
>>>> > >
>>>> > > I know that it is kind out of the what RFC3261 instructs, but did
>>>> anyone
>>>> > > think on giving the option in configuration not to hang up calls in
>>>> case
>>>> > > of
>>>> > > an ACK does not arrive?
>>>> > > I know that it has the risk of open sessions but there some other
>>>> ways to
>>>> > > handle those cases
>>>> > >
>>>> > > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>>>> > > http://krice at freeswitch.org> > wrote:
>>>> > >
>>>> > > This is probably the same scenario as this is exactly what to
>>>> expect...
>>>> > > Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
>>>> > > hangsup
>>>> > > the call....
>>>> > >
>>>> > > Quite common on networks with NAT issues or broken endpoints
>>>> > >
>>>> > >
>>>> > > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>>>> > > http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>>> > >
>>>> > > I saw this happened earlier when the remote party does not send SIP
>>>> ACK
>>>> > > after receiving SIP OK, so the call is being disconnected after
>>>> exactly 32
>>>> > > seconds.
>>>> > > Not sure if this is exact same scenario here, but just something to
>>>> > > consider...
>>>> > >
>>>> > > Regards.
>>>> > > Vitalie
>>>> > >
>>>> > >
>>>> > > 2012/11/15 kaleem rehman <k4kaleem at gmail.com <
>>>> http://k4kaleem at gmail.com>
>>>> > >
>>>> > >  <http://k4kaleem at gmail.com> >
>>>> > >
>>>> > > Hi All,
>>>> > >
>>>> > > my inbound calls are fine with no issues, my outbound calls get
>>>> > > disconnected after 32 seconds and its on all calls. i tried 2
>>>> different
>>>> > > suppliers and its same result.
>>>> > > please find the attached log file with sofia in debug mode. -
>>>> caller was
>>>> > > extension 1234 and desination was 01908321682
>>>> > >
>>>> > > your help will be greately appreciated.
>>>> > >
>>>> > > regards,
>>>> > > Kaleem
>>>> > >
>>>> > >
>>>> _________________________________________________________________________
>>>> > > Professional FreeSWITCH Consulting Services:
>>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>>> > > http://consulting at freeswitch.org>
>>>> > > http://www.freeswitchsolutions.com
>>>> > >
>>>> > > 
>>>> > > 
>>>> > >
>>>> > > Official FreeSWITCH Sites
>>>> > > http://www.freeswitch.org
>>>> > > http://wiki.freeswitch.org
>>>> > > http://www.cluecon.com
>>>> > >
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > > http://www.freeswitch.org
>>>> > >
>>>> > >
>>>> > >
>>>> > > ------------------------------
>>>> > >
>>>> _________________________________________________________________________
>>>> > > Professional FreeSWITCH Consulting Services:
>>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>>> > > http://consulting at freeswitch.org>
>>>> > > http://www.freeswitchsolutions.com
>>>> > >
>>>> > > 
>>>> > > 
>>>> > >
>>>> > > Official FreeSWITCH Sites
>>>> > > http://www.freeswitch.org
>>>> > > http://wiki.freeswitch.org
>>>> > > http://www.cluecon.com
>>>> > >
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > > http://www.freeswitch.org
>>>> > >
>>>> > >
>>>> > > --
>>>> > > Ken
>>>> > > *http://www.FreeSWITCH.org
>>>> > > http://www.ClueCon.com
>>>> > > http://www.OSTAG.org
>>>> > > *irc.freenode.net #freeswitch
>>>> > >
>>>> > >
>>>> _________________________________________________________________________
>>>> > > Professional FreeSWITCH Consulting Services:
>>>> > > consulting at freeswitch.org
>>>> > > http://www.freeswitchsolutions.com
>>>> > >
>>>> > > 
>>>> > > 
>>>> > >
>>>> > > Official FreeSWITCH Sites
>>>> > > http://www.freeswitch.org
>>>> > > http://wiki.freeswitch.org
>>>> > > http://www.cluecon.com
>>>> > >
>>>> > > FreeSWITCH-users mailing list
>>>> > > FreeSWITCH-users at lists.freeswitch.org
>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > > http://www.freeswitch.org
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Nov 17 22:03:00 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Nov 2012 13:03:00 -0600
Subject: [Freeswitch-users] uuid_media leaves behind zombie call
In-Reply-To: <027901cdc4de$6e305ee0$4a911ca0$@com>
References: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>
	<CAKbxfG-YU0Sz2wd0mZ9+rv3MON13f6AYbJcwxPa1GdmsuGaRoQ@mail.gmail.com>
	<870204F45EE7D34E8D27CC0E602E11A1B46979@EX01.semafone.local>
	<CAKbxfG8q5qn5XPeyNL2nhOmGzUjvbJMJ0qMZTEVnhwfYb5NwHw@mail.gmail.com>
	<027901cdc4de$6e305ee0$4a911ca0$@com>
Message-ID: <CAKbxfG_8wC=jFfRDE=ji0y_XnmN8tCnfAUMOdL0MMmnWNFM28Q@mail.gmail.com>

I thought I explained it rather thoroughly....

If someone has a problem, the chance of getting it fixed over the list is
significantly smaller because its difficult to track amongst the other 500
emails a day we field.

So we try to make a policy to better HELP people but you were compelled to
make a comment as if its wrong for us to decide how to best manage a very
large community.....

If real problems that require code changes were omitted from the list and
submitted to Jira, it would reduce the traffic here about 5% at best and
only improve the resolution rate. so, yes, there are still plenty of
reasons to still have the list.

This kind of comment really discourages me.......
On Nov 17, 2012 12:22 PM, "Bote Man" <bote_radio at botecomm.com> wrote:

> Well, if you want the additional workload, fine.****
>
> ** **
>
> One might reasonably wonder why this user mailing list exists in the first
> place, if not to ask questions and learn about FreeSWITCH?****
>
> ** **
>
> Bote****
>
> ** **
>
> ** **
>
> *From:* Anthony Minessale
> *Sent:* Friday, 16 November, 2012 11:14
>
> No problem, I just want to try and trace peoples path into the community
> and make sure we have all the proper information at first glance.****
>
> ** **
>
> Even in a case of uncertainty, Jira is better because its a database of
> "issues" and we always have a "not a bug" resolution.****
>
> Even in cases of mistaken configuration, it can be considered a valid
> issue and we can change defaults etc.****
>
> ** **
>
> Jiras are not expensive to create and easy to close, just a lot more
> organized.****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hakkie42 at gmail.com  Sat Nov 17 22:53:02 2012
From: hakkie42 at gmail.com (Jim)
Date: Sat, 17 Nov 2012 20:53:02 +0100
Subject: [Freeswitch-users] call disconnects after 32 seconds issue -
 FreeSWITCH-users Digest, Vol 77, Issue 117
In-Reply-To: <CAFWsKqzkeURVdTvmTKvVxBFhtKHpLRE0LcKB9VATjhsJiNKUyg@mail.gmail.com>
References: <CAFWsKqzkeURVdTvmTKvVxBFhtKHpLRE0LcKB9VATjhsJiNKUyg@mail.gmail.com>
Message-ID: <50A7EB1E.1050604@gmail.com>

On 17-11-2012 18:14, kaleem rehman wrote:
> Hi All,
>  
> Its kaleem, i reported the issue with call disconnecting after 32
> seconds, i have managed to fix the issue. please find the detailed fix
> below;
>  
> i had to get the external Interface (internet facing) to pass 5060
> TCP/UDP & 5080 TCP/UDP to freeswitch server.

<snip>

> now log on to your router and change all port fowarding (please refer to
> documentation for your router) forward port you can see in log file(in
> my case 5060 & 5080) to freeswitch server.
Aren't those ports defined in your profile? Seems there's no need to go
hunting through the log if you can fish them out of your fs config?
See here:
http://wiki.freeswitch.org/wiki/Firewall

<snip>

> i dont want to post messy documentation so will clean up a bit and
> hopefully FreeSwitch admins will import to WIKI

Actually, you can (and are encouraged to) edit the wiki yourself. Having
this info on the mailing list is nice but not as useful.


From anthony.minessale at gmail.com  Sun Nov 18 00:45:52 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Nov 2012 15:45:52 -0600
Subject: [Freeswitch-users] call disconnects after 32 seconds
In-Reply-To: <CAFVUaV-8HUU++2m=6sah4ovQz6sde4-hL4Hv2qDKdzF00WXFwA@mail.gmail.com>
References: <CAFVUaV8UNom2yZnYxtRsjJJT99Rz2hSsCMuWqbbmbBwCvbtGNQ@mail.gmail.com>
	<CCCC0505.60B94%krice@freeswitch.org>
	<CAFVUaV87xz0_8uijmCqKa+oaKtWk_6UGKUhDUBuMS7mEp7nUnQ@mail.gmail.com>
	<1546109.RqNAKPZJSR@sos>
	<CAFVUaV9VnVwVeNn_KFHZwV3Q4N5u8rrcMv5RbPzfh143j29KTA@mail.gmail.com>
	<CAFVUaV-hPaPUZLQtp+J_VLwHfYNnMQE_TzcUymPqurmbm6AmAQ@mail.gmail.com>
	<CAKbxfG-dWuNCgykuNWHV=QxTkwHkFhjKDwC0wK6ezTFVPhhcWg@mail.gmail.com>
	<CAFVUaV-8HUU++2m=6sah4ovQz6sde4-hL4Hv2qDKdzF00WXFwA@mail.gmail.com>
Message-ID: <CAKbxfG-GjZ4YANKBoNP7kCWo5LG0_kB41-Wi+nbRVQ-yeek4JA@mail.gmail.com>

Great.. The best news is that the reporter found his problem and fixed it
despite the tangents......
On Nov 17, 2012 2:33 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:

> Again Anthony I am not trying to push you to change anything just wanted
> to have a discussion if I want to change anything I would change in the
> code by myself
> You are looking for an ulterior motive when there is none
> I get it that you want this forum to be more questions and answers about
> FreeSWITCH configurations and as I stated  before I will not correct people
> or give my thoughts on what I think even if they are missing some facts in
> their assumptions (including in this mail) which they consider as premise
>
> On Sat, Nov 17, 2012 at 7:21 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> The only reason that there would be a failure to get the ACK would be if
>> the 2 endpoints have lost communication.
>> If you can't get an ACK to the host, you can't get a BYE to them either.
>>  Even if you hacked it to tolerate no ACK, the session timers would fail in
>> another 30 seconds.
>>
>> You are just focused on your business needs and just disregarding logic.
>>  You ask if you are missing something and several individuals are telling
>> you YES.  This is the 4th time I am telling you that you should focus
>> your enthusiasm on learning rather than affirming a m00t point because we
>> are not going to change the behavior.
>>
>> I have not seen a seen a single case of missing ACK that did not point
>> out a real problem that can be simply fixed by either using some NAT
>> related configuration and proper formed Contact hosts.
>>
>> If this argument continues it will be ended with moderation.....
>>
>>
>> On Fri, Nov 16, 2012 at 6:07 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>>
>>> Here is the reason from the RFC3261 :
>>>    "...The reason for this separation is rooted in the importance of
>>>       delivering all 200 (OK) responses to an INVITE to the UAC.  To
>>>       deliver them all to the UAC, the UAS alone takes responsibility
>>>       for retransmitting them (see Section 13.3.1.4), and the UAC alone
>>>       takes responsibility for acknowledging them with ACK (see Section
>>>       13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
>>>       effectively considered its own transaction..."
>>>
>>>
>>> On Fri, Nov 16, 2012 at 3:50 PM, Yiftach Golan <yiftah at choochee.com>wrote:
>>>
>>>> Yes I you are right if the RTP is not tied with SIP this theory is not
>>>> really valid
>>>> But this is again true about BYE that does not reach to the destination
>>>> so the risk of not closing the dialog still exist
>>>> In most of the voice implementation that I saw there are three options
>>>> :
>>>> 1. Softswitch (FreeSWITCH, Asterisk, etc)
>>>> 2. MGW (Avaya, Cisco, etc)
>>>> 3. Media release but only between two phones
>>>> You can tie your RTP to the SIP with option 1 and option 2
>>>> Option 3 is the problematic one but you will never release your media
>>>> for billing purpose so usually your risk will be extension to extension
>>>> open dialog which is less riskier
>>>> Again as I said it is pretty philosophical debate but from my long
>>>> experience in SIP I do not think that there is a practical use for it but
>>>> maybe I am missing something
>>>>
>>>> On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin <
>>>> sos at sokhapkin.dyndns.org> wrote:
>>>>
>>>>>  Don't mix signaling (SIP) and media (RTP). Signaling and media could
>>>>> run
>>>>> different ways. Why do you think FS will always be in media path?
>>>>>
>>>>> On Friday 16 November 2012 14:56:58 Yiftach Golan wrote:
>>>>> > This is where I am getting a bit confused, if the 200OK arrived to
>>>>> the
>>>>> > other side and we checked that the RTP exists (with mod_sofia
>>>>> option) we
>>>>> > will not get to hours of calls (unless the other side did not hanged
>>>>> up)
>>>>> > In any case there is a good chance that the BYE is getting lost, so
>>>>> the
>>>>> > danger exist even without the ACK
>>>>> >
>>>>> > I am guessing that the designers of the SIP protocol came up with
>>>>> the ACK
>>>>> > because there is a potential for open dialog that is not bound with
>>>>> time
>>>>> > and they therefore wanted to know that the other side actually ACKs
>>>>> the
>>>>> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE
>>>>> always
>>>>> > tied with RTP (at least in most normal SIP implementations) I'm not
>>>>> sure
>>>>> > that this ACK is that needed
>>>>> > but again as I said it is more of philosophical debate, maybe a
>>>>> potential
>>>>> > request in the new RFC
>>>>> >
>>>>> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice at freeswitch.org>
>>>>> wrote:
>>>>> > >  In this case masking the issue can lead to massive bills...
>>>>> Imaging
>>>>> > >
>>>>> > > paying by the minute... All of a sudden you are now leaving 2
>>>>> minute calls
>>>>> > > up for hours on end... And continuing to get billed for them... Or
>>>>> you are
>>>>> > > not continuing to bill a customer for them... And now you have
>>>>> unexpected
>>>>> > > HUGE bills coming in... Masking it is far worse then just fixing
>>>>> it...
>>>>> > >
>>>>> > > If we mask this one issue, we might as well mask memory leaks, or
>>>>> > > passwords that don?t work, etc... Sure sometimes we might have to
>>>>> mask an
>>>>> > > issue for production to work in the short term, but that is never
>>>>> the
>>>>> > > correct answer fix a problem
>>>>> > >
>>>>> > >
>>>>> > > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:
>>>>> > >
>>>>> > > While I agree on the details I disagree on the solution
>>>>> > > Sometimes masking the problems can be a good solution but I guess
>>>>> it is a
>>>>> > > philosophical debate
>>>>> > >
>>>>> > >
>>>>> > > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice at freeswitch.org>
>>>>> wrote:
>>>>> > >
>>>>> > > That leaves to big a risk of open sessions and only masks the true
>>>>> issue
>>>>> > > which is a problem with FS getting the ACK back...
>>>>> > >
>>>>> > > Theres a reason FS is not getting the ACK, and FS will make several
>>>>> > > attempts to get an ack by retransmitting the 200 OK several times
>>>>> before
>>>>> > > that timeout occurs.
>>>>> > >
>>>>> > > The real fix here is to fix the underlying cause, not masking
>>>>> it....
>>>>> > >
>>>>> > >
>>>>> > > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah at choochee.com <
>>>>> > > http://yiftah at choochee.com> > wrote:
>>>>> > >
>>>>> > > I know that it is kind out of the what RFC3261 instructs, but did
>>>>> anyone
>>>>> > > think on giving the option in configuration not to hang up calls
>>>>> in case
>>>>> > > of
>>>>> > > an ACK does not arrive?
>>>>> > > I know that it has the risk of open sessions but there some other
>>>>> ways to
>>>>> > > handle those cases
>>>>> > >
>>>>> > > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice at freeswitch.org <
>>>>> > > http://krice at freeswitch.org> > wrote:
>>>>> > >
>>>>> > > This is probably the same scenario as this is exactly what to
>>>>> expect...
>>>>> > > Call gets answered far end doesn?t ACK FS sending them a 200OK , fs
>>>>> > > hangsup
>>>>> > > the call....
>>>>> > >
>>>>> > > Quite common on networks with NAT issues or broken endpoints
>>>>> > >
>>>>> > >
>>>>> > > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100 at gmail.com <
>>>>> > > http://vetali100 at gmail.com>  <http://vetali100 at gmail.com> > wrote:
>>>>> > >
>>>>> > > I saw this happened earlier when the remote party does not send
>>>>> SIP ACK
>>>>> > > after receiving SIP OK, so the call is being disconnected after
>>>>> exactly 32
>>>>> > > seconds.
>>>>> > > Not sure if this is exact same scenario here, but just something to
>>>>> > > consider...
>>>>> > >
>>>>> > > Regards.
>>>>> > > Vitalie
>>>>> > >
>>>>> > >
>>>>> > > 2012/11/15 kaleem rehman <k4kaleem at gmail.com <
>>>>> http://k4kaleem at gmail.com>
>>>>> > >
>>>>> > >  <http://k4kaleem at gmail.com> >
>>>>> > >
>>>>> > > Hi All,
>>>>> > >
>>>>> > > my inbound calls are fine with no issues, my outbound calls get
>>>>> > > disconnected after 32 seconds and its on all calls. i tried 2
>>>>> different
>>>>> > > suppliers and its same result.
>>>>> > > please find the attached log file with sofia in debug mode. -
>>>>> caller was
>>>>> > > extension 1234 and desination was 01908321682
>>>>> > >
>>>>> > > your help will be greately appreciated.
>>>>> > >
>>>>> > > regards,
>>>>> > > Kaleem
>>>>> > >
>>>>> > >
>>>>> _________________________________________________________________________
>>>>> > > Professional FreeSWITCH Consulting Services:
>>>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>>>> > > http://consulting at freeswitch.org>
>>>>> > > http://www.freeswitchsolutions.com
>>>>> > >
>>>>> > > 
>>>>> > > 
>>>>> > >
>>>>> > > Official FreeSWITCH Sites
>>>>> > > http://www.freeswitch.org
>>>>> > > http://wiki.freeswitch.org
>>>>> > > http://www.cluecon.com
>>>>> > >
>>>>> > > FreeSWITCH-users mailing list
>>>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > > http://www.freeswitch.org
>>>>> > >
>>>>> > >
>>>>> > >
>>>>> > > ------------------------------
>>>>> > >
>>>>> _________________________________________________________________________
>>>>> > > Professional FreeSWITCH Consulting Services:
>>>>> > > consulting at freeswitch.org <http://consulting at freeswitch.org>  <
>>>>> > > http://consulting at freeswitch.org>
>>>>> > > http://www.freeswitchsolutions.com
>>>>> > >
>>>>> > > 
>>>>> > > 
>>>>> > >
>>>>> > > Official FreeSWITCH Sites
>>>>> > > http://www.freeswitch.org
>>>>> > > http://wiki.freeswitch.org
>>>>> > > http://www.cluecon.com
>>>>> > >
>>>>> > > FreeSWITCH-users mailing list
>>>>> > > FreeSWITCH-users at lists.freeswitch.org <
>>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>  <
>>>>> > > http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > > http://www.freeswitch.org
>>>>> > >
>>>>> > >
>>>>> > > --
>>>>> > > Ken
>>>>> > > *http://www.FreeSWITCH.org
>>>>> > > http://www.ClueCon.com
>>>>> > > http://www.OSTAG.org
>>>>> > > *irc.freenode.net #freeswitch
>>>>> > >
>>>>> > >
>>>>> _________________________________________________________________________
>>>>> > > Professional FreeSWITCH Consulting Services:
>>>>> > > consulting at freeswitch.org
>>>>> > > http://www.freeswitchsolutions.com
>>>>> > >
>>>>> > > 
>>>>> > > 
>>>>> > >
>>>>> > > Official FreeSWITCH Sites
>>>>> > > http://www.freeswitch.org
>>>>> > > http://wiki.freeswitch.org
>>>>> > > http://www.cluecon.com
>>>>> > >
>>>>> > > FreeSWITCH-users mailing list
>>>>> > > FreeSWITCH-users at lists.freeswitch.org
>>>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jh.zhou at outlook.com  Sat Nov 17 16:48:34 2012
From: jh.zhou at outlook.com (ZhouJianhua)
Date: Sat, 17 Nov 2012 13:48:34 +0000
Subject: [Freeswitch-users] How the IAX2 is supported in freeswitch?
Message-ID: <COL002-W992796C4D7183B4D4649B2F8500@phx.gbl>

Hi,
I need to reduce the bandwidth of media stream, and RTP is too heavy for us.IAX can carry more data in one packet, but the protocol is not supported as well as SIP in FS.
I just want to know that is there anyone used IAX in freeswitch ?And is it worth a try?
Thanks. 		 	   		  
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From andrew at cassidywebservices.co.uk  Sun Nov 18 01:27:53 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sat, 17 Nov 2012 22:27:53 +0000
Subject: [Freeswitch-users] call disconnects after 32 seconds issue -
 FreeSWITCH-users Digest, Vol 77, Issue 117
In-Reply-To: <50A7EB1E.1050604@gmail.com>
References: <CAFWsKqzkeURVdTvmTKvVxBFhtKHpLRE0LcKB9VATjhsJiNKUyg@mail.gmail.com>
	<50A7EB1E.1050604@gmail.com>
Message-ID: <CAMbW9o+_px50wfuCx_nFFXbmu4NeO2Lz6AYZQW0Z2Dh=oe=h=g@mail.gmail.com>

I just use stun to get around these issues, does half of what you've just
described automatically.

<param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
<param name="ext-sip-ip" value="stun:stun.freeswitch.org"/>

in my sip profiles did the trick.

On 17 November 2012 19:53, Jim <hakkie42 at gmail.com> wrote:

> On 17-11-2012 18:14, kaleem rehman wrote:
> > Hi All,
> >
> > Its kaleem, i reported the issue with call disconnecting after 32
> > seconds, i have managed to fix the issue. please find the detailed fix
> > below;
> >
> > i had to get the external Interface (internet facing) to pass 5060
> > TCP/UDP & 5080 TCP/UDP to freeswitch server.
>
> <snip>
>
> > now log on to your router and change all port fowarding (please refer to
> > documentation for your router) forward port you can see in log file(in
> > my case 5060 & 5080) to freeswitch server.
> Aren't those ports defined in your profile? Seems there's no need to go
> hunting through the log if you can fish them out of your fs config?
> See here:
> http://wiki.freeswitch.org/wiki/Firewall
>
> <snip>
>
> > i dont want to post messy documentation so will clean up a bit and
> > hopefully FreeSwitch admins will import to WIKI
>
> Actually, you can (and are encouraged to) edit the wiki yourself. Having
> this info on the mailing list is nice but not as useful.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From sos at sokhapkin.dyndns.org  Sun Nov 18 01:49:56 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 17 Nov 2012 17:49:56 -0500
Subject: [Freeswitch-users] How the IAX2 is supported in freeswitch?
In-Reply-To: <COL002-W992796C4D7183B4D4649B2F8500@phx.gbl>
References: <COL002-W992796C4D7183B4D4649B2F8500@phx.gbl>
Message-ID: <3202266.kfoy5uzG5i@sos>

 I'd suggest to use standard high compression codecs instead of non-standard 
IAX protocol.

On Saturday 17 November 2012 13:48:34 ZhouJianhua wrote:
> Hi,
> I need to reduce the bandwidth of media stream, and RTP is too heavy for
> us.IAX can carry more data in one packet, but the protocol is not supported
> as well as SIP in FS.
 I just want to know that is there anyone used IAX in
> freeswitch ?And is it worth a try? Thanks.


From krice at freeswitch.org  Sun Nov 18 03:36:55 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 17 Nov 2012 18:36:55 -0600
Subject: [Freeswitch-users] How the IAX2 is supported in freeswitch?
In-Reply-To: <COL002-W992796C4D7183B4D4649B2F8500@phx.gbl>
Message-ID: <CCCD89C7.60BDC%krice@freeswitch.org>

IAX2 doesn?t really offer that much lower bandwidth then SIP... Adjusting
the ptime on calls can achieve the same effect as enabling IAX2 trunking.
example:

G711 PCMU or PCMA at 20ms ptime is about 80kbps,

Move to G729 and this drops to 24kbps, about a 70% savings in bandwidth, now
add a 60ms ptime to this, and the bandwidth requirement drops to 13.3kbps
nearly an over all 84% savings in bandwidth...

So the who IAX argument is killed at that point

On 11/17/12 7:48 AM, "ZhouJianhua" <jh.zhou at outlook.com> wrote:

> Hi,
> 
> I need to reduce the bandwidth of media stream, and RTP is too heavy for us.
> IAX can carry more data in one packet, but the protocol is not supported as
> well as SIP in FS.
> 
> I just want to know that is there anyone used IAX in freeswitch ?
> And is it worth a try?
> 
> Thanks.
>        
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From gabe at gundy.org  Sun Nov 18 04:08:08 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 17 Nov 2012 18:08:08 -0700
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
	<CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
	<CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>
Message-ID: <CAMwZ-t+hkyfuf87b=48MfTMALDSyB7Efa-6BkfkKmMiLrx7+pA@mail.gmail.com>

On Wed, Aug 29, 2012 at 7:36 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> the stun / ice stuff is mandatory so ya.  turn I have no exp with.
> you need stun/ice/srtp as a prereq.  There are lot of changes to do
> into to core to refactor and avoid code dup and do it right that come
> before the new module its a giant undertaking so we are looking for
> sponsors.

I'd love to see this happen. My company, Izeni, would also love to
contribute some cash to it. However, WebRTC seems like a bigger deal
than bounties are typically used for.

Whatever the case, we'll put $1000 toward the effort. This is not a
bounty (conditional based on finishing the feature). It's a
no-obligation, thanks-for-an-awesome-product,
good-luck-with-future-development, contribution.  I hope other
companies that benefit daily from FreeSWITCH consider following suit.
Let's make this happen :)

FreeSWITCH and WebRTC FTW!

Best,
Gabe (and all the FreeSWITCH fans at Izeni)


From kaushalshriyan at gmail.com  Sun Nov 18 04:13:39 2012
From: kaushalshriyan at gmail.com (Kaushal Shriyan)
Date: Sun, 18 Nov 2012 06:43:39 +0530
Subject: [Freeswitch-users] Pager Duty Service on FreeSWITCH
In-Reply-To: <CAF-ZooSBy1z_mwS1F8a1JNwjqBtq4_8ow2LHNyXmDe2UQtbhtg@mail.gmail.com>
References: <CAD7Ssm_iS83R2Hm9e52nxaNZjN9DwfB0KsWyZRaXzywZP5iWdQ@mail.gmail.com>
	<CAF-ZooSBy1z_mwS1F8a1JNwjqBtq4_8ow2LHNyXmDe2UQtbhtg@mail.gmail.com>
Message-ID: <CAD7Ssm8cppQ1x281w5vME=Y4Wj4BtCo_9hbXBo_4=XtVRdQa1w@mail.gmail.com>

On Sat, Nov 17, 2012 at 11:43 PM, William Suffill <william.suffill at gmail.com
> wrote:

> Not really sure what you are after. If you could clarify what you are
> trying to do someone might be able to give you better insight on how to get
> what you are after.
>
>
Hi William

This product is a http://www.pagerduty.com commercial and is there a open
source product similar to this one and Freeswitch has support for time
based operations.

Regards

Kaushal
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From 8f27e956 at gmail.com  Sun Nov 18 04:42:21 2012
From: 8f27e956 at gmail.com (Scott)
Date: Sat, 17 Nov 2012 20:42:21 -0500
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
	method
Message-ID: <CA+Y7Lf4+gfzWayBbLLGj6x4ckfOqMbXRB9tOniQUkXe0uU+T0w@mail.gmail.com>

Are there reasons why the function 'hash' (non-persistent storage) and
function 'db' (persistent storage) share a look-a-like user interface (e.g.
select/insert/delete) but do not work-a-like.  In particular, in 'hash',
insert overwrites an identical realm/data_key pair whereas 'db' , in so far
as I can tell, just blindly adds, and adds, adds, the insert(s).  However,
the 'db' select method returns one record only even if the db has
accumulated many realm/data_key records, including data_value duplicates.

NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
non-persistent 'hash' with the persistent 'db'  I *think* is straight
forward ...

(1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
index, as follows,

CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
('hostname','realm','data_key');

(1) At the c-language embedded SQL string-level,

change the existing INSERT to be the following,

INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
(%s,%s,%s,%s);

The index will help speed UP the reads.  Neither change is "exotic" and
should be portable across standard sql implementations.

;-) ... thoughts ?

Thanks,
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From bote_radio at botecomm.com  Sun Nov 18 05:15:42 2012
From: bote_radio at botecomm.com (Bote Man)
Date: Sat, 17 Nov 2012 21:15:42 -0500
Subject: [Freeswitch-users] uuid_media leaves behind zombie call
In-Reply-To: <CAKbxfG_8wC=jFfRDE=ji0y_XnmN8tCnfAUMOdL0MMmnWNFM28Q@mail.gmail.com>
References: <870204F45EE7D34E8D27CC0E602E11A1B468D5@EX01.semafone.local>
	<CAKbxfG-YU0Sz2wd0mZ9+rv3MON13f6AYbJcwxPa1GdmsuGaRoQ@mail.gmail.com>
	<870204F45EE7D34E8D27CC0E602E11A1B46979@EX01.semafone.local>
	<CAKbxfG8q5qn5XPeyNL2nhOmGzUjvbJMJ0qMZTEVnhwfYb5NwHw@mail.gmail.com>
	<027901cdc4de$6e305ee0$4a911ca0$@com>
	<CAKbxfG_8wC=jFfRDE=ji0y_XnmN8tCnfAUMOdL0MMmnWNFM28Q@mail.gmail.com>
Message-ID: <039f01cdc532$9d77e750$d867b5f0$@com>

No need to be discouraged. Maybe my message was off-base or poorly
communicated, apologies.

 

In any case, I will continue to contribute in what little way I can by
updating the wiki from solutions found here in the hopes that it will
obviate the need for questions to this list nor open tickets on Jira,
thus reducing your support burden.

 

Regards,

Bote

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Saturday, 17 November, 2012 14:03
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] uuid_media leaves behind zombie call

 

I thought I explained it rather thoroughly....

If someone has a problem, the chance of getting it fixed over the list
is significantly smaller because its difficult to track amongst the
other 500 emails a day we field. 

So we try to make a policy to better HELP people but you were compelled
to make a comment as if its wrong for us to decide how to best manage a
very large community.....

If real problems that require code changes were omitted from the list
and submitted to Jira, it would reduce the traffic here about 5% at best
and only improve the resolution rate. so, yes, there are still plenty of
reasons to still have the list.  

This kind of comment really discourages me.......

On Nov 17, 2012 12:22 PM, "Bote Man" <bote_radio at botecomm.com> wrote:

Well, if you want the additional workload, fine.

 

One might reasonably wonder why this user mailing list exists in the
first place, if not to ask questions and learn about FreeSWITCH?

 

Bote

 

 

From: Anthony Minessale
Sent: Friday, 16 November, 2012 11:14

No problem, I just want to try and trace peoples path into the community
and make sure we have all the proper information at first glance.

 

Even in a case of uncertainty, Jira is better because its a database of
"issues" and we always have a "not a bug" resolution.

Even in cases of mistaken configuration, it can be considered a valid
issue and we can change defaults etc.

 

Jiras are not expensive to create and easy to close, just a lot more
organized.

 


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




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

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

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From gerald.weber at besharp.at  Sun Nov 18 10:50:36 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Sun, 18 Nov 2012 07:50:36 +0000
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db'
	insert	method
In-Reply-To: <CA+Y7Lf4+gfzWayBbLLGj6x4ckfOqMbXRB9tOniQUkXe0uU+T0w@mail.gmail.com>
References: <CA+Y7Lf4+gfzWayBbLLGj6x4ckfOqMbXRB9tOniQUkXe0uU+T0w@mail.gmail.com>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2D3EC3@exchange.besharp.at>

AFAIK, "if not exists" and "or replace" is not ansi sql  syntax, so you will get in trouble if
someone uses odbc to a db that doesn't support non standard syntax.

The insert or replace could be replaced with the merge statement.

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Scott
Gesendet: Sonntag, 18. November 2012 02:42
An: FreeSWITCH Users Help
Betreff: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert method

Are there reasons why the function 'hash' (non-persistent storage) and function 'db' (persistent storage) share a look-a-like user interface (e.g. select/insert/delete) but do not work-a-like.  In particular, in 'hash', insert overwrites an identical realm/data_key pair whereas 'db' , in so far as I can tell, just blindly adds, and adds, adds, the insert(s).  However, the 'db' select method returns one record only even if the db has accumulated many realm/data_key records, including data_value duplicates.

NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the non-persistent 'hash' with the persistent 'db'  I *think* is straight forward ...

(1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND index, as follows,

CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data' ('hostname','realm','data_key');

(1) At the c-language embedded SQL string-level,

change the existing INSERT to be the following,

INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES (%s,%s,%s,%s);

The index will help speed UP the reads.  Neither change is "exotic" and should be portable across standard sql implementations.

;-) ... thoughts ?

Thanks,


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From 8f27e956 at gmail.com  Sun Nov 18 14:25:30 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Sun, 18 Nov 2012 06:25:30 -0500
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
	method
In-Reply-To: <D1925DECD316894DAADB84076C1B7D827E2D3EC3@exchange.besharp.at>
References: <CA+Y7Lf4+gfzWayBbLLGj6x4ckfOqMbXRB9tOniQUkXe0uU+T0w@mail.gmail.com>
	<D1925DECD316894DAADB84076C1B7D827E2D3EC3@exchange.besharp.at>
Message-ID: <-1713583309525454317@unknownmsgid>

The IF NOT EXITS term is easily omitted.  It's a helper term for batch
scripted SQL to prevent an error being thrown in the case of the index
already exists as in the case where the make schema script is run more than
once.  Easily omitted and, perhaps more properly, replaced with a DROP
INDEX stmt and CREATE UNIQUE INDEX stmt pair.

Yeah, i checked further, the OR REPLACE modifier is a little more Hit/miss.
 The capability is there typically present, but the syntax varies -- e.g
sqlite3 ON REPLACE is mySQL's ON DUPLICATE KEY UPDATE.

Darn shame sqlite3 doesn't support stored procedures 'cause then it's more
easily abstracted for portability up the SQL server engine curve.

On 2012-11-18, at 3:53, Gerald Weber <gerald.weber at besharp.at> wrote:

AFAIK, ?if not exists? and ?or replace? is not ansi sql  syntax, so you
will get in trouble if

someone uses odbc to a db that doesn?t support non standard syntax.



The insert or replace could be replaced with the merge statement.



*Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Scott
*Gesendet:* Sonntag, 18. November 2012 02:42
*An:* FreeSWITCH Users Help
*Betreff:* [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
method



Are there reasons why the function 'hash' (non-persistent storage) and
function 'db' (persistent storage) share a look-a-like user interface (e.g.
select/insert/delete) but do not work-a-like.  In particular, in 'hash',
insert overwrites an identical realm/data_key pair whereas 'db' , in so far
as I can tell, just blindly adds, and adds, adds, the insert(s).  However,
the 'db' select method returns one record only even if the db has
accumulated many realm/data_key records, including data_value duplicates.

NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
non-persistent 'hash' with the persistent 'db'  I *think* is straight
forward ...

(1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
index, as follows,

CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
('hostname','realm','data_key');

(1) At the c-language embedded SQL string-level,

change the existing INSERT to be the following,

INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
(%s,%s,%s,%s);

The index will help speed UP the reads.  Neither change is "exotic" and
should be portable across standard sql implementations.

;-) ... thoughts ?

Thanks,


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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From 8f27e956 at gmail.com  Sun Nov 18 16:26:07 2012
From: 8f27e956 at gmail.com (Scott)
Date: Sun, 18 Nov 2012 08:26:07 -0500
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
	method
In-Reply-To: <-1713583309525454317@unknownmsgid>
References: <CA+Y7Lf4+gfzWayBbLLGj6x4ckfOqMbXRB9tOniQUkXe0uU+T0w@mail.gmail.com>
	<D1925DECD316894DAADB84076C1B7D827E2D3EC3@exchange.besharp.at>
	<-1713583309525454317@unknownmsgid>
Message-ID: <CA+Y7Lf42nq1=snVW-7nFXHRqvvW9WC8FSSkOy6wjTuSYXGtAaw@mail.gmail.com>

OK, with the one-time CREATE UNIQUE INDEX on 'table' (
hostname,realm,data_key); stmt in place at the schema-level, a portable
work-a-like of INSERT OR REPLACE stmt is the UPDATE-INSERT (i.e. "UPSERT"
lol) stmt pair that follows,

BEGIN
TRANSACTION;

UPDATE 'table' SET data_value='value_data' WHERE hostname='value_hostname'
AND realm='value_realm' AND data_key='value_data_key' ;
INSERT INTO 'table' (hostname,realm,data_key,data_value) SELECT
'value_hostname', 'value_realm', 'value_data_key', 'value_data_value'
   WHERE NOT EXISTS (SELECT 1 FROM 'table' WHERE hostname='value_hostname'
AND realm='value_realm' AND data_key='value_data_key' );
COMMIT;


The two WHERE clauses must be identical and the variables should be in the
same order as the unique compound index is ordered.
The UPDATE fails quietly and benignly if the record does NOT already exist
and succeeds if it does.
The INSERT ... WHERE NOT EXISTS stmt succeeds if the record does NOT
already exist fails quietly and benignly if it does.
The BEGIN and COMMIT aren't typically necessary but they're over-safe just
in case the same record is attempted by another thread.

I think this get's it done.

:-)


On 18 November 2012 06:25, S. Scott <8f27e956 at gmail.com> wrote:

> The IF NOT EXITS term is easily omitted.  It's a helper term for batch
> scripted SQL to prevent an error being thrown in the case of the index
> already exists as in the case where the make schema script is run more than
> once.  Easily omitted and, perhaps more properly, replaced with a DROP
> INDEX stmt and CREATE UNIQUE INDEX stmt pair.
>
> Yeah, i checked further, the OR REPLACE modifier is a little more
> Hit/miss.  The capability is there typically present, but the syntax varies
> -- e.g sqlite3 ON REPLACE is mySQL's ON DUPLICATE KEY UPDATE.
>
> Darn shame sqlite3 doesn't support stored procedures 'cause then it's more
> easily abstracted for portability up the SQL server engine curve.
>
> On 2012-11-18, at 3:53, Gerald Weber <gerald.weber at besharp.at> wrote:
>
> AFAIK, ?if not exists? and ?or replace? is not ansi sql  syntax, so you
> will get in trouble if
>
> someone uses odbc to a db that doesn?t support non standard syntax.
>
>
>
> The insert or replace could be replaced with the merge statement.
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Scott
> *Gesendet:* Sonntag, 18. November 2012 02:42
> *An:* FreeSWITCH Users Help
> *Betreff:* [Freeswitch-users] Suggestion to harmonize 'hash' & 'db'
> insert method
>
>
>
> Are there reasons why the function 'hash' (non-persistent storage) and
> function 'db' (persistent storage) share a look-a-like user interface (e.g.
> select/insert/delete) but do not work-a-like.  In particular, in 'hash',
> insert overwrites an identical realm/data_key pair whereas 'db' , in so far
> as I can tell, just blindly adds, and adds, adds, the insert(s).  However,
> the 'db' select method returns one record only even if the db has
> accumulated many realm/data_key records, including data_value duplicates.
>
> NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
> non-persistent 'hash' with the persistent 'db'  I *think* is straight
> forward ...
>
> (1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
> index, as follows,
>
> CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
> ('hostname','realm','data_key');
>
> (1) At the c-language embedded SQL string-level,
>
> change the existing INSERT to be the following,
>
> INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
> (%s,%s,%s,%s);
>
> The index will help speed UP the reads.  Neither change is "exotic" and
> should be portable across standard sql implementations.
>
> ;-) ... thoughts ?
>
> Thanks,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mitch.capper at gmail.com  Sun Nov 18 17:30:59 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sun, 18 Nov 2012 06:30:59 -0800
Subject: [Freeswitch-users] G729 File version of $${us-ring}
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AD9A4EC@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AD9A4EC@MAIL.millicorp.com>
Message-ID: <CALn233qk8dMRD2hmNdn-f5_R5GJdBxyrQDycV2JJqaOO3Mmktg@mail.gmail.com>

As long as you have a single G729 license just create it yourself
us_ring is just the tone stream used so create a g729 call, record it,
and play the tone stream.

~mitch

On Sat, Nov 17, 2012 at 8:25 AM, Tim Meade <Tim.Meade at millicorp.com> wrote:
>
>
> I was wondering if anyone had a G729 file of the $${us-ring}
>
>
>
> We want to use it for playing the ringback with G729 pass through.
>
>
>
> <action application="set"  data="ringback=SomeRingBackFile.G729"/>
>
>
>
>
>
> Thanks
>
>
>
> Tim
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From krice at freeswitch.org  Sun Nov 18 18:41:53 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 18 Nov 2012 09:41:53 -0600
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
 method
In-Reply-To: <CA+Y7Lf42nq1=snVW-7nFXHRqvvW9WC8FSSkOy6wjTuSYXGtAaw@mail.gmail.com>
Message-ID: <CCCE5DE1.60BFA%krice@freeswitch.org>

You have to keep in mind that we have to support more then just mysql or
sqlite with the the db interface... This means using SQL that?s common to
ALL platforms include MSSQL, PostgreSQL, Oracle and any other database
someone might want to use via ODBC...

So to do this completely would require modification to the code on insert to
first check if the row exists for the key, then deleting it, then inserting
the new row...  This is going to be the most generic supportable method to
make them work consistently across the board...

However, Keep in mind that what you are calling a bug, could be treated by
many people as a feature... Don?t allow me to create/replace and already
existing key...




On 11/18/12 7:26 AM, "Scott" <8f27e956 at gmail.com> wrote:

> OK, with the one-time CREATE UNIQUE INDEX on 'table'
> (hostname,realm,data_key); stmt in place at the schema-level, a portable
> work-a-like of INSERT OR REPLACE stmt is the UPDATE-INSERT (i.e. "UPSERT" lol)
> stmt pair that follows,
> 
> BEGIN 
> TRANSACTION;??????????????????????????????????????????????????????????????????
> ??????????????????????????????????????????????
> UPDATE 'table' SET data_value='value_data' WHERE hostname='value_hostname' AND
> realm='value_realm' AND data_key='value_data_key' ;
> INSERT INTO 'table' (hostname,realm,data_key,data_value) SELECT
> 'value_hostname', 'value_realm', 'value_data_key', 'value_data_value'
> ?? WHERE NOT EXISTS (SELECT 1 FROM 'table' WHERE hostname='value_hostname' AND
> realm='value_realm' AND data_key='value_data_key' );
> COMMIT;???????????????????????????????????????????????????????????????????????
> ?????????????????????????????????????????????????????
> 
> The two WHERE clauses must be identical and the variables should be in the
> same order as the unique compound index is ordered.
> The UPDATE fails quietly and benignly if the record does NOT already exist and
> succeeds if it does.
> The INSERT ... WHERE NOT EXISTS stmt succeeds if the record does NOT already
> exist fails quietly and benignly if it does.
> The BEGIN and COMMIT aren't typically necessary but they're over-safe just in
> case the same record is attempted by another thread.
> 
> I think this get's it done.
> 
> :-)
> 
> 
> On 18 November 2012 06:25, S. Scott <8f27e956 at gmail.com> wrote:
>> The IF NOT EXITS term is easily omitted. ?It's a helper term for batch
>> scripted SQL to prevent an error being thrown in the case of the index
>> already exists as in the case where the make schema script is run more than
>> once. ?Easily omitted and, perhaps more properly, replaced with a DROP INDEX
>> stmt and CREATE UNIQUE INDEX stmt pair.
>> 
>> Yeah, i checked further, the OR REPLACE modifier is a little more Hit/miss.
>> ?The capability is there typically present, but the syntax varies -- e.g
>> sqlite3 ON REPLACE is mySQL's ON DUPLICATE KEY UPDATE.
>> 
>> Darn shame sqlite3 doesn't support stored procedures 'cause then it's more
>> easily abstracted for portability up the SQL server engine curve.
>> 
>> On 2012-11-18, at 3:53, Gerald Weber <gerald.weber at besharp.at> wrote:
>> 
>>> AFAIK, ?if not exists? and ?or replace? is not ansi sql ?syntax, so you will
>>> get in trouble if
>>> someone uses odbc to a db that doesn?t support non standard syntax.
>>> ?
>>> The insert or replace could be replaced with the merge statement.
>>> ?
>>> Von: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Scott
>>> Gesendet: Sonntag, 18. November 2012 02:42
>>> An: FreeSWITCH Users Help
>>> Betreff: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
>>> method
>>> ?
>>> Are there reasons why the function 'hash' (non-persistent storage) and
>>> function 'db' (persistent storage) share a look-a-like user interface (e.g.
>>> select/insert/delete) but do not work-a-like.? In particular, in 'hash',
>>> insert overwrites an identical realm/data_key pair whereas 'db' , in so far
>>> as I can tell, just blindly adds, and adds, adds, the insert(s).? However,
>>> the 'db' select method returns one record only even if the db has
>>> accumulated many realm/data_key records, including data_value duplicates.?
>>> 
>>> NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
>>> non-persistent 'hash' with the persistent 'db'? I *think* is straight
>>> forward ...
>>> 
>>> (1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
>>> index, as follows,
>>> 
>>> CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
>>> ('hostname','realm','data_key');
>>> 
>>> (1) At the c-language embedded SQL string-level,
>>> 
>>> change the existing INSERT to be the following,
>>> 
>>> INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
>>> (%s,%s,%s,%s);
>>> 
>>> The index will help speed UP the reads.? Neither change is "exotic" and
>>> should be portable across standard sql implementations.
>>> 
>>> ;-) ... thoughts ?
>>> 
>>> Thanks,
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From ahmed at netelsat.net  Sun Nov 18 13:21:35 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Sun, 18 Nov 2012 15:21:35 +0500
Subject: [Freeswitch-users] Freeswitch Sonus bypass media
In-Reply-To: <CALFFxt3OoGhop_PKwKNKk6p-F-xdo4+HgNfyxCzHXcaO26S4hg@mail.gmail.com>
References: <CALFFxt3OoGhop_PKwKNKk6p-F-xdo4+HgNfyxCzHXcaO26S4hg@mail.gmail.com>
Message-ID: <CALFFxt0OobB4cb0CL1ucN3y+AVOamHj-8oV7+udd-pzaJCB94w@mail.gmail.com>

Dear All,

i've read in past or may be still Freeswitch and sonus are not good with
each other . but every post shows its always rtp between FS and Sonus. In
my case i am bypassing all media , so just signalling between US (using FS)
and Provider (Sonus) . And ACD is extreme low as 4-5 minutes due to 100s of
short calls ending with normal call clearing.
Can any one Please help if in bypass still FS and Sonus can have some
problem.

Thanks in advance
regards
soFh
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From rajprithiv88 at gmail.com  Sun Nov 18 18:18:25 2012
From: rajprithiv88 at gmail.com (Rajkumar K)
Date: Sun, 18 Nov 2012 20:48:25 +0530
Subject: [Freeswitch-users] FreeSWITCH High
 availability(Normal_temporary_failure problem)
In-Reply-To: <CAD-6L_Xf8eD2=MTKscrZxwkpPBafe3-j5dSXV9RM2jjbrgZc2Q@mail.gmail.com>
References: <CAFYz+X=+BrXLChSegXxOw4hD+qFWdOM=V-_sOMhRYyt4A6Y1+A@mail.gmail.com>
	<CAD-6L_Xf8eD2=MTKscrZxwkpPBafe3-j5dSXV9RM2jjbrgZc2Q@mail.gmail.com>
Message-ID: <CAFYz+Xk06HQXgS_S4LA0Py+Cm=tDfmMBBPZnhs1B7mxH5Kgw3Q@mail.gmail.com>

Hi,

Thanks for your response.. i got the solution it was the timing mismatch
between the two servers...
Thanks for your time Eliot...




On Sat, Nov 10, 2012 at 4:33 AM, Eliot Gable <egable+freeswitch at gmail.com>wrote:

> I have not looked at your invalid CSeq error, but since you are using
> pacemaker, you might find this resource agent useful. Here is an example of
> how to use it:
>
> primitive FreeSWITCH ocf:fssolutions:FreeSWITCH \
>
>         params ips="eth0/217.160.21.196/26:eth1/217.160.21.197/26"
> user="freeswitch" group="freeswitch" \
>
>         op monitor interval="3s" role="Master" on-fail="restart" depth="0"
> \
>
>         op monitor interval="10s" role="Slave" on-fail="restart" depth="0"
> \
>
>         op start interval="0" timeout="30" \
>
>         op stop interval="0" timeout="60"
>
> ms FreeSWITCH-MS FreeSWITCH \
>
>         meta master-max="1" master-node-max="1" clone-max="2"
> clone-node-max="1" \
>
>                 notify="false" target-role="Master"
>
> location FreeSWITCH-MS-on-ha02 FreeSWITCH-MS 50: ha02.bkw.org
>
> location FreeSWITCH-MS-on-ha01 FreeSWITCH-MS 50: ha01.bkw.org
>
> property $id="cib-bootstrap-options" \
>
>         dc-version="1.1.7-6.el6-148fccfd5985c5590cc601123c6c16e966b85d14" \
>
>         cluster-infrastructure="corosync" \
>
>         expected-quorum-votes="2" \
>
>         stonith-enabled="false" \
>
>         no-quorum-policy="ignore" \
>
>         last-lrm-refresh="1347305291"
>
> rsc_defaults $id="rsc-options" \
>
>         resource-stickiness="100"
>
>
> On Wed, Nov 7, 2012 at 12:03 AM, Rajkumar K <rajprithiv88 at gmail.com>wrote:
>
>> Hi,
>>
>> I am trying to achieve high availability in FreeSWITCH using heartbeat
>> and pacemaker and I am able to switch between the server whenever one of
>> the servers crashes. But the problem is one server is able to recover the
>> calls when im invoking sofia recover, but another server is recovering the
>> call one few times.(Mostly not able to recover).
>>
>>
>> I have primary and secondary server installed in two different
>> machines(centos), the FreeSWITCH instances are always running in both the
>> PCs. I am running heartbeat and pacemaker to monitor the IP or sofia
>> fail-over. A floating IP is configure in heartbeat to reach the active
>> server.
>> I succeeded in switching between the servers whenever IP or FreeSWITCH
>> and once it reaches the another server it invokes sofia recover to recover
>> the calls.
>>
>> Both freeswtich instances are using the same configuration and database
>> is shared using ODBC connectivity.
>>
>> Problem is:
>>
>> Call is made using the primary server, and i did fsctl crash in primary
>> server cli. Heartbeat resource switches to secondary server and it invokes
>> "sofia profile internal restart" and "sofia recover". and it recovers the
>> call. The call gets recovered in 4-5 seconds.
>>
>> At the same time i will start the freeswitch instance in the primary
>> server. Now if i crash the secondary server using fsctl crash, the
>> resources switches to primary server and it invokes "sofia profile internal
>> restart" and "sofia recover". Also the server sends invite request to the
>> clients But it ends in NORMAL_TEMPORARY_FAILURE. Wireshark log says Client
>> is responding with "Invalid CSeq" for the Server's INVITE request. This
>> happens always with the primary server and very few times primary server is
>> also able to recover the calls.
>>
>> I have the same configurations in both the servers
>> And also i checked by stopping the heartbeat switching, and crashed the
>> primary server's freeswitch. Then if i start freeswitch again in the same
>> server and invoking sofia recover will recover the calls without any
>> problem.
>>
>>
>> I have also attached the cli logs of primary and secondary servers.
>> I am not able to identify the exact problem in this, Please help me out
>> in this problem.
>>
>>
>> Thanks
>> Rajkumar
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even considered
> to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
> not live to eat.) ~Marcus Tullius Cicero
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From itispip-qq at hotmail.com  Sun Nov 18 19:44:13 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Mon, 19 Nov 2012 00:44:13 +0800
Subject: [Freeswitch-users] How to enable unicode in CLI command?
Message-ID: <BLU169-W654F5F82AD03E6AC000B8F8C570@phx.gbl>


I tried to use chat / skypopen_chat from CLI console to send a message to receiver, the message is written by unicode charactor (Korean or Thai), the receiver cannot get correct charators;
Tried to call chat / skypopen_chat API from LUA/JScript, ended the same;
I believe Freeswitch itself support unicode, as when I use an SIP client as "Bria" to send SIP chat message between 2 clients, unicode charactor displays correctly.
Is it just the CLI command by design not support unicode, or I have to do something in config xml?
 		 	   		  
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From avi at avimarcus.net  Sun Nov 18 19:48:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 18 Nov 2012 18:48:53 +0200
Subject: [Freeswitch-users] Freeswitch Sonus bypass media
In-Reply-To: <CALFFxt0OobB4cb0CL1ucN3y+AVOamHj-8oV7+udd-pzaJCB94w@mail.gmail.com>
References: <CALFFxt3OoGhop_PKwKNKk6p-F-xdo4+HgNfyxCzHXcaO26S4hg@mail.gmail.com>
	<CALFFxt0OobB4cb0CL1ucN3y+AVOamHj-8oV7+udd-pzaJCB94w@mail.gmail.com>
Message-ID: <CALNZuwmpd2Qe_CgCOmFd+P96f2dz0TTHaPx134raUUQD7=c=Sw@mail.gmail.com>

I was under the impression that an ACD of over 3 minutes was good.
-Avi


On Sun, Nov 18, 2012 at 12:21 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Dear All,
>
> i've read in past or may be still Freeswitch and sonus are not good with
> each other . but every post shows its always rtp between FS and Sonus. In
> my case i am bypassing all media , so just signalling between US (using FS)
> and Provider (Sonus) . And ACD is extreme low as 4-5 minutes due to 100s of
> short calls ending with normal call clearing.
> Can any one Please help if in bypass still FS and Sonus can have some
> problem.
>
> Thanks in advance
> regards
> soFh
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Sun Nov 18 20:42:06 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 18 Nov 2012 11:42:06 -0600
Subject: [Freeswitch-users] Freeswitch Sonus bypass media
In-Reply-To: <CALFFxt0OobB4cb0CL1ucN3y+AVOamHj-8oV7+udd-pzaJCB94w@mail.gmail.com>
Message-ID: <CCCE7A0E.60C01%krice@freeswitch.org>

FreeSWITCH has built in compensation for sonus endpoints... It will fix
sonus? screw ups...


On 11/18/12 4:21 AM, "Ahmed Sboor" <ahmed at netelsat.net> wrote:

> Dear All,
> 
> i've read in past or may be still Freeswitch and sonus are not good with each
> other . but every post shows its always rtp between FS and Sonus. In my case i
> am bypassing all media , so just signalling between US (using FS) and Provider
> (Sonus) . And ACD is extreme low as 4-5 minutes due to 100s of short calls
> ending with normal call clearing.
> Can any one Please help if in bypass still FS and Sonus can have some problem.
> 
> Thanks in advance
> regards
> soFh
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From yiftah at choochee.com  Sun Nov 18 23:04:15 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sun, 18 Nov 2012 12:04:15 -0800
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
	method
In-Reply-To: <CCCE5DE1.60BFA%krice@freeswitch.org>
References: <CA+Y7Lf42nq1=snVW-7nFXHRqvvW9WC8FSSkOy6wjTuSYXGtAaw@mail.gmail.com>
	<CCCE5DE1.60BFA%krice@freeswitch.org>
Message-ID: <CAFVUaV8mjzYsN0UnQSJnTvACgEGV_BgLRUSeeDGpRHMc8FGa1g@mail.gmail.com>

We have the exact same problem and we solve it pretty nicely but due to the
policy "do not say anything that is different than what we did" I will not
put it in public if you want Scott we can continue with you and Ken in
private
Also I would please like to ask you not to use inappropriate words like
"MySQL" in this forum as it may cause people to feel uncomfortable

On Sun, Nov 18, 2012 at 7:41 AM, Ken Rice <krice at freeswitch.org> wrote:

>  You have to keep in mind that we have to support more then just mysql or
> sqlite with the the db interface... This means using SQL that?s common to
> ALL platforms include MSSQL, PostgreSQL, Oracle and any other database
> someone might want to use via ODBC...
>
> So to do this completely would require modification to the code on insert
> to first check if the row exists for the key, then deleting it, then
> inserting the new row...  This is going to be the most generic supportable
> method to make them work consistently across the board...
>
> However, Keep in mind that what you are calling a bug, could be treated by
> many people as a feature... Don?t allow me to create/replace and already
> existing key...
>
>
>
>
> On 11/18/12 7:26 AM, "Scott" <8f27e956 at gmail.com> wrote:
>
> OK, with the one-time CREATE UNIQUE INDEX on 'table'
> (hostname,realm,data_key); stmt in place at the schema-level, a portable
> work-a-like of INSERT OR REPLACE stmt is the UPDATE-INSERT (i.e. "UPSERT"
> lol) stmt pair that follows,
>
> BEGIN
> TRANSACTION;
>
> UPDATE 'table' SET data_value='value_data' WHERE hostname='value_hostname'
> AND realm='value_realm' AND data_key='value_data_key' ;
> INSERT INTO 'table' (hostname,realm,data_key,data_value) SELECT
> 'value_hostname', 'value_realm', 'value_data_key', 'value_data_value'
>    WHERE NOT EXISTS (SELECT 1 FROM 'table' WHERE hostname='value_hostname'
> AND realm='value_realm' AND data_key='value_data_key' );
> COMMIT;
>
>
> The two WHERE clauses must be identical and the variables should be in the
> same order as the unique compound index is ordered.
> The UPDATE fails quietly and benignly if the record does NOT already exist
> and succeeds if it does.
> The INSERT ... WHERE NOT EXISTS stmt succeeds if the record does NOT
> already exist fails quietly and benignly if it does.
> The BEGIN and COMMIT aren't typically necessary but they're over-safe just
> in case the same record is attempted by another thread.
>
> I think this get's it done.
>
> :-)
>
>
> On 18 November 2012 06:25, S. Scott <8f27e956 at gmail.com> wrote:
>
> The IF NOT EXITS term is easily omitted.  It's a helper term for batch
> scripted SQL to prevent an error being thrown in the case of the index
> already exists as in the case where the make schema script is run more than
> once.  Easily omitted and, perhaps more properly, replaced with a DROP
> INDEX stmt and CREATE UNIQUE INDEX stmt pair.
>
> Yeah, i checked further, the OR REPLACE modifier is a little more
> Hit/miss.  The capability is there typically present, but the syntax varies
> -- e.g sqlite3 ON REPLACE is mySQL's ON DUPLICATE KEY UPDATE.
>
> Darn shame sqlite3 doesn't support stored procedures 'cause then it's more
> easily abstracted for portability up the SQL server engine curve.
>
> On 2012-11-18, at 3:53, Gerald Weber <gerald.weber at besharp.at> wrote:
>
> AFAIK, ?if not exists? and ?or replace? is not ansi sql  syntax, so you
> will get in trouble if
> someone uses odbc to a db that doesn?t support non standard syntax.
>
> The insert or replace could be replaced with the merge statement.
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *Im Auftrag von *Scott
> *Gesendet:* Sonntag, 18. November 2012 02:42
> *An:* FreeSWITCH Users Help
> *Betreff:* [Freeswitch-users] Suggestion to harmonize 'hash' & 'db'
> insert method
>
> Are there reasons why the function 'hash' (non-persistent storage) and
> function 'db' (persistent storage) share a look-a-like user interface (e.g.
> select/insert/delete) but do not work-a-like.  In particular, in 'hash',
> insert overwrites an identical realm/data_key pair whereas 'db' , in so far
> as I can tell, just blindly adds, and adds, adds, the insert(s).  However,
> the 'db' select method returns one record only even if the db has
> accumulated many realm/data_key records, including data_value duplicates.
>
> NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
> non-persistent 'hash' with the persistent 'db'  I *think* is straight
> forward ...
>
> (1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
> index, as follows,
>
> CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
> ('hostname','realm','data_key');
>
> (1) At the c-language embedded SQL string-level,
>
> change the existing INSERT to be the following,
>
> INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
> (%s,%s,%s,%s);
>
> The index will help speed UP the reads.  Neither change is "exotic" and
> should be portable across standard sql implementations.
>
> ;-) ... thoughts ?
>
> Thanks,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yiftah at choochee.com  Sun Nov 18 23:11:06 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Sun, 18 Nov 2012 12:11:06 -0800
Subject: [Freeswitch-users] Freeswitch Sonus bypass media
In-Reply-To: <CALFFxt0OobB4cb0CL1ucN3y+AVOamHj-8oV7+udd-pzaJCB94w@mail.gmail.com>
References: <CALFFxt3OoGhop_PKwKNKk6p-F-xdo4+HgNfyxCzHXcaO26S4hg@mail.gmail.com>
	<CALFFxt0OobB4cb0CL1ucN3y+AVOamHj-8oV7+udd-pzaJCB94w@mail.gmail.com>
Message-ID: <CAFVUaV9Eg3g+Uxn60nckMx5Dk80zkRas7XUY9ETYKY1ueev4rA@mail.gmail.com>

We are using extensively FS and Sonus (not for signaling but only for RTP)
and it works great  no problems at all

On Sun, Nov 18, 2012 at 2:21 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Dear All,
>
> i've read in past or may be still Freeswitch and sonus are not good with
> each other . but every post shows its always rtp between FS and Sonus. In
> my case i am bypassing all media , so just signalling between US (using FS)
> and Provider (Sonus) . And ACD is extreme low as 4-5 minutes due to 100s of
> short calls ending with normal call clearing.
> Can any one Please help if in bypass still FS and Sonus can have some
> problem.
>
> Thanks in advance
> regards
> soFh
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ahmed at netelsat.net  Sun Nov 18 20:25:12 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Sun, 18 Nov 2012 22:25:12 +0500
Subject: [Freeswitch-users] Freeswitch Sonus bypass media
In-Reply-To: <CALNZuwmpd2Qe_CgCOmFd+P96f2dz0TTHaPx134raUUQD7=c=Sw@mail.gmail.com>
References: <CALFFxt3OoGhop_PKwKNKk6p-F-xdo4+HgNfyxCzHXcaO26S4hg@mail.gmail.com>
	<CALFFxt0OobB4cb0CL1ucN3y+AVOamHj-8oV7+udd-pzaJCB94w@mail.gmail.com>
	<CALNZuwmpd2Qe_CgCOmFd+P96f2dz0TTHaPx134raUUQD7=c=Sw@mail.gmail.com>
Message-ID: <CALFFxt2Q39mZO95qwSkx2yDnBHXhdpz+oiBkxFoHmsfCcfuDBQ@mail.gmail.com>

Hi Avi,
Thats good May be on non cli routes , but on route like India that too
direct from TATA . When people don't disconnect call for 2 hrs , 13-14 min
ACD is normally what TATA delivers.
So definitely there is something wrong .


On Sun, Nov 18, 2012 at 9:48 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I was under the impression that an ACD of over 3 minutes was good.
> -Avi
>
>
> On Sun, Nov 18, 2012 at 12:21 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Dear All,
>>
>> i've read in past or may be still Freeswitch and sonus are not good with
>> each other . but every post shows its always rtp between FS and Sonus. In
>> my case i am bypassing all media , so just signalling between US (using FS)
>> and Provider (Sonus) . And ACD is extreme low as 4-5 minutes due to 100s of
>> short calls ending with normal call clearing.
>> Can any one Please help if in bypass still FS and Sonus can have some
>> problem.
>>
>> Thanks in advance
>> regards
>> soFh
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Mon Nov 19 00:16:15 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 18 Nov 2012 15:16:15 -0600
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
 method
In-Reply-To: <CAFVUaV8mjzYsN0UnQSJnTvACgEGV_BgLRUSeeDGpRHMc8FGa1g@mail.gmail.com>
Message-ID: <CCCEAC3F.60C07%krice@freeswitch.org>

Ok that?s enough Yiftach...

No need to me a schmuck... Tony asked that the previous thread die for a
reason...

There is no policy about don?t say something different then what we did,
when the fact is there are probably 10 different ways to do most things
actually working within the framework that is freeswitch....

What you were asked to do is quit advocating for implementing things that
break the established protocols...

Enough is enough let it die... Further comments on that subject will result
in moderation...

K


On 11/18/12 2:04 PM, "Yiftach Golan" <yiftah at choochee.com> wrote:

> We have the exact same problem and we solve it pretty nicely but due to the
> policy "do not say anything that is different than what we did" I will not put
> it in public if you want Scott we can continue with you and Ken in private
> Also I would please like to ask you not to use inappropriate words like
> "MySQL" in this forum as it may cause people to feel uncomfortable
> ? 
> On Sun, Nov 18, 2012 at 7:41 AM, Ken Rice <krice at freeswitch.org> wrote:
>> You have to keep in mind that we have to support more then just mysql or
>> sqlite with the the db interface... This means using SQL that?s common to ALL
>> platforms include MSSQL, PostgreSQL, Oracle and any other database someone
>> might want to use via ODBC...
>> 
>> So to do this completely would require modification to the code on insert to
>> first check if the row exists for the key, then deleting it, then inserting
>> the new row... ?This is going to be the most generic supportable method to
>> make them work consistently across the board...
>> 
>> However, Keep in mind that what you are calling a bug, could be treated by
>> many people as a feature... Don?t allow me to create/replace and already
>> existing key...
>> 
>> 
>> 
>> 
>> On 11/18/12 7:26 AM, "Scott" <8f27e956 at gmail.com <http://8f27e956 at gmail.com>
>> > wrote:
>> 
>>> OK, with the one-time CREATE UNIQUE INDEX on 'table'
>>> (hostname,realm,data_key); stmt in place at the schema-level, a portable
>>> work-a-like of INSERT OR REPLACE stmt is the UPDATE-INSERT (i.e. "UPSERT"
>>> lol) stmt pair that follows,
>>> 
>>> BEGIN 
>>> TRANSACTION;????????????????????????????????????????????????????????????????
>>> ????????????????????????????????????????????????
>>> UPDATE 'table' SET data_value='value_data' WHERE hostname='value_hostname'
>>> AND realm='value_realm' AND data_key='value_data_key' ;
>>> INSERT INTO 'table' (hostname,realm,data_key,data_value) SELECT
>>> 'value_hostname', 'value_realm', 'value_data_key', 'value_data_value'
>>> ?? WHERE NOT EXISTS (SELECT 1 FROM 'table' WHERE hostname='value_hostname'
>>> AND realm='value_realm' AND data_key='value_data_key' );
>>> COMMIT;?????????????????????????????????????????????????????????????????????
>>> ???????????????????????????????????????????????????????
>>> 
>>> The two WHERE clauses must be identical and the variables should be in the
>>> same order as the unique compound index is ordered.
>>> The UPDATE fails quietly and benignly if the record does NOT already exist
>>> and succeeds if it does.
>>> The INSERT ... WHERE NOT EXISTS stmt succeeds if the record does NOT already
>>> exist fails quietly and benignly if it does.
>>> The BEGIN and COMMIT aren't typically necessary but they're over-safe just
>>> in case the same record is attempted by another thread.
>>> 
>>> I think this get's it done.
>>> 
>>> :-)
>>> 
>>> 
>>> On 18 November 2012 06:25, S. Scott <8f27e956 at gmail.com
>>> <http://8f27e956 at gmail.com> > wrote:
>>>> The IF NOT EXITS term is easily omitted. ?It's a helper term for batch
>>>> scripted SQL to prevent an error being thrown in the case of the index
>>>> already exists as in the case where the make schema script is run more than
>>>> once. ?Easily omitted and, perhaps more properly, replaced with a DROP
>>>> INDEX stmt and CREATE UNIQUE INDEX stmt pair.
>>>> 
>>>> Yeah, i checked further, the OR REPLACE modifier is a little more Hit/miss.
>>>> ?The capability is there typically present, but the syntax varies -- e.g
>>>> sqlite3 ON REPLACE is mySQL's ON DUPLICATE KEY UPDATE.
>>>> 
>>>> Darn shame sqlite3 doesn't support stored procedures 'cause then it's more
>>>> easily abstracted for portability up the SQL server engine curve.
>>>> 
>>>> On 2012-11-18, at 3:53, Gerald Weber <gerald.weber at besharp.at
>>>> <http://gerald.weber at besharp.at> > wrote:
>>>> 
>>>>> AFAIK, ?if not exists? and ?or replace? is not ansi sql ?syntax, so you
>>>>> will get in trouble if
>>>>> someone uses odbc to a db that doesn?t support non standard syntax.
>>>>> ?
>>>>> The insert or replace could be replaced with the merge statement.
>>>>> ?
>>>>> Von: freeswitch-users-bounces at lists.freeswitch.org
>>>>> <http://freeswitch-users-bounces at lists.freeswitch.org>
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
>>>>> Scott
>>>>> Gesendet: Sonntag, 18. November 2012 02:42
>>>>> An: FreeSWITCH Users Help
>>>>> Betreff: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
>>>>> method
>>>>> ?
>>>>> Are there reasons why the function 'hash' (non-persistent storage) and
>>>>> function 'db' (persistent storage) share a look-a-like user interface
>>>>> (e.g. select/insert/delete) but do not work-a-like.? In particular, in
>>>>> 'hash', insert overwrites an identical realm/data_key pair whereas 'db' ,
>>>>> in so far as I can tell, just blindly adds, and adds, adds, the
>>>>> insert(s).? However, the 'db' select method returns one record only even
>>>>> if the db has accumulated many realm/data_key records, including
>>>>> data_value duplicates.?
>>>>> 
>>>>> NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
>>>>> non-persistent 'hash' with the persistent 'db'? I *think* is straight
>>>>> forward ...
>>>>> 
>>>>> (1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
>>>>> index, as follows,
>>>>> 
>>>>> CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
>>>>> ('hostname','realm','data_key');
>>>>> 
>>>>> (1) At the c-language embedded SQL string-level,
>>>>> 
>>>>> change the existing INSERT to be the following,
>>>>> 
>>>>> INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
>>>>> (%s,%s,%s,%s);
>>>>> 
>>>>> The index will help speed UP the reads.? Neither change is "exotic" and
>>>>> should be portable across standard sql implementations.
>>>>> 
>>>>> ;-) ... thoughts ?
>>>>> 
>>>>> Thanks,
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org

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

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From 8f27e956 at gmail.com  Mon Nov 19 00:22:16 2012
From: 8f27e956 at gmail.com (Scott)
Date: Sun, 18 Nov 2012 16:22:16 -0500
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
	method
In-Reply-To: <CCCE5DE1.60BFA%krice@freeswitch.org>
References: <CA+Y7Lf42nq1=snVW-7nFXHRqvvW9WC8FSSkOy6wjTuSYXGtAaw@mail.gmail.com>
	<CCCE5DE1.60BFA%krice@freeswitch.org>
Message-ID: <CA+Y7Lf7f_2hT-0=00PPFutJ9PCk42Cta4HD9DTy7a9_=J+zSdQ@mail.gmail.com>

On 18 November 2012 10:41, Ken Rice <krice at freeswitch.org> wrote:

>  So to do this completely would require modification to the code on
> insert to first check if the row exists for the key, then deleting it, then
> inserting the new row...
>

The offered sql "upsert" stmt pair achieves natively what you're saying,
without the need for discrete check and delete and then insert logic c- or
p-language.  The sql is undergoing conversational refinement (than's
Gerald) to -- hopefully -- alight on the portable sql.

The reflection version of your statement is that in order for -- the user
-- to have harmonized hash-db behavior then -- the user -- has to do all
you prescribe -- in script or xml -- to one-only method (which is, in fact,
I've done now in my running use of persistent storage db dp_tool).


> However, Keep in mind that what you are calling a bug,
>
>
Two things  (1) In the op, I did concede "NOTWITHSTANDING
reasons-unknown-to-me;" and (2) I did not call it a "bug."

I, (we), were discussing harmonization of 'hash' and 'db' with specific
regard to the INSERT method -- GIVEN THAT the unbalanced db user interface
where SELECT method RETURNS one record only and DELETE method shot gun
removes ALL the realm/data_key records.  There *may* be compelling reasons
not to.  The "almost" work-a-like nature of the hash-db tools /suggests/ to
me that possibly at the once-upon-a-time of code commit an sql technique
just simply wasn't then known.  Perhaps -- and possibly a still-fledgling
perhaps at that -- a sql technique has emerged.  Perhaps not.

Whatever, I guess, if there's no traction, then there's no traction. It's a
common-enough sql challange that, perhaps, the sql fragments will help
someone somewhere in the future when google sussess this convo thread.

Cheers,


On 11/18/12 7:26 AM, "Scott" <8f27e956 at gmail.com> wrote:

OK, with the one-time CREATE UNIQUE INDEX on 'table'
> (hostname,realm,data_key); stmt in place at the schema-level, a portable
> work-a-like of INSERT OR REPLACE stmt is the UPDATE-INSERT (i.e. "UPSERT"
> lol) stmt pair that follows,
>
> BEGIN
> TRANSACTION;
>
> UPDATE 'table' SET data_value='value_data' WHERE hostname='value_hostname'
> AND realm='value_realm' AND data_key='value_data_key' ;
> INSERT INTO 'table' (hostname,realm,data_key,data_value) SELECT
> 'value_hostname', 'value_realm', 'value_data_key', 'value_data_value'
>    WHERE NOT EXISTS (SELECT 1 FROM 'table' WHERE hostname='value_hostname'
> AND realm='value_realm' AND data_key='value_data_key' );
> COMMIT;
>
>
> The two WHERE clauses must be identical and the variables should be in the
> same order as the unique compound index is ordered.
> The UPDATE fails quietly and benignly if the record does NOT already exist
> and succeeds if it does.
> The INSERT ... WHERE NOT EXISTS stmt succeeds if the record does NOT
> already exist fails quietly and benignly if it does.
> The BEGIN and COMMIT aren't typically necessary but they're over-safe just
> in case the same record is attempted by another thread.
>
> I think this get's it done.
>
> :-)
>
>
>
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From luis.daniel.lucio at gmail.com  Mon Nov 19 01:08:32 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sun, 18 Nov 2012 17:08:32 -0500
Subject: [Freeswitch-users] Packing for Mageia
In-Reply-To: <8981939843652678481@unknownmsgid>
References: <CAFLo2QyjHiReXtneD-ms-sYXR2v5ZQbayuE6VFQ3Ys_-c-k=zw@mail.gmail.com>
	<CAMwZ-tJAZDDEXqFrXqZxVFaUbNy3GenmMwhEp-NTqcffWYWxHQ@mail.gmail.com>
	<CAFLo2QzHQDHDqspKUkQ1TFgbJ_uFhqDrSLCZ2aqHjZsSOn-E8A@mail.gmail.com>
	<8981939843652678481@unknownmsgid>
Message-ID: <CAFLo2QwjP9w5DntEKTRSyWc8P8R-8J-9i02mx7UExs3eauRYeA@mail.gmail.com>

Hey, Im gack

http://pkgsubmit.mageia.org/uploads/failure/cauldron/core/release/20121118155430.dlucio.valstar.29275/log/freeswitch-1.2.3-1.mga3/build.0.20121118155506.log

It doesnt link, i read about underlink:
https://wiki.mageia.org/en/Underlinking_issues_in_packaging#Perl_modules

originaly i was having problems with mod_perl but it compiles now
after applying some ld flags, but now mod_snmp doesnt. I dont have
idea why it tries to link to perl.  I have already place the
unlinking,

any advice?

2012/3/24 Brian West <brian at freeswitch.org>:
> Peas and carrots, peas and carrots?  Stay tuned we will have such a
> thing soonish!
>
> Yours Truely,
> Lolly pop
>
> Sent from my eyePad
>
> On Mar 24, 2012, at 10:32 PM, Luis Daniel Lucio Quiroz
> <luis.daniel.lucio at gmail.com> wrote:
>
>> Changelog file says it is 1.0.7, but
>> configure.in says 1.1.beta1
>>
>> also in files.freeswitch.org i found rpms saying 1.1.beta2 dated march 7
>>
>> So, according your recomendations what release do you recommend me?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From 8f27e956 at gmail.com  Mon Nov 19 02:32:10 2012
From: 8f27e956 at gmail.com (Scott)
Date: Sun, 18 Nov 2012 18:32:10 -0500
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
	method
In-Reply-To: <CAFVUaV8mjzYsN0UnQSJnTvACgEGV_BgLRUSeeDGpRHMc8FGa1g@mail.gmail.com>
References: <CA+Y7Lf42nq1=snVW-7nFXHRqvvW9WC8FSSkOy6wjTuSYXGtAaw@mail.gmail.com>
	<CCCE5DE1.60BFA%krice@freeswitch.org>
	<CAFVUaV8mjzYsN0UnQSJnTvACgEGV_BgLRUSeeDGpRHMc8FGa1g@mail.gmail.com>
Message-ID: <CA+Y7Lf49Ddf5C6bXp32HpjDbWsHbfSQ_uMJ-NDVhHEDPa=MwaQ@mail.gmail.com>

On 18 November 2012 15:04, Yiftach Golan <yiftah at choochee.com> wrote:

> We have the exact same problem and we solve it pretty nicely but due to
> the policy "do not say anything that is different than what we did" I will
> not put it in public if you want Scott we can continue with you and Ken in
> private
>

Cool and kind of you.  Thanks, Yiftach.

Right now, on our easy stuff, we're ok with our work-around.  We just
delete THEN insert.  But we have more coming up and we enjoy seeing others'
cleverness-es!

    <entry action="menu-exec-app" digits="0" param="db
delete/ss_fmfm/${caller_id_number}"/>
    <entry action="menu-exec-app" digits="0" param="db
insert/ss_fmfm/${caller_id_number}/off"/>
    <entry action="menu-exec-app" digits="0" param="playback shout://
translate.google.com/translate_tts?tl=en&q=Option:+now+$db(select/..."/>
    <entry action="menu-exec-app" digits="1" param="db
delete/ss_fmfm/${caller_id_number}"/>
    <entry action="menu-exec-app" digits="1" param="db
insert/ss_fmfm/${caller_id_number}/on"/>
    <entry action="menu-exec-app" digits="1" param="playback shout://
translate.google.com/translate_tts?tl=en&q=Option:+now+${db(select/.."/>


Also I would please like to ask you not to use inappropriate words like
> "MySQL" in this forum as it may cause people to feel uncomfortable


Hmm?  I had no idea -- and how could one.  In my defense, I didn't assert
"its" use.  I just showed (intended) its comparative syntax and the rest
was English just being funny-that-way about nouns.  ;-)
Not-so-understood, but nevertheless acknowledged and guided accordingly.

(I'm actually a postgreSQL and huge DB2 EXPRESS-C fan!)

Again, thank you, on both counts.
/Scott


> On Sun, Nov 18, 2012 at 7:41 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  You have to keep in mind that we have to support more then just mysql
>> or sqlite with the the db interface... This means using SQL that?s common
>> to ALL platforms include MSSQL, PostgreSQL, Oracle and any other database
>> someone might want to use via ODBC...
>>
>> So to do this completely would require modification to the code on insert
>> to first check if the row exists for the key, then deleting it, then
>> inserting the new row...  This is going to be the most generic supportable
>> method to make them work consistently across the board...
>>
>> However, Keep in mind that what you are calling a bug, could be treated
>> by many people as a feature... Don?t allow me to create/replace and already
>> existing key...
>>
>>
>>
>>
>> On 11/18/12 7:26 AM, "Scott" <8f27e956 at gmail.com> wrote:
>>
>> OK, with the one-time CREATE UNIQUE INDEX on 'table'
>> (hostname,realm,data_key); stmt in place at the schema-level, a portable
>> work-a-like of INSERT OR REPLACE stmt is the UPDATE-INSERT (i.e. "UPSERT"
>> lol) stmt pair that follows,
>>
>> BEGIN
>> TRANSACTION;
>>
>> UPDATE 'table' SET data_value='value_data' WHERE
>> hostname='value_hostname' AND realm='value_realm' AND
>> data_key='value_data_key' ;
>> INSERT INTO 'table' (hostname,realm,data_key,data_value) SELECT
>> 'value_hostname', 'value_realm', 'value_data_key', 'value_data_value'
>>    WHERE NOT EXISTS (SELECT 1 FROM 'table' WHERE
>> hostname='value_hostname' AND realm='value_realm' AND
>> data_key='value_data_key' );
>> COMMIT;
>>
>>
>> The two WHERE clauses must be identical and the variables should be in
>> the same order as the unique compound index is ordered.
>> The UPDATE fails quietly and benignly if the record does NOT already
>> exist and succeeds if it does.
>> The INSERT ... WHERE NOT EXISTS stmt succeeds if the record does NOT
>> already exist fails quietly and benignly if it does.
>> The BEGIN and COMMIT aren't typically necessary but they're over-safe
>> just in case the same record is attempted by another thread.
>>
>> I think this get's it done.
>>
>> :-)
>>
>>
>> On 18 November 2012 06:25, S. Scott <8f27e956 at gmail.com> wrote:
>>
>> The IF NOT EXITS term is easily omitted.  It's a helper term for batch
>> scripted SQL to prevent an error being thrown in the case of the index
>> already exists as in the case where the make schema script is run more than
>> once.  Easily omitted and, perhaps more properly, replaced with a DROP
>> INDEX stmt and CREATE UNIQUE INDEX stmt pair.
>>
>> Yeah, i checked further, the OR REPLACE modifier is a little more
>> Hit/miss.  The capability is there typically present, but the syntax varies
>> -- e.g sqlite3 ON REPLACE is mySQL's ON DUPLICATE KEY UPDATE.
>>
>> Darn shame sqlite3 doesn't support stored procedures 'cause then it's
>> more easily abstracted for portability up the SQL server engine curve.
>>
>> On 2012-11-18, at 3:53, Gerald Weber <gerald.weber at besharp.at> wrote:
>>
>> AFAIK, ?if not exists? and ?or replace? is not ansi sql  syntax, so you
>> will get in trouble if
>> someone uses odbc to a db that doesn?t support non standard syntax.
>>
>> The insert or replace could be replaced with the merge statement.
>>
>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [
>> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
>> *Im Auftrag von *Scott
>> *Gesendet:* Sonntag, 18. November 2012 02:42
>> *An:* FreeSWITCH Users Help
>> *Betreff:* [Freeswitch-users] Suggestion to harmonize 'hash' & 'db'
>> insert method
>>
>> Are there reasons why the function 'hash' (non-persistent storage) and
>> function 'db' (persistent storage) share a look-a-like user interface (e.g.
>> select/insert/delete) but do not work-a-like.  In particular, in 'hash',
>> insert overwrites an identical realm/data_key pair whereas 'db' , in so far
>> as I can tell, just blindly adds, and adds, adds, the insert(s).  However,
>> the 'db' select method returns one record only even if the db has
>> accumulated many realm/data_key records, including data_value duplicates.
>>
>> NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
>> non-persistent 'hash' with the persistent 'db'  I *think* is straight
>> forward ...
>>
>> (1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
>> index, as follows,
>>
>> CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
>> ('hostname','realm','data_key');
>>
>> (1) At the c-language embedded SQL string-level,
>>
>> change the existing INSERT to be the following,
>>
>> INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
>> (%s,%s,%s,%s);
>>
>> The index will help speed UP the reads.  Neither change is "exotic" and
>> should be portable across standard sql implementations.
>>
>> ;-) ... thoughts ?
>>
>> Thanks,
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Ken
>> *http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> *irc.freenode.net #freeswitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.villasmil.work at gmail.com  Mon Nov 19 02:48:55 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 19 Nov 2012 00:48:55 +0100
Subject: [Freeswitch-users] Logging
Message-ID: <CAFGRPVoWqXZnuaDqePzmBBhorxS0RqLhhrYXV_No=xLAHmxPSA@mail.gmail.com>

Hello guys,

I'm going crazy with the logging feature.

If I understand correctly, you set the minimum logging in switch.conf.xml,
i.e.: debug.

I set debug because I want to be able to do "console loglevel debug" for
debugging later... but I don't want it all the time!

FS is creating a log file every 2 minutes... I don't want that, I want to
have "err" as the normal logging level and be able to do "debug" on the
console whenever I need it...

Is this possible?

Thanks

David
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From krice at freeswitch.org  Mon Nov 19 03:19:29 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 18 Nov 2012 18:19:29 -0600
Subject: [Freeswitch-users] Logging
In-Reply-To: <CAFGRPVoWqXZnuaDqePzmBBhorxS0RqLhhrYXV_No=xLAHmxPSA@mail.gmail.com>
Message-ID: <CCCED731.60C0E%krice@freeswitch.org>

Yes this is possible...

Set the loglevel in switch.conf.xml to err, then if you need to change it
you can change it at any time from the CLI with fsctl loglevel <LOG_LEVEL>
..

You can also adjust it in logfile.conf.xml for whats logged in the files and
adjust it in console.conf.xml (or vars.xml) for the console loglevel ....

Keep in mind that the loglevel in switch.conf.xml is the core logger
loglevel, setting this to say err, makes the core logger drop everything
higher then the ERR level and things will not even make it to the console or
logfile. 

K




On 11/18/12 5:48 PM, "David Villasmil" <david.villasmil.work at gmail.com>
wrote:

> Hello guys,
> 
> I'm going crazy with the logging feature.
> 
> If I understand correctly, you set the minimum logging in?switch.conf.xml,
> i.e.: debug.
> 
> I set debug because I want to be able to do "console loglevel debug" for
> debugging later... but I don't want it all the time!
> 
> FS is creating a log file every 2 minutes... I don't want that, I want to have
> "err" as the normal logging level and be able to do "debug" on the console
> whenever I need it...
> 
> Is this possible?
> 
> Thanks
> 
> David
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From cal.leeming at simplicitymedialtd.co.uk  Mon Nov 19 07:34:13 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 19 Nov 2012 04:34:13 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable (domain
 of originating user for blind transfer)
Message-ID: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>

Hi guys,

In a nut shell, it appears that when attempting to perform a blind transfer
under certain conditions (explained below), mod_xml_curl does not expose
the originating domain in a clean format.

My initial plan was to find the point where these variable were being
generated, look at what was available, then add an extra variable for the
domain and submit a patch.

Sadly my C isn't great and I hit a brick wall, so if anyone can help out, I
will ensure the mod_xml_curl documentation is updated and/or assist with
any patching/testing required.

Please take the following scenario;

* Extension 2000 calls an external number via a gateway (i.e. bridge
sofia/gateway/name/e164_number_here).
* Call connects fine, audio stays good, no disconnection problems etc.
* Call is blind transferred to another extension

As a result, the following is determined;

* User initiating the blind transfer is 2000
* Domain initiating the blind transfer is c1881.voiceflare.co.uk
* Destination number of the call is 447866123456
* Number to blind transfer to is 2001
* Call to mod_xml_curl is made

It makes reference to the User in the following 'clean' variables (by
clean, I mean variables that just contain 2000 and don't require mangling
to extract the info);

 u'Caller-ANI': u'2000',
 u'Caller-Username': u'2000',
 u'Caller-Caller-ID-Number': u'2000',
 u'Hunt-ANI': u'2000',
 u'Hunt-Caller-ID-Number': u'2000',
 u'Hunt-Username': u'2000',
 u'variable_last_sent_callee_id_number': u'2000',
 u'variable_sip_from_user': u'2000',

It also has the User in the following unclean variables;

 u'variable_bridge_channel': u'sofia/external/
2000 at c1881.voiceflare.co.uk:5060',
 u'variable_sip_from_uri': u'2000 at 89.238.182.137',
 u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>;tag=XryjFQp3rB2NF',
 u'variable_sip_h_Referred-By': u'"foxx" <
sip:2000 at c1881.voiceflare.co.uk:5060>',

However, it only references the domain in the following unclean variables;

 u'variable_bridge_channel': u'sofia/external/
2000 at c1881.voiceflare.co.uk:5060',
 u'variable_sip_h_Referred-By': u'"foxx" <
sip:2000 at c1881.voiceflare.co.uk:5060>',
 u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',

Lets say that we want to determine the user/domain that has initiated this
transfer, doing so would mean mangling with one of those above variables,
which seems a bit dirty (plus it is not clear which is the correct one to
use).

I have attached the SIP trace of the entire blind transfer event, and the
full mod_xml_curl request being sent.

If anyone could answer the following, it'd be much appreciated;

* Should there be an improvement patch that exposes the domain of the user
that originated the blind transfer?
* Are there better/alternative ways to extracting the domain of the user
that originated the blind transfer?

Many thanks

Cal
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{u'Answer-State': u'answered',
 u'Call-Direction': u'outbound',
 u'Caller-ANI': u'2000',
 u'Caller-Callee-ID-Name': u'Outbound Call',
 u'Caller-Callee-ID-Number': u'447866123456',
 u'Caller-Caller-ID-Name': u'foxx',
 u'Caller-Caller-ID-Number': u'2000',
 u'Caller-Channel-Answered-Time': u'1353296828185007',
 u'Caller-Channel-Created-Time': u'1353296819484965',
 u'Caller-Channel-Hangup-Time': u'0',
 u'Caller-Channel-Name': u'sofia/external/447866123456',
 u'Caller-Channel-Progress-Media-Time': u'1353296824324993',
 u'Caller-Channel-Progress-Time': u'0',
 u'Caller-Channel-Transfer-Time': u'0',
 u'Caller-Context': u'default',
 u'Caller-Destination-Number': u'2001',
 u'Caller-Dialplan': u'XML',
 u'Caller-Direction': u'outbound',
 u'Caller-Network-Addr': u'80.93.165.111',
 u'Caller-Privacy-Hide-Name': u'false',
 u'Caller-Privacy-Hide-Number': u'false',
 u'Caller-Profile-Created-Time': u'1353296834585013',
 u'Caller-Profile-Index': u'2',
 u'Caller-RDNIS': u'447866123456',
 u'Caller-Screen-Bit': u'true',
 u'Caller-Source': u'mod_sofia',
 u'Caller-Transfer-Source': u'1353296834:ce841838-31fb-11e2-a40f-bdc0bb753e98:bl_xfer:2001/default/XML',
 u'Caller-Unique-ID': u'c58483e4-31fb-11e2-a406-bdc0bb753e98',
 u'Caller-Username': u'2000',
 u'Channel-Call-State': u'RINGING',
 u'Channel-Call-UUID': u'c58483e4-31fb-11e2-a406-bdc0bb753e98',
 u'Channel-HIT-Dialplan': u'false',
 u'Channel-Name': u'sofia/external/447866123456',
 u'Channel-Read-Codec-Bit-Rate': u'64000',
 u'Channel-Read-Codec-Name': u'PCMA',
 u'Channel-Read-Codec-Rate': u'8000',
 u'Channel-State': u'CS_ROUTING',
 u'Channel-State-Number': u'2',
 u'Channel-Write-Codec-Bit-Rate': u'64000',
 u'Channel-Write-Codec-Name': u'PCMA',
 u'Channel-Write-Codec-Rate': u'8000',
 u'Core-UUID': u'65606776-2ed1-11e2-92ce-bdc0bb753e98',
 u'Event-Calling-File': u'mod_dialplan_xml.c',
 u'Event-Calling-Function': u'dialplan_xml_locate',
 u'Event-Calling-Line-Number': u'456',
 u'Event-Date-GMT': u'Mon, 19 Nov 2012 03:47:14 GMT',
 u'Event-Date-Local': u'2012-11-19 03:47:14',
 u'Event-Date-Timestamp': u'1353296834605025',
 u'Event-Name': u'REQUEST_PARAMS',
 u'Event-Sequence': u'54154',
 u'FreeSWITCH-Hostname': u'vded213',
 u'FreeSWITCH-IPv4': u'89.238.182.137',
 u'FreeSWITCH-IPv6': u'::1',
 u'FreeSWITCH-Switchname': u'vded213',
 u'Hunt-ANI': u'2000',
 u'Hunt-Callee-ID-Name': u'Outbound Call',
 u'Hunt-Callee-ID-Number': u'447866123456',
 u'Hunt-Caller-ID-Name': u'foxx',
 u'Hunt-Caller-ID-Number': u'2000',
 u'Hunt-Channel-Answered-Time': u'1353296828185007',
 u'Hunt-Channel-Created-Time': u'1353296819484965',
 u'Hunt-Channel-Hangup-Time': u'0',
 u'Hunt-Channel-Name': u'sofia/external/447866123456',
 u'Hunt-Channel-Progress-Media-Time': u'1353296824324993',
 u'Hunt-Channel-Progress-Time': u'0',
 u'Hunt-Channel-Transfer-Time': u'0',
 u'Hunt-Context': u'default',
 u'Hunt-Destination-Number': u'2001',
 u'Hunt-Dialplan': u'XML',
 u'Hunt-Direction': u'outbound',
 u'Hunt-Network-Addr': u'80.93.165.111',
 u'Hunt-Privacy-Hide-Name': u'false',
 u'Hunt-Privacy-Hide-Number': u'false',
 u'Hunt-Profile-Created-Time': u'1353296834585013',
 u'Hunt-Profile-Index': u'2',
 u'Hunt-RDNIS': u'447866123456',
 u'Hunt-Screen-Bit': u'true',
 u'Hunt-Source': u'mod_sofia',
 u'Hunt-Transfer-Source': u'1353296834:ce841838-31fb-11e2-a40f-bdc0bb753e98:bl_xfer:2001/default/XML',
 u'Hunt-Unique-ID': u'c58483e4-31fb-11e2-a406-bdc0bb753e98',
 u'Hunt-Username': u'2000',
 u'Presence-Call-Direction': u'outbound',
 u'Unique-ID': u'c58483e4-31fb-11e2-a406-bdc0bb753e98',
 u'hostname': u'vded213',
 u'key_name': u'',
 u'key_value': u'',
 u'section': u'dialplan',
 u'tag_name': u'',
 u'variable_absolute_codec_string': u'PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b,GSM at 8000h@20i at 13200b',
 u'variable_advertised_media_ip': u'89.238.182.137',
 u'variable_bridge_channel': u'sofia/external/2000 at c1881.voiceflare.co.uk:5060',
 u'variable_bridge_uuid': u'c57cb114-31fb-11e2-a400-bdc0bb753e98',
 u'variable_call_uuid': u'c58483e4-31fb-11e2-a406-bdc0bb753e98',
 u'variable_channel_name': u'sofia/external/447866123456',
 u'variable_current_application': u'playback',
 u'variable_current_application_response': u'PLAYBACK ERROR',
 u'variable_direction': u'outbound',
 u'variable_endpoint_disposition': u'ANSWER',
 u'variable_ep_codec_string': u'PCMA at 8000h@20i at 64000b',
 u'variable_is_outbound': u'true',
 u'variable_last_bridge_to': u'c57cb114-31fb-11e2-a400-bdc0bb753e98',
 u'variable_last_sent_callee_id_name': u'foxx',
 u'variable_last_sent_callee_id_number': u'2000',
 u'variable_local_media_ip': u'89.238.182.137',
 u'variable_local_media_port': u'17622',
 u'variable_max_forwards': u'68',
 u'variable_originate_early_media': u'true',
 u'variable_originating_leg_uuid': u'c57cb114-31fb-11e2-a400-bdc0bb753e98',
 u'variable_originator': u'c57cb114-31fb-11e2-a400-bdc0bb753e98',
 u'variable_originator_codec': u'PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b,GSM at 8000h@20i at 13200b',
 u'variable_read_codec': u'PCMA',
 u'variable_read_rate': u'8000',
 u'variable_recovery_profile_name': u'external',
 u'variable_remote_media_ip': u'80.93.165.111',
 u'variable_remote_media_port': u'28436',
 u'variable_rtp_use_ssrc': u'1387842288',
 u'variable_session_id': u'115',
 u'variable_sip_2833_recv_payload': u'101',
 u'variable_sip_2833_send_payload': u'101',
 u'variable_sip_audio_recv_pt': u'8',
 u'variable_sip_call_id': u'9ce2e3bf-ac9e-1230-eead-000c299684b0',
 u'variable_sip_contact_host': u'80.93.165.111',
 u'variable_sip_contact_params': u'transport=udp',
 u'variable_sip_contact_port': u'5060',
 u'variable_sip_contact_uri': u'447866123456 at 80.93.165.111:5060',
 u'variable_sip_contact_user': u'447866123456',
 u'variable_sip_cseq': u'36312346',
 u'variable_sip_destination_url': u'sip:447866123456 at sip.numbergroup-services.com',
 u'variable_sip_from_display': u'foxx',
 u'variable_sip_from_host': u'89.238.182.137',
 u'variable_sip_from_tag': u'XryjFQp3rB2NF',
 u'variable_sip_from_uri': u'2000 at 89.238.182.137',
 u'variable_sip_from_user': u'2000',
 u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137>;tag=XryjFQp3rB2NF',
 u'variable_sip_full_to': u'<sip:447866123456 at sip.numbergroup-services.com>;tag=39FNHXrcry73S',
 u'variable_sip_full_via': u'SIP/2.0/UDP 89.238.182.137;rport=5060;branch=z9hG4bKeHtmQ2KK0er8r',
 u'variable_sip_gateway_name': u'numbergroup',
 u'variable_sip_h_Referred-By': u'"foxx" <sip:2000 at c1881.voiceflare.co.uk:5060>',
 u'variable_sip_local_network_addr': u'89.238.182.137',
 u'variable_sip_local_sdp_str': u'v=0\no=FreeSWITCH 1353279197 1353279198 IN IP4 89.238.182.137\ns=FreeSWITCH\nc=IN IP4 89.238.182.137\nt=0 0\nm=audio 17622 RTP/AVP 0 8 3 101 13\na=rtpmap:101 telephone-event/8000\na=fmtp:101 0-16\na=ptime:20\na=sendrecv\n',
 u'variable_sip_network_ip': u'80.93.165.111',
 u'variable_sip_network_port': u'5060',
 u'variable_sip_outgoing_contact_uri': u'<sip:gw+numbergroup at 89.238.182.137:5060;transport=udp;gw=numbergroup>',
 u'variable_sip_ph_P-Charging-Vector': u'icid-value=c58d77ce-31fb-11e2-a21a-99ec1789df65;icid-generated-at=80.93.165.111;orig-ioi=numbergroup.com',
 u'variable_sip_profile_name': u'gateway',
 u'variable_sip_recover_contact': u'<sip:447866123456 at 80.93.165.111:5060;transport=udp>',
 u'variable_sip_recover_via': u'SIP/2.0/UDP 89.238.182.137;rport=5060;branch=z9hG4bKeHtmQ2KK0er8r',
 u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',
 u'variable_sip_reply_host': u'80.93.165.111',
 u'variable_sip_reply_port': u'5060',
 u'variable_sip_req_uri': u'447866123456 at sip.numbergroup-services.com',
 u'variable_sip_to_host': u'sip.numbergroup-services.com',
 u'variable_sip_to_tag': u'39FNHXrcry73S',
 u'variable_sip_to_uri': u'447866123456 at sip.numbergroup-services.com',
 u'variable_sip_to_user': u'447866123456',
 u'variable_sip_use_codec_name': u'PCMA',
 u'variable_sip_use_codec_ptime': u'20',
 u'variable_sip_use_codec_rate': u'8000',
 u'variable_sip_use_pt': u'8',
 u'variable_sip_user_agent': u'numbergroup.com',
 u'variable_sofia_profile_name': u'external',
 u'variable_switch_m_sdp': u'v=0\r\no=3cxVCE 263380575 226789996 IN IP4 82.30.159.182\r\ns=3cxVCE Audio Call\r\nc=IN IP4 82.30.159.182\r\nt=0 0\r\nm=audio 40046 RTP/AVP 0 8 3 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=sendonly\r\na=ptime:20\r\nm=video 40000 RTP/AVP 34\r\nc=IN IP4 82.30.159.182\r\na=rtpmap:34 H263/90000\r\na=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1\r\n',
 u'variable_switch_r_sdp': u'v=0\r\no=numbergroup 1353268388 1353268389 IN IP4 80.93.165.111\r\ns=numbergroup\r\nc=IN IP4 80.93.165.111\r\nt=0 0\r\nm=audio 28436 RTP/AVP 8 101 13\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=rtpmap:13 CN/8000\r\na=ptime:20\r\n',
 u'variable_transfer_history': u'ARRAY::1353296834:ce841838-31fb-11e2-a40f-bdc0bb753e98:bl_xfer:2001/default/XML',
 u'variable_transfer_source': u'1353296834:ce841838-31fb-11e2-a40f-bdc0bb753e98:bl_xfer:2001/default/XML',
 u'variable_uuid': u'c58483e4-31fb-11e2-a406-bdc0bb753e98',
 u'variable_write_codec': u'PCMA',
 u'variable_write_rate': u'8000'}



freeswitch at vded213> 
recv 614 bytes from udp/[82.30.159.182]:60585 at 04:24:34.648703:
   ------------------------------------------------------------------------
   REFER sip:07866123456 at 89.238.182.137:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 82.30.159.182:60585;branch=z9hG4bK-d8754z-da5faa120000ed7a-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:2000 at 82.30.159.182:60585;rinstance=b0415b098f47701f>
   To: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   From: "foxx"<sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 3 REFER
   User-Agent: 3CXPhone 6.0.25732.0
   Refer-To: <sip:2001 at c1881.voiceflare.co.uk:5060>
   Referred-By: "foxx"<sip:2000 at c1881.voiceflare.co.uk:5060>
   Content-Length: 0

   ------------------------------------------------------------------------
2012-11-19 04:24:34.644982 [DEBUG] switch_core_session.c:905 Send signal sofia/external/2000 at c1881.voiceflare.co.uk:5060 [BREAK]
2012-11-19 04:24:34.664991 [DEBUG] sofia.c:7308 Process REFER to [2001 at c1881.voiceflare.co.uk]
2012-11-19 04:24:34.664991 [DEBUG] switch_ivr.c:1742 (sofia/external/447866123456) State Change CS_EXCHANGE_MEDIA -> CS_ROUTING
2012-11-19 04:24:34.664991 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/447866123456 [BREAK]
2012-11-19 04:24:34.664991 [DEBUG] switch_core_session.c:759 Send signal sofia/external/447866123456 [BREAK]
2012-11-19 04:24:34.664991 [NOTICE] switch_ivr.c:1748 Transfer sofia/external/447866123456 to XML[2001 at default]
send 646 bytes to udp/[82.30.159.182]:60585 at 04:24:34.665258:

   ------------------------------------------------------------------------
   SIP/2.0 202 Accepted
   Via: SIP/2.0/UDP 82.30.159.182:60585;branch=z9hG4bK-d8754z-da5faa120000ed7a-1---d8754z-;rport=60585
   From: "foxx"<sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   To: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 3 REFER
   Contact: <sip:07866123456 at 89.238.182.137:5060;transport=udp>
   Expires: 60
   User-Agent: FreeSWITCH-mod_sofia/1.2.3
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Length: 0

   ------------------------------------------------------------------------
send 805 bytes to udp/[82.30.159.182]:60585 at 04:24:34.665421:
   ------------------------------------------------------------------------
   NOTIFY sip:2000 at 82.30.159.182:60585;rinstance=b0415b098f47701f SIP/2.0
   Via: SIP/2.0/UDP 89.238.182.137;rport;branch=z9hG4bKptDmyvN0cKKmr
   Max-Forwards: 70
   From: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   To: "foxx" <sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 36313473 NOTIFY
   Contact: <sip:07866123456 at 89.238.182.137:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.2.3
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Event: refer;id=3
   Allow-Events: talk, hold, conference, refer
   Subscription-State: terminated;reason=noresource
   Content-Type: message/sipfrag;version=2.0
   Content-Length: 16

   SIP/2.0 200 OK

   ------------------------------------------------------------------------
2012-11-19 04:24:34.685056 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/2000 at c1881.voiceflare.co.uk:5060]
2012-11-19 04:24:34.685056 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/447866123456 [BREAK]
2012-11-19 04:24:34.685056 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/447866123456]
2012-11-19 04:24:34.685056 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/2000 at c1881.voiceflare.co.uk:5060 [BREAK]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:456 (sofia/external/447866123456) State EXCHANGE_MEDIA going to sleep
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:398 (sofia/external/447866123456) Running State Change CS_ROUTING
2012-11-19 04:24:34.685056 [DEBUG] switch_channel.c:1964 (sofia/external/447866123456) Callstate Change ACTIVE -> RINGING
2012-11-19 04:24:34.685056 [DEBUG] switch_core_session.c:759 Send signal sofia/external/447866123456 [BREAK]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_session.c:759 Send signal sofia/external/2000 at c1881.voiceflare.co.uk:5060 [BREAK]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:446 (sofia/external/447866123456) State ROUTING
2012-11-19 04:24:34.685056 [DEBUG] mod_sofia.c:149 sofia/external/447866123456 SOFIA ROUTING
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:117 sofia/external/447866123456 Standard ROUTING
2012-11-19 04:24:34.685056 [INFO] mod_dialplan_xml.c:485 Processing foxx <2000>->2001 in context default
2012-11-19 04:24:34.685056 [NOTICE] switch_core_state_machine.c:262 sofia/external/2000 at c1881.voiceflare.co.uk:5060 has executed the last dialplan instruction, hanging up.
2012-11-19 04:24:34.685056 [DEBUG] switch_channel.c:2950 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) Callstate Change HELD -> HANGUP
2012-11-19 04:24:34.685056 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/external/2000 at c1881.voiceflare.co.uk:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2012-11-19 04:24:34.685056 [DEBUG] switch_channel.c:2973 Send signal sofia/external/2000 at c1881.voiceflare.co.uk:5060 [KILL]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/2000 at c1881.voiceflare.co.uk:5060 [BREAK]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:453 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State EXECUTE going to sleep
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:398 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) Running State Change CS_HANGUP
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:638 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State HANGUP
2012-11-19 04:24:34.685056 [DEBUG] mod_sofia.c:483 Channel sofia/external/2000 at c1881.voiceflare.co.uk:5060 hanging up, cause: NORMAL_CLEARING
2012-11-19 04:24:34.685056 [DEBUG] mod_sofia.c:532 Sending BYE to sofia/external/2000 at c1881.voiceflare.co.uk:5060
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:48 sofia/external/2000 at c1881.voiceflare.co.uk:5060 Standard HANGUP, cause: NORMAL_CLEARING
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:638 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State HANGUP going to sleep
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:429 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State Change CS_HANGUP -> CS_REPORTING
2012-11-19 04:24:34.685056 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/2000 at c1881.voiceflare.co.uk:5060 [BREAK]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:398 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) Running State Change CS_REPORTING
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:703 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State REPORTING
2012-11-19 04:24:34.685056 [ERR] mod_xml_cdr.c:247 Error writing [/usr/local/freeswitch/log/xml_cdr/a_fb363230-3200-11e2-a468-bdc0bb753e98.cdr.xml][No such file or directory]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:92 sofia/external/2000 at c1881.voiceflare.co.uk:5060 Standard REPORTING, cause: NORMAL_CLEARING
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:703 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State REPORTING going to sleep
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:423 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State Change CS_REPORTING -> CS_DESTROY
2012-11-19 04:24:34.685056 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/2000 at c1881.voiceflare.co.uk:5060 [BREAK]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_session.c:1415 Session 120 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) Locked, Waiting on external entities
2012-11-19 04:24:34.685056 [NOTICE] switch_core_session.c:1433 Session 120 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) Ended
2012-11-19 04:24:34.685056 [NOTICE] switch_core_session.c:1437 Close Channel sofia/external/2000 at c1881.voiceflare.co.uk:5060 [CS_DESTROY]
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:527 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) Callstate Change HANGUP -> DOWN
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:530 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) Running State Change CS_DESTROY
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:540 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State DESTROY
2012-11-19 04:24:34.685056 [DEBUG] mod_sofia.c:376 sofia/external/2000 at c1881.voiceflare.co.uk:5060 SOFIA DESTROY
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:99 sofia/external/2000 at c1881.voiceflare.co.uk:5060 Standard DESTROY
2012-11-19 04:24:34.685056 [DEBUG] switch_core_state_machine.c:540 (sofia/external/2000 at c1881.voiceflare.co.uk:5060) State DESTROY going to sleep
recv 423 bytes from udp/[82.30.159.182]:60585 at 04:24:34.815422:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 89.238.182.137;rport=5060;branch=z9hG4bKptDmyvN0cKKmr
   Contact: <sip:2000 at 82.30.159.182:60585;rinstance=b0415b098f47701f>
   To: "foxx"<sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   From: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 36313473 NOTIFY
   User-Agent: 3CXPhone 6.0.25732.0
   Content-Length: 0
   ------------------------------------------------------------------------
send 672 bytes to udp/[82.30.159.182]:60585 at 04:24:34.815653:
   ------------------------------------------------------------------------
   BYE sip:2000 at 82.30.159.182:60585;rinstance=b0415b098f47701f SIP/2.0
   Via: SIP/2.0/UDP 89.238.182.137;rport;branch=z9hG4bKQ36c0Q639U96K
   Max-Forwards: 70
   From: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   To: "foxx" <sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 36313474 BYE
   Contact: <sip:07866123456 at 89.238.182.137:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.2.3
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 501 bytes from udp/[82.30.159.182]:60585 at 04:24:34.815705:
   ------------------------------------------------------------------------
   BYE sip:07866123456 at 89.238.182.137:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 82.30.159.182:60585;branch=z9hG4bK-d8754z-cb28cb688f6de166-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:2000 at 82.30.159.182:60585;rinstance=b0415b098f47701f>
   To: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   From: "foxx"<sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 5 BYE
   User-Agent: 3CXPhone 6.0.25732.0
   Content-Length: 0

   ------------------------------------------------------------------------
recv 420 bytes from udp/[82.30.159.182]:60585 at 04:24:34.942566:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 89.238.182.137;rport=5060;branch=z9hG4bKQ36c0Q639U96K
   Contact: <sip:2000 at 82.30.159.182:60585;rinstance=b0415b098f47701f>
   To: "foxx"<sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   From: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 36313474 BYE
   User-Agent: 3CXPhone 6.0.25732.0
   Content-Length: 0

   ------------------------------------------------------------------------
send 340 bytes to udp/[82.30.159.182]:60585 at 04:24:34.942843:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 82.30.159.182:60585;branch=z9hG4bK-d8754z-cb28cb688f6de166-1---d8754z-;rport=60585
   From: "foxx"<sip:2000 at c1881.voiceflare.co.uk:5060>;tag=f5168a69
   To: <sip:07866123456 at c1881.voiceflare.co.uk:5060>;tag=1yN876FKm9cDK
   Call-ID: ZWZhYTdhODk2NDgxODU4MjM5OTYyYjVlMTk4NTEwYWI.
   CSeq: 5 BYE
   Content-Length: 0

   ------------------------------------------------------------------------
2012-11-19 04:24:34.944968 [ERR] mod_xml_curl.c:310 Received HTTP error 500 trying to fetch http://localhost:80/api/1.1/fscurl
data: [hostname=vded213&section=dialplan&tag_name=&key_name=&key_value=&Event-Name=REQUEST_PARAMS&Core-UUID=65606776-2ed1-11e2-92ce-bdc0bb753e98&FreeSWITCH-Hostname=vded213&FreeSWITCH-Switchname=vded213&FreeSWITCH-IPv
4=89.238.182.137&FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2012-11-19%2004%3A24%3A34&Event-Date-GMT=Mon,%2019%20Nov%202012%2004%3A24%3A34%20GMT&Event-Date-Timestamp=1353299074685056&Event-Calling-File=mod_dialplan_xml.
c&Event-Calling-Function=dialplan_xml_locate&Event-Calling-Line-Number=456&Event-Sequence=54747&Channel-State=CS_ROUTING&Channel-Call-State=RINGING&Channel-State-Number=2&Channel-Name=sofia/external/447866123456&Uniqu
e-ID=fb3e0014-3200-11e2-a46e-bdc0bb753e98&Call-Direction=outbound&Presence-Call-Direction=outbound&Channel-HIT-Dialplan=false&Channel-Call-UUID=fb3e0014-3200-11e2-a46e-bdc0bb753e98&Answer-State=answered&Channel-Read-C
odec-Name=PCMA&Channel-Read-Codec-Rate=8000&Channel-Read-Codec-Bit-Rate=64000&Channel-Write-Codec-Name=PCMA&Channel-Write-Codec-Rate=8000&Channel-Write-Codec-Bit-Rate=64000&Caller-Direction=outbound&Caller-Username=20
00&Caller-Dialplan=XML&Caller-Caller-ID-Name=foxx&Caller-Caller-ID-Number=2000&Caller-Callee-ID-Name=Outbound%20Call&Caller-Callee-ID-Number=447866123456&Caller-Network-Addr=80.93.165.111&Caller-ANI=2000&Caller-Destin
ation-Number=2001&Caller-Unique-ID=fb3e0014-3200-11e2-a46e-bdc0bb753e98&Caller-Source=mod_sofia&Caller-Transfer-Source=1353299074%3A05b52eaa-3201-11e2-a474-bdc0bb753e98%3Abl_xfer%3A2001/default/XML&Caller-Context=default&Caller-RDNIS=447866123456&Caller-Channel-Name=sofia/external/447866123456&Caller-Profile-Index=2&Caller-Profile-Created-Time=1353299074664991&Caller-Channel-Created-Time=1353299057104964&Caller-Channel-Answered-Time=1353299065304972&Caller-Channel-Progress-Time=0&Caller-Channel-Progress-Media-Time=1353299061845008&Caller-Channel-Hangup-Time=0&Caller-Channel-Transfer-Time=0&Caller-Screen-Bit=true&Caller-Privacy-Hide-Name=false&Caller-Privacy-Hide-Number=false&variable_direction=outbound&variable_is_outbound=true&variable_uuid=fb3e0014-3200-11e2-a46e-bdc0bb753e98&variable_session_id=121&variable_sip_gateway_name=numbergroup&variable_sip_profile_name=gateway&variable_channel_name=sofia/external/447866123456&variable_sip_destination_url=sip%3A447866123456%40sip.numbergroup-services.com&variable_absolute_codec_string=PCMU%408000h%4020i%4064000b,PCMA%408000h%4020i%4064000b,GSM%408000h%4020i%4013200b&variable_originator_codec=PCMU%408000h%4020i%4064000b,PCMA%408000h%4020i%4064000b,GSM%408000h%4020i%4013200b&variable_originator=fb363230-3200-11e2-a468-bdc0bb753e98&variable_originate_early_media=true&variable_originating_leg_uuid=fb363230-3200-11e2-a468-bdc0bb753e98&variable_sip_local_sdp_str=v%3D0%0Ao%3DFreeSWITCH%201353266655%201353266656%20IN%20IP4%2089.238.182.137%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2089.238.182.137%0At%3D0%200%0Am%3Daudio%2032402%20RTP/AVP%200%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A&variable_sip_outgoing_contact_uri=%3Csip%3Agw%2Bnumbergroup%4089.238.182.137%3A5060%3Btransport%3Dudp%3Bgw%3Dnumbergroup%3E&variable_sip_req_uri=447866123456%40sip.numbergroup-services.com&variable_sofia_profile_name=external&variable_recovery_profile_name=external&variable_sip_ph_P-Charging-Vector=icid-value%3Dfb4572a4-3200-11e2-89d6-99ec1789df65%3Bicid-generated-at%3D80.93.165.111%3Borig-ioi%3Dnumbergroup.com&variable_switch_r_sdp=v%3D0%0D%0Ao%3Dnumbergroup%201353252075%201353252076%20IN%20IP4%2080.93.165.111%0D%0As%3Dnumbergroup%0D%0Ac%3DIN%20IP4%2080.93.165.111%0D%0At%3D0%200%0D%0Am%3Daudio%2046986%20RTP/AVP%208%20101%2013%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Drtpmap%3A13%20CN/8000%0D%0Aa%3Dptime%3A20%0D%0A&variable_ep_codec_string=PCMA%408000h%4020i%4064000b&variable_sip_audio_recv_pt=8&variable_sip_use_codec_name=PCMA&variable_sip_use_codec_rate=8000&variable_sip_use_codec_ptime=20&variable_read_codec=PCMA&variable_read_rate=8000&variable_write_codec=PCMA&variable_write_rate=8000&variable_local_media_ip=89.238.182.137&variable_local_media_port=32402&variable_advertised_media_ip=89.238.182.137&variable_sip_use_pt=8&variable_rtp_use_ssrc=3232801725&variable_sip_2833_send_payload=101&variable_sip_2833_recv_payload=101&variable_remote_media_ip=80.93.165.111&variable_remote_media_port=46986&variable_last_bridge_to=fb363230-3200-11e2-a468-bdc0bb753e98&variable_bridge_channel=sofia/external/2000%40c1881.voiceflare.co.uk%3A5060&variable_bridge_uuid=fb363230-3200-11e2-a468-bdc0bb753e98&variable_sip_local_network_addr=89.238.182.137&variable_sip_reply_host=80.93.165.111&variable_sip_reply_port=5060&variable_sip_network_ip=80.93.165.111&variable_sip_network_port=5060&variable_sip_user_agent=numbergroup.com&variable_sip_recover_contact=%3Csip%3A447866123456%4080.93.165.111%3A5060%3Btransport%3Dudp%3E&variable_sip_full_via=SIP/2.0/UDP%2089.238.182.137%3Brport%3D5060%3Bbranch%3Dz9hG4bKKZ19rB3NNrgvN&variable_sip_recover_via=SIP/2.0/UDP%2089.238.182.137%3Brport%3D5060%3Bbranch%3Dz9hG4bKKZ19rB3NNrgvN&variable_sip_from_display=foxx&variable_sip_full_from=%22foxx%22%20%3Csip%3A2000%4089.238.182.137%3E%3Btag%3D27e1910pHj3Ze&variable_sip_full_to=%3Csip%3A447866123456%40sip.numbergroup-services.com%3E%3Btag%3DrrSHS2S5ZeZ9g&variable_sip_from_user=2000&variable_sip_from_uri=2000%4089.238.182.137&variable_sip_from_host=89.238.182.137&variable_sip_to_user=447866123456&variable_sip_to_uri=447866123456%40sip.numbergroup-services.com&variable_sip_to_host=sip.numbergroup-services.com&variable_sip_contact_params=transport%3Dudp&variable_sip_contact_user=447866123456&variable_sip_contact_port=5060&variable_sip_contact_uri=447866123456%4080.93.165.111%3A5060&variable_sip_contact_host=80.93.165.111&variable_sip_to_tag=rrSHS2S5ZeZ9g&variable_sip_from_tag=27e1910pHj3Ze&variable_sip_cseq=36313465&variable_sip_call_id=d29c5ff4-aca3-1230-eead-000c299684b0&variable_endpoint_disposition=ANSWER&variable_last_sent_callee_id_name=foxx&variable_last_sent_callee_id_number=2000&variable_switch_m_sdp=v%3D0%0D%0Ao%3D3cxVCE%20215333835%2095945341%20IN%20IP4%2082.30.159.182%0D%0As%3D3cxVCE%20Audio%20Call%0D%0Ac%3DIN%20IP4%2082.30.159.182%0D%0At%3D0%200%0D%0Am%3Daudio%2040020%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Dsendonly%0D%0Aa%3Dptime%3A20%0D%0Am%3Dvideo%2040018%20RTP/AVP%2034%0D%0Ac%3DIN%20IP4%2082.30.159.182%0D%0Aa%3Drtpmap%3A34%20H263/90000%0D%0Aa%3Dfmtp%3A34%20QCIF%3D1%3BCIF%3D1%3BSQCIF%3D1%3BCIF4%3D1%0D%0A&variable_current_application=playback&variable_current_application_response=PLAYBACK%20ERROR&variable_sip_h_Referred-By=%22foxx%22%20%3Csip%3A2000%40c1881.voiceflare.co.uk%3A5060%3E&variable_sip_refer_to=%3Csip%3A2001%40c1881.voiceflare.co.uk%3A5060%3E&variable_max_forwards=68&variable_transfer_history=ARRAY%3A%3A1353299074%3A05b52eaa-3201-11e2-a474-bdc0bb753e98%3Abl_xfer%3A2001/default/XML&variable_transfer_source=1353299074%3A05b52eaa-3201-11e2-a474-bdc0bb753e98%3Abl_xfer%3A2001/default/XML&variable_call_uuid=fb3e0014-3200-11e2-a46e-bdc0bb753e98&Hunt-Direction=outbound&Hunt-Username=2000&Hunt-Dialplan=XML&Hunt-Caller-ID-Name=foxx&Hunt-Caller-ID-Number=2000&Hunt-Callee-ID-Name=Outbound%20Call&Hunt-Callee-ID-Number=447866123456&Hunt-Network-Addr=80.93.165.111&Hunt-ANI=2000&Hunt-Destination-Number=2001&Hunt-Unique-ID=fb3e0014-3200-11e2-a46e-bdc0bb753e98&Hunt-Source=mod_sofia&Hunt-Transfer-Source=1353299074%3A05b52eaa-3201-11e2-a474-bdc0bb753e98%3Abl_xfer%3A2001/default/XML&Hunt-Context=default&Hunt-RDNIS=447866123456&Hunt-Channel-Name=sofia/external/447866123456&Hunt-Profile-Index=2&Hunt-Profile-Created-Time=1353299074664991&Hunt-Channel-Created-Time=1353299057104964&Hunt-Channel-Answered-Time=1353299065304972&Hunt-Channel-Progress-Time=0&Hunt-Channel-Progress-Media-Time=1353299061845008&Hunt-Channel-Hangup-Time=0&Hunt-Channel-Transfer-Time=0&Hunt-Screen-Bit=true&Hunt-Privacy-Hide-Name=false&Hunt-Privacy-Hide-Number=false]
2012-11-19 04:24:34.944968 [ERR] mod_dialplan_xml.c:507 Open of dialplan failed

2012-11-19 04:24:34.944968 [INFO] switch_core_state_machine.c:192 No Route, Aborting
2012-11-19 04:24:34.944968 [DEBUG] switch_channel.c:2950 (sofia/external/447866123456) Callstate Change RINGING -> HANGUP
2012-11-19 04:24:34.944968 [NOTICE] switch_core_state_machine.c:193 Hangup sofia/external/447866123456 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2012-11-19 04:24:34.944968 [DEBUG] switch_channel.c:2973 Send signal sofia/external/447866123456 [KILL]
2012-11-19 04:24:34.944968 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/447866123456 [BREAK]
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:446 (sofia/external/447866123456) State ROUTING going to sleep
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:398 (sofia/external/447866123456) Running State Change CS_HANGUP
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:638 (sofia/external/447866123456) State HANGUP
2012-11-19 04:24:34.944968 [DEBUG] mod_sofia.c:483 Channel sofia/external/447866123456 hanging up, cause: NO_ROUTE_DESTINATION
2012-11-19 04:24:34.944968 [DEBUG] mod_sofia.c:532 Sending BYE to sofia/external/447866123456
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:48 sofia/external/447866123456 Standard HANGUP, cause: NO_ROUTE_DESTINATION
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:638 (sofia/external/447866123456) State HANGUP going to sleep
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:429 (sofia/external/447866123456) State Change CS_HANGUP -> CS_REPORTING
2012-11-19 04:24:34.944968 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/447866123456 [BREAK]
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:398 (sofia/external/447866123456) Running State Change CS_REPORTING
send 988 bytes to udp/[80.93.165.111]:5060 at 04:24:34.946395:
   ------------------------------------------------------------------------
   BYE sip:447866123456 at 80.93.165.111:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 89.238.182.137;rport;branch=z9hG4bKrc051jQ764ZSF
   Max-Forwards: 70
   From: "foxx" <sip:2000 at 89.238.182.137>;tag=27e1910pHj3Ze
   To: <sip:447866123456 at sip.numbergroup-services.com>;tag=rrSHS2S5ZeZ9g
   Call-ID: d29c5ff4-aca3-1230-eead-000c299684b0
   CSeq: 36313467 BYE
   Contact: <sip:gw+numbergroup at 89.238.182.137:5060;transport=udp;gw=numbergroup>
   User-Agent: FreeSWITCH-mod_sofia/1.2.3
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Proxy-Authorization: ***STRIPPED***
   Reason: Q.850;cause=3;text="NO_ROUTE_DESTINATION"
   Content-Length: 0

   ------------------------------------------------------------------------
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:703 (sofia/external/447866123456) State REPORTING
2012-11-19 04:24:34.944968 [ERR] mod_xml_cdr.c:247 Error writing [/usr/local/freeswitch/log/xml_cdr/a_fb3e0014-3200-11e2-a46e-bdc0bb753e98.cdr.xml][No such file or directory]
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:92 sofia/external/447866123456 Standard REPORTING, cause: NO_ROUTE_DESTINATION
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:703 (sofia/external/447866123456) State REPORTING going to sleep
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:423 (sofia/external/447866123456) State Change CS_REPORTING -> CS_DESTROY
2012-11-19 04:24:34.944968 [DEBUG] switch_core_session.c:1210 Send signal sofia/external/447866123456 [BREAK]
2012-11-19 04:24:34.944968 [DEBUG] switch_core_session.c:1415 Session 121 (sofia/external/447866123456) Locked, Waiting on external entities
2012-11-19 04:24:34.944968 [NOTICE] switch_core_session.c:1433 Session 121 (sofia/external/447866123456) Ended
2012-11-19 04:24:34.944968 [NOTICE] switch_core_session.c:1437 Close Channel sofia/external/447866123456 [CS_DESTROY]
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:527 (sofia/external/447866123456) Callstate Change HANGUP -> DOWN
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:530 (sofia/external/447866123456) Running State Change CS_DESTROY
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:540 (sofia/external/447866123456) State DESTROY
2012-11-19 04:24:34.944968 [DEBUG] mod_sofia.c:376 sofia/external/447866123456 SOFIA DESTROY
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:99 sofia/external/447866123456 Standard DESTROY
2012-11-19 04:24:34.944968 [DEBUG] switch_core_state_machine.c:540 (sofia/external/447866123456) State DESTROY going to sleep
recv 472 bytes from udp/[80.93.165.111]:5060 at 04:24:34.971946:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 89.238.182.137;rport=5060;branch=z9hG4bKrc051jQ764ZSF
   From: "foxx" <sip:2000 at 89.238.182.137>;tag=27e1910pHj3Ze
   To: <sip:447866123456 at sip.numbergroup-services.com>;tag=rrSHS2S5ZeZ9g
   Call-ID: d29c5ff4-aca3-1230-eead-000c299684b0
   CSeq: 36313467 BYE
   User-Agent: numbergroup.com
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

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

From cal.leeming at simplicitymedialtd.co.uk  Mon Nov 19 07:38:28 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 19 Nov 2012 04:38:28 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
References: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
Message-ID: <CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>

Not sure if this is relevant but thought I'd point it out.

The following field seems to contain the IP of the domain we were expecting
('c1881.voiceflare.co.uk')

 u'variable_sip_from_host': u'89.238.182.137',

Normally, this field would contain the hostname and not the IP.

Cal

On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hi guys,
>
> In a nut shell, it appears that when attempting to perform a blind
> transfer under certain conditions (explained below), mod_xml_curl does not
> expose the originating domain in a clean format.
>
> My initial plan was to find the point where these variable were being
> generated, look at what was available, then add an extra variable for the
> domain and submit a patch.
>
> Sadly my C isn't great and I hit a brick wall, so if anyone can help out,
> I will ensure the mod_xml_curl documentation is updated and/or assist with
> any patching/testing required.
>
> Please take the following scenario;
>
> * Extension 2000 calls an external number via a gateway (i.e. bridge
> sofia/gateway/name/e164_number_here).
> * Call connects fine, audio stays good, no disconnection problems etc.
> * Call is blind transferred to another extension
>
> As a result, the following is determined;
>
> * User initiating the blind transfer is 2000
> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
> * Destination number of the call is 447866123456
> * Number to blind transfer to is 2001
> * Call to mod_xml_curl is made
>
> It makes reference to the User in the following 'clean' variables (by
> clean, I mean variables that just contain 2000 and don't require mangling
> to extract the info);
>
>  u'Caller-ANI': u'2000',
>  u'Caller-Username': u'2000',
>  u'Caller-Caller-ID-Number': u'2000',
>  u'Hunt-ANI': u'2000',
>  u'Hunt-Caller-ID-Number': u'2000',
>  u'Hunt-Username': u'2000',
>  u'variable_last_sent_callee_id_number': u'2000',
>  u'variable_sip_from_user': u'2000',
>
> It also has the User in the following unclean variables;
>
>  u'variable_bridge_channel': u'sofia/external/
> 2000 at c1881.voiceflare.co.uk:5060',
>  u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
> >;tag=XryjFQp3rB2NF',
>  u'variable_sip_h_Referred-By': u'"foxx" <
> sip:2000 at c1881.voiceflare.co.uk:5060>',
>
> However, it only references the domain in the following unclean variables;
>
>  u'variable_bridge_channel': u'sofia/external/
> 2000 at c1881.voiceflare.co.uk:5060',
>  u'variable_sip_h_Referred-By': u'"foxx" <
> sip:2000 at c1881.voiceflare.co.uk:5060>',
>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',
>
> Lets say that we want to determine the user/domain that has initiated this
> transfer, doing so would mean mangling with one of those above variables,
> which seems a bit dirty (plus it is not clear which is the correct one to
> use).
>
> I have attached the SIP trace of the entire blind transfer event, and the
> full mod_xml_curl request being sent.
>
> If anyone could answer the following, it'd be much appreciated;
>
> * Should there be an improvement patch that exposes the domain of the user
> that originated the blind transfer?
> * Are there better/alternative ways to extracting the domain of the user
> that originated the blind transfer?
>
> Many thanks
>
> Cal
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Mon Nov 19 07:58:41 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 19 Nov 2012 04:58:41 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>
References: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
	<CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>
Message-ID: <CAHKQagFrWbRft0wLiPsML22K61mfM+Z3Qh1ZJC+aKiebnNB+GA@mail.gmail.com>

Another quick update on this before I pass out from lack of sleep..!

Just had a look through the mod_sofia.c/h source and found the following;

mod_sofia.c/mod_sofia.h
#define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
if (!zstr(full_ref_by)) {
switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
"Referred-By", full_ref_by);
}
if (!zstr(full_ref_to)) {
switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
full_ref_to);
}
if (!zstr(full_ref_to)) {
switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
full_ref_to);
}
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
"Process REFER to [%s@%s]\n", exten, (char *) refer_to->r_url->url_host);

If the correct approach is deemed to be patching code, then I figured it
could be as simple as this;

switch_channel_set_variable(t_channel, "Referred-By-User", exten);
switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *)
refer_to->r_url->url_host);

This is pretty much where my knowledge of C ends, I can (just about) read
and copy chunks of C code, but that's about it :)

Cal

On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Not sure if this is relevant but thought I'd point it out.
>
> The following field seems to contain the IP of the domain we were
> expecting ('c1881.voiceflare.co.uk')
>
>  u'variable_sip_from_host': u'89.238.182.137',
>
> Normally, this field would contain the hostname and not the IP.
>
> Cal
>
> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Hi guys,
>>
>> In a nut shell, it appears that when attempting to perform a blind
>> transfer under certain conditions (explained below), mod_xml_curl does not
>> expose the originating domain in a clean format.
>>
>> My initial plan was to find the point where these variable were being
>> generated, look at what was available, then add an extra variable for the
>> domain and submit a patch.
>>
>> Sadly my C isn't great and I hit a brick wall, so if anyone can help out,
>> I will ensure the mod_xml_curl documentation is updated and/or assist with
>> any patching/testing required.
>>
>>  Please take the following scenario;
>>
>> * Extension 2000 calls an external number via a gateway (i.e. bridge
>> sofia/gateway/name/e164_number_here).
>> * Call connects fine, audio stays good, no disconnection problems etc.
>> * Call is blind transferred to another extension
>>
>> As a result, the following is determined;
>>
>> * User initiating the blind transfer is 2000
>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>> * Destination number of the call is 447866123456
>> * Number to blind transfer to is 2001
>> * Call to mod_xml_curl is made
>>
>> It makes reference to the User in the following 'clean' variables (by
>> clean, I mean variables that just contain 2000 and don't require mangling
>> to extract the info);
>>
>>  u'Caller-ANI': u'2000',
>>  u'Caller-Username': u'2000',
>>  u'Caller-Caller-ID-Number': u'2000',
>>  u'Hunt-ANI': u'2000',
>>  u'Hunt-Caller-ID-Number': u'2000',
>>  u'Hunt-Username': u'2000',
>>  u'variable_last_sent_callee_id_number': u'2000',
>>  u'variable_sip_from_user': u'2000',
>>
>> It also has the User in the following unclean variables;
>>
>>  u'variable_bridge_channel': u'sofia/external/
>> 2000 at c1881.voiceflare.co.uk:5060',
>>  u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>> >;tag=XryjFQp3rB2NF',
>>  u'variable_sip_h_Referred-By': u'"foxx" <
>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>
>> However, it only references the domain in the following unclean variables;
>>
>>  u'variable_bridge_channel': u'sofia/external/
>> 2000 at c1881.voiceflare.co.uk:5060',
>>  u'variable_sip_h_Referred-By': u'"foxx" <
>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',
>>
>> Lets say that we want to determine the user/domain that has initiated
>> this transfer, doing so would mean mangling with one of those above
>> variables, which seems a bit dirty (plus it is not clear which is the
>> correct one to use).
>>
>> I have attached the SIP trace of the entire blind transfer event, and the
>> full mod_xml_curl request being sent.
>>
>> If anyone could answer the following, it'd be much appreciated;
>>
>> * Should there be an improvement patch that exposes the domain of the
>> user that originated the blind transfer?
>> * Are there better/alternative ways to extracting the domain of the user
>> that originated the blind transfer?
>>
>> Many thanks
>>
>> Cal
>>
>>
>
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From cal.leeming at simplicitymedialtd.co.uk  Mon Nov 19 08:52:56 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 19 Nov 2012 05:52:56 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagFrWbRft0wLiPsML22K61mfM+Z3Qh1ZJC+aKiebnNB+GA@mail.gmail.com>
References: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
	<CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>
	<CAHKQagFrWbRft0wLiPsML22K61mfM+Z3Qh1ZJC+aKiebnNB+GA@mail.gmail.com>
Message-ID: <CAHKQagHvQwWCOvdM0abqrM1xz65kFz9hY6J+fMKbJjH=Ot=kgQ@mail.gmail.com>

Sorry, another update.. after tinkering with the SIP headers, we found that
we're able to pass any user/host along in an INVITE, and this is passed
onto mod_xml_curl.

To fix this particular part of the problem, we enabled auth_calls and this
gives us correct/clean variables we can trust, specifically;

 u'variable_sip_auth_username': u'2000',
 u'variable_user_name': u'2000',

However, when attempting to do the blind transfer again, this variables are
all missing.

At this point I'm convinced that attempting to extract the user/domain from
the Refer headers is probably not the right thing to do... But I'm still no
closer to figuring out what the correct approach should be to expose the
authenticated user/domain to mod_xml_curl.

Cal


On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Another quick update on this before I pass out from lack of sleep..!
>
> Just had a look through the mod_sofia.c/h source and found the following;
>
> mod_sofia.c/mod_sofia.h
> #define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
> if (!zstr(full_ref_by)) {
> switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
> "Referred-By", full_ref_by);
> }
> if (!zstr(full_ref_to)) {
> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
> full_ref_to);
> }
> if (!zstr(full_ref_to)) {
>  switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
> full_ref_to);
> }
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
> "Process REFER to [%s@%s]\n", exten, (char *) refer_to->r_url->url_host);
>
> If the correct approach is deemed to be patching code, then I figured it
> could be as simple as this;
>
> switch_channel_set_variable(t_channel, "Referred-By-User", exten);
> switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *)
> refer_to->r_url->url_host);
>
> This is pretty much where my knowledge of C ends, I can (just about) read
> and copy chunks of C code, but that's about it :)
>
> Cal
>
> On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Not sure if this is relevant but thought I'd point it out.
>>
>> The following field seems to contain the IP of the domain we were
>> expecting ('c1881.voiceflare.co.uk')
>>
>>  u'variable_sip_from_host': u'89.238.182.137',
>>
>> Normally, this field would contain the hostname and not the IP.
>>
>> Cal
>>
>> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Hi guys,
>>>
>>> In a nut shell, it appears that when attempting to perform a blind
>>> transfer under certain conditions (explained below), mod_xml_curl does not
>>> expose the originating domain in a clean format.
>>>
>>> My initial plan was to find the point where these variable were being
>>> generated, look at what was available, then add an extra variable for the
>>> domain and submit a patch.
>>>
>>> Sadly my C isn't great and I hit a brick wall, so if anyone can help
>>> out, I will ensure the mod_xml_curl documentation is updated and/or assist
>>> with any patching/testing required.
>>>
>>>  Please take the following scenario;
>>>
>>> * Extension 2000 calls an external number via a gateway (i.e. bridge
>>> sofia/gateway/name/e164_number_here).
>>> * Call connects fine, audio stays good, no disconnection problems etc.
>>> * Call is blind transferred to another extension
>>>
>>> As a result, the following is determined;
>>>
>>> * User initiating the blind transfer is 2000
>>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>>> * Destination number of the call is 447866123456
>>> * Number to blind transfer to is 2001
>>> * Call to mod_xml_curl is made
>>>
>>> It makes reference to the User in the following 'clean' variables (by
>>> clean, I mean variables that just contain 2000 and don't require mangling
>>> to extract the info);
>>>
>>>  u'Caller-ANI': u'2000',
>>>  u'Caller-Username': u'2000',
>>>  u'Caller-Caller-ID-Number': u'2000',
>>>  u'Hunt-ANI': u'2000',
>>>  u'Hunt-Caller-ID-Number': u'2000',
>>>  u'Hunt-Username': u'2000',
>>>  u'variable_last_sent_callee_id_number': u'2000',
>>>  u'variable_sip_from_user': u'2000',
>>>
>>> It also has the User in the following unclean variables;
>>>
>>>  u'variable_bridge_channel': u'sofia/external/
>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>  u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>>> >;tag=XryjFQp3rB2NF',
>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>
>>> However, it only references the domain in the following unclean
>>> variables;
>>>
>>>  u'variable_bridge_channel': u'sofia/external/
>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',
>>>
>>> Lets say that we want to determine the user/domain that has initiated
>>> this transfer, doing so would mean mangling with one of those above
>>> variables, which seems a bit dirty (plus it is not clear which is the
>>> correct one to use).
>>>
>>> I have attached the SIP trace of the entire blind transfer event, and
>>> the full mod_xml_curl request being sent.
>>>
>>> If anyone could answer the following, it'd be much appreciated;
>>>
>>> * Should there be an improvement patch that exposes the domain of the
>>> user that originated the blind transfer?
>>> * Are there better/alternative ways to extracting the domain of the user
>>> that originated the blind transfer?
>>>
>>> Many thanks
>>>
>>> Cal
>>>
>>>
>>
>
-------------- next part --------------
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From cal.leeming at simplicitymedialtd.co.uk  Mon Nov 19 10:34:04 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 19 Nov 2012 07:34:04 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagHvQwWCOvdM0abqrM1xz65kFz9hY6J+fMKbJjH=Ot=kgQ@mail.gmail.com>
References: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
	<CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>
	<CAHKQagFrWbRft0wLiPsML22K61mfM+Z3Qh1ZJC+aKiebnNB+GA@mail.gmail.com>
	<CAHKQagHvQwWCOvdM0abqrM1xz65kFz9hY6J+fMKbJjH=Ot=kgQ@mail.gmail.com>
Message-ID: <CAHKQagHowbH_Zr8YPWGDsbug4m1ggcaTY=LdgzYf9yvJq15qFA@mail.gmail.com>

Last update, then I'm really going to sleep!

Apologies for the noise btw, in hindsight I should have collected all this
info and posted in one go.

I've tried enabling auth-all-packets (along with auth_calls), as I thought
maybe having authentication on REFER packets might make a difference, but
sadly it had no effect (the SIP headers show proxy-authorization in the
REFER, but nothing extra showed in mod_xml_curl)

I've managed to narrow down the circumstances in which this happens;

CORRECT:

* User receives call from gateway, blind transfer to another user (shows
correctly as variable_dialed_user/variable_dialed_domain)
* User receives call from another user, blind transfer to gateway (shows
correctly as variable_dialed_user/variable_dialed_domain)
* User receives call from another user, blind transfer to another
user (shows correctly as variable_dialed_user/variable_dialed_domain)
* User makes call to another user, blind transfer to another user (shows
correctly as variable_dialed_user/variable_dialed_domain)
* User makes call to another user, blind transfer to a gateway (shows
correctly as variable_dialed_user/variable_dialed_domain)

MISSING:

* User makes call to a gateway, blind transfer to another gateway (no clean
variables for domain)
* User makes call to a gateway, blind transfer to another user (no clean
variables for domain)

So, the problem seems to happen specifically when you blind transfer a call
that is already in progress on a gateway, where the call was originated by
a user and not a gateway.

I did a bit more looking through code, added a few switch_log_printf()
calls, and found that the following method is NOT called in those two
scenarios where these variables are missing;
mod_dptools.c: "switch_call_cause_t user_outgoing_channel"

This is about as far as I can go on this, as I just don't know enough about
C to give any more in-depth info :/

Cal

On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Sorry, another update.. after tinkering with the SIP headers, we found
> that we're able to pass any user/host along in an INVITE, and this is
> passed onto mod_xml_curl.
>
> To fix this particular part of the problem, we enabled auth_calls and this
> gives us correct/clean variables we can trust, specifically;
>
>  u'variable_sip_auth_username': u'2000',
>  u'variable_user_name': u'2000',
>
> However, when attempting to do the blind transfer again, this variables
> are all missing.
>
> At this point I'm convinced that attempting to extract the user/domain
> from the Refer headers is probably not the right thing to do... But I'm
> still no closer to figuring out what the correct approach should be to
> expose the authenticated user/domain to mod_xml_curl.
>
> Cal
>
>
> On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Another quick update on this before I pass out from lack of sleep..!
>>
>> Just had a look through the mod_sofia.c/h source and found the following;
>>
>> mod_sofia.c/mod_sofia.h
>> #define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
>> if (!zstr(full_ref_by)) {
>> switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
>> "Referred-By", full_ref_by);
>> }
>> if (!zstr(full_ref_to)) {
>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>> full_ref_to);
>> }
>> if (!zstr(full_ref_to)) {
>>  switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>> full_ref_to);
>> }
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>> "Process REFER to [%s@%s]\n", exten, (char *) refer_to->r_url->url_host);
>>
>> If the correct approach is deemed to be patching code, then I figured it
>> could be as simple as this;
>>
>> switch_channel_set_variable(t_channel, "Referred-By-User", exten);
>> switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *)
>> refer_to->r_url->url_host);
>>
>> This is pretty much where my knowledge of C ends, I can (just about) read
>> and copy chunks of C code, but that's about it :)
>>
>> Cal
>>
>> On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Not sure if this is relevant but thought I'd point it out.
>>>
>>> The following field seems to contain the IP of the domain we were
>>> expecting ('c1881.voiceflare.co.uk')
>>>
>>>  u'variable_sip_from_host': u'89.238.182.137',
>>>
>>> Normally, this field would contain the hostname and not the IP.
>>>
>>> Cal
>>>
>>> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>>> Hi guys,
>>>>
>>>> In a nut shell, it appears that when attempting to perform a blind
>>>> transfer under certain conditions (explained below), mod_xml_curl does not
>>>> expose the originating domain in a clean format.
>>>>
>>>> My initial plan was to find the point where these variable were being
>>>> generated, look at what was available, then add an extra variable for the
>>>> domain and submit a patch.
>>>>
>>>> Sadly my C isn't great and I hit a brick wall, so if anyone can help
>>>> out, I will ensure the mod_xml_curl documentation is updated and/or assist
>>>> with any patching/testing required.
>>>>
>>>>  Please take the following scenario;
>>>>
>>>> * Extension 2000 calls an external number via a gateway (i.e. bridge
>>>> sofia/gateway/name/e164_number_here).
>>>> * Call connects fine, audio stays good, no disconnection problems etc.
>>>> * Call is blind transferred to another extension
>>>>
>>>> As a result, the following is determined;
>>>>
>>>> * User initiating the blind transfer is 2000
>>>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>>>> * Destination number of the call is 447866123456
>>>> * Number to blind transfer to is 2001
>>>> * Call to mod_xml_curl is made
>>>>
>>>> It makes reference to the User in the following 'clean' variables (by
>>>> clean, I mean variables that just contain 2000 and don't require mangling
>>>> to extract the info);
>>>>
>>>>  u'Caller-ANI': u'2000',
>>>>  u'Caller-Username': u'2000',
>>>>  u'Caller-Caller-ID-Number': u'2000',
>>>>  u'Hunt-ANI': u'2000',
>>>>  u'Hunt-Caller-ID-Number': u'2000',
>>>>  u'Hunt-Username': u'2000',
>>>>  u'variable_last_sent_callee_id_number': u'2000',
>>>>  u'variable_sip_from_user': u'2000',
>>>>
>>>> It also has the User in the following unclean variables;
>>>>
>>>>  u'variable_bridge_channel': u'sofia/external/
>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>  u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>>>> >;tag=XryjFQp3rB2NF',
>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>
>>>> However, it only references the domain in the following unclean
>>>> variables;
>>>>
>>>>  u'variable_bridge_channel': u'sofia/external/
>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',
>>>>
>>>> Lets say that we want to determine the user/domain that has initiated
>>>> this transfer, doing so would mean mangling with one of those above
>>>> variables, which seems a bit dirty (plus it is not clear which is the
>>>> correct one to use).
>>>>
>>>> I have attached the SIP trace of the entire blind transfer event, and
>>>> the full mod_xml_curl request being sent.
>>>>
>>>> If anyone could answer the following, it'd be much appreciated;
>>>>
>>>> * Should there be an improvement patch that exposes the domain of the
>>>> user that originated the blind transfer?
>>>> * Are there better/alternative ways to extracting the domain of the
>>>> user that originated the blind transfer?
>>>>
>>>> Many thanks
>>>>
>>>> Cal
>>>>
>>>>
>>>
>>
>
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From hiryu23 at gmail.com  Mon Nov 19 08:37:18 2012
From: hiryu23 at gmail.com (hiryu23)
Date: Sun, 18 Nov 2012 21:37:18 -0800 (PST)
Subject: [Freeswitch-users] Subscribe for MWI
In-Reply-To: <11AAB3D9-F17F-4AC1-B66C-591CF93155D1@5ninesolutions.com>
References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com>
	<CAKbxfG8YwzEHCMTXNC065hd5EOjA9Cjs5bJaXJsHnqYsHrA87g@mail.gmail.com>
	<8b33f052-3895-4252-8f47-6ce672e810c6@blur>
	<850DE19F-0F8B-4E4F-9B22-534311287478@5ninesolutions.com>
	<1338282624250-7579201.post@n2.nabble.com>
	<1338282786982-7579202.post@n2.nabble.com>
	<F2AF4752-E260-44CE-8230-1EF58A168855@5ninesolutions.com>
	<FF236F95-370C-492A-9EA5-B62B1EFAB893@5ninesolutions.com>
	<1342167849415-7580806.post@n2.nabble.com>
	<11AAB3D9-F17F-4AC1-B66C-591CF93155D1@5ninesolutions.com>
Message-ID: <1353303438948-7584698.post@n2.nabble.com>

I notice when i register with UDP/TCP, i am able to receive the NOTIFY packet
from FreeSwitch.
However, when i register with TLS, I do not see FreeSwitch generating any
NOTIFY packet. 

any help?



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Subscribe-for-MWI-tp7557448p7584698.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Mon Nov 19 14:19:30 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 19 Nov 2012 12:19:30 +0100
Subject: [Freeswitch-users] How to enable unicode in CLI command?
In-Reply-To: <BLU169-W654F5F82AD03E6AC000B8F8C570@phx.gbl>
References: <BLU169-W654F5F82AD03E6AC000B8F8C570@phx.gbl>
Message-ID: <CALXCt0p9dR63AfScg5V6=4jugWCnEharx=tOKLLtXqrzXnf7eA@mail.gmail.com>

from CLI you can't.
You can use ESL, eg telnet to port 8021 (IIRC).
Anyway, look in the wiki page of gsmopen and/or skypopen for an example howto.
Via ESL unicode is supported.
-giovanni

On 11/18/12, ?? <itispip-qq at hotmail.com> wrote:
>
> I tried to use chat / skypopen_chat from CLI console to send a message to
> receiver, the message is written by unicode charactor (Korean or Thai), the
> receiver cannot get correct charators;
> Tried to call chat / skypopen_chat API from LUA/JScript, ended the same;
> I believe Freeswitch itself support unicode, as when I use an SIP client as
> "Bria" to send SIP chat message between 2 clients, unicode charactor
> displays correctly.
> Is it just the CLI command by design not support unicode, or I have to do
> something in config xml?
>  		 	   		


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Rob.Moore at Aeriandi.com  Mon Nov 19 14:51:19 2012
From: Rob.Moore at Aeriandi.com (Rob Moore)
Date: Mon, 19 Nov 2012 11:51:19 +0000
Subject: [Freeswitch-users] Error on Call Recording - Windows, FS V1.2.4
Message-ID: <49C5FCA19A8A114493EBAACA42FE5899105E9041@1AERDCEXCHMBX1.AER.AERCO.local>

Hi Everyone,

Just a minor one here I'm checking on.

Running the latest release of Freeswitch V1.2.4 (on Windows Server 2008R2, yes I know!) and running through some tests with our system that uses Lua scripts to request actions from the Server.

Whilst recording calls I'm getting the following error, the call recording files is still created (or appended to if I choose to call this).

2012-11-19 11:29:22.666979 [ERR] switch_ivr_async.c:1840 Error finding the folder path section in 'C:\FreeSwitch\recordings\RobsTESTRecording5762fadd-7b5-8abf-e61cb1633827.wav'

Does anyone have any idea what could be causing this?

I've had a dig around in the documentation for switch_ivr_async.c. It looks like and error relating to a check on creating a directory but If I'm honest it's a little beyond my current skill set.

Really I'm just trying to get shot of the error message and at the moment I'm a little stumped as to what could be causing this (although I wouldn't be surprised if it's the windows file system)

Thanks in advanced

Rob

Rob Moore

Telephony Systems Infrastructure Manager

[Description: Aeriandi]

m. +44 (0)7766 838 040
t.   +44 (0)845 108 0308

Aeriandi Ltd / Prama House, 267 Banbury Road, Oxford OX2 7HT / +44 (0)845 108 0308 / info at aeriandi.com<mailto:info at aeriandi.com> / aeriandi.com<http://www.aeriandi.com/>

This transmission is intended solely for the addressee and may be confidential. If you are not the named addressee, or if the message has been addressed to you in error, you must not read, disclose, reproduce, distribute or use this transmission. Please delete the message or contact the sender. Delivery of this message to any person other than the named addressee is not intended in any way to waive confidentiality. Please note that neither Aeriandi Ltd nor the sender accepts any responsibility for any viruses, which may be transmitted, and it is your responsibility to scan attachments (if any).



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From NuwanW at unifybusiness.co.uk  Mon Nov 19 15:35:46 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 19 Nov 2012 12:35:46 +0000
Subject: [Freeswitch-users] [Confidential] - RE: Error on Call Recording -
	Windows, FS V1.2.4
In-Reply-To: <49C5FCA19A8A114493EBAACA42FE5899105E9041@1AERDCEXCHMBX1.AER.AERCO.local>
References: <49C5FCA19A8A114493EBAACA42FE5899105E9041@1AERDCEXCHMBX1.AER.AERCO.local>
Message-ID: <78990CE7CC964442A7C2CA5F4689695E99BCB82B@BARXB0003.UnifyBusiness.local>

Rob,

I had the same issue with FreeSwitch .net sockets. To fix this, replace "\" with "/". So in your record request, file path should be  'C:/FreeSwitch/recordings/RobsTESTRecording5762fadd-7b5-8abf-e61cb1633827.wav'


Nuwan

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Moore
Sent: 19 November 2012 11:51
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Error on Call Recording - Windows, FS V1.2.4

Hi Everyone,

Just a minor one here I'm checking on.

Running the latest release of Freeswitch V1.2.4 (on Windows Server 2008R2, yes I know!) and running through some tests with our system that uses Lua scripts to request actions from the Server.

Whilst recording calls I'm getting the following error, the call recording files is still created (or appended to if I choose to call this).

2012-11-19 11:29:22.666979 [ERR] switch_ivr_async.c:1840 Error finding the folder path section in 'C:\FreeSwitch\recordings\RobsTESTRecording5762fadd-7b5-8abf-e61cb1633827.wav'

Does anyone have any idea what could be causing this?

I've had a dig around in the documentation for switch_ivr_async.c. It looks like and error relating to a check on creating a directory but If I'm honest it's a little beyond my current skill set.

Really I'm just trying to get shot of the error message and at the moment I'm a little stumped as to what could be causing this (although I wouldn't be surprised if it's the windows file system)

Thanks in advanced

Rob

Rob Moore

Telephony Systems Infrastructure Manager

[Description: Aeriandi]

m. +44 (0)7766 838 040
t.   +44 (0)845 108 0308

Aeriandi Ltd / Prama House, 267 Banbury Road, Oxford OX2 7HT / +44 (0)845 108 0308 / info at aeriandi.com<mailto:info at aeriandi.com> / aeriandi.com<http://www.aeriandi.com/>

This transmission is intended solely for the addressee and may be confidential. If you are not the named addressee, or if the message has been addressed to you in error, you must not read, disclose, reproduce, distribute or use this transmission. Please delete the message or contact the sender. Delivery of this message to any person other than the named addressee is not intended in any way to waive confidentiality. Please note that neither Aeriandi Ltd nor the sender accepts any responsibility for any viruses, which may be transmitted, and it is your responsibility to scan attachments (if any).



This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From peter.olsson at visionutveckling.se  Mon Nov 19 15:41:05 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 19 Nov 2012 12:41:05 +0000
Subject: [Freeswitch-users] Error on Call Recording - Windows, FS V1.2.4
Message-ID: <1FFF97C269757C458224B7C895F35F151C5CA0@cantor.std.visionutv.se>

It might be one of those Windows path things... Try using / instead of \ as a path separator. Both of them works in Windows, but FS probably only checks the "/" in the current code.

Also, the only problem here is that if the directory wouldn't exist already, it would fail - besides that there is no real problem, except the error log message.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Rob Moore
Skickat: den 19 november 2012 12:51
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Error on Call Recording - Windows, FS V1.2.4

Hi Everyone,

Just a minor one here I'm checking on.

Running the latest release of Freeswitch V1.2.4 (on Windows Server 2008R2, yes I know!) and running through some tests with our system that uses Lua scripts to request actions from the Server.

Whilst recording calls I'm getting the following error, the call recording files is still created (or appended to if I choose to call this).

2012-11-19 11:29:22.666979 [ERR] switch_ivr_async.c:1840 Error finding the folder path section in 'C:\FreeSwitch\recordings\RobsTESTRecording5762fadd-7b5-8abf-e61cb1633827.wav'

Does anyone have any idea what could be causing this?

I've had a dig around in the documentation for switch_ivr_async.c. It looks like and error relating to a check on creating a directory but If I'm honest it's a little beyond my current skill set.

Really I'm just trying to get shot of the error message and at the moment I'm a little stumped as to what could be causing this (although I wouldn't be surprised if it's the windows file system)

Thanks in advanced

Rob

Rob Moore

Telephony Systems Infrastructure Manager

[Description: Aeriandi]

m. +44 (0)7766 838 040
t.   +44 (0)845 108 0308

Aeriandi Ltd / Prama House, 267 Banbury Road, Oxford OX2 7HT / +44 (0)845 108 0308 / info at aeriandi.com<mailto:info at aeriandi.com> / aeriandi.com<http://www.aeriandi.com/>

This transmission is intended solely for the addressee and may be confidential. If you are not the named addressee, or if the message has been addressed to you in error, you must not read, disclose, reproduce, distribute or use this transmission. Please delete the message or contact the sender. Delivery of this message to any person other than the named addressee is not intended in any way to waive confidentiality. Please note that neither Aeriandi Ltd nor the sender accepts any responsibility for any viruses, which may be transmitted, and it is your responsibility to scan attachments (if any).



!DSPAM:50aa21e332761953521012!
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From mnrao2001 at gmail.com  Mon Nov 19 15:45:18 2012
From: mnrao2001 at gmail.com (Nageshwara Rao Moova)
Date: Mon, 19 Nov 2012 18:15:18 +0530
Subject: [Freeswitch-users] Log Rollover issue
Message-ID: <CAPDWOejFaGi6OmavH2Q9nKYRzn6zpz1gDbiSVqEcaCV678+bbg@mail.gmail.com>

Hi all,

I have modified my default logconf file for rollover to be restricted to
100. But sometimes freeswitch is unable to rename rollover say ?cannot
rename freeswitch.log.41? and fails. But the fail follows with serious
issue of writing all the logs to "freeswitch.log" and ends up filling the
disk space.

I have not change the default file size i.e 10MB.

How does 10MB size overridden by freeswitch?

-- 
regards & thanks
--
m nageshwara rao
99891 86280
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From Tim.Meade at Millicorp.com  Mon Nov 19 16:14:15 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Mon, 19 Nov 2012 13:14:15 +0000
Subject: [Freeswitch-users] G729 File version of $${us-ring}
In-Reply-To: <CALn233qk8dMRD2hmNdn-f5_R5GJdBxyrQDycV2JJqaOO3Mmktg@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AD9A4EC@MAIL.millicorp.com>
	<CALn233qk8dMRD2hmNdn-f5_R5GJdBxyrQDycV2JJqaOO3Mmktg@mail.gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350ADA59B6@MAIL.millicorp.com>

Thanks Mitch.  On today's (or tomorrow) todo list.   Was just wondering if someone had a nice clean one out there.

Tim

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
Sent: Sunday, November 18, 2012 9:31 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] G729 File version of $${us-ring}

As long as you have a single G729 license just create it yourself us_ring is just the tone stream used so create a g729 call, record it, and play the tone stream.

~mitch

On Sat, Nov 17, 2012 at 8:25 AM, Tim Meade <Tim.Meade at millicorp.com> wrote:
>
>
> I was wondering if anyone had a G729 file of the $${us-ring}
>
>
>
> We want to use it for playing the ringback with G729 pass through.
>
>
>
> <action application="set"  data="ringback=SomeRingBackFile.G729"/>
>
>
>
>
>
> Thanks
>
>
>
> Tim
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>

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

 

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 bret at ticm.com  Mon Nov 19 16:41:47 2012
From: bret at ticm.com (Bret Watson)
Date: Mon, 19 Nov 2012 21:41:47 +0800
Subject: [Freeswitch-users] how to force a fixed request timer on a gateway?
Message-ID: <CAN4s1rgqA2o5zwTeiO8bSdz=BQDj+gRtWoCYYM6YqFHUbQ_NBw@mail.gmail.com>

I think I've gotten to the root cause of the pennytel failure problems.

Basically Pennytel is handing the wrong refresh time when my end makes the
registration attempt. The first attempt if the box has been off for a day -
always works, but not long afterwards its fails. with a series of 200 ok
issues and finally a 400.

It seems that freeswitch assumes Pennytel is asking for shorter and shorter
refresh periods, and pennytel interprets my end do more frequent
registrations as flooding and bans me.

So I figure if I can override what Pennytel is sending with my own timeout
period, it should stay happy - I've tried

<param name="expire-seconds" value="360"/>
but with no effect.

Thanks

Bret
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From philq at qsystemsengineering.com  Mon Nov 19 17:45:46 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Mon, 19 Nov 2012 09:45:46 -0500
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
	contact IP/port
Message-ID: <054001cdc664$92486660$b6d93320$@com>

Is there a way to get FreeSWITCH to rewrite the media address as well as the
contact header info coming from connected endpoints?  With the proper
settings, FS does a great job of taking care of just about any NAT-related
issue on the SIP side, but going through the documentation I've been unable
to find a way to get it to touch the SDP.  Is there a setting to get it to
use the same address for media?

Thanks,

Phil Quesinberry
Q Systems Engineering, Inc.
Embedded Systems Development and VoIP Business Telephone Hosting
Improve your business telephone services and save money
(410) 969-8002
http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/> 


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From david.villasmil.work at gmail.com  Mon Nov 19 17:59:17 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 19 Nov 2012 15:59:17 +0100
Subject: [Freeswitch-users] FreeSWITCH and Digium T410P
Message-ID: <CAFGRPVrko1_T_Oa8B7Rzh=WDdz73WOuy7OOGPe1aq-q=w_+60Q@mail.gmail.com>

Hello Guys,

Is there any *clear* documentation to install a Digium T410P? there's so
many stuff I don't have it clear whether libpri is needed or not, whether I
need some sangoma header, etc..

Also, I understand no SS7 is supported on Digium?

Anything clear?

Thanks!!

David
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From sterned at xakep.ru  Mon Nov 19 18:01:49 2012
From: sterned at xakep.ru (sterned)
Date: Mon, 19 Nov 2012 07:01:49 -0800 (PST)
Subject: [Freeswitch-users] problem with receiving and sending faxes
In-Reply-To: <CAKzWOxVvGmNznXcuSw+BEJmR0ZwvU7zijo5ynLMO2HUa5X2VGQ@mail.gmail.com>
References: <1352398794952-7584436.post@n2.nabble.com>
	<1353080771990-7584627.post@n2.nabble.com>
	<CAKzWOxVvGmNznXcuSw+BEJmR0ZwvU7zijo5ynLMO2HUa5X2VGQ@mail.gmail.com>
Message-ID: <1353337309443-7584709.post@n2.nabble.com>

Hi, here to sip trace logs.
http://pastebin.freeswitch.org/20233



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problem-with-receiving-and-sending-faxes-tp7584436p7584709.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rob.moore at aeriandi.com  Mon Nov 19 19:12:30 2012
From: rob.moore at aeriandi.com (Rob Moore)
Date: Mon, 19 Nov 2012 16:12:30 +0000 (UTC)
Subject: [Freeswitch-users]
	=?utf-8?q?=5BConfidential=5D_-_RE=3A_Error_on_?=
	=?utf-8?q?Call_Recording_-=09Windows=2C_FS_V1=2E2=2E4?=
References: <49C5FCA19A8A114493EBAACA42FE5899105E9041@1AERDCEXCHMBX1.AER.AERCO.local>
	<78990CE7CC964442A7C2CA5F4689695E99BCB82B@BARXB0003.UnifyBusiness.local>
Message-ID: <loom.20121119T165734-788@post.gmane.org>



Aww Nuwan you absolute legend! That worked perfectly. Many Thanks for the help!



Rob.



From msc at freeswitch.org  Mon Nov 19 19:28:52 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 08:28:52 -0800
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAMwZ-t+hkyfuf87b=48MfTMALDSyB7Efa-6BkfkKmMiLrx7+pA@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
	<CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
	<CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>
	<CAMwZ-t+hkyfuf87b=48MfTMALDSyB7Efa-6BkfkKmMiLrx7+pA@mail.gmail.com>
Message-ID: <CAKzWOxU+3RkOW1fO26y9gZ9eB8Bk4AEzEhe3=whJx7mTzYVf5g@mail.gmail.com>

Whatever the case, we'll put $1000 toward the effort. This is not a

> bounty (conditional based on finishing the feature). It's a
> no-obligation, thanks-for-an-awesome-product,
> good-luck-with-future-development, contribution.  I hope other
> companies that benefit daily from FreeSWITCH consider following suit.
> Let's make this happen :)
>
+1

Thanks for your support and setting such a great example!
-MC
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From anthony.minessale at gmail.com  Mon Nov 19 19:31:20 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Nov 2012 10:31:20 -0600
Subject: [Freeswitch-users] Suggestion to harmonize 'hash' & 'db' insert
	method
In-Reply-To: <CA+Y7Lf4+gfzWayBbLLGj6x4ckfOqMbXRB9tOniQUkXe0uU+T0w@mail.gmail.com>
References: <CA+Y7Lf4+gfzWayBbLLGj6x4ckfOqMbXRB9tOniQUkXe0uU+T0w@mail.gmail.com>
Message-ID: <CAKbxfG9gXT3q4JAUxAsCzrDhTd7HVC6MDRM4t-6q2P_OLfwrDg@mail.gmail.com>

I think this is already the intended behavior of the db app.

under insert it does the following:

sql = switch_mprintf("delete from db_data where realm='%q' and
data_key='%q'", argv[1], argv[2]);

then:

sql = switch_mprintf("insert into db_data (hostname, realm, data_key, data)
values('%q','%q','%q','%q');", globals.hostname, argv[1], argv[2], argv[3]);


If you are getting differing results, can you report it under Jira and post
examples on how to reproduce using GIT HEAD so we can get to the bottom of
it.  There must be a bug if this is happening.


Also .. Yiftach Golan.

Why do you continue to make childish comments?  Especially on other
people's thread.  Asking people to go offline with you so you can feed them
FUD about how we don't listen to people, meanwhile here we are every day
answering hundreds of emails.    Do you really not understand?  I am
seriously trying to help you here.  You need to learn a few things
about etiquette to work in a community setting.  There is a difference
between saying your opinion and arguing a moot point into the ground when
its not even your thread......  Being annoyed because we want to use Jira
which is the intended resource for development and issue tracking makes no
sense and is a bit selfish.  If we are going to share our project with you,
the least you can do is follow our minimal set of guidelines to make it
easier for a handful of people to manage a community of several thousand
people.....

=/






On Sat, Nov 17, 2012 at 7:42 PM, Scott <8f27e956 at gmail.com> wrote:

> Are there reasons why the function 'hash' (non-persistent storage) and
> function 'db' (persistent storage) share a look-a-like user interface (e.g.
> select/insert/delete) but do not work-a-like.  In particular, in 'hash',
> insert overwrites an identical realm/data_key pair whereas 'db' , in so far
> as I can tell, just blindly adds, and adds, adds, the insert(s).  However,
> the 'db' select method returns one record only even if the db has
> accumulated many realm/data_key records, including data_value duplicates.
>
> NOTWITHSTANDING reasons-unknow-to-me, a remedy to harmonize the
> non-persistent 'hash' with the persistent 'db'  I *think* is straight
> forward ...
>
> (1) At the call_limit.db schema-level, a one-time create UNIQUE COMPOUND
> index, as follows,
>
> CREATE UNIQUE INDEX IF NOT EXISTS 'idx_db_data_HostRealmDK' ON 'db_data'
> ('hostname','realm','data_key');
>
> (1) At the c-language embedded SQL string-level,
>
> change the existing INSERT to be the following,
>
> INSERT OR REPLACE INTO db_data (hostname,realm,data_key,data) VALUES
> (%s,%s,%s,%s);
>
> The index will help speed UP the reads.  Neither change is "exotic" and
> should be portable across standard sql implementations.
>
> ;-) ... thoughts ?
>
> Thanks,
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From msc at freeswitch.org  Mon Nov 19 20:04:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 09:04:46 -0800
Subject: [Freeswitch-users] extension name="Your_Are_Here"
In-Reply-To: <CA+Y7Lf7cfqigqjnK+t8XpNnR5Gx54MsGxSKv9Ofm8m1tHfqcLg@mail.gmail.com>
References: <CA+Y7Lf7cfqigqjnK+t8XpNnR5Gx54MsGxSKv9Ofm8m1tHfqcLg@mail.gmail.com>
Message-ID: <CAKzWOxUFSOYuuiXZVOV9ZwSvDJYnaCC6LR7nknSjSOVYopGPXQ@mail.gmail.com>

I don't see any chan vars that correspond to the extension name. However,
you can just create one:

<action application="set" data="ext_name=You_Are_Here" inline="true"/>
<action application="log" data="INFO this message emitted from [${ext_name}]"
/>

-MC

On Fri, Nov 16, 2012 at 7:15 PM, Scott <8f27e956 at gmail.com> wrote:

> In the dialplan xml, for the <extension name="You_Are_Here" > clause, how
> does one reference the value of the name= ?
>
> e.g. <action application="log" data="INFO this message emitted from [
> ${name}]" />
>
> It's in the debug output so it "in there somewhere."
>
> :-)
>
> With thanks,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mike at jerris.com  Mon Nov 19 20:07:00 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Nov 2012 12:07:00 -0500
Subject: [Freeswitch-users] changing vm default announcement
In-Reply-To: <50A710A2.90606@gmail.com>
References: 50A5BA00.5050209@gmail.com <50A710A2.90606@gmail.com>
Message-ID: <CD41D402-816D-4D45-BDCA-1241A0FB4078@jerris.com>

I do recall an issue with this being fixed, have you tried 1.2.3?

Mike


On Nov 16, 2012, at 11:20 PM, andy <packetandy at gmail.com> wrote:

> hi Nick, thanks for the reply. 
> 
> Greetings are in  $${base_dir}/storage/voicemail/default/$${domain}/<user>/greeting_1.wav as specified in the wiki.
> trying to  force freeswitch to use this custom greeting by using <action application="set" data="voicemail_greeting_number=1"/> 
> does not work. Default greeting from Callie still plays.
> 
> If one records a greeting and then overwrites the recorded greeting with another file in the same directory - that works. 
> 
> Guys, is this a bug, or do I not understand what the 'voicemail_greeting_number variable is supposed to do?
> 
> drew
> 
> BTW - I am on FS 1.2.1

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From msc at freeswitch.org  Mon Nov 19 20:11:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 09:11:20 -0800
Subject: [Freeswitch-users] changing vm default announcement
In-Reply-To: <50A710A2.90606@gmail.com>
References: <50A710A2.90606@gmail.com>
Message-ID: <CAKzWOxWdBNpMyfn0dbb6t_Pz=qc3U3-i4a3z0_J93D9F0dHYVA@mail.gmail.com>

Can you do a quick test? Dial into the voicemail box in question and go
into the options menu (5) and then do two things:
1 - record greeting number 4
2 - choose greeting number 4

Then send a call to that user's voicemail box and see if greeting number 4
plays. Look at the console to make sure it attempts to play greeting 4.
Let's go from there.

-MC

On Fri, Nov 16, 2012 at 8:20 PM, andy <packetandy at gmail.com> wrote:

>  hi Nick, thanks for the reply.
>
> Greetings are in
> $${base_dir}/storage/voicemail/default/$${domain}/<user>/greeting_1.wav as
> specified in the wiki.
>
> trying to  force freeswitch to use this custom greeting by using *<action application="set" data="voicemail_greeting_number=1"/> *does not work*. *Default greeting from Callie still plays*.
> *If one records a greeting and then overwrites the recorded greeting with another file in the same directory - that works.
>
> Guys, is this a bug, or do I not understand what the 'voicemail_greeting_number variable is supposed to do?
>
> drew
>
> BTW - I am on FS 1.2.1
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 19 20:13:12 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 09:13:12 -0800
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
In-Reply-To: <6626166B66164AB4B5C0E344762D7E4A@bob>
References: <6626166B66164AB4B5C0E344762D7E4A@bob>
Message-ID: <CAKzWOxXBT+ZnVVS0e-K9qh2Fc-SPEWpQ962iYZWZDVL5WSb0qA@mail.gmail.com>

Is this device on the same LAN as FreeSWITCH? Get a console log and SIP
trace and drop it on pastebin.freeswitch.org and the gang here will offer
some insights.

-MC

On Fri, Nov 16, 2012 at 10:33 PM, Jason Holden <jason.holden at start.ca>wrote:

>  Hi.****
>
> When accessing voicemail to listen to messages I am finding that it is
> dropping at 30 seconds each time with a message of 100 sleep timer.****
>
> Does anyone have any recommendations?****
>
> I am using a Sipura 3000 connected to my freeswitch server.****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From marketing at cluecon.com  Mon Nov 19 21:13:57 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 19 Nov 2012 10:13:57 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVB-kjZ-s1ESuQfhkQ8QhCULymdZvD9DDNEeHjdh0PZOg@mail.gmail.com>

Happy short week to those of you in North America!

The weekly FreeSWITCH news and notes took a hiatus while I was out on a
medical leave. I am happy to report that I am back to work and recovering
nicely. Many thanks to those who sent their well-wishes and happy thoughts.
We have a great community and I am glad to be a part of it!

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_14>
 we covered some Linux/FreeSWITCH install and configuration tips. A special
thanks to Ken Rice for giving us some practical information on many of the
useful files and utility items that are available in the FreeSWITCH source
tree and how to implement them, including FreeSWITCH init
scripts<http://fisheye.freeswitch.org/browse/freeswitch.git/build/freeswitch.init.redhat?hb=true>,
Anthony's .emacs
file<http://fisheye.freeswitch.org/browse/freeswitch.git/support-d/.emacs?hb=true>,
and even a monit <http://mmonit.com/monit/> configuration
example<http://fisheye.freeswitch.org/browse/freeswitch.git/build/freeswitch.monitrc?hb=true>.
I hope you found these items as useful as I did.

We recently released FreeSWITCH 1.2.4 and Ken Rice tells me that more
updates are in the works. More information will be available on this week's
conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21>.
This week I will be presenting a Wiki how-to: adding a channel
variable page<http://wiki.freeswitch.org/wiki/Variable_skeleton>.
This will be especially useful because it illustrates a number of
Mediawiki<http://www.mediawiki.org/>
 concepts. Also, we have a lot of missing channel variables so if everyone
picks one or two to add we'll be able to expand the wiki coverage.

Have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From msc at freeswitch.org  Mon Nov 19 21:34:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 10:34:35 -0800
Subject: [Freeswitch-users] Pager Duty Service on FreeSWITCH
In-Reply-To: <CAD7Ssm8cppQ1x281w5vME=Y4Wj4BtCo_9hbXBo_4=XtVRdQa1w@mail.gmail.com>
References: <CAD7Ssm_iS83R2Hm9e52nxaNZjN9DwfB0KsWyZRaXzywZP5iWdQ@mail.gmail.com>
	<CAF-ZooSBy1z_mwS1F8a1JNwjqBtq4_8ow2LHNyXmDe2UQtbhtg@mail.gmail.com>
	<CAD7Ssm8cppQ1x281w5vME=Y4Wj4BtCo_9hbXBo_4=XtVRdQa1w@mail.gmail.com>
Message-ID: <CAKzWOxWPeEpb+3h-221FwhtDTENyta=XU7p1L3zrOpERrvPBYw@mail.gmail.com>

Well, nothing pre-programmed. However, if you use the originate command you
can fire off phone calls at will. An example of sending a call from the
Linux command line would be something like this:

fs_cli -x 'originate
{origination_caller_id_number=1234567890}sofia/gateway/my_gateway/18005551212
&playback("/path/to/soundfile.wav")'

There are other ways to do it, like using ESL (event socket library). "Some
assembly required" though, as you'll need to put all the pieces together.

-MC

On Sat, Nov 17, 2012 at 5:13 PM, Kaushal Shriyan
<kaushalshriyan at gmail.com>wrote:

>
>
> On Sat, Nov 17, 2012 at 11:43 PM, William Suffill <
> william.suffill at gmail.com> wrote:
>
>> Not really sure what you are after. If you could clarify what you are
>> trying to do someone might be able to give you better insight on how to get
>> what you are after.
>>
>>
> Hi William
>
> This product is a http://www.pagerduty.com commercial and is there a open
> source product similar to this one and Freeswitch has support for time
> based operations.
>
> Regards
>
> Kaushal
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From gabe at gundy.org  Mon Nov 19 22:07:18 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 19 Nov 2012 12:07:18 -0700
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAKzWOxU+3RkOW1fO26y9gZ9eB8Bk4AEzEhe3=whJx7mTzYVf5g@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
	<CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
	<CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>
	<CAMwZ-t+hkyfuf87b=48MfTMALDSyB7Efa-6BkfkKmMiLrx7+pA@mail.gmail.com>
	<CAKzWOxU+3RkOW1fO26y9gZ9eB8Bk4AEzEhe3=whJx7mTzYVf5g@mail.gmail.com>
Message-ID: <CAMwZ-tJ4ijj18HmSZR4WjcuNtuswHcxERNdcBPw12z+dF7Oh_g@mail.gmail.com>

On Mon, Nov 19, 2012 at 9:28 AM, Michael Collins <msc at freeswitch.org> wrote:
>> Whatever the case, we'll put $1000 toward the effort. This is not a
>> bounty (conditional based on finishing the feature). It's a
>> no-obligation, thanks-for-an-awesome-product,
>> good-luck-with-future-development, contribution.  I hope other
>> companies that benefit daily from FreeSWITCH consider following suit.
>> Let's make this happen :)
>
> +1
>
> Thanks for your support and setting such a great example!


It's our pleasure. I wish we could do more. One day we will be able to! :)

I'm hoping other companies see the value of WebRTC and see what they
can do to support it too.

Gabe


From jeff at jefflenk.com  Mon Nov 19 22:12:18 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 19 Nov 2012 11:12:18 -0800 (PST)
Subject: [Freeswitch-users] [Confidential] - RE: Error on Call Recording
 -	Windows, FS V1.2.4
In-Reply-To: <loom.20121119T165734-788@post.gmane.org>
References: <49C5FCA19A8A114493EBAACA42FE5899105E9041@1AERDCEXCHMBX1.AER.AERCO.local>
	<78990CE7CC964442A7C2CA5F4689695E99BCB82B@BARXB0003.UnifyBusiness.local>
	<loom.20121119T165734-788@post.gmane.org>
Message-ID: <1353352337984-7584717.post@n2.nabble.com>

This is a very common source of problems with windows and fs; with fs being
cross platform with *nix based environments that use the forward slash
exclusively for path separators. The issue involves backslash "\" encoded
escape entries. (ex. \r being an encoded return character hex 0xd or \n an
encoded newline hex 0xa) there are many more examples of this too.

The best way to deal with this in a cross platform way is to use windows
ability to use forward slashes "/" for path separators. All releases of
windows can do this without difficulty(and least for 10 years or so).

All internal path handing in FreeSWITCH has been changed(last few months) to
manipulate paths with forward slashes to minimize this very problem. So if
you supply external paths to fs make sure you always supply them with
forward slashes to avoid this problem.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-on-Call-Recording-Windows-FS-V1-2-4-tp7584705p7584717.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From paul at iamfine.com  Mon Nov 19 22:43:06 2012
From: paul at iamfine.com (Paul)
Date: Mon, 19 Nov 2012 11:43:06 -0800 (PST)
Subject: [Freeswitch-users] Problems with icall carrier in Texas
Message-ID: <1353354186557-7584722.post@n2.nabble.com>

We have a number of DID's with icall that we forward to a FS box.We started
getting reports from our users that they are getting busy signals starting
Sat morning. We have alternate numbers that we handed out to our users
through a different carrier.

This outage is a serious issue, and I cant seem to get anyone there to
discuss it. The main number rings busy. The trouble ticket system cant be
accessed as the account login fails.

I would like to know if this is isolated to us or system wide.

Is anyone else experiencing issues with them or know if they have a status
page.
I have tried to contact them on their main number and get a busy signal.

Thanks in advance






--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problems-with-icall-carrier-in-Texas-tp7584722.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Nov 19 23:05:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 12:05:27 -0800
Subject: [Freeswitch-users] [Confidential] - RE: Error on Call Recording
 - Windows, FS V1.2.4
In-Reply-To: <1353352337984-7584717.post@n2.nabble.com>
References: <49C5FCA19A8A114493EBAACA42FE5899105E9041@1AERDCEXCHMBX1.AER.AERCO.local>
	<78990CE7CC964442A7C2CA5F4689695E99BCB82B@BARXB0003.UnifyBusiness.local>
	<loom.20121119T165734-788@post.gmane.org>
	<1353352337984-7584717.post@n2.nabble.com>
Message-ID: <CAKzWOxU6FL_CD2txfo13pnPc3C7GGuVTYBTPM9aoukHHvDNYHg@mail.gmail.com>

Thanks Jeff!

I codified this in the Windows quickstart section:
http://wiki.freeswitch.org/wiki/Installation_for_Windows#Path_Separator

HtH the Windows users...

-MC

On Mon, Nov 19, 2012 at 11:12 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> This is a very common source of problems with windows and fs; with fs being
> cross platform with *nix based environments that use the forward slash
> exclusively for path separators. The issue involves backslash "\" encoded
> escape entries. (ex. \r being an encoded return character hex 0xd or \n an
> encoded newline hex 0xa) there are many more examples of this too.
>
> The best way to deal with this in a cross platform way is to use windows
> ability to use forward slashes "/" for path separators. All releases of
> windows can do this without difficulty(and least for 10 years or so).
>
> All internal path handing in FreeSWITCH has been changed(last few months)
> to
> manipulate paths with forward slashes to minimize this very problem. So if
> you supply external paths to fs make sure you always supply them with
> forward slashes to avoid this problem.
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Error-on-Call-Recording-Windows-FS-V1-2-4-tp7584705p7584717.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Mon Nov 19 23:20:58 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Nov 2012 14:20:58 -0600
Subject: [Freeswitch-users] Log Rollover issue
In-Reply-To: <CAPDWOejFaGi6OmavH2Q9nKYRzn6zpz1gDbiSVqEcaCV678+bbg@mail.gmail.com>
References: <CAPDWOejFaGi6OmavH2Q9nKYRzn6zpz1gDbiSVqEcaCV678+bbg@mail.gmail.com>
Message-ID: <CAKbxfG8krEgxHPWeWi263ubeiducGnsvPUBJhKs=K-JxtX-9Gg@mail.gmail.com>

When it says it cannot, is that because of file permissions? Dos FS have
permission to write to the directory?
Can you reproduce this with logs and if so can you report it to Jira
http://jira.freeswitch.org



On Mon, Nov 19, 2012 at 6:45 AM, Nageshwara Rao Moova
<mnrao2001 at gmail.com>wrote:

> Hi all,
>
> I have modified my default logconf file for rollover to be restricted to
> 100. But sometimes freeswitch is unable to rename rollover say ?cannot
> rename freeswitch.log.41? and fails. But the fail follows with serious
> issue of writing all the logs to "freeswitch.log" and ends up filling the
> disk space.
>
> I have not change the default file size i.e 10MB.
>
> How does 10MB size overridden by freeswitch?
>
> --
> regards & thanks
> --
> m nageshwara rao
> 99891 86280
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From msc at freeswitch.org  Mon Nov 19 23:33:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 12:33:59 -0800
Subject: [Freeswitch-users] Problems with icall carrier in Texas
In-Reply-To: <1353354186557-7584722.post@n2.nabble.com>
References: <1353354186557-7584722.post@n2.nabble.com>
Message-ID: <CAKzWOxUPNw6O9w06GW-8cMwSC+4zTRL6+v1-b7_bs_Gf5R-FGg@mail.gmail.com>

I think everyone with iCall is experiencing issues. We strongly recommend
you pursue other options ASAP.
-MC

On Mon, Nov 19, 2012 at 11:43 AM, Paul <paul at iamfine.com> wrote:

> We have a number of DID's with icall that we forward to a FS box.We started
> getting reports from our users that they are getting busy signals starting
> Sat morning. We have alternate numbers that we handed out to our users
> through a different carrier.
>
> This outage is a serious issue, and I cant seem to get anyone there to
> discuss it. The main number rings busy. The trouble ticket system cant be
> accessed as the account login fails.
>
> I would like to know if this is isolated to us or system wide.
>
> Is anyone else experiencing issues with them or know if they have a status
> page.
> I have tried to contact them on their main number and get a busy signal.
>
> Thanks in advance
>
>
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Problems-with-icall-carrier-in-Texas-tp7584722.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jason.holden at start.ca  Tue Nov 20 00:32:00 2012
From: jason.holden at start.ca (Jason Holden)
Date: Mon, 19 Nov 2012 16:32:00 -0500
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
Message-ID: <472138AA0F3C4B01A08CBF5CB2CFFB2D@bob>

I can not log on to the page but the following is the cli log.

Also I am on my local LAN.

 

 

2012-11-19 17:16:46.630373 [NOTICE] switch_channel.c:953 New Channel
sofia/internal/201-entros at 192.168.15.9
[1107676f-36a3-4bc2-b413-525c898bb3b6]

2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_NEW

2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/201-entros at 192.168.15.9) State NEW

2012-11-19 17:16:46.650375 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected by
acl "domains". Falling back to Digest auth.

2012-11-19 17:16:46.650375 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.650375 [DEBUG] sofia.c:1755 detaching session
1107676f-36a3-4bc2-b413-525c898bb3b6

2012-11-19 17:16:46.650375 [WARNING] sofia_reg.c:1481 SIP auth challenge
(INVITE) on sofia profile 'internal' for [*97 at 192.168.15.9] from ip
192.168.15.2

2012-11-19 17:16:46.670376 [DEBUG] sofia.c:1847 Re-attaching to session
1107676f-36a3-4bc2-b413-525c898bb3b6

2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected by
acl "domains". Falling back to Digest auth.

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5596 Channel
sofia/internal/201-entros at 192.168.15.9 entering state [received][100]

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5607 Remote SDP:

v=0

o=- 6486898 6486898 IN IP4 192.168.15.2

s=-

c=IN IP4 192.168.15.2

t=0 0

m=audio 16392 RTP/AVP 0 100 101

a=rtpmap:0 PCMU/8000

a=rtpmap:100 NSE/8000

a=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:3093 Set Codec
sofia/internal/201-entros at 192.168.15.9 PCMU/8000 20 ms 160 samples 64000
bits

2012-11-19 17:16:46.690372 [DEBUG] switch_core_codec.c:111
sofia/internal/201-entros at 192.168.15.9 Original read codec set to PCMU:0

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5265 Set 2833 dtmf send/recv
payload to 101

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5824
(sofia/internal/201-entros at 192.168.15.9) State Change CS_NEW -> CS_INIT

2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_INIT

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
(sofia/internal/201-entros at 192.168.15.9) State INIT

2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:86
sofia/internal/201-entros at 192.168.15.9 SOFIA INIT

2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:126
(sofia/internal/201-entros at 192.168.15.9) State Change CS_INIT -> CS_ROUTING

2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
(sofia/internal/201-entros at 192.168.15.9) State INIT going to sleep

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_ROUTING

2012-11-19 17:16:46.690372 [DEBUG] switch_channel.c:1988
(sofia/internal/201-entros at 192.168.15.9) Callstate Change DOWN -> RINGING

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/201-entros at 192.168.15.9) State ROUTING

2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:149
sofia/internal/201-entros at 192.168.15.9 SOFIA ROUTING

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:117
sofia/internal/201-entros at 192.168.15.9 Standard ROUTING

2012-11-19 17:16:46.690372 [INFO] mod_dialplan_xml.c:498 Processing home
<201-entros>->*97 in context default

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.911] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.911] destination_number(*97) =~ /^(911)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.10d] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.10d] destination_number(*97) =~ /^(\d{10})$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.11d] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.11d] destination_number(*97) =~ /^\+?(\d{11})$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.tollfree] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.tollfree] destination_number(*97) =~
/^1?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->call-direction] continue=true

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 ANTI-Action
set(call_direction=local) 

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->group-intercept] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[group-intercept] destination_number(*97) =~ /^\*8$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->redial]
continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [redial]
destination_number(*97) =~ /^(redial|\*870)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->call_privacy] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [call_privacy]
destination_number(*97) =~ /^\*67(\d+)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->call_return] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [call_return]
destination_number(*97) =~ /^\*69$|^lcr$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->intercept-ext] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[intercept-ext] destination_number(*97) =~ /^\*\*(\d+)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->extension-intercom] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[extension-intercom] destination_number(*97) =~ /^\*8(\d{2,7})$/
break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->send_to_voicemail] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[send_to_voicemail] destination_number(*97) =~ /^\*99(\d{2,7})$/
break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->vmain]
continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [vmain]
destination_number(*97) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->vmain_user] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (PASS) [vmain_user]
destination_number(*97) =~ /^\*97$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Action answer() 

Dialplan: sofia/internal/201-entros at 192.168.15.9 Action sleep(1000) 

Dialplan: sofia/internal/201-entros at 192.168.15.9 Action voicemail(check
default ${domain_name} ${caller_id_number}) 

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:167
(sofia/internal/201-entros at 192.168.15.9) State Change CS_ROUTING ->
CS_EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/201-entros at 192.168.15.9) State ROUTING going to sleep

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/201-entros at 192.168.15.9) State EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:242
sofia/internal/201-entros at 192.168.15.9 SOFIA EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:209
sofia/internal/201-entros at 192.168.15.9 Standard EXECUTE

EXECUTE sofia/internal/201-entros at 192.168.15.9 set(call_direction=local)

2012-11-19 17:16:46.710380 [DEBUG] mod_dptools.c:1344
sofia/internal/201-entros at 192.168.15.9 SET [call_direction]=[local]

EXECUTE sofia/internal/201-entros at 192.168.15.9 answer()

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3350 AUDIO RTP
[sofia/internal/201-entros at 192.168.15.9] 192.168.15.9 port 24942 ->
192.168.15.2 port 16392 codec: 0 ms: 20

2012-11-19 17:16:46.710380 [DEBUG] switch_rtp.c:1927 Starting timer [soft]
160 bytes per 20ms

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3614 Set 2833 dtmf send
payload to 101

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3620 Set 2833 dtmf receive
payload to 101

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3647
sofia/internal/201-entros at 192.168.15.9 Set rtp dtmf delay to 40

2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:856 Local SDP
sofia/internal/201-entros at 192.168.15.9:

v=0

o=FreeSWITCH 1353300664 1353300665 IN IP4 192.168.15.9

s=FreeSWITCH

c=IN IP4 192.168.15.9

t=0 0

m=audio 24942 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

a=sendrecv

 

2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.710380 [DEBUG] sofia.c:5596 Channel
sofia/internal/201-entros at 192.168.15.9 entering state [completed][200]

2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:830 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.710380 [DEBUG] switch_channel.c:3380
(sofia/internal/201-entros at 192.168.15.9) Callstate Change RINGING -> ACTIVE

2012-11-19 17:16:46.710380 [NOTICE] mod_dptools.c:1176 Channel
[sofia/internal/201-entros at 192.168.15.9] has been answered

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(1000)

2012-11-19 17:16:46.750379 [DEBUG] switch_rtp.c:3606 Correct ip/port
confirmed.

EXECUTE sofia/internal/201-entros at 192.168.15.9 voicemail(check default
192.168.15.9 201-entros)

2012-11-19 17:16:47.730206 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:47.990165 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:48.010904 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-enter_pass.wav] (en:en)

2012-11-19 17:16:48.030171 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:48.930025 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1604

2012-11-19 17:16:48.930025 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-enter_pass.wav

2012-11-19 17:16:49.289963 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 9:1444

2012-11-19 17:16:49.709899 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 2:1524

2012-11-19 17:16:50.169828 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 7:1524

2012-11-19 17:16:55.151063 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:1604

2012-11-19 17:16:55.271043 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:55.291060 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-you_have.wav] (en:en)

2012-11-19 17:16:55.311050 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:55.869947 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav

2012-11-19 17:16:55.989933 [DEBUG] switch_ivr_play_say.c:244 Handle say:[0]
(en:en)

2012-11-19 17:16:56.009933 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:56.769808 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/0.wav

2012-11-19 17:16:56.889794 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-new.wav] (en:en)

2012-11-19 17:16:56.909794 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:57.269730 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav

2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-messages.wav] (en:en)

2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:58.049611 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav

2012-11-19 17:16:58.170590 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-you_have.wav] (en:en)

2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:58.730505 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav

2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:244 Handle say:[0]
(en:en)

2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:59.630368 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/0.wav

2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-saved.wav] (en:en)

2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:00.230274 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-saved.wav

2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-messages.wav] (en:en)

2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:01.010156 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav

2012-11-19 17:17:01.250116 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-listen_new.wav] (en:en)

2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:02.609908 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav

2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:03.129826 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
(en:en)

2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:03.949702 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/1.wav

2012-11-19 17:17:04.069679 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-listen_saved.wav] (en:en)

2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:05.910397 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_saved.wav

2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:06.430318 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2]
(en:en)

2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:07.190204 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/2.wav

2012-11-19 17:17:07.310180 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-advanced.wav] (en:en)

2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:08.709964 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-advanced.wav

2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:09.229884 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:244 Handle say:[5]
(en:en)

2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:10.129747 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/5.wav

2012-11-19 17:17:10.249724 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-to_exit.wav] (en:en)

2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:11.069598 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_exit.wav

2012-11-19 17:17:11.089603 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1524

2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:11.410578 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-to_record_greeting.wav] (en:en)

2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:12.450387 [DEBUG] switch_ivr_play_say.c:1682 done playing
file
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_record_greeting.wa
v

2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:12.970304 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
(en:en)

2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:13.790181 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/1.wav

2012-11-19 17:17:13.910188 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-choose_greeting.wav] (en:en)

2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:15.089978 [DEBUG] switch_ivr_play_say.c:1682 done playing
file
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-choose_greeting.wav

2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:15.609897 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2]
(en:en)

2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:16.369782 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/2.wav

2012-11-19 17:17:16.489760 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-record_name2.wav] (en:en)

2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:17.770561 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name2.wav

2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:18.270487 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:244 Handle say:[3]
(en:en)

2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.730420 [DEBUG] sofia.c:5596 Channel
sofia/internal/201-entros at 192.168.15.9 entering state [terminating][0]

2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:2979
(sofia/internal/201-entros at 192.168.15.9) Callstate Change ACTIVE -> HANGUP

2012-11-19 17:17:18.730420 [NOTICE] sofia.c:6380 Hangup
sofia/internal/201-entros at 192.168.15.9 [CS_EXECUTE] [NORMAL_UNSPECIFIED]

2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:3002 Send signal
sofia/internal/201-entros at 192.168.15.9 [KILL]

2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.730420 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/3.wav

2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:2685
sofia/internal/201-entros at 192.168.15.9 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/201-entros at 192.168.15.9) State EXECUTE going to sleep

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_HANGUP

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
(sofia/internal/201-entros at 192.168.15.9) State HANGUP

2012-11-19 17:17:18.730420 [DEBUG] mod_sofia.c:503 Channel
sofia/internal/201-entros at 192.168.15.9 hanging up, cause: NORMAL_UNSPECIFIED

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:48
sofia/internal/201-entros at 192.168.15.9 Standard HANGUP, cause:
NORMAL_UNSPECIFIED

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
(sofia/internal/201-entros at 192.168.15.9) State HANGUP going to sleep

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/201-entros at 192.168.15.9) State Change CS_HANGUP ->
CS_REPORTING

2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_REPORTING

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:749
(sofia/internal/201-entros at 192.168.15.9) State REPORTING

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:92
sofia/internal/201-entros at 192.168.15.9 Standard REPORTING, cause:
NORMAL_UNSPECIFIED

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:749
(sofia/internal/201-entros at 192.168.15.9) State REPORTING going to sleep

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:440
(sofia/internal/201-entros at 192.168.15.9) State Change CS_REPORTING ->
CS_DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1492 Session 2
(sofia/internal/201-entros at 192.168.15.9) Locked, Waiting on external
entities

2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1510 Session 2
(sofia/internal/201-entros at 192.168.15.9) Ended

2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1514 Close Channel
sofia/internal/201-entros at 192.168.15.9 [CS_DESTROY]

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:556
(sofia/internal/201-entros at 192.168.15.9) Callstate Change HANGUP -> DOWN

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:559
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/201-entros at 192.168.15.9) State DESTROY

2012-11-19 17:17:19.010389 [DEBUG] mod_sofia.c:396
sofia/internal/201-entros at 192.168.15.9 SOFIA DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:99
sofia/internal/201-entros at 192.168.15.9 Standard DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/201-entros at 192.168.15.9) State DESTROY going to sleep

 

 

  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Monday, November 19, 2012 12:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] retreiving voicemail dropping after 30
seconds

 

Is this device on the same LAN as FreeSWITCH? Get a console log and SIP
trace and drop it on pastebin.freeswitch.org and the gang here will offer
some insights.

-MC

On Fri, Nov 16, 2012 at 10:33 PM, Jason Holden <jason.holden at start.ca>
wrote:

Hi.

When accessing voicemail to listen to messages I am finding that it is
dropping at 30 seconds each time with a message of 100 sleep timer.

Does anyone have any recommendations?

I am using a Sipura 3000 connected to my freeswitch server.

 


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




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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

 

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From blee at gocentrix.com  Tue Nov 20 01:02:37 2012
From: blee at gocentrix.com (Bryant Lee)
Date: Mon, 19 Nov 2012 17:02:37 -0500
Subject: [Freeswitch-users] Problems with icall carrier in Texas
In-Reply-To: <1353354186557-7584722.post@n2.nabble.com>
References: <1353354186557-7584722.post@n2.nabble.com>
Message-ID: <CAGodFm_Q=EJMrnz2TsGYKuiYvkRN2qB=kzYEdKf10Lf0XUXmkA@mail.gmail.com>

Paul,

Once upon a time we had iCall but we were forced to move off because of
things like this.  I also got good at finding other numbers that for some
reason worked when their main number didnt.  Give this a shot: 214-377-3111

Best regards,




On Mon, Nov 19, 2012 at 2:43 PM, Paul <paul at iamfine.com> wrote:

> We have a number of DID's with icall that we forward to a FS box.We started
> getting reports from our users that they are getting busy signals starting
> Sat morning. We have alternate numbers that we handed out to our users
> through a different carrier.
>
> This outage is a serious issue, and I cant seem to get anyone there to
> discuss it. The main number rings busy. The trouble ticket system cant be
> accessed as the account login fails.
>
> I would like to know if this is isolated to us or system wide.
>
> Is anyone else experiencing issues with them or know if they have a status
> page.
> I have tried to contact them on their main number and get a busy signal.
>
> Thanks in advance
>
>
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Problems-with-icall-carrier-in-Texas-tp7584722.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Tue Nov 20 01:11:04 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 19 Nov 2012 16:11:04 -0600
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
In-Reply-To: <472138AA0F3C4B01A08CBF5CB2CFFB2D@bob>
Message-ID: <CCD00A98.60D0D%krice@freeswitch.org>

You sir, failed the bot test... It tells you right there in the login prompt
what the login info is


On 11/19/12 3:32 PM, "Jason Holden" <jason.holden at start.ca> wrote:

> I can not log on to the page but the following is the cli log.
> Also I am on my local LAN.
>  
>  
> 2012-11-19 17:16:46.630373 [NOTICE] switch_channel.c:953 New Channel
> sofia/internal/201-entros at 192.168.15.9 [1107676f-36a3-4bc2-b413-525c898bb3b6]
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_NEW
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:433
> (sofia/internal/201-entros at 192.168.15.9) State NEW
> 2012-11-19 17:16:46.650375 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected by
> acl "domains". Falling back to Digest auth.
> 2012-11-19 17:16:46.650375 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.650375 [DEBUG] sofia.c:1755 detaching session
> 1107676f-36a3-4bc2-b413-525c898bb3b6
> 2012-11-19 17:16:46.650375 [WARNING] sofia_reg.c:1481 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [*97 at 192.168.15.9] from ip
> 192.168.15.2
> 2012-11-19 17:16:46.670376 [DEBUG] sofia.c:1847 Re-attaching to session
> 1107676f-36a3-4bc2-b413-525c898bb3b6
> 2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected by
> acl "domains". Falling back to Digest auth.
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5596 Channel
> sofia/internal/201-entros at 192.168.15.9 entering state [received][100]
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5607 Remote SDP:
> v=0
> o=- 6486898 6486898 IN IP4 192.168.15.2
> s=-
> c=IN IP4 192.168.15.2
> t=0 0
> m=audio 16392 RTP/AVP 0 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:100 NSE/8000
> a=fmtp:100 192-193
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>  
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:3093 Set Codec
> sofia/internal/201-entros at 192.168.15.9 PCMU/8000 20 ms 160 samples 64000 bits
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_codec.c:111
> sofia/internal/201-entros at 192.168.15.9 Original read codec set to PCMU:0
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5265 Set 2833 dtmf send/recv
> payload to 101
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5824
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_NEW -> CS_INIT
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_INIT
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
> (sofia/internal/201-entros at 192.168.15.9) State INIT
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:86
> sofia/internal/201-entros at 192.168.15.9 SOFIA INIT
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:126
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_INIT -> CS_ROUTING
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
> (sofia/internal/201-entros at 192.168.15.9) State INIT going to sleep
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_ROUTING
> 2012-11-19 17:16:46.690372 [DEBUG] switch_channel.c:1988
> (sofia/internal/201-entros at 192.168.15.9) Callstate Change DOWN -> RINGING
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:470
> (sofia/internal/201-entros at 192.168.15.9) State ROUTING
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:149
> sofia/internal/201-entros at 192.168.15.9 SOFIA ROUTING
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:117
> sofia/internal/201-entros at 192.168.15.9 Standard ROUTING
> 2012-11-19 17:16:46.690372 [INFO] mod_dialplan_xml.c:498 Processing home
> <201-entros>->*97 in context default
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.911] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [AtlasVoice.911]
> destination_number(*97) =~ /^(911)$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.10d] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [AtlasVoice.10d]
> destination_number(*97) =~ /^(\d{10})$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.11d] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [AtlasVoice.11d]
> destination_number(*97) =~ /^\+?(\d{11})$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.tollfree] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.tollfree] destination_number(*97) =~
> /^1?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->call-direction] continue=true
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [call-direction]
> ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 ANTI-Action
> set(call_direction=local)
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->group-intercept] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [group-intercept] destination_number(*97) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->redial]
> continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [redial]
> destination_number(*97) =~ /^(redial|\*870)$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->call_privacy] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [call_privacy]
> destination_number(*97) =~ /^\*67(\d+)$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->call_return] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [call_return]
> destination_number(*97) =~ /^\*69$|^lcr$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->intercept-ext] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [intercept-ext]
> destination_number(*97) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->extension-intercom] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [extension-intercom] destination_number(*97) =~ /^\*8(\d{2,7})$/
> break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->send_to_voicemail] continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [send_to_voicemail] destination_number(*97) =~ /^\*99(\d{2,7})$/
> break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->vmain]
> continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [vmain]
> destination_number(*97) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->vmain_user]
> continue=false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (PASS) [vmain_user]
> destination_number(*97) =~ /^\*97$/ break=on-false
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Action answer()
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Action sleep(1000)
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Action voicemail(check
> default ${domain_name} ${caller_id_number})
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:167
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_ROUTING -> CS_EXECUTE
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:470
> (sofia/internal/201-entros at 192.168.15.9) State ROUTING going to sleep
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_EXECUTE
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:477
> (sofia/internal/201-entros at 192.168.15.9) State EXECUTE
> 2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:242
> sofia/internal/201-entros at 192.168.15.9 SOFIA EXECUTE
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:209
> sofia/internal/201-entros at 192.168.15.9 Standard EXECUTE
> EXECUTE sofia/internal/201-entros at 192.168.15.9 set(call_direction=local)
> 2012-11-19 17:16:46.710380 [DEBUG] mod_dptools.c:1344
> sofia/internal/201-entros at 192.168.15.9 SET [call_direction]=[local]
> EXECUTE sofia/internal/201-entros at 192.168.15.9 answer()
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3350 AUDIO RTP
> [sofia/internal/201-entros at 192.168.15.9] 192.168.15.9 port 24942 ->
> 192.168.15.2 port 16392 codec: 0 ms: 20
> 2012-11-19 17:16:46.710380 [DEBUG] switch_rtp.c:1927 Starting timer [soft] 160
> bytes per 20ms
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3614 Set 2833 dtmf send
> payload to 101
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3620 Set 2833 dtmf receive
> payload to 101
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3647
> sofia/internal/201-entros at 192.168.15.9 Set rtp dtmf delay to 40
> 2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:856 Local SDP
> sofia/internal/201-entros at 192.168.15.9:
> v=0
> o=FreeSWITCH 1353300664 1353300665 IN IP4 192.168.15.9
> s=FreeSWITCH
> c=IN IP4 192.168.15.9
> t=0 0
> m=audio 24942 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
> a=sendrecv
>  
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.710380 [DEBUG] sofia.c:5596 Channel
> sofia/internal/201-entros at 192.168.15.9 entering state [completed][200]
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:830 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:16:46.710380 [DEBUG] switch_channel.c:3380
> (sofia/internal/201-entros at 192.168.15.9) Callstate Change RINGING -> ACTIVE
> 2012-11-19 17:16:46.710380 [NOTICE] mod_dptools.c:1176 Channel
> [sofia/internal/201-entros at 192.168.15.9] has been answered
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(1000)
> 2012-11-19 17:16:46.750379 [DEBUG] switch_rtp.c:3606 Correct ip/port
> confirmed.
> EXECUTE sofia/internal/201-entros at 192.168.15.9 voicemail(check default
> 192.168.15.9 201-entros)
> 2012-11-19 17:16:47.730206 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-19 17:16:47.990165 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-19 17:16:48.010904 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-enter_pass.wav] (en:en)
> 2012-11-19 17:16:48.030171 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:16:48.930025 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1604
> 2012-11-19 17:16:48.930025 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-enter_pass.wav
> 2012-11-19 17:16:49.289963 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 9:1444
> 2012-11-19 17:16:49.709899 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 2:1524
> 2012-11-19 17:16:50.169828 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 7:1524
> 2012-11-19 17:16:55.151063 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:1604
> 2012-11-19 17:16:55.271043 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-19 17:16:55.291060 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-you_have.wav] (en:en)
> 2012-11-19 17:16:55.311050 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:16:55.869947 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav
> 2012-11-19 17:16:55.989933 [DEBUG] switch_ivr_play_say.c:244 Handle say:[0]
> (en:en)
> 2012-11-19 17:16:56.009933 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:16:56.769808 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/0.wav
> 2012-11-19 17:16:56.889794 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-new.wav] (en:en)
> 2012-11-19 17:16:56.909794 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:16:57.269730 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav
> 2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-messages.wav] (en:en)
> 2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:16:58.049611 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav
> 2012-11-19 17:16:58.170590 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-you_have.wav] (en:en)
> 2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:16:58.730505 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav
> 2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:244 Handle say:[0]
> (en:en)
> 2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:16:59.630368 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/0.wav
> 2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-saved.wav] (en:en)
> 2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:00.230274 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-saved.wav
> 2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-messages.wav] (en:en)
> 2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:01.010156 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav
> 2012-11-19 17:17:01.250116 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-listen_new.wav] (en:en)
> 2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:02.609908 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav
> 2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:03.129826 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
> 2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
> (en:en)
> 2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:03.949702 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/1.wav
> 2012-11-19 17:17:04.069679 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)
> 2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-listen_saved.wav] (en:en)
> 2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:05.910397 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_saved.wav
> 2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:06.430318 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
> 2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2]
> (en:en)
> 2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:07.190204 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/2.wav
> 2012-11-19 17:17:07.310180 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)
> 2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-advanced.wav] (en:en)
> 2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:08.709964 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-advanced.wav
> 2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:09.229884 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
> 2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:244 Handle say:[5]
> (en:en)
> 2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:10.129747 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/5.wav
> 2012-11-19 17:17:10.249724 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)
> 2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-to_exit.wav] (en:en)
> 2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:11.069598 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_exit.wav
> 2012-11-19 17:17:11.089603 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1524
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
> 2012-11-19 17:17:11.410578 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-to_record_greeting.wav] (en:en)
> 2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:12.450387 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file 
> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_record_greeting.wav
> 2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:12.970304 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
> 2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
> (en:en)
> 2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:13.790181 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/1.wav
> 2012-11-19 17:17:13.910188 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)
> 2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-choose_greeting.wav] (en:en)
> 2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:15.089978 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file 
> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-choose_greeting.wav
> 2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:15.609897 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
> 2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2]
> (en:en)
> 2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:16.369782 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/2.wav
> 2012-11-19 17:17:16.489760 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)
> 2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-record_name2.wav] (en:en)
> 2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:17.770561 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name2.wav
> 2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:18.270487 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
> 2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:244 Handle say:[3]
> (en:en)
> 2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated
> L16 at 8000hz 1 channels 20ms
> 2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:17:18.730420 [DEBUG] sofia.c:5596 Channel
> sofia/internal/201-entros at 192.168.15.9 entering state [terminating][0]
> 2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:2979
> (sofia/internal/201-entros at 192.168.15.9) Callstate Change ACTIVE -> HANGUP
> 2012-11-19 17:17:18.730420 [NOTICE] sofia.c:6380 Hangup
> sofia/internal/201-entros at 192.168.15.9 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:3002 Send signal
> sofia/internal/201-entros at 192.168.15.9 [KILL]
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:17:18.730420 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/3.wav
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:2685
> sofia/internal/201-entros at 192.168.15.9 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:477
> (sofia/internal/201-entros at 192.168.15.9) State EXECUTE going to sleep
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_HANGUP
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
> (sofia/internal/201-entros at 192.168.15.9) State HANGUP
> 2012-11-19 17:17:18.730420 [DEBUG] mod_sofia.c:503 Channel
> sofia/internal/201-entros at 192.168.15.9 hanging up, cause: NORMAL_UNSPECIFIED
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:48
> sofia/internal/201-entros at 192.168.15.9 Standard HANGUP, cause:
> NORMAL_UNSPECIFIED
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
> (sofia/internal/201-entros at 192.168.15.9) State HANGUP going to sleep
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_HANGUP ->
> CS_REPORTING
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_REPORTING
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:749
> (sofia/internal/201-entros at 192.168.15.9) State REPORTING
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:92
> sofia/internal/201-entros at 192.168.15.9 Standard REPORTING, cause:
> NORMAL_UNSPECIFIED
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:749
> (sofia/internal/201-entros at 192.168.15.9) State REPORTING going to sleep
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:440
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_REPORTING ->
> CS_DESTROY
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1492 Session 2
> (sofia/internal/201-entros at 192.168.15.9) Locked, Waiting on external entities
> 2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1510 Session 2
> (sofia/internal/201-entros at 192.168.15.9) Ended
> 2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1514 Close Channel
> sofia/internal/201-entros at 192.168.15.9 [CS_DESTROY]
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:556
> (sofia/internal/201-entros at 192.168.15.9) Callstate Change HANGUP -> DOWN
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_DESTROY
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/201-entros at 192.168.15.9) State DESTROY
> 2012-11-19 17:17:19.010389 [DEBUG] mod_sofia.c:396
> sofia/internal/201-entros at 192.168.15.9 SOFIA DESTROY
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:99
> sofia/internal/201-entros at 192.168.15.9 Standard DESTROY
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/201-entros at 192.168.15.9) State DESTROY going to sleep
>  
>  
> 
> 
> From: Michael Collins [mailto:msc at freeswitch.org]
> Sent: Monday, November 19, 2012 12:13 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] retreiving voicemail dropping after 30 seconds
>  
> Is this device on the same LAN as FreeSWITCH? Get a console log and SIP trace
> and drop it on pastebin.freeswitch.org <http://pastebin.freeswitch.org> and
> the gang here will offer some insights.
> 
> -MC
> On Fri, Nov 16, 2012 at 10:33 PM, Jason Holden <jason.holden at start.ca> wrote:
> Hi.
> When accessing voicemail to listen to messages I am finding that it is
> dropping at 30 seconds each time with a message of 100 sleep timer.
> Does anyone have any recommendations?
> I am using a Sipura 3000 connected to my freeswitch server.
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch

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From brian at freeswitch.org  Tue Nov 20 01:19:09 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Nov 2012 16:19:09 -0600
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
	contact IP/port
In-Reply-To: <054001cdc664$92486660$b6d93320$@com>
References: <054001cdc664$92486660$b6d93320$@com>
Message-ID: <DB46FE12-EB6B-47BD-B516-5EB41994CB87@freeswitch.org>

Are you talking about the inbound SDP from the broken endpoint behind nat that has little to no clue that its behind nat?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 19, 2012, at 8:45 AM, Phil Quesinberry <philq at qsystemsengineering.com> wrote:

> Is there a way to get FreeSWITCH to rewrite the media address as well as the contact header info coming from connected endpoints?  With the proper settings, FS does a great job of taking care of just about any NAT-related issue on the SIP side, but going through the documentation I?ve been unable to find a way to get it to touch the SDP.  Is there a setting to get it to use the same address for media?
> 
> 



From msc at freeswitch.org  Tue Nov 20 01:38:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 14:38:43 -0800
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
In-Reply-To: <472138AA0F3C4B01A08CBF5CB2CFFB2D@bob>
References: <472138AA0F3C4B01A08CBF5CB2CFFB2D@bob>
Message-ID: <CAKzWOxVk-tx7NEX2=3JyhP8w-9XqmcpLNL69vs6N2n8bsRr0uA@mail.gmail.com>

On Mon, Nov 19, 2012 at 1:32 PM, Jason Holden <jason.holden at start.ca> wrote:

> ** **
>
> I can not log on to the page but the following is the cli log.
>
Sure you can! Just read the challenge dialog a bit more closely. ;)
Also, select "FreeSWITCH Log" as the syntax highlighting. Don't forget to
turn on SIP trace:
sofia profile internal siptrace on

-MC


> ****
>
> Also I am on my local LAN.****
>
> ** **
>
> ** **
>
> 2012-11-19 17:16:46.630373 [NOTICE] switch_channel.c:953 New Channel
> sofia/internal/201-entros at 192.168.15.9[1107676f-36a3-4bc2-b413-525c898bb3b6]
> ****
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_NEW****
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:433
> (sofia/internal/201-entros at 192.168.15.9) State NEW****
>
> 2012-11-19 17:16:46.650375 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected
> by acl "domains". Falling back to Digest auth.****
>
> 2012-11-19 17:16:46.650375 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.650375 [DEBUG] sofia.c:1755 detaching session
> 1107676f-36a3-4bc2-b413-525c898bb3b6****
>
> 2012-11-19 17:16:46.650375 [WARNING] sofia_reg.c:1481 SIP auth challenge
> (INVITE) on ****sofia**** profile 'internal' for [*97 at 192.168.15.9] from
> ip 192.168.15.2****
>
> 2012-11-19 17:16:46.670376 [DEBUG] sofia.c:1847 Re-attaching to session
> 1107676f-36a3-4bc2-b413-525c898bb3b6****
>
> 2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected
> by acl "domains". Falling back to Digest auth.****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5596 Channel sofia/internal/
> 201-entros at 192.168.15.9 entering state [received][100]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5607 Remote SDP:****
>
> v=0****
>
> o=- 6486898 6486898 IN IP4 192.168.15.2****
>
> s=-****
>
> c=IN IP4 192.168.15.2****
>
> t=0 0****
>
> m=audio 16392 RTP/AVP 0 100 101****
>
> a=rtpmap:0 PCMU/8000****
>
> a=rtpmap:100 NSE/8000****
>
> a=fmtp:100 192-193****
>
> a=rtpmap:101 telephone-event/8000****
>
> a=fmtp:101 0-15****
>
> a=ptime:20****
>
> ** **
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:3093 Set Codec
> sofia/internal/201-entros at 192.168.15.9 PCMU/8000 20 ms 160 samples 64000
> bits****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_codec.c:111 sofia/internal/
> 201-entros at 192.168.15.9 Original read codec set to PCMU:0****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5265 Set 2833 dtmf
> send/recv payload to 101****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5824 (sofia/internal/
> 201-entros at 192.168.15.9) State Change CS_NEW -> CS_INIT****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_INIT****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
> (sofia/internal/201-entros at 192.168.15.9) State INIT****
>
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:86 sofia/internal/
> 201-entros at 192.168.15.9 ****SOFIA**** INIT****
>
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:126 (sofia/internal/
> 201-entros at 192.168.15.9) State Change CS_INIT -> CS_ROUTING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
> (sofia/internal/201-entros at 192.168.15.9) State INIT going to sleep****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_ROUTING**
> **
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_channel.c:1988 (sofia/internal/
> 201-entros at 192.168.15.9) Callstate Change DOWN -> RINGING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:470
> (sofia/internal/201-entros at 192.168.15.9) State ROUTING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:149 sofia/internal/
> 201-entros at 192.168.15.9 ****SOFIA**** ROUTING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:117
> sofia/internal/201-entros at 192.168.15.9 Standard ROUTING****
>
> 2012-11-19 17:16:46.690372 [INFO] mod_dialplan_xml.c:498 Processing home
> <201-entros>->*97 in context default****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.911] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.911] destination_number(*97) =~ /^(911)$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.10d] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.10d] destination_number(*97) =~ /^(\d{10})$/ break=on-false***
> *
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.11d] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.11d] destination_number(*97) =~ /^\+?(\d{11})$/ break=on-false
> ****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.tollfree] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.tollfree] destination_number(*97) =~
> /^1?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->call-direction] continue=true****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
> break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 ANTI-Action
> set(call_direction=local) ****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->group-intercept] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [group-intercept] destination_number(*97) =~ /^\*8$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->redial] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [redial]
> destination_number(*97) =~ /^(redial|\*870)$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->call_privacy] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [call_privacy] destination_number(*97) =~ /^\*67(\d+)$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->call_return] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [call_return] destination_number(*97) =~ /^\*69$|^lcr$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->intercept-ext] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [intercept-ext] destination_number(*97) =~ /^\*\*(\d+)$/ break=on-false***
> *
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->extension-intercom] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [extension-intercom] destination_number(*97) =~ /^\*8(\d{2,7})$/
> break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->send_to_voicemail] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [send_to_voicemail] destination_number(*97) =~ /^\*99(\d{2,7})$/
> break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->vmain]
> continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [vmain]
> destination_number(*97) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
> [default->vmain_user] continue=false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (PASS)
> [vmain_user] destination_number(*97) =~ /^\*97$/ break=on-false****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Action answer() ****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Action sleep(1000) ****
>
> Dialplan: sofia/internal/201-entros at 192.168.15.9 Action voicemail(check
> default ${domain_name} ${caller_id_number}) ****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:167
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_ROUTING ->
> CS_EXECUTE****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:470
> (sofia/internal/201-entros at 192.168.15.9) State ROUTING going to sleep****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_EXECUTE**
> **
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:477
> (sofia/internal/201-entros at 192.168.15.9) State EXECUTE****
>
> 2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:242 sofia/internal/
> 201-entros at 192.168.15.9 SOFIA EXECUTE****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:209
> sofia/internal/201-entros at 192.168.15.9 Standard EXECUTE****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 set(call_direction=local)**
> **
>
> 2012-11-19 17:16:46.710380 [DEBUG] mod_dptools.c:1344 sofia/internal/
> 201-entros at 192.168.15.9 SET [call_direction]=[local]****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 answer()****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3350 AUDIO RTP
> [sofia/internal/201-entros at 192.168.15.9] 192.168.15.9 port 24942 ->
> 192.168.15.2 port 16392 codec: 0 ms: 20****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_rtp.c:1927 Starting timer [soft]
> 160 bytes per 20ms****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3614 Set 2833 dtmf send
> payload to 101****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3620 Set 2833 dtmf receive
> payload to 101****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3647 sofia/internal/
> 201-entros at 192.168.15.9 Set rtp dtmf delay to 40****
>
> 2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:856 Local SDP
> sofia/internal/201-entros at 192.168.15.9:****
>
> v=0****
>
> o=FreeSWITCH 1353300664 1353300665 IN IP4 192.168.15.9****
>
> s=FreeSWITCH****
>
> c=IN IP4 192.168.15.9****
>
> t=0 0****
>
> m=audio 24942 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****
>
> a=sendrecv****
>
> ** **
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia.c:5596 Channel sofia/internal/
> 201-entros at 192.168.15.9 entering state [completed][200]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:830 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_channel.c:3380 (sofia/internal/
> 201-entros at 192.168.15.9) Callstate Change RINGING -> ACTIVE****
>
> 2012-11-19 17:16:46.710380 [NOTICE] mod_dptools.c:1176 Channel
> [sofia/internal/201-entros at 192.168.15.9] has been answered****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(1000)****
>
> 2012-11-19 17:16:46.750379 [DEBUG] switch_rtp.c:3606 Correct ip/port
> confirmed.****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 voicemail(check default
> 192.168.15.9 201-entros)****
>
> 2012-11-19 17:16:47.730206 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:47.990165 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:48.010904 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-enter_pass.wav] (en:en)****
>
> 2012-11-19 17:16:48.030171 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:48.930025 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1604*
> ***
>
> 2012-11-19 17:16:48.930025 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-enter_pass.wav
> ****
>
> 2012-11-19 17:16:49.289963 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 9:1444*
> ***
>
> 2012-11-19 17:16:49.709899 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 2:1524*
> ***
>
> 2012-11-19 17:16:50.169828 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 7:1524*
> ***
>
> 2012-11-19 17:16:55.151063 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:1604*
> ***
>
> 2012-11-19 17:16:55.271043 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:55.291060 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-you_have.wav] (en:en)****
>
> 2012-11-19 17:16:55.311050 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:55.869947 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav**
> **
>
> 2012-11-19 17:16:55.989933 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[0] (en:en)****
>
> 2012-11-19 17:16:56.009933 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:56.769808 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/0.wav****
>
> 2012-11-19 17:16:56.889794 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-new.wav] (en:en)****
>
> 2012-11-19 17:16:56.909794 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:57.269730 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav****
>
> 2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-messages.wav] (en:en)****
>
> 2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:58.049611 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav**
> **
>
> 2012-11-19 17:16:58.170590 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-you_have.wav] (en:en)****
>
> 2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:58.730505 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav**
> **
>
> 2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[0] (en:en)****
>
> 2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:59.630368 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/0.wav****
>
> 2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-saved.wav] (en:en)****
>
> 2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:00.230274 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-saved.wav****
>
> 2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-messages.wav] (en:en)****
>
> 2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:01.010156 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav**
> **
>
> 2012-11-19 17:17:01.250116 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-listen_new.wav] (en:en)****
>
> 2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:02.609908 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav
> ****
>
> 2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:03.129826 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[1] (en:en)****
>
> 2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:03.949702 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/1.wav****
>
> 2012-11-19 17:17:04.069679 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-listen_saved.wav] (en:en)****
>
> 2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:05.910397 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_saved.wav
> ****
>
> 2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:06.430318 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[2] (en:en)****
>
> 2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:07.190204 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/2.wav****
>
> 2012-11-19 17:17:07.310180 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-advanced.wav] (en:en)****
>
> 2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:08.709964 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-advanced.wav**
> **
>
> 2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:09.229884 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[5] (en:en)****
>
> 2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:10.129747 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/5.wav****
>
> 2012-11-19 17:17:10.249724 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-to_exit.wav] (en:en)****
>
> 2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:11.069598 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_exit.wav***
> *
>
> 2012-11-19 17:17:11.089603 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1524*
> ***
>
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:11.410578 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-to_record_greeting.wav] (en:en)****
>
> 2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:12.450387 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file
> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_record_greeting.wav
> ****
>
> 2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:12.970304 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[1] (en:en)****
>
> 2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:13.790181 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/1.wav****
>
> 2012-11-19 17:17:13.910188 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-choose_greeting.wav] (en:en)****
>
> 2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:15.089978 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file
> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-choose_greeting.wav
> ****
>
> 2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:15.609897 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[2] (en:en)****
>
> 2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:16.369782 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/2.wav****
>
> 2012-11-19 17:17:16.489760 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-record_name2.wav] (en:en)****
>
> 2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:17.770561 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name2.wav
> ****
>
> 2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:18.270487 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[3] (en:en)****
>
> 2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] sofia.c:5596 Channel sofia/internal/
> 201-entros at 192.168.15.9 entering state [terminating][0]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:2979 (sofia/internal/
> 201-entros at 192.168.15.9) Callstate Change ACTIVE -> HANGUP****
>
> 2012-11-19 17:17:18.730420 [NOTICE] sofia.c:6380 Hangup sofia/internal/
> 201-entros at 192.168.15.9 [CS_EXECUTE] [NORMAL_UNSPECIFIED]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:3002 Send signal
> sofia/internal/201-entros at 192.168.15.9 [KILL]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/3.wav****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:2685
> sofia/internal/201-entros at 192.168.15.9 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:477
> (sofia/internal/201-entros at 192.168.15.9) State EXECUTE going to sleep****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_HANGUP***
> *
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
> (sofia/internal/201-entros at 192.168.15.9) State HANGUP****
>
> 2012-11-19 17:17:18.730420 [DEBUG] mod_sofia.c:503 Channel sofia/internal/
> 201-entros at 192.168.15.9 hanging up, cause: NORMAL_UNSPECIFIED****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:48
> sofia/internal/201-entros at 192.168.15.9 Standard HANGUP, cause:
> NORMAL_UNSPECIFIED****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
> (sofia/internal/201-entros at 192.168.15.9) State HANGUP going to sleep****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:446
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_HANGUP ->
> CS_REPORTING****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_REPORTING
> ****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:749
> (sofia/internal/201-entros at 192.168.15.9) State REPORTING****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:92
> sofia/internal/201-entros at 192.168.15.9 Standard REPORTING, cause:
> NORMAL_UNSPECIFIED****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:749
> (sofia/internal/201-entros at 192.168.15.9) State REPORTING going to sleep***
> *
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:440
> (sofia/internal/201-entros at 192.168.15.9) State Change CS_REPORTING ->
> CS_DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1287 Send signal
> sofia/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1492 Session 2
> (sofia/internal/201-entros at 192.168.15.9) Locked, Waiting on external
> entities****
>
> 2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1510 Session 2
> (sofia/internal/201-entros at 192.168.15.9) Ended****
>
> 2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1514 Close
> Channel sofia/internal/201-entros at 192.168.15.9 [CS_DESTROY]****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:556
> (sofia/internal/201-entros at 192.168.15.9) Callstate Change HANGUP -> DOWN**
> **
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/201-entros at 192.168.15.9) Running State Change CS_DESTROY**
> **
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/201-entros at 192.168.15.9) State DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] mod_sofia.c:396 sofia/internal/
> 201-entros at 192.168.15.9 SOFIA DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:99
> sofia/internal/201-entros at 192.168.15.9 Standard DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/201-entros at 192.168.15.9) State DESTROY going to sleep****
>
> ** **
>
> ** **
>  ------------------------------
>
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Monday, November 19, 2012 12:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] retreiving voicemail dropping after 30
> seconds****
>
> ** **
>
> Is this device on the same LAN as FreeSWITCH? Get a console log and SIP
> trace and drop it on pastebin.freeswitch.org and the gang here will offer
> some insights.
>
> -MC****
>
> On Fri, Nov 16, 2012 at 10:33 PM, Jason Holden <jason.holden at start.ca>
> wrote:****
>
> Hi.****
>
> When accessing voicemail to listen to messages I am finding that it is
> dropping at 30 seconds each time with a message of 100 sleep timer.****
>
> Does anyone have any recommendations?****
>
> I am using a Sipura 3000 connected to my freeswitch server.****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Tue Nov 20 02:36:04 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 19 Nov 2012 23:36:04 +0000
Subject: [Freeswitch-users] G729 File version of $${us-ring}
In-Reply-To: <804D48104511D4468F0D60DF9D3100350ADA59B6@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AD9A4EC@MAIL.millicorp.com>
	<CALn233qk8dMRD2hmNdn-f5_R5GJdBxyrQDycV2JJqaOO3Mmktg@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350ADA59B6@MAIL.millicorp.com>
Message-ID: <CAFiqYu=j1Vzny3=VgkpxTv9d8kVL2DVSV2grA3jjXwptYN5XMQ@mail.gmail.com>

You can create your own using fs_encode from a .wav recording, but it does
need a single mod_com_g729 license.

http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding

-Steve



On 19 November 2012 13:14, Tim Meade <Tim.Meade at millicorp.com> wrote:

> Thanks Mitch.  On today's (or tomorrow) todo list.   Was just wondering if
> someone had a nice clean one out there.
>
> Tim
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
> Sent: Sunday, November 18, 2012 9:31 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] G729 File version of $${us-ring}
>
> As long as you have a single G729 license just create it yourself us_ring
> is just the tone stream used so create a g729 call, record it, and play the
> tone stream.
>
> ~mitch
>
> On Sat, Nov 17, 2012 at 8:25 AM, Tim Meade <Tim.Meade at millicorp.com>
> wrote:
> >
> >
> > I was wondering if anyone had a G729 file of the $${us-ring}
> >
> >
> >
> > We want to use it for playing the ringback with G729 pass through.
> >
> >
> >
> > <action application="set"  data="ringback=SomeRingBackFile.G729"/>
> >
> >
> >
> >
> >
> > Thanks
> >
> >
> >
> > Tim
> >
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Tim.Meade at Millicorp.com  Tue Nov 20 02:36:45 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Mon, 19 Nov 2012 23:36:45 +0000
Subject: [Freeswitch-users] xml_curl requests multiple times per call
Message-ID: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>

Is there any way or method to have multiple xml_curl calls per call?

Something like

Call comes in
xml_curl gets dialplan xml
At some point in the dialplan xml something triggers a second call to xml_curl with current dial plan parameters.


Hope that makes some kind of sense.

Thanks

Tim
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From steveayre at gmail.com  Tue Nov 20 02:45:54 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 19 Nov 2012 23:45:54 +0000
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
 contact IP/port
In-Reply-To: <054001cdc664$92486660$b6d93320$@com>
References: <054001cdc664$92486660$b6d93320$@com>
Message-ID: <CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>

What're you trying to achieve?

The FS NAT handling should take care of clients that send the wrong
(internal) IP in the SDP.

FS will return its public IP in the returned SDP, they then send from their
RTP port to the FS port. FS will see the IP/port the packets are arriving
from (the external side of their NAT) and auto-adjust to sending packets to
there. You'll see a auto-adjusting rtp log message when that happens. That
can't happen until they send media though, which can cause a problem during
call setup, eg for ringback.

FS behind NAT would be slightly different, there are multiple options there
including the ext-rtp-ip param.

For rewriting packets it's worth bearing in mind that FS is a B2BUA, not a
SIP proxy.

Rewriting the SDP is possible from the dialplan, but I have no experience
in that. http://wiki.freeswitch.org/wiki/Codec_negotiation#Rewriting_SDP

-Steve



On 19 November 2012 14:45, Phil Quesinberry
<philq at qsystemsengineering.com>wrote:

> **
>
> Is there a way to get FreeSWITCH to rewrite the media address as well as
> the contact header info coming from connected endpoints?  With the proper
> settings, FS does a great job of taking care of just about any
> NAT-related issue on the SIP side, but going through the documentation I?ve
> been unable to find a way to get it to touch the SDP.  Is there a setting
> to get it to use the same address for media?
>
> Thanks,
>
> *******Phil Quesinberry*
>
> Q Systems Engineering, Inc.
>
> Embedded Systems Development and VoIP Business Telephone Hosting
>
> *******Improve your business telephone services and save money*
>
> (410) 969-8002
>
> *****http://www.qsystemsengineering.com*<http://www.qsystemsengineering.com/>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Nov 20 03:24:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Nov 2012 16:24:09 -0800
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
Message-ID: <CAKzWOxUg2-hZbJpWpsGs-MObMSWvJxPXMuQMFEUi151X5iDftg@mail.gmail.com>

It does make sense, but I don't know that you can use xml_curl for that.
However, it sounds suspiciously similar to how mod_httapi works. Have you
ever looked at that?

-MC

On Mon, Nov 19, 2012 at 3:36 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:

>  Is there any way or method to have multiple xml_curl calls per call?
>
> ****
>
> Something like****
>
> ** **
>
> Call comes in****
>
> xml_curl gets dialplan xml****
>
> At some point in the dialplan xml something triggers a second call to
> xml_curl with current dial plan parameters.****
>
> ** **
>
> ** **
>
> Hope that makes some kind of sense.****
>
> ** **
>
> Thanks****
>
> ** **
>
> Tim ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jmesquita at freeswitch.org  Tue Nov 20 03:38:27 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Mon, 19 Nov 2012 21:38:27 -0300
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
Message-ID: <AD367A20-70E7-4015-97F2-5786AF9D02E2@freeswitch.org>

Transfer or execute extension. 

Sent from my iPhone

On Nov 19, 2012, at 8:36 PM, Tim Meade <Tim.Meade at Millicorp.com> wrote:

> Is there any way or method to have multiple xml_curl calls per call?
> 
> Something like
>  
> Call comes in
> xml_curl gets dialplan xml
> At some point in the dialplan xml something triggers a second call to xml_curl with current dial plan parameters.
>  
>  
> Hope that makes some kind of sense.
>  
> Thanks
>  
> Tim
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From krice at freeswitch.org  Tue Nov 20 04:09:40 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 19 Nov 2012 19:09:40 -0600
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
Message-ID: <CCD03474.60D45%krice@freeswitch.org>

If you use the transfer app in the dialplan, this should induce it to
re-fire the curl... However this is not stateless and it would look like a
new call unless your http service is keeping track of things

Also check out mod_httapi, this is more like ESL, but it asks a webserver
what to do each step of the call

K

On 11/19/12 5:36 PM, "Tim Meade" <Tim.Meade at Millicorp.com> wrote:

> Is there any way or method to have multiple xml_curl calls per call?
> 
> 
> Something like
>  
> Call comes in
> xml_curl gets dialplan xml
> At some point in the dialplan xml something triggers a second call to xml_curl
> with current dial plan parameters.
>  
>  
> Hope that makes some kind of sense.
>  
> Thanks
>  
> Tim 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From anthony.minessale at gmail.com  Tue Nov 20 04:55:58 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Nov 2012 19:55:58 -0600
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
Message-ID: <CAKbxfG8PJinkzFhYVooB5pXV_SGQeZcuYUKRQgBwudbYNDoTyA@mail.gmail.com>

Yes,

Either the transfer app or the execute_extension will both do it.
One transfers the call to a whole new extension and the other requests an
extension and executes it in place like a gosub.



On Mon, Nov 19, 2012 at 5:36 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:

>  Is there any way or method to have multiple xml_curl calls per call?
>
> ****
>
> Something like****
>
> ** **
>
> Call comes in****
>
> xml_curl gets dialplan xml****
>
> At some point in the dialplan xml something triggers a second call to
> xml_curl with current dial plan parameters.****
>
> ** **
>
> ** **
>
> Hope that makes some kind of sense.****
>
> ** **
>
> Thanks****
>
> ** **
>
> Tim ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Nov 20 05:55:12 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Nov 2012 20:55:12 -0600
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagHowbH_Zr8YPWGDsbug4m1ggcaTY=LdgzYf9yvJq15qFA@mail.gmail.com>
References: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
	<CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>
	<CAHKQagFrWbRft0wLiPsML22K61mfM+Z3Qh1ZJC+aKiebnNB+GA@mail.gmail.com>
	<CAHKQagHvQwWCOvdM0abqrM1xz65kFz9hY6J+fMKbJjH=Ot=kgQ@mail.gmail.com>
	<CAHKQagHowbH_Zr8YPWGDsbug4m1ggcaTY=LdgzYf9yvJq15qFA@mail.gmail.com>
Message-ID: <CAKbxfG8vrDExvtbt-W--u0AizwTR+S0x7XqoPrA1FXms=Yq-gw@mail.gmail.com>

One thing you can do is set the variable when you do know it, from the
dialplan.  So it will be there in all the subsequent events.



On Mon, Nov 19, 2012 at 1:34 AM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Last update, then I'm really going to sleep!
>
> Apologies for the noise btw, in hindsight I should have collected all this
> info and posted in one go.
>
> I've tried enabling auth-all-packets (along with auth_calls), as I thought
> maybe having authentication on REFER packets might make a difference, but
> sadly it had no effect (the SIP headers show proxy-authorization in the
> REFER, but nothing extra showed in mod_xml_curl)
>
> I've managed to narrow down the circumstances in which this happens;
>
> CORRECT:
>
> * User receives call from gateway, blind transfer to another user (shows
> correctly as variable_dialed_user/variable_dialed_domain)
> * User receives call from another user, blind transfer to gateway (shows
> correctly as variable_dialed_user/variable_dialed_domain)
> * User receives call from another user, blind transfer to another
> user (shows correctly as variable_dialed_user/variable_dialed_domain)
> * User makes call to another user, blind transfer to another user (shows
> correctly as variable_dialed_user/variable_dialed_domain)
> * User makes call to another user, blind transfer to a gateway (shows
> correctly as variable_dialed_user/variable_dialed_domain)
>
> MISSING:
>
> * User makes call to a gateway, blind transfer to another gateway (no
> clean variables for domain)
> * User makes call to a gateway, blind transfer to another user (no clean
> variables for domain)
>
> So, the problem seems to happen specifically when you blind transfer a
> call that is already in progress on a gateway, where the call was
> originated by a user and not a gateway.
>
> I did a bit more looking through code, added a few switch_log_printf()
> calls, and found that the following method is NOT called in those two
> scenarios where these variables are missing;
> mod_dptools.c: "switch_call_cause_t user_outgoing_channel"
>
> This is about as far as I can go on this, as I just don't know enough
> about C to give any more in-depth info :/
>
> Cal
>
> On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Sorry, another update.. after tinkering with the SIP headers, we found
>> that we're able to pass any user/host along in an INVITE, and this is
>> passed onto mod_xml_curl.
>>
>> To fix this particular part of the problem, we enabled auth_calls and
>> this gives us correct/clean variables we can trust, specifically;
>>
>>  u'variable_sip_auth_username': u'2000',
>>  u'variable_user_name': u'2000',
>>
>> However, when attempting to do the blind transfer again, this variables
>> are all missing.
>>
>> At this point I'm convinced that attempting to extract the user/domain
>> from the Refer headers is probably not the right thing to do... But I'm
>> still no closer to figuring out what the correct approach should be to
>> expose the authenticated user/domain to mod_xml_curl.
>>
>> Cal
>>
>>
>> On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Another quick update on this before I pass out from lack of sleep..!
>>>
>>> Just had a look through the mod_sofia.c/h source and found the following;
>>>
>>> mod_sofia.c/mod_sofia.h
>>> #define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
>>> if (!zstr(full_ref_by)) {
>>> switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
>>> "Referred-By", full_ref_by);
>>> }
>>> if (!zstr(full_ref_to)) {
>>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>> full_ref_to);
>>> }
>>> if (!zstr(full_ref_to)) {
>>>  switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>> full_ref_to);
>>> }
>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>>> "Process REFER to [%s@%s]\n", exten, (char *)
>>> refer_to->r_url->url_host);
>>>
>>> If the correct approach is deemed to be patching code, then I figured it
>>> could be as simple as this;
>>>
>>> switch_channel_set_variable(t_channel, "Referred-By-User", exten);
>>> switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *)
>>> refer_to->r_url->url_host);
>>>
>>> This is pretty much where my knowledge of C ends, I can (just about)
>>> read and copy chunks of C code, but that's about it :)
>>>
>>> Cal
>>>
>>> On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>>> Not sure if this is relevant but thought I'd point it out.
>>>>
>>>> The following field seems to contain the IP of the domain we were
>>>> expecting ('c1881.voiceflare.co.uk')
>>>>
>>>>  u'variable_sip_from_host': u'89.238.182.137',
>>>>
>>>> Normally, this field would contain the hostname and not the IP.
>>>>
>>>> Cal
>>>>
>>>> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <
>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>
>>>>> Hi guys,
>>>>>
>>>>> In a nut shell, it appears that when attempting to perform a blind
>>>>> transfer under certain conditions (explained below), mod_xml_curl does not
>>>>> expose the originating domain in a clean format.
>>>>>
>>>>> My initial plan was to find the point where these variable were being
>>>>> generated, look at what was available, then add an extra variable for the
>>>>> domain and submit a patch.
>>>>>
>>>>> Sadly my C isn't great and I hit a brick wall, so if anyone can help
>>>>> out, I will ensure the mod_xml_curl documentation is updated and/or assist
>>>>> with any patching/testing required.
>>>>>
>>>>>  Please take the following scenario;
>>>>>
>>>>> * Extension 2000 calls an external number via a gateway (i.e. bridge
>>>>> sofia/gateway/name/e164_number_here).
>>>>> * Call connects fine, audio stays good, no disconnection problems etc.
>>>>> * Call is blind transferred to another extension
>>>>>
>>>>> As a result, the following is determined;
>>>>>
>>>>> * User initiating the blind transfer is 2000
>>>>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>>>>> * Destination number of the call is 447866123456
>>>>> * Number to blind transfer to is 2001
>>>>> * Call to mod_xml_curl is made
>>>>>
>>>>> It makes reference to the User in the following 'clean' variables (by
>>>>> clean, I mean variables that just contain 2000 and don't require mangling
>>>>> to extract the info);
>>>>>
>>>>>  u'Caller-ANI': u'2000',
>>>>>  u'Caller-Username': u'2000',
>>>>>  u'Caller-Caller-ID-Number': u'2000',
>>>>>  u'Hunt-ANI': u'2000',
>>>>>  u'Hunt-Caller-ID-Number': u'2000',
>>>>>  u'Hunt-Username': u'2000',
>>>>>  u'variable_last_sent_callee_id_number': u'2000',
>>>>>  u'variable_sip_from_user': u'2000',
>>>>>
>>>>> It also has the User in the following unclean variables;
>>>>>
>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>  u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>>>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>>>>> >;tag=XryjFQp3rB2NF',
>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>
>>>>> However, it only references the domain in the following unclean
>>>>> variables;
>>>>>
>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',
>>>>>
>>>>> Lets say that we want to determine the user/domain that has initiated
>>>>> this transfer, doing so would mean mangling with one of those above
>>>>> variables, which seems a bit dirty (plus it is not clear which is the
>>>>> correct one to use).
>>>>>
>>>>> I have attached the SIP trace of the entire blind transfer event, and
>>>>> the full mod_xml_curl request being sent.
>>>>>
>>>>> If anyone could answer the following, it'd be much appreciated;
>>>>>
>>>>> * Should there be an improvement patch that exposes the domain of the
>>>>> user that originated the blind transfer?
>>>>> * Are there better/alternative ways to extracting the domain of the
>>>>> user that originated the blind transfer?
>>>>>
>>>>> Many thanks
>>>>>
>>>>> Cal
>>>>>
>>>>>
>>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From drk at drkngs.net  Tue Nov 20 06:05:11 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Mon, 19 Nov 2012 19:05:11 -0800
Subject: [Freeswitch-users]
	=?iso-8859-1?q?=5BConfidential=5D_-_RE=3A_Erro?=
	=?iso-8859-1?q?r_on_Call_Recording_-=09Windows=2C_FS_V1=2E2=2E4?=
In-Reply-To: <1353352337984-7584717.post@n2.nabble.com>
Message-ID: <20121120030511.2375413e@mail.tritonwest.net>

Jeff,  
   
Thanks for this info, but now that I remember from reading this post, do you know how I can specify a UNC path that FS will like?  
   
--Dave
      _____  

  From: Jeff Lenk [mailto:jeff at jefflenk.com]
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 19 Nov 2012 11:12:18 -0800
Subject: Re: [Freeswitch-users] [Confidential] - RE: Error on Call Recording - Windows, FS V1.2.4

This is a very common source of problems with windows and fs; with fs being
cross platform with *nix based environments that use the forward slash
exclusively for path separators. The issue involves backslash "\" encoded
escape entries. (ex. \r being an encoded return character hex 0xd or \n an
encoded newline hex 0xa) there are many more examples of this too.

The best way to deal with this in a cross platform way is to use windows
ability to use forward slashes "/" for path separators. All releases of
windows can do this without difficulty(and least for 10 years or so).

All internal path handing in FreeSWITCH has been changed(last few months) to
manipulate paths with forward slashes to minimize this very problem. So if
you supply external paths to fs make sure you always supply them with
forward slashes to avoid this problem.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-on-Call-Recording-Windows-FS-V1-2-4-tp7584705p7584717.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
      
   
 
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From jason.holden at start.ca  Tue Nov 20 06:35:09 2012
From: jason.holden at start.ca (Jason Holden)
Date: Mon, 19 Nov 2012 22:35:09 -0500
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
References: <472138AA0F3C4B01A08CBF5CB2CFFB2D@bob>
	<CAKzWOxVk-tx7NEX2=3JyhP8w-9XqmcpLNL69vs6N2n8bsRr0uA@mail.gmail.com>
Message-ID: <E66DCAE9702E4F869A4A2AE4431F8265@bob>

alrighty.

I added the log with the sip trace to pastebin.

Guess I was having a case of the Mondays.

Anyone able to give me their opinions on what is going on?

The one thing I notice is for some reason my public IP is showing up in the
trace. Wouldn't think it should be though since I am communicating on my
LAN.

 

  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Monday, November 19, 2012 5:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] retreiving voicemail dropping after 30
seconds

 

 

On Mon, Nov 19, 2012 at 1:32 PM, Jason Holden <jason.holden at start.ca> wrote:

I can not log on to the page but the following is the cli log.

Sure you can! Just read the challenge dialog a bit more closely. ;)
Also, select "FreeSWITCH Log" as the syntax highlighting. Don't forget to
turn on SIP trace:
sofia profile internal siptrace on

-MC
 

Also I am on my local LAN.

 

 

2012-11-19 17:16:46.630373 [NOTICE] switch_channel.c:953 New Channel
sofia/internal/201-entros at 192.168.15.9
[1107676f-36a3-4bc2-b413-525c898bb3b6]

2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_NEW

2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/201-entros at 192.168.15.9) State NEW

2012-11-19 17:16:46.650375 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected by
acl "domains". Falling back to Digest auth.

2012-11-19 17:16:46.650375 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.650375 [DEBUG] sofia.c:1755 detaching session
1107676f-36a3-4bc2-b413-525c898bb3b6

2012-11-19 17:16:46.650375 [WARNING] sofia_reg.c:1481 SIP auth challenge
(INVITE) on sofia profile 'internal' for [*97 at 192.168.15.9] from ip
192.168.15.2

2012-11-19 17:16:46.670376 [DEBUG] sofia.c:1847 Re-attaching to session
1107676f-36a3-4bc2-b413-525c898bb3b6

2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected by
acl "domains". Falling back to Digest auth.

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5596 Channel
sofia/internal/201-entros at 192.168.15.9 entering state [received][100]

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5607 Remote SDP:

v=0

o=- 6486898 6486898 IN IP4 192.168.15.2

s=-

c=IN IP4 192.168.15.2

t=0 0

m=audio 16392 RTP/AVP 0 100 101

a=rtpmap:0 PCMU/8000

a=rtpmap:100 NSE/8000

a=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:3093 Set Codec
sofia/internal/201-entros at 192.168.15.9 PCMU/8000 20 ms 160 samples 64000
bits

2012-11-19 17:16:46.690372 [DEBUG] switch_core_codec.c:111
sofia/internal/201-entros at 192.168.15.9 Original read codec set to PCMU:0

2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5265 Set 2833 dtmf send/recv
payload to 101

2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5824
(sofia/internal/201-entros at 192.168.15.9) State Change CS_NEW -> CS_INIT

2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_INIT

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
(sofia/internal/201-entros at 192.168.15.9) State INIT

2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:86
sofia/internal/201-entros at 192.168.15.9 SOFIA INIT

2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:126
(sofia/internal/201-entros at 192.168.15.9) State Change CS_INIT -> CS_ROUTING

2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454
(sofia/internal/201-entros at 192.168.15.9) State INIT going to sleep

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_ROUTING

2012-11-19 17:16:46.690372 [DEBUG] switch_channel.c:1988
(sofia/internal/201-entros at 192.168.15.9) Callstate Change DOWN -> RINGING

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/201-entros at 192.168.15.9) State ROUTING

2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:149
sofia/internal/201-entros at 192.168.15.9 SOFIA ROUTING

2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:117
sofia/internal/201-entros at 192.168.15.9 Standard ROUTING

2012-11-19 17:16:46.690372 [INFO] mod_dialplan_xml.c:498 Processing home
<201-entros>->*97 in context default

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.911] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.911] destination_number(*97) =~ /^(911)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.10d] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.10d] destination_number(*97) =~ /^(\d{10})$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.11d] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.11d] destination_number(*97) =~ /^\+?(\d{11})$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->AtlasVoice.tollfree] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[AtlasVoice.tollfree] destination_number(*97) =~
/^1?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->call-direction] continue=true

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 ANTI-Action
set(call_direction=local) 

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->group-intercept] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[group-intercept] destination_number(*97) =~ /^\*8$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->redial]
continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [redial]
destination_number(*97) =~ /^(redial|\*870)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->call_privacy] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [call_privacy]
destination_number(*97) =~ /^\*67(\d+)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->call_return] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [call_return]
destination_number(*97) =~ /^\*69$|^lcr$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->intercept-ext] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[intercept-ext] destination_number(*97) =~ /^\*\*(\d+)$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->extension-intercom] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[extension-intercom] destination_number(*97) =~ /^\*8(\d{2,7})$/
break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->send_to_voicemail] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL)
[send_to_voicemail] destination_number(*97) =~ /^\*99(\d{2,7})$/
break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing [default->vmain]
continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (FAIL) [vmain]
destination_number(*97) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 parsing
[default->vmain_user] continue=false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Regex (PASS) [vmain_user]
destination_number(*97) =~ /^\*97$/ break=on-false

Dialplan: sofia/internal/201-entros at 192.168.15.9 Action answer() 

Dialplan: sofia/internal/201-entros at 192.168.15.9 Action sleep(1000) 

Dialplan: sofia/internal/201-entros at 192.168.15.9 Action voicemail(check
default ${domain_name} ${caller_id_number}) 

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:167
(sofia/internal/201-entros at 192.168.15.9) State Change CS_ROUTING ->
CS_EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/201-entros at 192.168.15.9) State ROUTING going to sleep

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/201-entros at 192.168.15.9) State EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:242
sofia/internal/201-entros at 192.168.15.9 SOFIA EXECUTE

2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:209
sofia/internal/201-entros at 192.168.15.9 Standard EXECUTE

EXECUTE sofia/internal/201-entros at 192.168.15.9 set(call_direction=local)

2012-11-19 17:16:46.710380 [DEBUG] mod_dptools.c:1344
sofia/internal/201-entros at 192.168.15.9 SET [call_direction]=[local]

EXECUTE sofia/internal/201-entros at 192.168.15.9 answer()

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3350 AUDIO RTP
[sofia/internal/201-entros at 192.168.15.9] 192.168.15.9 port 24942 ->
192.168.15.2 port 16392 codec: 0 ms: 20

2012-11-19 17:16:46.710380 [DEBUG] switch_rtp.c:1927 Starting timer [soft]
160 bytes per 20ms

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3614 Set 2833 dtmf send
payload to 101

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3620 Set 2833 dtmf receive
payload to 101

2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3647
sofia/internal/201-entros at 192.168.15.9 Set rtp dtmf delay to 40

2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:856 Local SDP
sofia/internal/201-entros at 192.168.15.9:

v=0

o=FreeSWITCH 1353300664 1353300665 IN IP4 192.168.15.9

s=FreeSWITCH

c=IN IP4 192.168.15.9

t=0 0

m=audio 24942 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

a=sendrecv

 

2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.710380 [DEBUG] sofia.c:5596 Channel
sofia/internal/201-entros at 192.168.15.9 entering state [completed][200]

2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:830 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:16:46.710380 [DEBUG] switch_channel.c:3380
(sofia/internal/201-entros at 192.168.15.9) Callstate Change RINGING -> ACTIVE

2012-11-19 17:16:46.710380 [NOTICE] mod_dptools.c:1176 Channel
[sofia/internal/201-entros at 192.168.15.9] has been answered

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(1000)

2012-11-19 17:16:46.750379 [DEBUG] switch_rtp.c:3606 Correct ip/port
confirmed.

EXECUTE sofia/internal/201-entros at 192.168.15.9 voicemail(check default
192.168.15.9 201-entros)

2012-11-19 17:16:47.730206 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:47.990165 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:48.010904 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-enter_pass.wav] (en:en)

2012-11-19 17:16:48.030171 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:48.930025 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1604

2012-11-19 17:16:48.930025 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-enter_pass.wav

2012-11-19 17:16:49.289963 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 9:1444

2012-11-19 17:16:49.709899 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 2:1524

2012-11-19 17:16:50.169828 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 7:1524

2012-11-19 17:16:55.151063 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:1604

2012-11-19 17:16:55.271043 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:55.291060 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-you_have.wav] (en:en)

2012-11-19 17:16:55.311050 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:55.869947 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav

2012-11-19 17:16:55.989933 [DEBUG] switch_ivr_play_say.c:244 Handle say:[0]
(en:en)

2012-11-19 17:16:56.009933 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:56.769808 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/0.wav

2012-11-19 17:16:56.889794 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-new.wav] (en:en)

2012-11-19 17:16:56.909794 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:57.269730 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav

2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-messages.wav] (en:en)

2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:58.049611 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav

2012-11-19 17:16:58.170590 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-you_have.wav] (en:en)

2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:58.730505 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav

2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:244 Handle say:[0]
(en:en)

2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:16:59.630368 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/0.wav

2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-saved.wav] (en:en)

2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:00.230274 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-saved.wav

2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-messages.wav] (en:en)

2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:01.010156 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav

2012-11-19 17:17:01.250116 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-listen_new.wav] (en:en)

2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:02.609908 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav

2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:03.129826 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
(en:en)

2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:03.949702 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/1.wav

2012-11-19 17:17:04.069679 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-listen_saved.wav] (en:en)

2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:05.910397 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_saved.wav

2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:06.430318 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2]
(en:en)

2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:07.190204 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/2.wav

2012-11-19 17:17:07.310180 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-advanced.wav] (en:en)

2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:08.709964 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-advanced.wav

2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:09.229884 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:244 Handle say:[5]
(en:en)

2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:10.129747 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/5.wav

2012-11-19 17:17:10.249724 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-to_exit.wav] (en:en)

2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:11.069598 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_exit.wav

2012-11-19 17:17:11.089603 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1524

2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:11.410578 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]

2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-to_record_greeting.wav] (en:en)

2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:12.450387 [DEBUG] switch_ivr_play_say.c:1682 done playing
file
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_record_greeting.wa
v

2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:12.970304 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
(en:en)

2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:13.790181 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/1.wav

2012-11-19 17:17:13.910188 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-choose_greeting.wav] (en:en)

2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:15.089978 [DEBUG] switch_ivr_play_say.c:1682 done playing
file
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-choose_greeting.wav

2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:15.609897 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2]
(en:en)

2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:16.369782 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/2.wav

2012-11-19 17:17:16.489760 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(100)] (en:en)

EXECUTE sofia/internal/201-entros at 192.168.15.9 sleep(100)

2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-record_name2.wav] (en:en)

2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:17.770561 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name2.wav

2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)

2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:18.270487 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav

2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:244 Handle say:[3]
(en:en)

2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.730420 [DEBUG] sofia.c:5596 Channel
sofia/internal/201-entros at 192.168.15.9 entering state [terminating][0]

2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:2979
(sofia/internal/201-entros at 192.168.15.9) Callstate Change ACTIVE -> HANGUP

2012-11-19 17:17:18.730420 [NOTICE] sofia.c:6380 Hangup
sofia/internal/201-entros at 192.168.15.9 [CS_EXECUTE] [NORMAL_UNSPECIFIED]

2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:3002 Send signal
sofia/internal/201-entros at 192.168.15.9 [KILL]

2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.730420 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://digits/3.wav

2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:2685
sofia/internal/201-entros at 192.168.15.9 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/201-entros at 192.168.15.9) State EXECUTE going to sleep

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_HANGUP

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
(sofia/internal/201-entros at 192.168.15.9) State HANGUP

2012-11-19 17:17:18.730420 [DEBUG] mod_sofia.c:503 Channel
sofia/internal/201-entros at 192.168.15.9 hanging up, cause: NORMAL_UNSPECIFIED

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:48
sofia/internal/201-entros at 192.168.15.9 Standard HANGUP, cause:
NORMAL_UNSPECIFIED

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667
(sofia/internal/201-entros at 192.168.15.9) State HANGUP going to sleep

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/201-entros at 192.168.15.9) State Change CS_HANGUP ->
CS_REPORTING

2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_REPORTING

2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:749
(sofia/internal/201-entros at 192.168.15.9) State REPORTING

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:92
sofia/internal/201-entros at 192.168.15.9 Standard REPORTING, cause:
NORMAL_UNSPECIFIED

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:749
(sofia/internal/201-entros at 192.168.15.9) State REPORTING going to sleep

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:440
(sofia/internal/201-entros at 192.168.15.9) State Change CS_REPORTING ->
CS_DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1287 Send signal
sofia/internal/201-entros at 192.168.15.9 [BREAK]

2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1492 Session 2
(sofia/internal/201-entros at 192.168.15.9) Locked, Waiting on external
entities

2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1510 Session 2
(sofia/internal/201-entros at 192.168.15.9) Ended

2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1514 Close Channel
sofia/internal/201-entros at 192.168.15.9 [CS_DESTROY]

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:556
(sofia/internal/201-entros at 192.168.15.9) Callstate Change HANGUP -> DOWN

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:559
(sofia/internal/201-entros at 192.168.15.9) Running State Change CS_DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/201-entros at 192.168.15.9) State DESTROY

2012-11-19 17:17:19.010389 [DEBUG] mod_sofia.c:396
sofia/internal/201-entros at 192.168.15.9 SOFIA DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:99
sofia/internal/201-entros at 192.168.15.9 Standard DESTROY

2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/201-entros at 192.168.15.9) State DESTROY going to sleep

 

 


  _____  


From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Monday, November 19, 2012 12:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] retreiving voicemail dropping after 30
seconds

 

Is this device on the same LAN as FreeSWITCH? Get a console log and SIP
trace and drop it on pastebin.freeswitch.org and the gang here will offer
some insights.

-MC

On Fri, Nov 16, 2012 at 10:33 PM, Jason Holden <jason.holden at start.ca>
wrote:

Hi.

When accessing voicemail to listen to messages I am finding that it is
dropping at 30 seconds each time with a message of 100 sleep timer.

Does anyone have any recommendations?

I am using a Sipura 3000 connected to my freeswitch server.

 


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




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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

 


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




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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From jeff at jefflenk.com  Tue Nov 20 08:41:55 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 19 Nov 2012 21:41:55 -0800 (PST)
Subject: [Freeswitch-users] [Confidential] - RE: Error on Call Recording
 -	Windows, FS V1.2.4
In-Reply-To: <20121120030511.2375413e@mail.tritonwest.net>
References: <49C5FCA19A8A114493EBAACA42FE5899105E9041@1AERDCEXCHMBX1.AER.AERCO.local>
	<78990CE7CC964442A7C2CA5F4689695E99BCB82B@BARXB0003.UnifyBusiness.local>
	<loom.20121119T165734-788@post.gmane.org>
	<1353352337984-7584717.post@n2.nabble.com>
	<20121120030511.2375413e@mail.tritonwest.net>
Message-ID: <1353390115325-7584744.post@n2.nabble.com>

you should be able to double encode them

\\\\test\\name1\\name2



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-on-Call-Recording-Windows-FS-V1-2-4-tp7584705p7584744.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From regis.freeswitch.org at tornad.net  Tue Nov 20 09:47:57 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Tue, 20 Nov 2012 07:47:57 +0100
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <CAKzWOxUg2-hZbJpWpsGs-MObMSWvJxPXMuQMFEUi151X5iDftg@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
	<CAKzWOxUg2-hZbJpWpsGs-MObMSWvJxPXMuQMFEUi151X5iDftg@mail.gmail.com>
Message-ID: <CAHWtr1RNR27a8YvBp9TeVUHYavTaZLudWOMcDDwGg=VppAENwQ@mail.gmail.com>

If you transfer your call,  a new dialplan/ext will be called/build,via
xml_curl....
So you can transfer inside via action at the point you want or transfer
externaly via event_socket or xml_rpc.

2012/11/20 Michael Collins <msc at freeswitch.org>

> It does make sense, but I don't know that you can use xml_curl for that.
> However, it sounds suspiciously similar to how mod_httapi works. Have you
> ever looked at that?
>
> -MC
>
> On Mon, Nov 19, 2012 at 3:36 PM, Tim Meade <Tim.Meade at millicorp.com>wrote:
>
>>  Is there any way or method to have multiple xml_curl calls per call?
>>
>> ****
>>
>> Something like****
>>
>> ** **
>>
>> Call comes in****
>>
>> xml_curl gets dialplan xml****
>>
>> At some point in the dialplan xml something triggers a second call to
>> xml_curl with current dial plan parameters.****
>>
>> ** **
>>
>> ** **
>>
>> Hope that makes some kind of sense.****
>>
>> ** **
>>
>> Thanks****
>>
>> ** **
>>
>> Tim ****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From regis.freeswitch.org at tornad.net  Tue Nov 20 09:49:28 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Tue, 20 Nov 2012 07:49:28 +0100
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <CAHWtr1RNR27a8YvBp9TeVUHYavTaZLudWOMcDDwGg=VppAENwQ@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
	<CAKzWOxUg2-hZbJpWpsGs-MObMSWvJxPXMuQMFEUi151X5iDftg@mail.gmail.com>
	<CAHWtr1RNR27a8YvBp9TeVUHYavTaZLudWOMcDDwGg=VppAENwQ@mail.gmail.com>
Message-ID: <CAHWtr1Sk-L6oWP9YdZ=GS7F+7gqM=hv4_kCFyERQdLyD0UpPeQ@mail.gmail.com>

Pfff , my gmail interface wasn't refresh when i clicked answer so I don't
saw all the answer before mine :)
sorry for the doublon...

2012/11/20 Regis M <regis.freeswitch.org at tornad.net>

> If you transfer your call,  a new dialplan/ext will be called/build,via
> xml_curl....
> So you can transfer inside via action at the point you want or transfer
> externaly via event_socket or xml_rpc.
>
>
> 2012/11/20 Michael Collins <msc at freeswitch.org>
>
>> It does make sense, but I don't know that you can use xml_curl for that.
>> However, it sounds suspiciously similar to how mod_httapi works. Have you
>> ever looked at that?
>>
>> -MC
>>
>> On Mon, Nov 19, 2012 at 3:36 PM, Tim Meade <Tim.Meade at millicorp.com>wrote:
>>
>>>  Is there any way or method to have multiple xml_curl calls per call?
>>>
>>> ****
>>>
>>> Something like****
>>>
>>> ** **
>>>
>>> Call comes in****
>>>
>>> xml_curl gets dialplan xml****
>>>
>>> At some point in the dialplan xml something triggers a second call to
>>> xml_curl with current dial plan parameters.****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> Hope that makes some kind of sense.****
>>>
>>> ** **
>>>
>>> Thanks****
>>>
>>> ** **
>>>
>>> Tim ****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From avi at avimarcus.net  Tue Nov 20 10:24:18 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 20 Nov 2012 09:24:18 +0200
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <CAKbxfG8PJinkzFhYVooB5pXV_SGQeZcuYUKRQgBwudbYNDoTyA@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
	<CAKbxfG8PJinkzFhYVooB5pXV_SGQeZcuYUKRQgBwudbYNDoTyA@mail.gmail.com>
Message-ID: <CALNZuw=SbT1CmvMVd293jztyXDiV34QRS3OzoNjqdhVuenJV1Q@mail.gmail.com>

When you transfer, the xml_curl gets called again, and has access to all
current channel variables.

-Avi
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From prasd.d.b at gmail.com  Tue Nov 20 10:51:10 2012
From: prasd.d.b at gmail.com (Prasd D)
Date: Mon, 19 Nov 2012 23:51:10 -0800
Subject: [Freeswitch-users] NAT behavior
In-Reply-To: <079800ED-3488-4CB5-8FD8-0D286509FEE2@mojolingo.com>
References: <079800ED-3488-4CB5-8FD8-0D286509FEE2@mojolingo.com>
Message-ID: <CAPZnxCT4tLjNtJP1HC4EWtVGNMCwvrmk57+akCyeFy25q1PmPw@mail.gmail.com>

Try the auto-nat feature (see wiki), by default fs out of box is
configured that way

On 11/5/12, Luca Pradovera <lpradovera at mojolingo.com> wrote:
> Hello,
> without going in too much detail, we have a working setup with Asterisk over
> a pretty complex network topology.
> The setting that makes it work is nat=force_rport,comedia, and we have been
> trying to replicate that behavior on FreeSWITCH.
> We basically want t force all RTP traffic to go through a set IP, no matter
> what SDP says or wants to do.
>
> Thanks,
>
> Luca


-- 
Thanks,
Prasd


From prasd.d.b at gmail.com  Tue Nov 20 10:55:19 2012
From: prasd.d.b at gmail.com (Prasd D)
Date: Mon, 19 Nov 2012 23:55:19 -0800
Subject: [Freeswitch-users] E-mail voicemails - msmtp in Windows
In-Reply-To: <1352818521085-7584543.post@n2.nabble.com>
References: <CAG0Fp+_kQf1pZyZpCDWuqAVMpYsCfA3p1gUyrSzWK24VNi9Yjg@mail.gmail.com>
	<1352818521085-7584543.post@n2.nabble.com>
Message-ID: <CAPZnxCTE=kfMoYjrMFVDcsXLV_vO7ncEdxOPoG_z7rS9D3Ds0w@mail.gmail.com>

I am having trouble with voicemail using msmtp or anything else for
that matter too with the latest version.

Its as if the voicemail to email is just not even there.

I did add the following to the 1001.xml (recipient of voicemail).
<param name="vm-mailto" value="root at localhost"/>
<param name="vm-mailfrom" value="root at localhost"/>

But I don't see a single debug log message containing the word email
(or Email) or any indication of anything. Can you please help ?

On 11/13/12, Jeff Lenk <jeff at jefflenk.com> wrote:
> Please report this to Jira at jira.freeswitch.org
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/E-mail-voicemails-msmtp-in-Windows-tp7584525p7584543.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Thanks,
Prasd


From miha at softnet.si  Tue Nov 20 11:02:54 2012
From: miha at softnet.si (Miha)
Date: Tue, 20 Nov 2012 09:02:54 +0100
Subject: [Freeswitch-users] xml_curl
Message-ID: <50AB392E.4060006@softnet.si>

Hi,

I have set some part of my dialplan with culr. I would like to implement 
whole dialplan with curl but I need a little bit of help at the beginning.

Sometimes a need a information if user is on FS so I am using 
event_socket to call user_exists and some other api functions.

Is it possible to do radiu_auth with api or how what would you do to 
authenticate with radius? Install radiusclient on server where http 
server and do radius authorization on that server?

BR,
Miha


From tang.du at hotmail.com  Tue Nov 20 11:17:04 2012
From: tang.du at hotmail.com (tangdu)
Date: Tue, 20 Nov 2012 00:17:04 -0800 (PST)
Subject: [Freeswitch-users] errors with configure ftmod_libpri
Message-ID: <1353399424053-7584746.post@n2.nabble.com>

hi
  Scene?centos5.8 ? te110p  pri ? dahdi-linux-2.2.1?libpri-1.4.12.
  Installed no error. when load mod_freetdm ,Error messages as follows
 
2012-11-20 15:43:21.969763 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:21.969763 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:21.969763 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:21.989754 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:21.989754 [ERR] ftmod_zt.c:1119 [s1c16][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2012-11-20 15:43:21.989754 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:22.009782 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:22.009782 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:22.029753 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2012-11-20 15:43:22.029753 [ERR] ftmod_zt.c:1131 [s1c16][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
 
How to solve it?

Thanks
  
TangDu




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/errors-with-configure-ftmod-libpri-tp7584746.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From andrew at cassidywebservices.co.uk  Tue Nov 20 11:32:40 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 20 Nov 2012 08:32:40 +0000
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAMwZ-tJ4ijj18HmSZR4WjcuNtuswHcxERNdcBPw12z+dF7Oh_g@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
	<CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
	<CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>
	<CAMwZ-t+hkyfuf87b=48MfTMALDSyB7Efa-6BkfkKmMiLrx7+pA@mail.gmail.com>
	<CAKzWOxU+3RkOW1fO26y9gZ9eB8Bk4AEzEhe3=whJx7mTzYVf5g@mail.gmail.com>
	<CAMwZ-tJ4ijj18HmSZR4WjcuNtuswHcxERNdcBPw12z+dF7Oh_g@mail.gmail.com>
Message-ID: <CAMbW9oJOrrwoohjzOpf2BqHvrXG+QpZ2K=HijQfX3rKSiGJ8Rw@mail.gmail.com>

I'd love to set bounties/donate money, but $1000 is more than the total
value of my company! lol.

On 19 November 2012 19:07, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Nov 19, 2012 at 9:28 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> >> Whatever the case, we'll put $1000 toward the effort. This is not a
> >> bounty (conditional based on finishing the feature). It's a
> >> no-obligation, thanks-for-an-awesome-product,
> >> good-luck-with-future-development, contribution.  I hope other
> >> companies that benefit daily from FreeSWITCH consider following suit.
> >> Let's make this happen :)
> >
> > +1
> >
> > Thanks for your support and setting such a great example!
>
>
> It's our pleasure. I wish we could do more. One day we will be able to! :)
>
> I'm hoping other companies see the value of WebRTC and see what they
> can do to support it too.
>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From gabe at gundy.org  Tue Nov 20 14:06:43 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 20 Nov 2012 04:06:43 -0700
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAMbW9oJOrrwoohjzOpf2BqHvrXG+QpZ2K=HijQfX3rKSiGJ8Rw@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
	<CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
	<CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>
	<CAMwZ-t+hkyfuf87b=48MfTMALDSyB7Efa-6BkfkKmMiLrx7+pA@mail.gmail.com>
	<CAKzWOxU+3RkOW1fO26y9gZ9eB8Bk4AEzEhe3=whJx7mTzYVf5g@mail.gmail.com>
	<CAMwZ-tJ4ijj18HmSZR4WjcuNtuswHcxERNdcBPw12z+dF7Oh_g@mail.gmail.com>
	<CAMbW9oJOrrwoohjzOpf2BqHvrXG+QpZ2K=HijQfX3rKSiGJ8Rw@mail.gmail.com>
Message-ID: <CAMwZ-tLcziZd5Z4jVQbXM-yMQcucXcnHYiuOXf3UOmDmVPmC5A@mail.gmail.com>

On Tue, Nov 20, 2012 at 1:32 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> I'd love to set bounties/donate money, but $1000 is more than the total
> value of my company! lol.


+1 Points for having your heart in the right place ;)


Gabe
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From garbytrash at gmail.com  Tue Nov 20 14:13:34 2012
From: garbytrash at gmail.com (Zenny)
Date: Tue, 20 Nov 2012 12:13:34 +0100
Subject: [Freeswitch-users] Cluecon 2012 and Gemeinshaft 5.0
Message-ID: <CACuV5sBTDx6Pea82yhy8N2c4S25kdPaXtJ5OSkcBrorbJ6+cww@mail.gmail.com>

Hi:

I read at the Cluecon 2012 link below
http://www.cluecon.com/presentation/gemeinschaft-50-one-size-fits-all/
that Gemeinshaft 5.0 has already been released, but neither there is
any announcment like http://www.freeswitch.org/node/373 nor any
pointer in the official site amooma.de?

Has anyone clue about this ClueCon event? Or is there any audio or
video recording of that specific session of Stefan? If yes, please
share. Thanks!

/zenny


From lists at kavun.ch  Tue Nov 20 14:37:30 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 20 Nov 2012 06:37:30 -0500
Subject: [Freeswitch-users] Ringing instead of MOH for valet parking
Message-ID: <BC2DB117-1A46-47B6-B88F-3C2E8F48E57F@kavun.ch>

Hi all,

Is there a built-in way to have callers hear ringback instead of MOH when they are valet-parked?
I do not mean to replace the MOH string with a ringtone, which would ultimately cause the caller to hear ringing whenever put on hold?. I am looking for an actual option.

It is probably trivial to change in the code, but I thought I would ask anyway before I start messing around? 

Thanks,

All the best,
Emrah

From gerald.weber at besharp.at  Tue Nov 20 15:02:13 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Tue, 20 Nov 2012 12:02:13 +0000
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAMwZ-tLcziZd5Z4jVQbXM-yMQcucXcnHYiuOXf3UOmDmVPmC5A@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
	<CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
	<CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>
	<CAMwZ-t+hkyfuf87b=48MfTMALDSyB7Efa-6BkfkKmMiLrx7+pA@mail.gmail.com>
	<CAKzWOxU+3RkOW1fO26y9gZ9eB8Bk4AEzEhe3=whJx7mTzYVf5g@mail.gmail.com>
	<CAMwZ-tJ4ijj18HmSZR4WjcuNtuswHcxERNdcBPw12z+dF7Oh_g@mail.gmail.com>
	<CAMbW9oJOrrwoohjzOpf2BqHvrXG+QpZ2K=HijQfX3rKSiGJ8Rw@mail.gmail.com>
	<CAMwZ-tLcziZd5Z4jVQbXM-yMQcucXcnHYiuOXf3UOmDmVPmC5A@mail.gmail.com>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2D6E9B@exchange.besharp.at>

Just donated 100USD :)
Not much, but like Gabe said:
>no-obligation, thanks-for-an-awesome-product, good-luck-with-future-development, contribution

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gabriel Gunderson
Gesendet: Dienstag, 20. November 2012 12:07
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] webrtc2sip

On Tue, Nov 20, 2012 at 1:32 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk>> wrote:
I'd love to set bounties/donate money, but $1000 is more than the total value of my company! lol.

+1 Points for having your heart in the right place ;)


Gabe
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From slickwilly2000 at gmx.de  Tue Nov 20 15:23:51 2012
From: slickwilly2000 at gmx.de (=?Windows-1252?Q?Alex_M=FCller?=)
Date: Tue, 20 Nov 2012 13:23:51 +0100
Subject: [Freeswitch-users] Ringing instead of MOH for valet parking
In-Reply-To: <BC2DB117-1A46-47B6-B88F-3C2E8F48E57F@kavun.ch>
References: <BC2DB117-1A46-47B6-B88F-3C2E8F48E57F@kavun.ch>
Message-ID: <C7DD95104E584A1EB5458250F1E212D9@iMac>

Have you already seen the channel-variable ?valet_hold_music??

This only affects the valet_hold, not the MOH globally in FreeSWITCH...

From: Emrah 
Sent: Tuesday, November 20, 2012 12:37 PM
To: FreeSWITCH Users Help 
Subject: [Freeswitch-users] Ringing instead of MOH for valet parking

Hi all,

Is there a built-in way to have callers hear ringback instead of MOH when they are valet-parked?
I do not mean to replace the MOH string with a ringtone, which would ultimately cause the caller to hear ringing whenever put on hold?. I am looking for an actual option.

It is probably trivial to change in the code, but I thought I would ask anyway before I start messing around? 

Thanks,

All the best,
Emrah
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From Tim.Meade at Millicorp.com  Tue Nov 20 15:52:06 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Tue, 20 Nov 2012 12:52:06 +0000
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <CAHWtr1Sk-L6oWP9YdZ=GS7F+7gqM=hv4_kCFyERQdLyD0UpPeQ@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
	<CAKzWOxUg2-hZbJpWpsGs-MObMSWvJxPXMuQMFEUi151X5iDftg@mail.gmail.com>
	<CAHWtr1RNR27a8YvBp9TeVUHYavTaZLudWOMcDDwGg=VppAENwQ@mail.gmail.com>
	<CAHWtr1Sk-L6oWP9YdZ=GS7F+7gqM=hv4_kCFyERQdLyD0UpPeQ@mail.gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350ADB0176@MAIL.millicorp.com>

Thanks everyone.  The transfer option sounds interesting.   I will most certainly look at httapi first.  We looked at it back when but I'm not sure why we didn't use it.

Tim

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Regis M
Sent: Tuesday, November 20, 2012 1:49 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl requests multiple times per call

Pfff , my gmail interface wasn't refresh when i clicked answer so I don't saw all the answer before mine :)
sorry for the doublon...
2012/11/20 Regis M <regis.freeswitch.org at tornad.net<mailto:regis.freeswitch.org at tornad.net>>
If you transfer your call,  a new dialplan/ext will be called/build,via xml_curl....
So you can transfer inside via action at the point you want or transfer externaly via event_socket or xml_rpc.

2012/11/20 Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>>
It does make sense, but I don't know that you can use xml_curl for that. However, it sounds suspiciously similar to how mod_httapi works. Have you ever looked at that?

-MC
On Mon, Nov 19, 2012 at 3:36 PM, Tim Meade <Tim.Meade at millicorp.com<mailto:Tim.Meade at millicorp.com>> wrote:
Is there any way or method to have multiple xml_curl calls per call?
Something like

Call comes in
xml_curl gets dialplan xml
At some point in the dialplan xml something triggers a second call to xml_curl with current dial plan parameters.


Hope that makes some kind of sense.

Thanks

Tim

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




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

FreeSWITCH-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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



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




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

FreeSWITCH-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
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From Tim.Meade at Millicorp.com  Tue Nov 20 15:52:43 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Tue, 20 Nov 2012 12:52:43 +0000
Subject: [Freeswitch-users] G729 File version of $${us-ring}
In-Reply-To: <CAFiqYu=j1Vzny3=VgkpxTv9d8kVL2DVSV2grA3jjXwptYN5XMQ@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AD9A4EC@MAIL.millicorp.com>
	<CALn233qk8dMRD2hmNdn-f5_R5GJdBxyrQDycV2JJqaOO3Mmktg@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350ADA59B6@MAIL.millicorp.com>
	<CAFiqYu=j1Vzny3=VgkpxTv9d8kVL2DVSV2grA3jjXwptYN5XMQ@mail.gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350ADB0188@MAIL.millicorp.com>

Thanks Steven.

I did just that.  I do have one now is someone needs it, just let me know.

Tim

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Monday, November 19, 2012 6:36 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] G729 File version of $${us-ring}

You can create your own using fs_encode from a .wav recording, but it does need a single mod_com_g729 license.

http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding

-Steve


On 19 November 2012 13:14, Tim Meade <Tim.Meade at millicorp.com<mailto:Tim.Meade at millicorp.com>> wrote:
Thanks Mitch.  On today's (or tomorrow) todo list.   Was just wondering if someone had a nice clean one out there.

Tim

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Mitch Capper
Sent: Sunday, November 18, 2012 9:31 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] G729 File version of $${us-ring}

As long as you have a single G729 license just create it yourself us_ring is just the tone stream used so create a g729 call, record it, and play the tone stream.

~mitch

On Sat, Nov 17, 2012 at 8:25 AM, Tim Meade <Tim.Meade at millicorp.com<mailto:Tim.Meade at millicorp.com>> wrote:
>
>
> I was wondering if anyone had a G729 file of the $${us-ring}
>
>
>
> We want to use it for playing the ringback with G729 pass through.
>
>
>
> <action application="set"  data="ringback=SomeRingBackFile.G729"/>
>
>
>
>
>
> Thanks
>
>
>
> Tim
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-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-use
> rs
> http://www.freeswitch.org
>

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

 

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

FreeSWITCH-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

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




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

FreeSWITCH-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

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From cal.leeming at simplicitymedialtd.co.uk  Tue Nov 20 16:21:36 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Tue, 20 Nov 2012 13:21:36 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAKbxfG8vrDExvtbt-W--u0AizwTR+S0x7XqoPrA1FXms=Yq-gw@mail.gmail.com>
References: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
	<CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>
	<CAHKQagFrWbRft0wLiPsML22K61mfM+Z3Qh1ZJC+aKiebnNB+GA@mail.gmail.com>
	<CAHKQagHvQwWCOvdM0abqrM1xz65kFz9hY6J+fMKbJjH=Ot=kgQ@mail.gmail.com>
	<CAHKQagHowbH_Zr8YPWGDsbug4m1ggcaTY=LdgzYf9yvJq15qFA@mail.gmail.com>
	<CAKbxfG8vrDExvtbt-W--u0AizwTR+S0x7XqoPrA1FXms=Yq-gw@mail.gmail.com>
Message-ID: <CAHKQagG36-_01XRP_75VPqmN_2QxpSbvtCKm7NXEC=2+EczEWQ@mail.gmail.com>

Hi Anthony,

Thanks for the reply, yeah I spent some time looking at alternative ways
and came up with one that seems to get the job done

I am convinced now that there is no bug as such - but one thing that is
absolutely clear, is that mod_xml_curl is in desperate need of
normalization.. even if it was fully documented, some of the variants don't
make logical sense. However, the information is there, and as long as you
do the correct conditional checks, then it will work flawlessly.

To make the authenticated domain stick when passing through to a gateway,
we use the following;

<action application="bridge"
data="{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}sofia/gateway/{{
dst.gateway.name}}/{{dst.gateway.fs_dial_prefix}}{{dst.dst_number_e164}}"/>

The most important part of the above being;
{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}

However, this alone doesn't deal with the fact that you have to look in
different places for the correct variables depending on what the current
call context is.

To a certain extent, your own business logic will also determine which
variables should be used, and a combination of conditional checks may need
to be used in order to figure out which variable you should be using in the
first place lol.

Here are the patterns we have found so far - this is just an information
dump for now, and later down the road I will update the mod_xml_curl
documentation.

# Ensure that variable_sip_auth_username / variable_sip_auth_realm
#
# If Call-Direction is inbound, then;
# src_user = variable_sip_auth_username
# src_domain = variable_sip_auth_realm
# dst_user = variable_sip_to_user
#
# If Call-Direction is outbound, then;
# originate_user = variable_sip_auth_username
# originate_domain = variable_sip_auth_realm
# src_user = variable_sip_to_user
# src_domain = variable_sip_to_host
# dst_user = Caller-Destination-Number

* gateway to gateway (442476100401 > 442476100402)
* domain to gateway - blind xfer to gateway (2000 > 442476100401
> 442476100402)
* domain to gateway - blind xfer to domain (2000 > 442476100401 > 2002)
* domain to gateway (2000 > 442476100401)
* domain to domain (2000 > 2001)

----

# Check if variable_sip_to_host is present and known gateway
#
# If variable_dialed_user and variable_dialed_domain are present;
# originate_user = variable_dialed_user
# originate_domain = variable_dialed_domain
# src_user = variable_sip_from_user
# src_domain = variable_sip_to_host
# dst_user = Caller-Destination-Number
#
# If not present;
# src_user = variable_sip_from_user
# src_domain = variable_sip_to_host
# dst_user = variable_sip_to_user

* gateway to domain (442476100401 > 2002)
* gateway to domain - blind xfer to gateway (442476100401 > 2000
> 442476100402)
* gateway to domain - blind xfer to domain (442476100401 > 2000 > 2002)

----

# check for variable_dialed_domain
# If match;
# src_user = variable_dialed_user
# src_domain  = variable_dialed_domain
# dst_user = Caller-Destination-Number

* domain to domain to gateway (2000 > 2001 > 442476100402)


On Tue, Nov 20, 2012 at 2:55 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> One thing you can do is set the variable when you do know it, from the
> dialplan.  So it will be there in all the subsequent events.
>
>
>
> On Mon, Nov 19, 2012 at 1:34 AM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Last update, then I'm really going to sleep!
>>
>> Apologies for the noise btw, in hindsight I should have collected all
>> this info and posted in one go.
>>
>> I've tried enabling auth-all-packets (along with auth_calls), as I
>> thought maybe having authentication on REFER packets might make a
>> difference, but sadly it had no effect (the SIP headers show
>> proxy-authorization in the REFER, but nothing extra showed in mod_xml_curl)
>>
>> I've managed to narrow down the circumstances in which this happens;
>>
>> CORRECT:
>>
>> * User receives call from gateway, blind transfer to another user (shows
>> correctly as variable_dialed_user/variable_dialed_domain)
>> * User receives call from another user, blind transfer to gateway (shows
>> correctly as variable_dialed_user/variable_dialed_domain)
>> * User receives call from another user, blind transfer to another
>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>> * User makes call to another user, blind transfer to another user (shows
>> correctly as variable_dialed_user/variable_dialed_domain)
>> * User makes call to another user, blind transfer to a gateway (shows
>> correctly as variable_dialed_user/variable_dialed_domain)
>>
>> MISSING:
>>
>> * User makes call to a gateway, blind transfer to another gateway (no
>> clean variables for domain)
>> * User makes call to a gateway, blind transfer to another user (no clean
>> variables for domain)
>>
>> So, the problem seems to happen specifically when you blind transfer a
>> call that is already in progress on a gateway, where the call was
>> originated by a user and not a gateway.
>>
>> I did a bit more looking through code, added a few switch_log_printf()
>> calls, and found that the following method is NOT called in those two
>> scenarios where these variables are missing;
>> mod_dptools.c: "switch_call_cause_t user_outgoing_channel"
>>
>> This is about as far as I can go on this, as I just don't know enough
>> about C to give any more in-depth info :/
>>
>> Cal
>>
>> On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Sorry, another update.. after tinkering with the SIP headers, we found
>>> that we're able to pass any user/host along in an INVITE, and this is
>>> passed onto mod_xml_curl.
>>>
>>> To fix this particular part of the problem, we enabled auth_calls and
>>> this gives us correct/clean variables we can trust, specifically;
>>>
>>>  u'variable_sip_auth_username': u'2000',
>>>  u'variable_user_name': u'2000',
>>>
>>> However, when attempting to do the blind transfer again, this variables
>>> are all missing.
>>>
>>> At this point I'm convinced that attempting to extract the user/domain
>>> from the Refer headers is probably not the right thing to do... But I'm
>>> still no closer to figuring out what the correct approach should be to
>>> expose the authenticated user/domain to mod_xml_curl.
>>>
>>> Cal
>>>
>>>
>>> On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>>> Another quick update on this before I pass out from lack of sleep..!
>>>>
>>>> Just had a look through the mod_sofia.c/h source and found the
>>>> following;
>>>>
>>>> mod_sofia.c/mod_sofia.h
>>>> #define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
>>>> if (!zstr(full_ref_by)) {
>>>> switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
>>>> "Referred-By", full_ref_by);
>>>> }
>>>> if (!zstr(full_ref_to)) {
>>>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>> full_ref_to);
>>>> }
>>>> if (!zstr(full_ref_to)) {
>>>>  switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>> full_ref_to);
>>>> }
>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>>> SWITCH_LOG_DEBUG, "Process REFER to [%s@%s]\n", exten, (char *)
>>>> refer_to->r_url->url_host);
>>>>
>>>> If the correct approach is deemed to be patching code, then I figured
>>>> it could be as simple as this;
>>>>
>>>> switch_channel_set_variable(t_channel, "Referred-By-User", exten);
>>>> switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *)
>>>> refer_to->r_url->url_host);
>>>>
>>>> This is pretty much where my knowledge of C ends, I can (just about)
>>>> read and copy chunks of C code, but that's about it :)
>>>>
>>>> Cal
>>>>
>>>> On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <
>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>
>>>>> Not sure if this is relevant but thought I'd point it out.
>>>>>
>>>>> The following field seems to contain the IP of the domain we were
>>>>> expecting ('c1881.voiceflare.co.uk')
>>>>>
>>>>>  u'variable_sip_from_host': u'89.238.182.137',
>>>>>
>>>>> Normally, this field would contain the hostname and not the IP.
>>>>>
>>>>> Cal
>>>>>
>>>>> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <
>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>
>>>>>> Hi guys,
>>>>>>
>>>>>> In a nut shell, it appears that when attempting to perform a blind
>>>>>> transfer under certain conditions (explained below), mod_xml_curl does not
>>>>>> expose the originating domain in a clean format.
>>>>>>
>>>>>> My initial plan was to find the point where these variable were being
>>>>>> generated, look at what was available, then add an extra variable for the
>>>>>> domain and submit a patch.
>>>>>>
>>>>>> Sadly my C isn't great and I hit a brick wall, so if anyone can help
>>>>>> out, I will ensure the mod_xml_curl documentation is updated and/or assist
>>>>>> with any patching/testing required.
>>>>>>
>>>>>>  Please take the following scenario;
>>>>>>
>>>>>> * Extension 2000 calls an external number via a gateway (i.e. bridge
>>>>>> sofia/gateway/name/e164_number_here).
>>>>>> * Call connects fine, audio stays good, no disconnection problems etc.
>>>>>> * Call is blind transferred to another extension
>>>>>>
>>>>>> As a result, the following is determined;
>>>>>>
>>>>>> * User initiating the blind transfer is 2000
>>>>>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>>>>>> * Destination number of the call is 447866123456
>>>>>> * Number to blind transfer to is 2001
>>>>>> * Call to mod_xml_curl is made
>>>>>>
>>>>>> It makes reference to the User in the following 'clean' variables (by
>>>>>> clean, I mean variables that just contain 2000 and don't require mangling
>>>>>> to extract the info);
>>>>>>
>>>>>>  u'Caller-ANI': u'2000',
>>>>>>  u'Caller-Username': u'2000',
>>>>>>  u'Caller-Caller-ID-Number': u'2000',
>>>>>>  u'Hunt-ANI': u'2000',
>>>>>>  u'Hunt-Caller-ID-Number': u'2000',
>>>>>>  u'Hunt-Username': u'2000',
>>>>>>  u'variable_last_sent_callee_id_number': u'2000',
>>>>>>  u'variable_sip_from_user': u'2000',
>>>>>>
>>>>>> It also has the User in the following unclean variables;
>>>>>>
>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>  u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>>>>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>>>>>> >;tag=XryjFQp3rB2NF',
>>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>
>>>>>> However, it only references the domain in the following unclean
>>>>>> variables;
>>>>>>
>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',
>>>>>>
>>>>>> Lets say that we want to determine the user/domain that has initiated
>>>>>> this transfer, doing so would mean mangling with one of those above
>>>>>> variables, which seems a bit dirty (plus it is not clear which is the
>>>>>> correct one to use).
>>>>>>
>>>>>> I have attached the SIP trace of the entire blind transfer event, and
>>>>>> the full mod_xml_curl request being sent.
>>>>>>
>>>>>> If anyone could answer the following, it'd be much appreciated;
>>>>>>
>>>>>> * Should there be an improvement patch that exposes the domain of the
>>>>>> user that originated the blind transfer?
>>>>>> * Are there better/alternative ways to extracting the domain of the
>>>>>> user that originated the blind transfer?
>>>>>>
>>>>>> Many thanks
>>>>>>
>>>>>> Cal
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From shaheryarkh at gmail.com  Tue Nov 20 16:07:48 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Tue, 20 Nov 2012 14:07:48 +0100
Subject: [Freeswitch-users] FreeSWITCH and Digium T410P
In-Reply-To: <CAFGRPVrko1_T_Oa8B7Rzh=WDdz73WOuy7OOGPe1aq-q=w_+60Q@mail.gmail.com>
References: <CAFGRPVrko1_T_Oa8B7Rzh=WDdz73WOuy7OOGPe1aq-q=w_+60Q@mail.gmail.com>
Message-ID: <CAFZQphxW2gV64wsBcqdzX+uGbEwC0t1JxO=V2VaGHFMmMFALvQ@mail.gmail.com>

Well, you need to decide for what purpose or role you want to use this
card?

1. Do you want simple T1 or E1 configuration using CAS? Then libpri is not
needed.

OR

2. Do you want ISDN PRI (Q.931 protocol), which uses CCS? Then libpri is
required.

For SS7, you either use libss7 with Asterisk for Digium cards OR use
mod_freetdm with Freeswitch for Sangoma cards.This is a recommendation not
a requirement as i understand Sangoma cards are also supported on Asterisk
and Digium cards also work with Freeswitch. Sangoma and Digium both are
good competitors of each other. So they keep on developing and releasing
newer / better hardware and its support on various voip switches and so on.
Choose according to your requirements.

Thank you.


On Mon, Nov 19, 2012 at 3:59 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> Is there any *clear* documentation to install a Digium T410P? there's so
> many stuff I don't have it clear whether libpri is needed or not, whether I
> need some sangoma header, etc..
>
> Also, I understand no SS7 is supported on Digium?
>
> Anything clear?
>
> Thanks!!
>
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aladiensaleh at gmail.com  Tue Nov 20 16:52:43 2012
From: aladiensaleh at gmail.com (Alaa Saleh)
Date: Tue, 20 Nov 2012 15:52:43 +0200
Subject: [Freeswitch-users] Flex client and rtmp module
Message-ID: <CAOx1=Bu_H3Xg_6ok7MW7w_5eBQ41LOmVFMu3LEZLP82Rhw3iYw@mail.gmail.com>

Hello
I'm new to freeswitch. and have some problem in using rtmp.
I compiled the mod_rtmp ,installed and configured it.
and configured the flex client so I can make calls from the flex client to
the demo IVR or other sip client.
the problem is that I can't call the flex client from other client (sip or
flex clients). I've read the wiki over and over and used this instruction
in my dial plan:

<action application="bridge"
data="${rtmp_contact($${rtmp_profile}/${dialed_ext}@$${domain})}"/>

which I think it's wrong when I analysed the FS log I found it should be
<action application="bridge" data="${rtmp_contact
$${rtmp_profile}/${dialed_ext}@$${domain} }"/>
i.e. with out ()

and I found that rtmp_profile is null

I don't know what to do.
pls help me and thanks in advance.
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From Tim.Meade at Millicorp.com  Tue Nov 20 17:30:12 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Tue, 20 Nov 2012 14:30:12 +0000
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <CCD03474.60D45%krice@freeswitch.org>
References: <804D48104511D4468F0D60DF9D3100350ADAD0D1@MAIL.millicorp.com>
	<CCD03474.60D45%krice@freeswitch.org>
Message-ID: <804D48104511D4468F0D60DF9D3100350ADB0CE0@MAIL.millicorp.com>

Using mod_httapi is there a method to inject a call into a server?   Have the web server tell the Freeswitch server to make a call.....

Mod_xml_rpc can do it, but again we loss the state of the call.

In an ideal world I want to create the call and have 2 or 3 callbacks to the web app session.....

Possible?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Monday, November 19, 2012 8:10 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl requests multiple times per call

If you use the transfer app in the dialplan, this should induce it to re-fire the curl... However this is not stateless and it would look like a new call unless your http service is keeping track of things

Also check out mod_httapi, this is more like ESL, but it asks a webserver what to do each step of the call

K

On 11/19/12 5:36 PM, "Tim Meade" <Tim.Meade at Millicorp.com> wrote:
Is there any way or method to have multiple xml_curl calls per call?


Something like

Call comes in
xml_curl gets dialplan xml
At some point in the dialplan xml something triggers a second call to xml_curl with current dial plan parameters.


Hope that makes some kind of sense.

Thanks

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




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

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
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irc.freenode.net #freeswitch
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From abaci64 at gmail.com  Tue Nov 20 17:37:17 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 20 Nov 2012 09:37:17 -0500
Subject: [Freeswitch-users] Ringing instead of MOH for valet parking
In-Reply-To: <BC2DB117-1A46-47B6-B88F-3C2E8F48E57F@kavun.ch>
References: <BC2DB117-1A46-47B6-B88F-3C2E8F48E57F@kavun.ch>
Message-ID: <50AB959D.3010607@gmail.com>

Try setting the "valet_hold_music" variable before parking the call.

On 11/20/2012 6:37 AM, Emrah wrote:
> Hi all,
>
> Is there a built-in way to have callers hear ringback instead of MOH when they are valet-parked?
> I do not mean to replace the MOH string with a ringtone, which would ultimately cause the caller to hear ringing whenever put on hold?. I am looking for an actual option.
>
> It is probably trivial to change in the code, but I thought I would ask anyway before I start messing around?
>
> Thanks,
>
> All the best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From shaheryarkh at gmail.com  Tue Nov 20 17:36:48 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Tue, 20 Nov 2012 15:36:48 +0100
Subject: [Freeswitch-users] Problems with icall carrier in Texas
In-Reply-To: <CAGodFm_Q=EJMrnz2TsGYKuiYvkRN2qB=kzYEdKf10Lf0XUXmkA@mail.gmail.com>
References: <1353354186557-7584722.post@n2.nabble.com>
	<CAGodFm_Q=EJMrnz2TsGYKuiYvkRN2qB=kzYEdKf10Lf0XUXmkA@mail.gmail.com>
Message-ID: <CAFZQphyFJFyEfdvg7ukaxx26z47iPZ=bk7yR8jxMANJv0kHuvg@mail.gmail.com>

Yes this number is working, just tested it. But i am in Germany, so may be
possible i was connected to different office, if they have
geo-locationing enabled in their customer support centre.

Thank you.


On Mon, Nov 19, 2012 at 11:02 PM, Bryant Lee <blee at gocentrix.com> wrote:

> Paul,
>
> Once upon a time we had iCall but we were forced to move off because of
> things like this.  I also got good at finding other numbers that for some
> reason worked when their main number didnt.  Give this a shot: 214-377-3111
>
> Best regards,
>
>
>
>
> On Mon, Nov 19, 2012 at 2:43 PM, Paul <paul at iamfine.com> wrote:
>
>> We have a number of DID's with icall that we forward to a FS box.We
>> started
>> getting reports from our users that they are getting busy signals starting
>> Sat morning. We have alternate numbers that we handed out to our users
>> through a different carrier.
>>
>> This outage is a serious issue, and I cant seem to get anyone there to
>> discuss it. The main number rings busy. The trouble ticket system cant be
>> accessed as the account login fails.
>>
>> I would like to know if this is isolated to us or system wide.
>>
>> Is anyone else experiencing issues with them or know if they have a status
>> page.
>> I have tried to contact them on their main number and get a busy signal.
>>
>> Thanks in advance
>>
>>
>>
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Problems-with-icall-carrier-in-Texas-tp7584722.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
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>
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From meditel at gmail.com  Tue Nov 20 17:56:29 2012
From: meditel at gmail.com (Meditel)
Date: Tue, 20 Nov 2012 15:56:29 +0100
Subject: [Freeswitch-users] Mod nibblebill account sharing
Message-ID: <CAHkDqS9AU52RAK2xSgT=FWHRBRWbNPkCaZszwpQjbYknZhOJQQ@mail.gmail.com>

Hi,

I am wondering if Mod nibblebill is able to handle well the case when
multiple users are using the same account.

This is my setup : Freeswitch + Mod nibblebill + MySQL
I have add one account with 1$ cash
I have two users sharing the same account.

Mod nibblebill configuration is the same for both users :
-------------------------------------------------------------------------------------

<variable name="nibble_rate" value="1"/>
<variable name="nibble_account" value="555"/>
<variable name="nibble_increment" value="60" />

And

<param name="nobal_amt" value="0"/>
<param name="nobal_action" value="hangup XML default"/>

<extension name="hangup">
  <condition field="destination_number" expression="^(hangup)$">
    <action application="playback" data="no_more_funds.wav"/>
    <action application="hangup"/>
  </condition>
</extension>

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

All working well if users are not calling at the same time.

But when both users try to call at the same time, they can both make 60sec
call (so 60+60 = 120sec) .. they have done 60sec for free :-(

Questions:

   1. How to avoid the fraud when sharing the same account and knowing that
   users can call at the same time ? (i don't like a make a heartbeat of 1sec
   ... it is not good for our database ...) & (I don't like to block the
   account to be used by only one user at the same time)
   2. May be we need to forget mod nibblebill and try using an other
   solution ?


Thanks for sharing your knowledge
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From krice at freeswitch.org  Tue Nov 20 18:24:22 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 20 Nov 2012 09:24:22 -0600
Subject: [Freeswitch-users] xml_curl requests multiple times per call
In-Reply-To: <804D48104511D4468F0D60DF9D3100350ADB0CE0@MAIL.millicorp.com>
Message-ID: <CCD0FCC6.60D9A%krice@freeswitch.org>

What you do is use mod_xml_rpc to start the call, then have it sent to
mod_httapi for control...

Example using mod_xml_rpc, send an originate
Originate sofia/profile/Aleg_dest at somehost httpapi_exten XML context
CLID_NAME CLID_NUMBER

Then the httapi exten takes over the B leg and controls the call

You can exhibit the same control on the A leg using the
loopback/exten/context/dialplan syntax (note: we usually recommend against
using loopbacks as they can be evil beasts at times)

K


On 11/20/12 8:30 AM, "Tim Meade" <Tim.Meade at Millicorp.com> wrote:

> Using mod_httapi is there a method to inject a call into a server?   Have the
> web server tell the Freeswitch server to make a call?..
>  
> Mod_xml_rpc can do it, but again we loss the state of the call.
>  
> In an ideal world I want to create the call and have 2 or 3 callbacks to the
> web app session?..
>  
> Possible?
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Monday, November 19, 2012 8:10 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] xml_curl requests multiple times per call
>  
> If you use the transfer app in the dialplan, this should induce it to re-fire
> the curl... However this is not stateless and it would look like a new call
> unless your http service is keeping track of things
> 
> Also check out mod_httapi, this is more like ESL, but it asks a webserver what
> to do each step of the call
> 
> K
> 
> On 11/19/12 5:36 PM, "Tim Meade" <Tim.Meade at Millicorp.com> wrote:
> Is there any way or method to have multiple xml_curl calls per call?
> 
> 
> Something like
>  
> Call comes in
> xml_curl gets dialplan xml
> At some point in the dialplan xml something triggers a second call to xml_curl
> with current dial plan parameters.
>  
>  
> Hope that makes some kind of sense.
>  
> Thanks
>  
> Tim 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch

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From a.venugopan at mundio.com  Tue Nov 20 19:18:58 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Tue, 20 Nov 2012 16:18:58 +0000
Subject: [Freeswitch-users] voicemail flow
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337353@Mail-Kilo.squay.com>

Hi,

While dialing voicemail, mod_voicemail.c file plays the primary role. I know it considers sounds.xml,tts.xml(/usr/local/freeswitch/conf/lang/en/vm) and voicemail.conf.xml (/usr/local/freeswitch/conf/autoload_configs).

Once the c file is hit after that what happens and how the flow goes? Can anyone please explain? As am not that good in C, I don't understand the flow.

Thanks.
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From robert.hadley at teotech.com  Tue Nov 20 19:44:21 2012
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 20 Nov 2012 16:44:21 +0000
Subject: [Freeswitch-users] Ringing instead of MOH for valet parking
In-Reply-To: <C7DD95104E584A1EB5458250F1E212D9@iMac>
References: <BC2DB117-1A46-47B6-B88F-3C2E8F48E57F@kavun.ch>
	<C7DD95104E584A1EB5458250F1E212D9@iMac>
Message-ID: <71943DD5C22943448A24B7C5CDC2380730707DE1@CH1PRD0411MB430.namprd04.prod.outlook.com>


Also check out ring_ready application: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready


From: Alex M?ller [mailto:slickwilly2000 at gmx.de]
Sent: Tuesday, November 20, 2012 4:24 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Ringing instead of MOH for valet parking

Have you already seen the channel-variable "valet_hold_music"?

This only affects the valet_hold, not the MOH globally in FreeSWITCH...

From: Emrah<mailto:lists at kavun.ch>
Sent: Tuesday, November 20, 2012 12:37 PM
To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Ringing instead of MOH for valet parking

Hi all,

Is there a built-in way to have callers hear ringback instead of MOH when they are valet-parked?
I do not mean to replace the MOH string with a ringtone, which would ultimately cause the caller to hear ringing whenever put on hold.... I am looking for an actual option.

It is probably trivial to change in the code, but I thought I would ask anyway before I start messing around...

Thanks,

All the best,
Emrah
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




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

FreeSWITCH-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
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From itispip-qq at hotmail.com  Tue Nov 20 20:17:46 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Wed, 21 Nov 2012 01:17:46 +0800
Subject: [Freeswitch-users] How to enable unicode in CLI command?
In-Reply-To: <CALXCt0p9dR63AfScg5V6=4jugWCnEharx=tOKLLtXqrzXnf7eA@mail.gmail.com>
References: <BLU169-W654F5F82AD03E6AC000B8F8C570@phx.gbl>,
	<CALXCt0p9dR63AfScg5V6=4jugWCnEharx=tOKLLtXqrzXnf7eA@mail.gmail.com>
Message-ID: <BLU169-W270C6B026B66CA55088BCA8C550@phx.gbl>


Thanks Giovanni for the tip. In Wiki it says skypopen_chat supports full utf8 chat text through ESL or API; yet in Lua script I create freeswitc.API() and then executestring("skypopen_chat interface1 remote_skypeid some_unicode_messsage), the utf8 message was not received correctly. Does the wiki "API" not referring to freeswitch.API in Lua? > Date: Mon, 19 Nov 2012 12:19:30 +0100
> From: gmaruzz at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] How to enable unicode in CLI command?
> 
> from CLI you can't.
> You can use ESL, eg telnet to port 8021 (IIRC).
> Anyway, look in the wiki page of gsmopen and/or skypopen for an example howto.
> Via ESL unicode is supported.
> -giovanni
> 
> On 11/18/12, ?? <itispip-qq at hotmail.com> wrote:
> >
> > I tried to use chat / skypopen_chat from CLI console to send a message to
> > receiver, the message is written by unicode charactor (Korean or Thai), the
> > receiver cannot get correct charators;
> > Tried to call chat / skypopen_chat API from LUA/JScript, ended the same;
> > I believe Freeswitch itself support unicode, as when I use an SIP client as
> > "Bria" to send SIP chat message between 2 clients, unicode charactor
> > displays correctly.
> > Is it just the CLI command by design not support unicode, or I have to do
> > something in config xml?
> >  		 	   		
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From eburke at edge-net.net  Tue Nov 20 21:43:21 2012
From: eburke at edge-net.net (Eli Burke)
Date: Tue, 20 Nov 2012 13:43:21 -0500
Subject: [Freeswitch-users] Return code from ESL Message Sending
In-Reply-To: <CACyT-3mMQ8RJfD7J=X8ZZubgR5kKbUeRu_Lj_wYa=T3v+mQmJA@mail.gmail.com>
References: <mailman.1329.1352570404.576.freeswitch-users@lists.freeswitch.org>
	<3A82F56B-1332-485E-9F6C-D2126A859CC8@edge-net.net>
	<CACyT-3mMQ8RJfD7J=X8ZZubgR5kKbUeRu_Lj_wYa=T3v+mQmJA@mail.gmail.com>
Message-ID: <1388879B-963F-4127-B884-965671CD720D@edge-net.net>

Kurtis,

We've been working with FreeSWITCH Consulting to address some issues with MESSAGE delivery. A couple of patches were committed on Nov 13 and Nov 14 to trunk and they may help with your problem. These patches affect the following behavior:
* MESSAGEs fed through the chatplan are correctly delivered or ignored by sofia
* when blocking=False, "Delivery-Failure" is replaced with "Nonblocking-Delivery: true"
* when blocking=True, "Delivery-Failure" is correctly set to true or false
* when blocking=True, "Delivery-Result-Code" is added to the event

Some background explanation: MESSAGEs are normally delivered in non-blocking mode, which means FreeSWITCH makes no attempt to determine if they were successfully received. There is a variable that can be set ("blocking: true") to force FreeSWITCH to wait for a response. You can already see this in action using the chat command in fs_cli-- it will report success or failure. 

Unfortunately, "blocking" is not set by default. RIght now, the only way to get this behavior is to set it manually. For example, a chatplan rule to add the header to all inbound MESSAGEs:
    <extension name="add_blocking" continue="true">
      <condition>
          <action application="set" data="is_reg=${sofia_contact(${to_user}" inline="true"/>
          <action application="set" data="blocking=true"/>   
      </condition>
    </extension>

There is a potential (and untested!) downside to forcing blocking to be always-on. The MESSAGE delivery queue is currently handled by a single thread. Even if all MESSAGE objects are delivered successfully to the local switch, some amount of latency may be introduced. In a real-world high-throughput scenario, it's possible that this could cause noticeable delays in the time it takes to delivery a MESSAGE, creating an ever-growing backlog.

The "is_reg" variable in the rule above could be used to short circuit failed attempts by shunting MESSAGEs to a database, or dropping them on the floor, but this would not necessarily fix things. The good news is that if a high-volume user can demonstrate that there is a problem, it's fixable within FreeSWITCH by moving to a multi-threaded message delivery queue.

-Eli


> On Nov 10, 2012, at 1:00 PM, freeswitch-users-request at lists.freeswitch.org wrote:
> 
> From: Kurtis Heimerl <kheimerl at cs.berkeley.edu>
> Subject: [Freeswitch-users] Return code from ESL Message Sending
> Date: November 9, 2012 11:42:19 PM EST
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hello Freeswitch Users:
> 
> We're currently trying to get the return code from a MESSAGE we send using ESL. The closest we've found is this jira: http://jira.freeswitch.org/browse/FS-4453 which seems to provide similar functionality for the chat command, but nothing for ESL.
> 
> Here's a pastebin of our current code: http://pastebin.freeswitch.org/20201
> 
> The server we are hitting is returning a "415 Unsupported Content Type" (which is correct) and we're trying to discover that in freeswitch, instead of assuming the message was received correctly. Right now, we get that the recvEventTimed is returning None. This is all done on the a pull of FS from yesterday. 
> 
> Any suggestions?
> 

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From luis.daniel.lucio at gmail.com  Tue Nov 20 22:27:58 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Tue, 20 Nov 2012 14:27:58 -0500
Subject: [Freeswitch-users] Packing for Mageia FINALLY
Message-ID: <CAFLo2QzWKpZXJAY1tWM6df1QTg8Qsy3kO5o=0DZ9n5MQWy1rPw@mail.gmail.com>

Freeswitch is now at mageia 3!

:) Please check SPEC and give me advises, I have problems with
mod_snmp, for some stragereason it tries to link with libperl

LD

2012/11/18 Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com>:
> Hey, Im gack
>
> http://pkgsubmit.mageia.org/uploads/failure/cauldron/core/release/20121118155430.dlucio.valstar.29275/log/freeswitch-1.2.3-1.mga3/build.0.20121118155506.log
>
> It doesnt link, i read about underlink:
> https://wiki.mageia.org/en/Underlinking_issues_in_packaging#Perl_modules
>
> originaly i was having problems with mod_perl but it compiles now
> after applying some ld flags, but now mod_snmp doesnt. I dont have
> idea why it tries to link to perl.  I have already place the
> unlinking,
>
> any advice?
>
> 2012/3/24 Brian West <brian at freeswitch.org>:
>> Peas and carrots, peas and carrots?  Stay tuned we will have such a
>> thing soonish!
>>
>> Yours Truely,
>> Lolly pop
>>
>> Sent from my eyePad
>>
>> On Mar 24, 2012, at 10:32 PM, Luis Daniel Lucio Quiroz
>> <luis.daniel.lucio at gmail.com> wrote:
>>
>>> Changelog file says it is 1.0.7, but
>>> configure.in says 1.1.beta1
>>>
>>> also in files.freeswitch.org i found rpms saying 1.1.beta2 dated march 7
>>>
>>> So, according your recomendations what release do you recommend me?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From msc at freeswitch.org  Tue Nov 20 22:42:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Nov 2012 11:42:41 -0800
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
In-Reply-To: <E66DCAE9702E4F869A4A2AE4431F8265@bob>
References: <472138AA0F3C4B01A08CBF5CB2CFFB2D@bob>
	<CAKzWOxVk-tx7NEX2=3JyhP8w-9XqmcpLNL69vs6N2n8bsRr0uA@mail.gmail.com>
	<E66DCAE9702E4F869A4A2AE4431F8265@bob>
Message-ID: <CAKzWOxVzGHDwM0kiR4Zbj9d0+iNL6eZKgQoYZApcV7Ao1S8rPA@mail.gmail.com>

Which pb entry is it?
-MC

On Mon, Nov 19, 2012 at 7:35 PM, Jason Holden <jason.holden at start.ca> wrote:

> ** **
>
> alrighty.****
>
> I added the log with the sip trace to pastebin.****
>
> Guess I was having a case of the Mondays.****
>
> Anyone able to give me their opinions on what is going on?****
>
> The one thing I notice is for some reason my public IP is showing up in
> the trace. Wouldn?t think it should be though since I am communicating on
> my LAN.****
>
> ** **
>  ------------------------------
>
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Monday, November 19, 2012 5:39 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] retreiving voicemail dropping after 30
> seconds****
>
> ** **
>
> ** **
>
> On Mon, Nov 19, 2012 at 1:32 PM, Jason Holden <jason.holden at start.ca>
> wrote:****
>
> I can not log on to the page but the following is the cli log.****
>
> Sure you can! Just read the challenge dialog a bit more closely. ;)
> Also, select "FreeSWITCH Log" as the syntax highlighting. Don't forget to
> turn on SIP trace:
> ****sofia**** profile internal siptrace on
>
> -MC
>  ****
>
>  Also I am on my local LAN.****
>
>  ****
>
>  ****
>
> 2012-11-19 17:16:46.630373 [NOTICE] switch_channel.c:953 New Channel ****
> sofia****/internal/201-entros at 192.168.15.9[1107676f-36a3-4bc2-b413-525c898bb3b6]
> ****
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:415 (****
> sofia****/internal/201-entros at 192.168.15.9) Running State Change CS_NEW***
> *
>
> 2012-11-19 17:16:46.630373 [DEBUG] switch_core_state_machine.c:433 (****
> sofia****/internal/201-entros at 192.168.15.9) State NEW****
>
> 2012-11-19 17:16:46.650375 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected
> by acl "domains". Falling back to Digest auth.****
>
> 2012-11-19 17:16:46.650375 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.650375 [DEBUG] sofia.c:1755 detaching session
> 1107676f-36a3-4bc2-b413-525c898bb3b6****
>
> 2012-11-19 17:16:46.650375 [WARNING] sofia_reg.c:1481 SIP auth challenge
> (INVITE) on ****sofia**** profile 'internal' for [*97 at 192.168.15.9] from
> ip 192.168.15.2****
>
> 2012-11-19 17:16:46.670376 [DEBUG] sofia.c:1847 Re-attaching to session
> 1107676f-36a3-4bc2-b413-525c898bb3b6****
>
> 2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.670376 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:7726 IP 192.168.15.2 Rejected
> by acl "domains". Falling back to Digest auth.****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5596 Channel ****sofia****
> /internal/201-entros at 192.168.15.9 entering state [received][100]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5607 Remote SDP:****
>
> v=0****
>
> o=- 6486898 6486898 IN IP4 192.168.15.2****
>
> s=-****
>
> c=IN IP4 192.168.15.2****
>
> t=0 0****
>
> m=audio 16392 RTP/AVP 0 100 101****
>
> a=rtpmap:0 PCMU/8000****
>
> a=rtpmap:100 NSE/8000****
>
> a=fmtp:100 192-193****
>
> a=rtpmap:101 telephone-event/8000****
>
> a=fmtp:101 0-15****
>
> a=ptime:20****
>
>  ****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5136 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:3093 Set Codec
> sofia/internal/201-entros at 192.168.15.9 PCMU/8000 20 ms 160 samples 64000
> bits****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_codec.c:111 ****sofia****
> /internal/201-entros at 192.168.15.9 Original read codec set to PCMU:0****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia_glue.c:5265 Set 2833 dtmf
> send/recv payload to 101****
>
> 2012-11-19 17:16:46.690372 [DEBUG] sofia.c:5824 (****sofia****/internal/
> 201-entros at 192.168.15.9) State Change CS_NEW -> CS_INIT****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
> ****sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415 (****
> sofia****/internal/201-entros at 192.168.15.9) Running State Change CS_INIT**
> **
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454 (****
> sofia****/internal/201-entros at 192.168.15.9) State INIT****
>
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:86 ****sofia****/internal/
> 201-entros at 192.168.15.9 SOFIA INIT****
>
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:126 (****sofia****
> /internal/201-entros at 192.168.15.9) State Change CS_INIT -> CS_ROUTING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_session.c:1287 Send signal
> ****sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:454 (****
> sofia****/internal/201-entros at 192.168.15.9) State INIT going to sleep****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:415 (****
> sofia****/internal/201-entros at 192.168.15.9) Running State Change
> CS_ROUTING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_channel.c:1988 (****sofia****
> /internal/201-entros at 192.168.15.9) Callstate Change DOWN -> RINGING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:470 (****
> sofia****/internal/201-entros at 192.168.15.9) State ROUTING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] mod_sofia.c:149 **sofia**/internal/
> 201-entros at 192.168.15.9 ****SOFIA**** ROUTING****
>
> 2012-11-19 17:16:46.690372 [DEBUG] switch_core_state_machine.c:117 ****
> sofia****/internal/201-entros at 192.168.15.9 Standard ROUTING****
>
> 2012-11-19 17:16:46.690372 [INFO] mod_dialplan_xml.c:498 Processing home
> <201-entros>->*97 in context default****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.911] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.911] destination_number(*97) =~ /^(911)$/ break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.10d] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.10d] destination_number(*97) =~ /^(\d{10})$/ break=on-false***
> *
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.11d] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.11d] destination_number(*97) =~ /^\+?(\d{11})$/ break=on-false
> ****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->AtlasVoice.tollfree] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [AtlasVoice.tollfree] destination_number(*97) =~
> /^1?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->call-direction] continue=true****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
> break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 ANTI-Action
> set(call_direction=local) ****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->group-intercept] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [group-intercept] destination_number(*97) =~ /^\*8$/ break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->redial] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [redial] destination_number(*97) =~ /^(redial|\*870)$/ break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->call_privacy] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [call_privacy] destination_number(*97) =~ /^\*67(\d+)$/ break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->call_return] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [call_return] destination_number(*97) =~ /^\*69$|^lcr$/ break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->intercept-ext] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [intercept-ext] destination_number(*97) =~ /^\*\*(\d+)$/ break=on-false***
> *
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->extension-intercom] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [extension-intercom] destination_number(*97) =~ /^\*8(\d{2,7})$/
> break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->send_to_voicemail] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [send_to_voicemail] destination_number(*97) =~ /^\*99(\d{2,7})$/
> break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->vmain] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (FAIL)
> [vmain] destination_number(*97) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false
> ****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 parsing
> [default->vmain_user] continue=false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Regex (PASS)
> [vmain_user] destination_number(*97) =~ /^\*97$/ break=on-false****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Action answer() *
> ***
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Action
> sleep(1000) ****
>
> Dialplan: ****sofia****/internal/201-entros at 192.168.15.9 Action
> voicemail(check default ${domain_name} ${caller_id_number}) ****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:167 (****
> sofia****/internal/201-entros at 192.168.15.9) State Change CS_ROUTING ->
> CS_EXECUTE****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:1287 Send signal
> ****sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:470 (****
> sofia****/internal/201-entros at 192.168.15.9) State ROUTING going to sleep**
> **
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:415 (****
> sofia****/internal/201-entros at 192.168.15.9) Running State Change
> CS_EXECUTE****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:477 (****
> sofia****/internal/201-entros at 192.168.15.9) State EXECUTE****
>
> 2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:242 ****sofia****/internal/
> 201-entros at 192.168.15.9 SOFIA EXECUTE****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_state_machine.c:209 ****
> sofia****/internal/201-entros at 192.168.15.9 Standard EXECUTE****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9set(call_direction=local)
> ****
>
> 2012-11-19 17:16:46.710380 [DEBUG] mod_dptools.c:1344 ****sofia****
> /internal/201-entros at 192.168.15.9 SET [call_direction]=[local]****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 answer()****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3350 AUDIO RTP [****sofia*
> ***/internal/201-entros at 192.168.15.9] 192.168.15.9 port 24942 ->
> 192.168.15.2 port 16392 codec: 0 ms: 20****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_rtp.c:1927 Starting timer [soft]
> 160 bytes per 20ms****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3614 Set 2833 dtmf send
> payload to 101****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3620 Set 2833 dtmf receive
> payload to 101****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia_glue.c:3647 ****sofia****
> /internal/201-entros at 192.168.15.9 Set rtp dtmf delay to 40****
>
> 2012-11-19 17:16:46.710380 [DEBUG] mod_sofia.c:856 Local SDP ****sofia****
> /internal/201-entros at 192.168.15.9:****
>
> v=0****
>
> o=FreeSWITCH 1353300664 1353300665 IN IP4 192.168.15.9****
>
> s=FreeSWITCH****
>
> c=IN IP4 192.168.15.9****
>
> t=0 0****
>
> m=audio 24942 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****
>
> a=sendrecv****
>
>  ****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] sofia.c:5596 Channel ****sofia****
> /internal/201-entros at 192.168.15.9 entering state [completed][200]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_core_session.c:830 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:16:46.710380 [DEBUG] switch_channel.c:3380 (****sofia****
> /internal/201-entros at 192.168.15.9) Callstate Change RINGING -> ACTIVE****
>
> 2012-11-19 17:16:46.710380 [NOTICE] mod_dptools.c:1176 Channel [****sofia*
> ***/internal/201-entros at 192.168.15.9] has been answered****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 sleep(1000)****
>
> 2012-11-19 17:16:46.750379 [DEBUG] switch_rtp.c:3606 Correct ip/port
> confirmed.****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 voicemail(check
> default 192.168.15.9 201-entros)****
>
> 2012-11-19 17:16:47.730206 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:47.990165 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:48.010904 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-enter_pass.wav] (en:en)****
>
> 2012-11-19 17:16:48.030171 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:48.930025 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1604*
> ***
>
> 2012-11-19 17:16:48.930025 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-enter_pass.wav
> ****
>
> 2012-11-19 17:16:49.289963 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 9:1444*
> ***
>
> 2012-11-19 17:16:49.709899 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 2:1524*
> ***
>
> 2012-11-19 17:16:50.169828 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 7:1524*
> ***
>
> 2012-11-19 17:16:55.151063 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:1604*
> ***
>
> 2012-11-19 17:16:55.271043 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:55.291060 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-you_have.wav] (en:en)****
>
> 2012-11-19 17:16:55.311050 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:55.869947 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav**
> **
>
> 2012-11-19 17:16:55.989933 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[0] (en:en)****
>
> 2012-11-19 17:16:56.009933 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:56.769808 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/0.wav****
>
> 2012-11-19 17:16:56.889794 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-new.wav] (en:en)****
>
> 2012-11-19 17:16:56.909794 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:57.269730 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav****
>
> 2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-messages.wav] (en:en)****
>
> 2012-11-19 17:16:57.389720 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:58.049611 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav**
> **
>
> 2012-11-19 17:16:58.170590 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-you_have.wav] (en:en)****
>
> 2012-11-19 17:16:58.190605 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:58.730505 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-you_have.wav**
> **
>
> 2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[0] (en:en)****
>
> 2012-11-19 17:16:58.850487 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:16:59.630368 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/0.wav****
>
> 2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-saved.wav] (en:en)****
>
> 2012-11-19 17:16:59.750348 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:00.230274 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-saved.wav****
>
> 2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-messages.wav] (en:en)****
>
> 2012-11-19 17:17:00.350255 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:01.010156 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-messages.wav**
> **
>
> 2012-11-19 17:17:01.250116 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-listen_new.wav] (en:en)****
>
> 2012-11-19 17:17:01.270135 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:02.609908 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav
> ****
>
> 2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:02.729887 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:03.129826 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[1] (en:en)****
>
> 2012-11-19 17:17:03.249807 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:03.949702 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/1.wav****
>
> 2012-11-19 17:17:04.069679 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-listen_saved.wav] (en:en)****
>
> 2012-11-19 17:17:04.309644 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:05.910397 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-listen_saved.wav
> ****
>
> 2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:06.030378 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:06.430318 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[2] (en:en)****
>
> 2012-11-19 17:17:06.550297 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:07.190204 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/2.wav****
>
> 2012-11-19 17:17:07.310180 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-advanced.wav] (en:en)****
>
> 2012-11-19 17:17:07.550143 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:08.709964 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-advanced.wav**
> **
>
> 2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:08.829946 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:09.229884 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[5] (en:en)****
>
> 2012-11-19 17:17:09.349864 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:10.129747 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/5.wav****
>
> 2012-11-19 17:17:10.249724 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-to_exit.wav] (en:en)****
>
> 2012-11-19 17:17:10.489687 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:11.069598 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_exit.wav***
> *
>
> 2012-11-19 17:17:11.089603 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 5:1524*
> ***
>
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:11.170582 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:11.410578 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]****
>
> 2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-to_record_greeting.wav] (en:en)****
>
> 2012-11-19 17:17:11.430568 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:12.450387 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file
> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-to_record_greeting.wav
> ****
>
> 2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:12.570371 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:12.970304 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[1] (en:en)****
>
> 2012-11-19 17:17:13.090285 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:13.790181 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/1.wav****
>
> 2012-11-19 17:17:13.910188 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-choose_greeting.wav] (en:en)****
>
> 2012-11-19 17:17:14.150128 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:15.089978 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file
> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-choose_greeting.wav
> ****
>
> 2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:15.209957 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:15.609897 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[2] (en:en)****
>
> 2012-11-19 17:17:15.729878 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:16.369782 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/2.wav****
>
> 2012-11-19 17:17:16.489760 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(100)] (en:en)****
>
> EXECUTE ****sofia****/internal/201-entros at 192.168.15.9 sleep(100)****
>
> 2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-record_name2.wav] (en:en)****
>
> 2012-11-19 17:17:16.709727 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:17.770561 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name2.wav
> ****
>
> 2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)****
>
> 2012-11-19 17:17:17.870561 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:18.270487 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav****
>
> 2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[3] (en:en)****
>
> 2012-11-19 17:17:18.390470 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.710418 [DEBUG] switch_core_session.c:976 Send signal *
> ***sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] sofia.c:5596 Channel ****sofia****
> /internal/201-entros at 192.168.15.9 entering state [terminating][0]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:2979 (****sofia****
> /internal/201-entros at 192.168.15.9) Callstate Change ACTIVE -> HANGUP****
>
> 2012-11-19 17:17:18.730420 [NOTICE] sofia.c:6380 Hangup ****sofia****
> /internal/201-entros at 192.168.15.9 [CS_EXECUTE] [NORMAL_UNSPECIFIED]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_channel.c:3002 Send signal ****
> sofia****/internal/201-entros at 192.168.15.9 [KILL]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
> ****sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file file_string://digits/3.wav****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:2685 ****sofia***
> */internal/201-entros at 192.168.15.9 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:477 (****
> sofia****/internal/201-entros at 192.168.15.9) State EXECUTE going to sleep**
> **
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415 (****
> sofia****/internal/201-entros at 192.168.15.9) Running State Change CS_HANGUP
> ****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667 (****
> sofia****/internal/201-entros at 192.168.15.9) State HANGUP****
>
> 2012-11-19 17:17:18.730420 [DEBUG] mod_sofia.c:503 Channel ****sofia****
> /internal/201-entros at 192.168.15.9 hanging up, cause: NORMAL_UNSPECIFIED***
> *
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:48 ****
> sofia****/internal/201-entros at 192.168.15.9 Standard HANGUP, cause:
> NORMAL_UNSPECIFIED****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:667 (****
> sofia****/internal/201-entros at 192.168.15.9) State HANGUP going to sleep***
> *
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:446 (****
> sofia****/internal/201-entros at 192.168.15.9) State Change CS_HANGUP ->
> CS_REPORTING****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_session.c:1287 Send signal
> ****sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:415 (****
> sofia****/internal/201-entros at 192.168.15.9) Running State Change
> CS_REPORTING****
>
> 2012-11-19 17:17:18.730420 [DEBUG] switch_core_state_machine.c:749 (****
> sofia****/internal/201-entros at 192.168.15.9) State REPORTING****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:92 ****
> sofia****/internal/201-entros at 192.168.15.9 Standard REPORTING, cause:
> NORMAL_UNSPECIFIED****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:749 (****
> sofia****/internal/201-entros at 192.168.15.9) State REPORTING going to sleep
> ****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:440 (****
> sofia****/internal/201-entros at 192.168.15.9) State Change CS_REPORTING ->
> CS_DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1287 Send signal
> ****sofia****/internal/201-entros at 192.168.15.9 [BREAK]****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_session.c:1492 Session 2 (*
> ***sofia****/internal/201-entros at 192.168.15.9) Locked, Waiting on
> external entities****
>
> 2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1510 Session 2 (
> ****sofia****/internal/201-entros at 192.168.15.9) Ended****
>
> 2012-11-19 17:17:19.010389 [NOTICE] switch_core_session.c:1514 Close
> Channel ****sofia****/internal/201-entros at 192.168.15.9 [CS_DESTROY]****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:556 (****
> sofia****/internal/201-entros at 192.168.15.9) Callstate Change HANGUP ->
> DOWN****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:559 (****
> sofia****/internal/201-entros at 192.168.15.9) Running State Change
> CS_DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569 (****
> sofia****/internal/201-entros at 192.168.15.9) State DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] mod_sofia.c:396 ****sofia****/internal/
> 201-entros at 192.168.15.9 SOFIA DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:99 ****
> sofia****/internal/201-entros at 192.168.15.9 Standard DESTROY****
>
> 2012-11-19 17:17:19.010389 [DEBUG] switch_core_state_machine.c:569 (****
> sofia****/internal/201-entros at 192.168.15.9) State DESTROY going to sleep**
> **
>
>  ****
>
>  ****
>  ------------------------------
>
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Monday, November 19, 2012 12:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] retreiving voicemail dropping after 30
> seconds****
>
>  ****
>
> Is this device on the same LAN as FreeSWITCH? Get a console log and SIP
> trace and drop it on pastebin.freeswitch.org and the gang here will offer
> some insights.
>
> -MC****
>
> On Fri, Nov 16, 2012 at 10:33 PM, Jason Holden <jason.holden at start.ca>
> wrote:****
>
> Hi.****
>
> When accessing voicemail to listen to messages I am finding that it is
> dropping at 30 seconds each time with a message of 100 sleep timer.****
>
> Does anyone have any recommendations?****
>
> I am using a Sipura 3000 connected to my freeswitch server.****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Nov 20 22:49:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Nov 2012 11:49:32 -0800
Subject: [Freeswitch-users] Cluecon 2012 and Gemeinshaft 5.0
In-Reply-To: <CACuV5sBTDx6Pea82yhy8N2c4S25kdPaXtJ5OSkcBrorbJ6+cww@mail.gmail.com>
References: <CACuV5sBTDx6Pea82yhy8N2c4S25kdPaXtJ5OSkcBrorbJ6+cww@mail.gmail.com>
Message-ID: <CAKzWOxX=VxabdzH_r_9yQw8nL-3ej21phzbGb9=-jei5move+A@mail.gmail.com>

Actually, the audio and video recordings are in the works. We'll keep you
all posted. I know we keep saying that but really they are!
-MC

On Tue, Nov 20, 2012 at 3:13 AM, Zenny <garbytrash at gmail.com> wrote:

> Hi:
>
> I read at the Cluecon 2012 link below
> http://www.cluecon.com/presentation/gemeinschaft-50-one-size-fits-all/
> that Gemeinshaft 5.0 has already been released, but neither there is
> any announcment like http://www.freeswitch.org/node/373 nor any
> pointer in the official site amooma.de?
>
> Has anyone clue about this ClueCon event? Or is there any audio or
> video recording of that specific session of Stefan? If yes, please
> share. Thanks!
>
> /zenny
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jason.holden at start.ca  Tue Nov 20 23:10:26 2012
From: jason.holden at start.ca (Jason Holden)
Date: Tue, 20 Nov 2012 15:10:26 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 77, Issue 150
In-Reply-To: <mailman.2056.1353440595.576.freeswitch-users@lists.freeswitch.org>
References: <mailman.2056.1353440595.576.freeswitch-users@lists.freeswitch.org>
Message-ID: <3C6BB2DCC2764929BA2C79667B2697EC@bob>

It is labeled entros.

I saw in the sip signaling that it timed out since did not receive an ack
back.

Not sure why this is happening though as well as why my external ip is
showing up in the sip signaling in my lan.
I am sure that is part of the reason why its erroring out due to no tramsion
reply.
Let me know what you think.

Thanks.
 

 


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
freeswitch-users-request at lists.freeswitch.org
Sent: Tuesday, November 20, 2012 2:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 77, Issue 150

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

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

You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

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



From gmaruzz at celliax.org  Tue Nov 20 23:30:00 2012
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 20 Nov 2012 21:30:00 +0100
Subject: [Freeswitch-users] How to enable unicode in CLI command?
In-Reply-To: <BLU169-W270C6B026B66CA55088BCA8C550@phx.gbl>
References: <BLU169-W654F5F82AD03E6AC000B8F8C570@phx.gbl>
	<CALXCt0p9dR63AfScg5V6=4jugWCnEharx=tOKLLtXqrzXnf7eA@mail.gmail.com>
	<BLU169-W270C6B026B66CA55088BCA8C550@phx.gbl>
Message-ID: <CALXCt0pVhw_B+RyRvNEKouHDeULk_DV7pihVKHP=-2FbV-moJA@mail.gmail.com>

No idea if FreeSWITCH Lua supports unicode.

Maybe you can write another post, asking if Lua in FreeSWITCH supports
unicode...

-giovanni

On Tue, Nov 20, 2012 at 6:17 PM, ?? <itispip-qq at hotmail.com> wrote:
> Thanks Giovanni for the tip. In Wiki it says
>
> skypopen_chat supports full utf8 chat text through ESL or API; yet in Lua
> script I create freeswitc.API() and then executestring("skypopen_chat
> interface1 remote_skypeid some_unicode_messsage), the utf8 message was not
> received correctly.
>
> Does the wiki "API" not referring to freeswitch.API in Lua?
>
>> Date: Mon, 19 Nov 2012 12:19:30 +0100
>> From: gmaruzz at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] How to enable unicode in CLI command?
>
>>
>> from CLI you can't.
>> You can use ESL, eg telnet to port 8021 (IIRC).
>> Anyway, look in the wiki page of gsmopen and/or skypopen for an example
>> howto.
>> Via ESL unicode is supported.
>> -giovanni
>>
>> On 11/18/12, ?? <itispip-qq at hotmail.com> wrote:
>> >
>> > I tried to use chat / skypopen_chat from CLI console to send a message
>> > to
>> > receiver, the message is written by unicode charactor (Korean or Thai),
>> > the
>> > receiver cannot get correct charators;
>> > Tried to call chat / skypopen_chat API from LUA/JScript, ended the same;
>> > I believe Freeswitch itself support unicode, as when I use an SIP client
>> > as
>> > "Bria" to send SIP chat message between 2 clients, unicode charactor
>> > displays correctly.
>> > Is it just the CLI command by design not support unicode, or I have to
>> > do
>> > something in config xml?
>> >
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From philq at qsystemsengineering.com  Tue Nov 20 23:35:27 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Tue, 20 Nov 2012 12:35:27 -0800 (PST)
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
	contact IP/port
In-Reply-To: <CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
References: <054001cdc664$92486660$b6d93320$@com>
	<CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
Message-ID: <1353443727657-7584773.post@n2.nabble.com>

We're running in bypass media mode with aggressive-nat-detection enabled.  FS
handles the signaling but passes the internal 10.x.x.x addresses for media
along unchanged, even between different sites.

Any recommendations for SIP proxies if FS can't do it automatically? 
Rewriting the SDP from the dialplan might be worth looking into depending
upon how it's implemented.

Thanks for the info.

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Rewriting-media-address-in-SDP-as-well-as-contact-IP-port-tp7584712p7584773.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From philq at qsystemsengineering.com  Tue Nov 20 23:36:05 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Tue, 20 Nov 2012 12:36:05 -0800 (PST)
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
	contact IP/port
In-Reply-To: <DB46FE12-EB6B-47BD-B516-5EB41994CB87@freeswitch.org>
References: <054001cdc664$92486660$b6d93320$@com>
	<DB46FE12-EB6B-47BD-B516-5EB41994CB87@freeswitch.org>
Message-ID: <1353443765604-7584774.post@n2.nabble.com>

Yes, exactly.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Rewriting-media-address-in-SDP-as-well-as-contact-IP-port-tp7584712p7584774.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue Nov 20 23:35:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Nov 2012 12:35:43 -0800
Subject: [Freeswitch-users] voicemail flow
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337353@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337353@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxXs4Y3V+_18kvttHA63aiU+Kd20BOJk34om+wE1f3++zQ@mail.gmail.com>

This question is a bit broad, but I'll add some info that I hope is useful.

When mod_voicemail.c is first loaded (or when it is re-loaded) then it will
read from voicemail.conf.xml to get its base configuration. That
configuration file defines voicemail "profiles" that can be used. The
example configuration provides a "default" profile that defines things like
the menu keys, skip greeting key, operator extension, etc. This "default"
value corresponds to this line from conf/dialplan/default.xml where the
Local_Extension sends a call to voicemail:

 <action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>

You could define other profiles if you so desire and then give them other
names.

The sounds.xml and tts.xml files both accomplish the same thing: they
define phrase macros that get played while a caller is connected to
voicemail. The sounds.xml file uses the pre-recorded sound files to stitch
together phrases whereas tts.xml uses a text-to-speech engine to accomplish
the same purpose. There really is no reason to use tts.xml unless you need
a voice or language where recordings are not available but for which you do
have a TTS engine.

The best way to see what's happening with mod_voicemail is to watch the
console debug log while you dial in and check your voicemail. The console
log will display what's going on. You'll notice that by far you see things
like "switch_ivr_play_say.c" and "switch_rtp.c". Consider this snippet of
me logging in to mailbox 1001:

2012-11-20 12:09:05.286038 [DEBUG] switch_rtp.c:3606 Correct ip/port
confirmed.
EXECUTE sofia/internal/1001 at 10.15.64.229 voicemail(check default
10.15.64.229)
2012-11-20 12:09:06.206030 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]
2012-11-20 12:09:06.446031 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]
2012-11-20 12:09:06.446031 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-enter_id.wav] (en:en)
2012-11-20 12:09:06.446031 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 16000hz 1 channels 20ms
2012-11-20 12:09:08.406030 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-enter_id.wav
2012-11-20 12:09:08.526030 [DEBUG] switch_ivr_play_say.c:244 Handle say:[#]
(en:en)
2012-11-20 12:09:08.526030 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 16000hz 1 channels 20ms
2012-11-20 12:09:09.186029 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://ascii/35.wav
2012-11-20 12:09:09.366029 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 1:1440
2012-11-20 12:09:09.786028 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 0:1120
2012-11-20 12:09:09.986027 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 0:1120
2012-11-20 12:09:10.526029 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 1:1280
2012-11-20 12:09:11.166034 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:1040
2012-11-20 12:09:11.166034 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]
2012-11-20 12:09:11.166034 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-enter_pass.wav] (en:en)
2012-11-20 12:09:11.166034 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 16000hz 1 channels 20ms
2012-11-20 12:09:13.106029 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-enter_pass.wav
2012-11-20 12:09:13.226027 [DEBUG] switch_ivr_play_say.c:244 Handle say:[#]
(en:en)
2012-11-20 12:09:13.226027 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 16000hz 1 channels 20ms
2012-11-20 12:09:13.886031 [DEBUG] switch_ivr_play_say.c:1682 done playing
file file_string://ascii/35.wav
2012-11-20 12:09:14.086030 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 1:1280
2012-11-20 12:09:14.586029 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 0:1280
2012-11-20 12:09:14.746031 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 0:800
2012-11-20 12:09:15.306037 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 1:1200
2012-11-20 12:09:15.766043 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:1200

Notice that there's no reference to "mod_voicemail.c" in there. That's
because mod_voicemail uses FS core APIs to perform basic functions like
playing sound files and collecting digits from the caller. That's ok - once
you get into the source code you'll start to see the connection. I'll get
you started. Open up src/mod/application/mod_voicemail/mod_voicemail.c and
find the function named "voicemail_check_main" near line 1850. When you log
in to check your voicemail this function controls the whole session from
the time you log in to the time you hang up. The switch statement near line
1903 checks the state and acts accordingly. Look at all the case statements
- they handle the various states, like "this person needs to log in"
(VM_CHECK_AUTH) or "play the messages in this folder"
(VM_CHECK_PLAY_MESSAGES).

I'm afraid I can't really go into any more detail without spending a lot of
time going line-by-line and translating the source code into plain-language
equivalents. However, I think you'll find that you are quite capable doing
this with a little practice and a lot of test calls. The FreeSWITCH source
code is extremely clean and in many cases is self-documenting. Many non-C
programmers are able to discern what a piece of code is doing simply by
looking at it and observing what it does when a call is in progress. Here's
a tip: you can add your own debug print lines to the code and recompile and
reload mod_voicemail. Look for places in the code where you see occurrences
of "switch_log_printf" and I think you'll be able to figure it out.

Last tip: to recompile mod_voicemail (or any other module) just go to the
root of the freeswitch source and type:
make mod_voicemail-install

That's it! Then go to fs_cli and type "reload mod_voicemail" and you're
ready to try out your changes.

I hope this helps all those who are curious about diving into the code.
It's not too scary as long as you don't mess with it on a production
machine. ;)

-Michael

P.S. - If you want to learn more about phrase macros then I highly
recommend chapter six of the original FreeSWITCH book. I cover part of the
sounds.xml phrase file, specifically the macro that handles pluralizing the
phrase "you have x new message(s)".



On Tue, Nov 20, 2012 at 8:18 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> While dialing voicemail, mod_voicemail.c file plays the primary role. I
> know it considers sounds.xml,tts.xml(/usr/local/freeswitch/conf/lang/en/vm)
> and voicemail.conf.xml (/usr/local/freeswitch/conf/autoload_configs).****
>
> ** **
>
> Once the c file is hit after that what happens and how the flow goes? Can
> anyone please explain? As am not that good in C, I don?t understand the
> flow.****
>
> ** **
>
> Thanks.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From Tim.Meade at Millicorp.com  Tue Nov 20 23:41:33 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Tue, 20 Nov 2012 20:41:33 +0000
Subject: [Freeswitch-users] sip_registrations column orig_server_host being
 set with internal IP in AWS
Message-ID: <804D48104511D4468F0D60DF9D3100350ADB56D7@MAIL.millicorp.com>

We are testing with Amazon AWS and have noticed that orig_server_host in sip_registrations table has the internal IP and not the external one.

Is there any way to have this assume the external IP as it's set in vars.conf with external_sip_ip?

Thanks

Tim
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From abaci64 at gmail.com  Wed Nov 21 00:26:24 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 20 Nov 2012 16:26:24 -0500
Subject: [Freeswitch-users] Return code from ESL Message Sending
In-Reply-To: <1388879B-963F-4127-B884-965671CD720D@edge-net.net>
References: <mailman.1329.1352570404.576.freeswitch-users@lists.freeswitch.org>
	<3A82F56B-1332-485E-9F6C-D2126A859CC8@edge-net.net>
	<CACyT-3mMQ8RJfD7J=X8ZZubgR5kKbUeRu_Lj_wYa=T3v+mQmJA@mail.gmail.com>
	<1388879B-963F-4127-B884-965671CD720D@edge-net.net>
Message-ID: <50ABF580.40502@gmail.com>

Would you mind documenting these option on the wiki 
(http://wiki.freeswitch.org/wiki/Mod_sms) so that people know about it.
Thanks

On 11/20/2012 1:43 PM, Eli Burke wrote:
> Kurtis,
>
> We've been working with FreeSWITCH Consulting to address some issues 
> with MESSAGE delivery. A couple of patches were committed on Nov 13 
> and Nov 14 to trunk and they may help with your problem. These patches 
> affect the following behavior:
> * MESSAGEs fed through the chatplan are correctly delivered or ignored 
> by sofia
> * when blocking=False, "Delivery-Failure" is replaced with 
> "Nonblocking-Delivery: true"
> * when blocking=True, "Delivery-Failure" is correctly set to true or false
> * when blocking=True, "Delivery-Result-Code" is added to the event
>
> Some background explanation: MESSAGEs are normally delivered in 
> non-blocking mode, which means FreeSWITCH makes no attempt to 
> determine if they were successfully received. There is a variable that 
> can be set ("blocking: true") to force FreeSWITCH to wait for a 
> response. You can already see this in action using the chat command in 
> fs_cli-- it will report success or failure.
>
> Unfortunately, "blocking" is not set by default. RIght now, the only 
> way to get this behavior is to set it manually. For example, a 
> chatplan rule to add the header to all inbound MESSAGEs:
>     <extension name="add_blocking" continue="true">
>       <condition>
>           <action application="set" 
> data="is_reg=${sofia_contact(${to_user}" inline="true"/>
>           <action application="set" data="blocking=true"/>
>       </condition>
>     </extension>
>
> There is a potential (and untested!) downside to forcing blocking to 
> be always-on. The MESSAGE delivery queue is currently handled by a 
> single thread. Even if all MESSAGE objects are delivered successfully 
> to the local switch, some amount of latency may be introduced. In a 
> real-world high-throughput scenario, it's possible that this could 
> cause noticeable delays in the time it takes to delivery a MESSAGE, 
> creating an ever-growing backlog.
>
> The "is_reg" variable in the rule above could be used to short circuit 
> failed attempts by shunting MESSAGEs to a database, or dropping them 
> on the floor, but this would not necessarily fix things. The good news 
> is that if a high-volume user can demonstrate that there is a problem, 
> it's fixable within FreeSWITCH by moving to a multi-threaded message 
> delivery queue.
>
> -Eli
>
>
>> On Nov 10, 2012, at 1:00 PM, 
>> freeswitch-users-request at lists.freeswitch.org <javascript:_e({}, 
>> 'cvml', 'freeswitch-users-request at lists.freeswitch.org');> wrote:
>>
>>     *From: *Kurtis Heimerl <kheimerl at cs.berkeley.edu
>>     <javascript:_e({}, 'cvml', 'kheimerl at cs.berkeley.edu');>>
>>     *Subject: **[Freeswitch-users] Return code from ESL Message Sending*
>>     *Date: *November 9, 2012 11:42:19 PM EST
>>     *To: *FreeSWITCH Users Help
>>     <freeswitch-users at lists.freeswitch.org <javascript:_e({}, 'cvml',
>>     'freeswitch-users at lists.freeswitch.org');>>
>>     *Reply-To: *FreeSWITCH Users Help
>>     <freeswitch-users at lists.freeswitch.org <javascript:_e({}, 'cvml',
>>     'freeswitch-users at lists.freeswitch.org');>>
>>
>>
>>     Hello Freeswitch Users:
>>
>>     We're currently trying to get the return code from a MESSAGE we
>>     send using ESL. The closest we've found is this jira:
>>     http://jira.freeswitch.org/browse/FS-4453 which seems to provide
>>     similar functionality for the chat command, but nothing for ESL.
>>
>>     Here's a pastebin of our current code:
>>     http://pastebin.freeswitch.org/20201
>>
>>     The server we are hitting is returning a "415 Unsupported Content
>>     Type" (which is correct) and we're trying to discover that in
>>     freeswitch, instead of assuming the message was received
>>     correctly. Right now, we get that the recvEventTimed is returning
>>     None. This is all done on the a pull of FS from yesterday.
>>
>>     Any suggestions?
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From yiftah at choochee.com  Wed Nov 21 00:55:07 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Tue, 20 Nov 2012 13:55:07 -0800
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
 contact IP/port
In-Reply-To: <1353443727657-7584773.post@n2.nabble.com>
References: <054001cdc664$92486660$b6d93320$@com>
	<CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
	<1353443727657-7584773.post@n2.nabble.com>
Message-ID: <CAFVUaV_mgPH8bSYkXaEETft8eYf+kLzBxWrjgoxj-=Rh+_QSCg@mail.gmail.com>

We use OpenSIPs as SIP proxy it has all the SDP rewriting and it works
great with FreeSWITCH
OpenSIPs is a great product

On Tue, Nov 20, 2012 at 12:35 PM, PhilQ <philq at qsystemsengineering.com>wrote:

> We're running in bypass media mode with aggressive-nat-detection enabled.
>  FS
> handles the signaling but passes the internal 10.x.x.x addresses for media
> along unchanged, even between different sites.
>
> Any recommendations for SIP proxies if FS can't do it automatically?
> Rewriting the SDP from the dialplan might be worth looking into depending
> upon how it's implemented.
>
> Thanks for the info.
>
> - Phil
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Rewriting-media-address-in-SDP-as-well-as-contact-IP-port-tp7584712p7584773.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Wed Nov 21 01:11:39 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 20 Nov 2012 16:11:39 -0600
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
 contact IP/port
In-Reply-To: <1353443727657-7584773.post@n2.nabble.com>
Message-ID: <CCD15C3B.60F22%krice@freeswitch.org>

IIRC freeswitch has never re-written this... In bypass media mode FreeSWITCH
can not accurately re-write the SDP for the RTP, it doesn't know what PORT
the NAT device is going to expose the RTP port on...

NAT devices have a bad habit of translating ports from whatever the host
behind them says to something completely different...

To handle this, FreeSWITCH adjusts media on the fly to send media back to
where it came from....


On 11/20/12 2:35 PM, "PhilQ" <philq at qsystemsengineering.com> wrote:

> We're running in bypass media mode with aggressive-nat-detection enabled.  FS
> handles the signaling but passes the internal 10.x.x.x addresses for media
> along unchanged, even between different sites.
> 
> Any recommendations for SIP proxies if FS can't do it automatically?
> Rewriting the SDP from the dialplan might be worth looking into depending
> upon how it's implemented.
> 
> Thanks for the info.
> 
> - Phil
> 
> 
> 
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Rewriting-media-address-in-SDP-a
> s-well-as-contact-IP-port-tp7584712p7584773.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From garmt.noname at gmail.com  Wed Nov 21 01:23:30 2012
From: garmt.noname at gmail.com (grmt)
Date: Tue, 20 Nov 2012 23:23:30 +0100
Subject: [Freeswitch-users] How to enable unicode in CLI command?
In-Reply-To: <BLU169-W270C6B026B66CA55088BCA8C550@phx.gbl>
References: <BLU169-W654F5F82AD03E6AC000B8F8C570@phx.gbl>,
	<CALXCt0p9dR63AfScg5V6=4jugWCnEharx=tOKLLtXqrzXnf7eA@mail.gmail.com>
	<BLU169-W270C6B026B66CA55088BCA8C550@phx.gbl>
Message-ID: <014101cdc76d$ac48b060$04da1120$@gmail.com>

Not 100% sure, but I think I may have had a similar problem.

I made a jira entry for this, FS-3679, which has a patch that contains a
special version of fs_cli_i (on linux) that supports utf-8, but it doesn?t
support command line editing and history, as the version of the lib that is
used for this (libedit) does not support unicode.

I?m not sure the patch can still be applied to the current release or git
head, but you can give it a try.



From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ??
Sent: Tuesday, November 20, 2012 18:18
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to enable unicode in CLI command?



Thanks Giovanni for the tip. In Wiki it says

skypopen_chat supports full utf8 chat text through ESL or API; yet in Lua
script I create freeswitc.API() and then executestring("skypopen_chat
interface1 remote_skypeid some_unicode_messsage), the utf8 message was not
received correctly.

Does the wiki "API" not referring to freeswitch.API in Lua?


> Date: Mon, 19 Nov 2012 12:19:30 +0100
> From: gmaruzz at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] How to enable unicode in CLI command?
>
> from CLI you can't.
> You can use ESL, eg telnet to port 8021 (IIRC).
> Anyway, look in the wiki page of gsmopen and/or skypopen for an example
howto.
> Via ESL unicode is supported.
> -giovanni
>
> On 11/18/12, ?? <itispip-qq at hotmail.com> wrote:
> >
> > I tried to use chat / skypopen_chat from CLI console to send a message
to
> > receiver, the message is written by unicode charactor (Korean or Thai),
the
> > receiver cannot get correct charators;
> > Tried to call chat / skypopen_chat API from LUA/JScript, ended the same;
> > I believe Freeswitch itself support unicode, as when I use an SIP client
as
> > "Bria" to send SIP chat message between 2 clients, unicode charactor
> > displays correctly.
> > Is it just the CLI command by design not support unicode, or I have to
do
> > something in config xml?
> >
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lazyvirus at gmx.com  Wed Nov 21 02:15:34 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Wed, 21 Nov 2012 00:15:34 +0100
Subject: [Freeswitch-users] How to enable unicode in CLI command?
In-Reply-To: <CALXCt0pVhw_B+RyRvNEKouHDeULk_DV7pihVKHP=-2FbV-moJA@mail.gmail.com>
References: <BLU169-W654F5F82AD03E6AC000B8F8C570@phx.gbl>
	<CALXCt0p9dR63AfScg5V6=4jugWCnEharx=tOKLLtXqrzXnf7eA@mail.gmail.com>
	<BLU169-W270C6B026B66CA55088BCA8C550@phx.gbl>
	<CALXCt0pVhw_B+RyRvNEKouHDeULk_DV7pihVKHP=-2FbV-moJA@mail.gmail.com>
Message-ID: <20121121001534.47397cc8@anubis.defcon1>

On Tue, 20 Nov 2012 21:30:00 +0100
Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:

> No idea if FreeSWITCH Lua supports unicode.
> 
> Maybe you can write another post, asking if Lua in FreeSWITCH supports
> unicode...

Yes, it does.

-- 
Debian Hint #19: If you're interested in building packages from source, you
should consider installing the apt-src package.


From 8f27e956 at gmail.com  Wed Nov 21 02:33:20 2012
From: 8f27e956 at gmail.com (Scott)
Date: Tue, 20 Nov 2012 18:33:20 -0500
Subject: [Freeswitch-users] Which one of the force-<>-domain params?
Message-ID: <CA+Y7Lf6+jC6ZSLt+GhTTWjcmpcMkzNst9Gy4itdippSCmZuxsQ@mail.gmail.com>

Hi, just about 100% feature-function:feature-function migrated off of that
other platform.  Bit of a learning curve but loving freeSWITCH.

Thanks for just a great project!

All is working.  Our edge router's NAT is very robust and configurable
(openBSD's pf) and we're using STUN.  However, in comparing the old sip
traces to
the fs sip traces, we have noticed something that I just don't know what to
tinker with.

With regard to the following fs sip trace output frag,

Contact: <sip:gw+trunk-voipms at 192.168.2.3
:5080;transport=udp;gw=trunk-voipms>
To: <sip:99.1.2.3:5080>;tag=0myeUN32NySac

With the 'other' config, we'd instead see,

Contact: <sip:gw+trunk-voipms at mydomain.ca
:5060;transport=udp;gw=trunk-voipms>
To: <sip:mydomain.ca:5060>;tag=0myeUN32NySac

Rolling forward, we'd prefer the mydomain.ca flavor.

We operate a DNS with split horizon and SRV records, meaning
public-outside, in fact, see's/resolves 99.1.2.3, and inside hosts and end
points, in fact, see/resolve 192.168.2.3.

We think it's one of the force-<>-domain or db-domain params but (our bad)
cannot "get it right."

Any thoughts as to the correct fs param="?" sip such that the fs
sip-domain-string is used where that sip-domain-string that also is
properly glued to a matching (resolvable) DNS-string?

Thanks,
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From 8f27e956 at gmail.com  Wed Nov 21 02:35:57 2012
From: 8f27e956 at gmail.com (Scott)
Date: Tue, 20 Nov 2012 18:35:57 -0500
Subject: [Freeswitch-users] Cluecon 2012 and Gemeinshaft 5.0
In-Reply-To: <CAKzWOxX=VxabdzH_r_9yQw8nL-3ej21phzbGb9=-jei5move+A@mail.gmail.com>
References: <CACuV5sBTDx6Pea82yhy8N2c4S25kdPaXtJ5OSkcBrorbJ6+cww@mail.gmail.com>
	<CAKzWOxX=VxabdzH_r_9yQw8nL-3ej21phzbGb9=-jei5move+A@mail.gmail.com>
Message-ID: <CA+Y7Lf6mmGqtq96sFcP0kyXjhm5MKnTqffvCvcc-PZEH_ZB7PA@mail.gmail.com>

HI Michael,

Is the 2012 Security presentation also included "in the works?"

:-)

Thanks,

On 20 November 2012 14:49, Michael Collins <msc at freeswitch.org> wrote:

> Actually, the audio and video recordings are in the works. We'll keep you
> all posted. I know we keep saying that but really they are!
> -MC
>
>
> On Tue, Nov 20, 2012 at 3:13 AM, Zenny <garbytrash at gmail.com> wrote:
>
>> Hi:
>>
>> I read at the Cluecon 2012 link below
>> http://www.cluecon.com/presentation/gemeinschaft-50-one-size-fits-all/
>> that Gemeinshaft 5.0 has already been released, but neither there is
>> any announcment like http://www.freeswitch.org/node/373 nor any
>> pointer in the official site amooma.de?
>>
>> Has anyone clue about this ClueCon event? Or is there any audio or
>> video recording of that specific session of Stefan? If yes, please
>> share. Thanks!
>>
>> /zenny
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Nov 21 03:47:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Nov 2012 16:47:21 -0800
Subject: [Freeswitch-users] Return code from ESL Message Sending
In-Reply-To: <50ABF580.40502@gmail.com>
References: <mailman.1329.1352570404.576.freeswitch-users@lists.freeswitch.org>
	<3A82F56B-1332-485E-9F6C-D2126A859CC8@edge-net.net>
	<CACyT-3mMQ8RJfD7J=X8ZZubgR5kKbUeRu_Lj_wYa=T3v+mQmJA@mail.gmail.com>
	<1388879B-963F-4127-B884-965671CD720D@edge-net.net>
	<50ABF580.40502@gmail.com>
Message-ID: <CAKzWOxUM-9-XxbY1L0CVZ8xnXxYic2978WaDVQuuXS-zp5OOTQ@mail.gmail.com>

Please do! If you have any questions about using the wiki please let me
know. Also, I will be doing a wiki tutorial during tomorrow's weekly
conference call.

-MC

On Tue, Nov 20, 2012 at 1:26 PM, Abaci <abaci64 at gmail.com> wrote:

>  Would you mind documenting these option on the wiki (
> http://wiki.freeswitch.org/wiki/Mod_sms) so that people know about it.
> Thanks
>
> On 11/20/2012 1:43 PM, Eli Burke wrote:
>
> Kurtis,
>
>  We've been working with FreeSWITCH Consulting to address some issues
> with MESSAGE delivery. A couple of patches were committed on Nov 13 and Nov
> 14 to trunk and they may help with your problem. These patches affect the
> following behavior:
> * MESSAGEs fed through the chatplan are correctly delivered or ignored by
> sofia
> * when blocking=False, "Delivery-Failure" is replaced with
> "Nonblocking-Delivery: true"
> * when blocking=True, "Delivery-Failure" is correctly set to true or false
> * when blocking=True, "Delivery-Result-Code" is added to the event
>
>  Some background explanation: MESSAGEs are normally delivered in
> non-blocking mode, which means FreeSWITCH makes no attempt to determine if
> they were successfully received. There is a variable that can be set
> ("blocking: true") to force FreeSWITCH to wait for a response. You can
> already see this in action using the chat command in fs_cli-- it will
> report success or failure.
>
>  Unfortunately, "blocking" is not set by default. RIght now, the only way
> to get this behavior is to set it manually. For example, a chatplan rule to
> add the header to all inbound MESSAGEs:
>      <extension name="add_blocking" continue="true">
>       <condition>
>           <action application="set"
> data="is_reg=${sofia_contact(${to_user}" inline="true"/>
>           <action application="set" data="blocking=true"/>
>       </condition>
>     </extension>
>
>  There is a potential (and untested!) downside to forcing blocking to be
> always-on. The MESSAGE delivery queue is currently handled by a single
> thread. Even if all MESSAGE objects are delivered successfully to the local
> switch, some amount of latency may be introduced. In a real-world
> high-throughput scenario, it's possible that this could cause noticeable
> delays in the time it takes to delivery a MESSAGE, creating an ever-growing
> backlog.
>
>  The "is_reg" variable in the rule above could be used to short circuit
> failed attempts by shunting MESSAGEs to a database, or dropping them on the
> floor, but this would not necessarily fix things. The good news is that if
> a high-volume user can demonstrate that there is a problem, it's fixable
> within FreeSWITCH by moving to a multi-threaded message delivery queue.
>
>  -Eli
>
>
>  On Nov 10, 2012, at 1:00 PM,
> freeswitch-users-request at lists.freeswitch.org wrote:
>
>   *From: *Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> *Subject: **[Freeswitch-users] Return code from ESL Message Sending*
>> *Date: *November 9, 2012 11:42:19 PM EST
>> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>
>>   Hello Freeswitch Users:
>>
>> We're currently trying to get the return code from a MESSAGE we send
>> using ESL. The closest we've found is this jira:
>> http://jira.freeswitch.org/browse/FS-4453 which seems to provide similar
>> functionality for the chat command, but nothing for ESL.
>>
>> Here's a pastebin of our current code:
>> http://pastebin.freeswitch.org/20201
>>
>> The server we are hitting is returning a "415 Unsupported Content Type"
>> (which is correct) and we're trying to discover that in freeswitch, instead
>> of assuming the message was received correctly. Right now, we get that the
>> recvEventTimed is returning None. This is all done on the a pull of FS from
>> yesterday.
>>
>>  Any suggestions?
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From packetandy at gmail.com  Wed Nov 21 04:00:34 2012
From: packetandy at gmail.com (andy)
Date: Tue, 20 Nov 2012 17:00:34 -0800
Subject: [Freeswitch-users] changing vm default announcement
In-Reply-To: <mailman.1957.1353348870.576.freeswitch-users@lists.freeswitch.org>
References: <mailman.1957.1353348870.576.freeswitch-users@lists.freeswitch.org>
Message-ID: <50AC27B2.9080905@gmail.com>

hi All,

thanks for the input, in the end I figured out the problem I have been 
having -

The following works as intended:

<action application="set" data="voicemail_greeting_number=1"/>
<action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>

However, in a lot of the of the example code,for instance default.xml for ext. 1000-1019, this is how voicemail is reached:

<action application="bridge" data="loopback/app=voicemail:default 
${domain_name} ${dialed_extension}"/>

In the second case,  <action application="set" 
data="voicemail_greeting_number=1"/>  will not work, probably because 
the channel variable is not set for the correct channel. I am guessing 
that "export" instead of "set" may fix this but I did not have a chance 
to try.

I would suggest that some clarification in the wiki between the two ways 
of calling voicemail would be very useful - when do you "bridge" and 
when do you just call the voicemail app?



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From sdevoy at bizfocused.com  Wed Nov 21 04:01:07 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 20 Nov 2012 20:01:07 -0500
Subject: [Freeswitch-users] Avaya SIP Phones?
Message-ID: <031701cdc783$b0eae640$12c0b2c0$@bizfocused.com>

Hi,

 

We have a potential customer asking us to give them a bid with AVAYA
phones!!!

 

Does AVAYA have any true SIP phones?  Do they work with FreeSwitch?

 

Thanks,

Sean

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From garbytrash at gmail.com  Wed Nov 21 04:10:28 2012
From: garbytrash at gmail.com (Zenny)
Date: Wed, 21 Nov 2012 02:10:28 +0100
Subject: [Freeswitch-users] Cluecon 2012 and Gemeinshaft 5.0
In-Reply-To: <CAKzWOxX=VxabdzH_r_9yQw8nL-3ej21phzbGb9=-jei5move+A@mail.gmail.com>
References: <CACuV5sBTDx6Pea82yhy8N2c4S25kdPaXtJ5OSkcBrorbJ6+cww@mail.gmail.com>
	<CAKzWOxX=VxabdzH_r_9yQw8nL-3ej21phzbGb9=-jei5move+A@mail.gmail.com>
Message-ID: <CACuV5sD9q76hOTkstS4=xEH131W4QAngPb9GWyek=zA7uuhXEg@mail.gmail.com>

Thanks, Michael for an update. I am aware of how time-consuming to
produce videos like that. Look forward to. Thanks in advance!

On 11/20/12, Michael Collins <msc at freeswitch.org> wrote:
> Actually, the audio and video recordings are in the works. We'll keep you
> all posted. I know we keep saying that but really they are!
> -MC
>
> On Tue, Nov 20, 2012 at 3:13 AM, Zenny <garbytrash at gmail.com> wrote:
>
>> Hi:
>>
>> I read at the Cluecon 2012 link below
>> http://www.cluecon.com/presentation/gemeinschaft-50-one-size-fits-all/
>> that Gemeinshaft 5.0 has already been released, but neither there is
>> any announcment like http://www.freeswitch.org/node/373 nor any
>> pointer in the official site amooma.de?
>>
>> Has anyone clue about this ClueCon event? Or is there any audio or
>> video recording of that specific session of Stefan? If yes, please
>> share. Thanks!
>>
>> /zenny
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>


From msc at freeswitch.org  Wed Nov 21 04:13:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Nov 2012 17:13:55 -0800
Subject: [Freeswitch-users] Which one of the force-<>-domain params?
In-Reply-To: <CA+Y7Lf6+jC6ZSLt+GhTTWjcmpcMkzNst9Gy4itdippSCmZuxsQ@mail.gmail.com>
References: <CA+Y7Lf6+jC6ZSLt+GhTTWjcmpcMkzNst9Gy4itdippSCmZuxsQ@mail.gmail.com>
Message-ID: <CAKzWOxWJKt_WMPofh48Uh+jR2puTVG5XNhTL7=fTuc35udz6sQ@mail.gmail.com>

Is this an outbound call made from a local phone out a gateway that you
configured?
-MC

On Tue, Nov 20, 2012 at 3:33 PM, Scott <8f27e956 at gmail.com> wrote:

> Hi, just about 100% feature-function:feature-function migrated off of that
> other platform.  Bit of a learning curve but loving freeSWITCH.
>
> Thanks for just a great project!
>
> All is working.  Our edge router's NAT is very robust and configurable
> (openBSD's pf) and we're using STUN.  However, in comparing the old sip
> traces to
> the fs sip traces, we have noticed something that I just don't know what
> to tinker with.
>
> With regard to the following fs sip trace output frag,
>
> Contact: <sip:gw+trunk-voipms at 192.168.2.3
> :5080;transport=udp;gw=trunk-voipms>
> To: <sip:99.1.2.3:5080>;tag=0myeUN32NySac
>
> With the 'other' config, we'd instead see,
>
> Contact: <sip:gw+trunk-voipms at mydomain.ca
> :5060;transport=udp;gw=trunk-voipms>
> To: <sip:mydomain.ca:5060>;tag=0myeUN32NySac
>
> Rolling forward, we'd prefer the mydomain.ca flavor.
>
> We operate a DNS with split horizon and SRV records, meaning
> public-outside, in fact, see's/resolves 99.1.2.3, and inside hosts and end
> points, in fact, see/resolve 192.168.2.3.
>
> We think it's one of the force-<>-domain or db-domain params but (our bad)
> cannot "get it right."
>
> Any thoughts as to the correct fs param="?" sip such that the fs
> sip-domain-string is used where that sip-domain-string that also is
> properly glued to a matching (resolvable) DNS-string?
>
> Thanks,
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jason.holden at start.ca  Wed Nov 21 06:13:38 2012
From: jason.holden at start.ca (Jason Holden)
Date: Tue, 20 Nov 2012 22:13:38 -0500
Subject: [Freeswitch-users] directory information pulled from db table
	rather then xml files
Message-ID: <B834F68C90C940369BBDD5E8B6A10722@bob>

Does anyone know if this is possible?

If so can you provide a web link with information?

I was unable to find any details on this in the wiki.

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From mnrao2001 at gmail.com  Wed Nov 21 07:23:51 2012
From: mnrao2001 at gmail.com (Nageshwara Rao Moova)
Date: Wed, 21 Nov 2012 09:53:51 +0530
Subject: [Freeswitch-users] Log Rollover issue
In-Reply-To: <CAKbxfG8krEgxHPWeWi263ubeiducGnsvPUBJhKs=K-JxtX-9Gg@mail.gmail.com>
References: <CAPDWOejFaGi6OmavH2Q9nKYRzn6zpz1gDbiSVqEcaCV678+bbg@mail.gmail.com>
	<CAKbxfG8krEgxHPWeWi263ubeiducGnsvPUBJhKs=K-JxtX-9Gg@mail.gmail.com>
Message-ID: <CAPDWOehb7p9JbO8DLqU=sYqSjx_v7jPumuR2zbQ47Wp_fggSSg@mail.gmail.com>

Hi Anthony,

I doubt any permission issue as the failure happened not initially but
after a while. The user we are using has sufficient permissions for doing
rwx operations. Below is the log snippet from freeswitch log.

If it?s any help, the freeswitch log had entries like this every few
seconds:

2012-10-26 11:53:16.653473 [CRIT] mod_logfile.c:164 Error renaming log from
/usr/local/freeswitch/log/freeswitch.log.40 to
/usr/local/freeswitch/log/freeswitch.log.41

??

2012-10-27 00:19:30.313497 [CRIT] mod_logfile.c:164 Error renaming log from
/usr/local/freeswitch/log/freeswitch.log.40 to
/usr/local/freeswitch/log/freeswitch.log.41


Until it ran out of disk space and crashed. No reason why it couldn?t
rename the file, though.




On Tue, Nov 20, 2012 at 1:50 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> When it says it cannot, is that because of file permissions? Dos FS have
> permission to write to the directory?
> Can you reproduce this with logs and if so can you report it to Jira
> http://jira.freeswitch.org
>
>
>
> On Mon, Nov 19, 2012 at 6:45 AM, Nageshwara Rao Moova <mnrao2001 at gmail.com
> > wrote:
>
>> Hi all,
>>
>> I have modified my default logconf file for rollover to be restricted to
>> 100. But sometimes freeswitch is unable to rename rollover say ?cannot
>> rename freeswitch.log.41? and fails. But the fail follows with serious
>> issue of writing all the logs to "freeswitch.log" and ends up filling the
>> disk space.
>>
>> I have not change the default file size i.e 10MB.
>>
>> How does 10MB size overridden by freeswitch?
>>
>> --
>> regards & thanks
>> --
>> m nageshwara rao
>> 99891 86280
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 & thanks
--
m nageshwara rao
99891 86280
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From kris at kriskinc.com  Wed Nov 21 07:49:52 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 20 Nov 2012 23:49:52 -0500
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
 contact IP/port
In-Reply-To: <1353443727657-7584773.post@n2.nabble.com>
References: <054001cdc664$92486660$b6d93320$@com>
	<CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
	<1353443727657-7584773.post@n2.nabble.com>
Message-ID: <CAKDfjgcmcA8tp-7WVEzxxHYtUXkyPxXmm3d0dSQqV54CNBwsRQ@mail.gmail.com>

You can rewrite the SDP to anything you want but that doesn't mean
you'll get media.

I suggest you do some research on the two main schools for SIP NAT
traversal (far end and near end).  In a nutshell, far end NAT
traversal is implemented by the server.  This is the FreeSWITCH
default.  The SDPs are rewritten to the address of the FreeSWITCH
server and passed to the remote endpoint.  This causes the media to be
relayed at the server between the endpoints.  This has the advantage
of being universally compatible - regardless of how dumb the
endpoint(s) is/are.  Disadvantages are many - increased sever load,
bandwidth, latency, etc.  However, with many endpoints this is the
only choice.

Near end NAT traversal relies on advances technologies such as STUN,
TURN, and ICE to be supported by the client (and the servers to enable
them configured and available).  TURN actually blurs the lines between
these two strategies, as do various hybrid approaches using local
proxies, etc but that's for another day.

Bypass media is incompatible with NAT unless you're using
STUN/TURN/ICE in your clients and even then I'm not sure FreeSWITCH
will completely and cleanly handle the various ICE/SIP/SDP
interactions.

On Tue, Nov 20, 2012 at 3:35 PM, PhilQ <philq at qsystemsengineering.com> wrote:
> We're running in bypass media mode with aggressive-nat-detection enabled.  FS
> handles the signaling but passes the internal 10.x.x.x addresses for media
> along unchanged, even between different sites.
>
> Any recommendations for SIP proxies if FS can't do it automatically?
> Rewriting the SDP from the dialplan might be worth looking into depending
> upon how it's implemented.
>
> Thanks for the info.
>
> - Phil
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Rewriting-media-address-in-SDP-as-well-as-contact-IP-port-tp7584712p7584773.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From yehavi.bourvine at gmail.com  Wed Nov 21 08:05:03 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 21 Nov 2012 07:05:03 +0200
Subject: [Freeswitch-users] directory information pulled from db table
 rather then xml files
In-Reply-To: <B834F68C90C940369BBDD5E8B6A10722@bob>
References: <B834F68C90C940369BBDD5E8B6A10722@bob>
Message-ID: <CAAtWKeHRf2n-qy17pOP+euiA6nMpmTLJ0MbDuZgY-Xvz0rPG-Q@mail.gmail.com>

You can't do that directly but need some agent to do it. See mod_curl_xml
in the WIKI. It is quite simple.

                  __Yehavi:

2012/11/21 Jason Holden <jason.holden at start.ca>

>  Does anyone know if this is possible?****
>
> If so can you provide a web link with information?****
>
> I was unable to find any details on this in the wiki.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mnrao2001 at gmail.com  Wed Nov 21 09:21:18 2012
From: mnrao2001 at gmail.com (Nageshwara Rao Moova)
Date: Wed, 21 Nov 2012 11:51:18 +0530
Subject: [Freeswitch-users] FreeTDM conf to support NSF
Message-ID: <CAPDWOejmch+QFYpvjtSBd8gj6ehm=HFNGXk9gpsy=MeNd4j=yw@mail.gmail.com>

Hi All,

We need to modify our freetdm dialer for outbound call when using AT&T
5ESS, which requires prefixing freetdm_isdn.netFac.spec channel variable to
freetdm.

Is it possible to set "nsf" in freetdm.conf per span. We need this option
because we will have assorted spans connected to the board.

Does freeTDM support similar conf. shown below of  zapata.conf? Is there
any configuration file for freeTDM to add this and use it?

Some switches (AT&T especially) require network specific facility IE
supported values are currently 'none', 'sdn', 'megacom', 'accunet'
nsf=none

;Sangoma A102 port 1 [slot:1 bus:3 span: 1]
context=from-outside
group=1
switchtype => 4ess*nsf=megacom*
signalling => pri_cpe
pridialplan => unknown
channel => 1-23
use_callerid => yes



-- 
regards & thanks
--
m nageshwara rao
99891 86280
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From yiftah at choochee.com  Wed Nov 21 06:45:40 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Tue, 20 Nov 2012 19:45:40 -0800
Subject: [Freeswitch-users] Avaya SIP Phones?
In-Reply-To: <031701cdc783$b0eae640$12c0b2c0$@bizfocused.com>
References: <031701cdc783$b0eae640$12c0b2c0$@bizfocused.com>
Message-ID: <CAFVUaV_y6JzbMJFjrn7Vu+AWHBEpHEO3ctVn2raCBMF0zQV5Og@mail.gmail.com>

I use to work for AVAYA, yes they have many SIP phones all the 96xx series
which is very popular
They have AVAYA sip implementation (which is very far from the standard) I
did not see any FreeSWITCH implementation for it


On Tue, Nov 20, 2012 at 5:01 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,****
>
> ** **
>
> We have a potential customer asking us to give them a bid with AVAYA
> phones!!!****
>
> ** **
>
> Does AVAYA have any true SIP phones?  Do they work with FreeSwitch?****
>
> ** **
>
> Thanks,****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yiftah at choochee.com  Wed Nov 21 08:22:06 2012
From: yiftah at choochee.com (Yiftach Golan)
Date: Tue, 20 Nov 2012 21:22:06 -0800
Subject: [Freeswitch-users] directory information pulled from db table
 rather then xml files
In-Reply-To: <B834F68C90C940369BBDD5E8B6A10722@bob>
References: <B834F68C90C940369BBDD5E8B6A10722@bob>
Message-ID: <CAFVUaV-BcviqyeajgkN53kdVT=c3WnfRmowDvLx90ONrJCVQmw@mail.gmail.com>

I was using the ${db(select/realm/key)} but for some reason it did not work
for variable in directory and I ended up adding the hook for variable as
well
http://wiki.freeswitch.org/wiki/Mod_db


On Tue, Nov 20, 2012 at 7:14 PM, Jason Holden <jason.holden at start.ca> wrote:

>  Does anyone know if this is possible?****
>
> If so can you provide a web link with information?****
>
> I was unable to find any details on this in the wiki.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Wed Nov 21 10:16:19 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 21 Nov 2012 09:16:19 +0200
Subject: [Freeswitch-users] directory information pulled from db table
 rather then xml files
In-Reply-To: <CAAtWKeHRf2n-qy17pOP+euiA6nMpmTLJ0MbDuZgY-Xvz0rPG-Q@mail.gmail.com>
References: <B834F68C90C940369BBDD5E8B6A10722@bob>
	<CAAtWKeHRf2n-qy17pOP+euiA6nMpmTLJ0MbDuZgY-Xvz0rPG-Q@mail.gmail.com>
Message-ID: <CALNZuwk-kGHMNUYcH1Mp8v7a_C2VDgKLX0mF6xg3WhTUE-PUBQ@mail.gmail.com>

I thought there was a link wiki entry with a Lua script that did this, but
again, I can't find it.
Also, fusionpbx has bundled with it a lua script that does this.

-Avi


On Wed, Nov 21, 2012 at 7:05 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> You can't do that directly but need some agent to do it. See mod_curl_xml
> in the WIKI. It is quite simple.
>
>                   __Yehavi:
>
> 2012/11/21 Jason Holden <jason.holden at start.ca>
>
>>   Does anyone know if this is possible?****
>>
>> If so can you provide a web link with information?****
>>
>> I was unable to find any details on this in the wiki.****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chaiyawut.so at gmail.com  Wed Nov 21 11:38:21 2012
From: chaiyawut.so at gmail.com (chaiyawut.so)
Date: Wed, 21 Nov 2012 00:38:21 -0800 (PST)
Subject: [Freeswitch-users] Can not use Python ESL with Django and apache
	web server
Message-ID: <1353487101708-7584800.post@n2.nabble.com>

I tried freeswitch python ESL library with Django and apache mod_wsgi today.
I already compiled and installed pymod in Freeswitch source folder. My
freeswitch server runs as root and my DJnago web server runs as www-data and
mod_wsgi. When used command "ESLconnection("localhost", "8021", "ClueCon")"
in my views.py to connect to freeswitch server, django froze permanently and
it said waiting for the request...



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-not-use-Python-ESL-with-Django-and-apache-web-server-tp7584800.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From stkn at openisdn.net  Wed Nov 21 12:23:28 2012
From: stkn at openisdn.net (Stefan Knoblich)
Date: Wed, 21 Nov 2012 10:23:28 +0100
Subject: [Freeswitch-users] FreeTDM conf to support NSF
In-Reply-To: <CAPDWOejmch+QFYpvjtSBd8gj6ehm=HFNGXk9gpsy=MeNd4j=yw@mail.gmail.com>
References: <CAPDWOejmch+QFYpvjtSBd8gj6ehm=HFNGXk9gpsy=MeNd4j=yw@mail.gmail.com>
Message-ID: <50AC9D90.7060705@openisdn.net>

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

On 21.11.2012 07:21, Nageshwara Rao Moova wrote:
> We need to modify our freetdm dialer for outbound call when using AT&T 5ESS, which requires prefixing freetdm_isdn.netFac.spec channel variable to
> freetdm.
> 
> Is it possible to set "nsf" in freetdm.conf per span. We need this option because we will have assorted spans connected to the board.
> 
> Does freeTDM support similar conf. shown below of  zapata.conf? Is there any configuration file for freeTDM to add this and use it?
> 
> Some switches (AT&T especially) require network specific facility IE supported values are currently 'none', 'sdn', 'megacom', 'accunet' nsf=none

Which type of span? libpri_spans (= ftmod_libpri) or sangoma_pri_spans (= ftmod_sng_isdn)?

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iEYEARECAAYFAlCsnZAACgkQjiIIAK4rYUqn5gCggvtyHk+h9LKnpvQJpYbw2mPd
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From stkn at openisdn.net  Wed Nov 21 13:11:23 2012
From: stkn at openisdn.net (Stefan Knoblich)
Date: Wed, 21 Nov 2012 11:11:23 +0100
Subject: [Freeswitch-users] FreeTDM conf to support NSF
In-Reply-To: <50AC9D90.7060705@openisdn.net>
References: <CAPDWOejmch+QFYpvjtSBd8gj6ehm=HFNGXk9gpsy=MeNd4j=yw@mail.gmail.com>
	<50AC9D90.7060705@openisdn.net>
Message-ID: <50ACA8CB.6050409@openisdn.net>

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

On 21.11.2012 10:50, Nageshwara Rao Moova wrote:
> Our span type is sangoma_pri_spans (= ftmod_sng_isdn)
> 
> 
> On Wed, Nov 21, 2012 at 2:53 PM, Stefan Knoblich <stkn at openisdn.net <mailto:stkn at openisdn.net>> wrote:
> 
> On 21.11.2012 07:21, Nageshwara Rao Moova wrote:
>> We need to modify our freetdm dialer for outbound call when using AT&T 5ESS, which requires prefixing freetdm_isdn.netFac.spec channel variable to 
>> freetdm.
> 
>> Is it possible to set "nsf" in freetdm.conf per span. We need this option because we will have assorted spans connected to the board.
> 
>> Does freeTDM support similar conf. shown below of  zapata.conf? Is there any configuration file for freeTDM to add this and use it?
> 
>> Some switches (AT&T especially) require network specific facility IE supported values are currently 'none', 'sdn', 'megacom', 'accunet' nsf=none
> 
> Which type of span? libpri_spans (= ftmod_libpri) or sangoma_pri_spans (= ftmod_sng_isdn)?


Please keep this discussion on the mailinglist, by replying to the list or all.

I don't see anything related to NSF / megacom in the configuration settings of ftmod_sangoma_isdn,
maybe one of the sangoma devs knows more...

[CCing moy]
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=fs44
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From andrew at cassidywebservices.co.uk  Wed Nov 21 13:31:07 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 21 Nov 2012 10:31:07 +0000
Subject: [Freeswitch-users] Can not use Python ESL with Django and
 apache web server
In-Reply-To: <1353487101708-7584800.post@n2.nabble.com>
References: <1353487101708-7584800.post@n2.nabble.com>
Message-ID: <CAMbW9oJpkmfrsZAh8QOLaRydZibP+YjhHEJYJrnfDS9SPbqp9Q@mail.gmail.com>

Last time I had a similar problem I had to increase the number of wsgi
worker processes. Use the processes=n option to the WSGIDaemonProcess
directive.

On 21 November 2012 08:38, chaiyawut.so <chaiyawut.so at gmail.com> wrote:

> I tried freeswitch python ESL library with Django and apache mod_wsgi
> today.
> I already compiled and installed pymod in Freeswitch source folder. My
> freeswitch server runs as root and my DJnago web server runs as www-data
> and
> mod_wsgi. When used command "ESLconnection("localhost", "8021", "ClueCon")"
> in my views.py to connect to freeswitch server, django froze permanently
> and
> it said waiting for the request...
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Can-not-use-Python-ESL-with-Django-and-apache-web-server-tp7584800.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From yehavi.bourvine at gmail.com  Wed Nov 21 16:08:02 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 21 Nov 2012 15:08:02 +0200
Subject: [Freeswitch-users] SRTP - how to enable on the other leg?
Message-ID: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>

Hi,

  I am trying to use SRTP without TLS. The initiating side asks and get
SRTP (two way) while the other side is not offered SRTP at all.

I've noted that ${sip_has_crypto} is null. I also tried setting
unconditionaly sip_secure_rtp (both set and export), but it doesn't help.

Any idea what I am doing wrong?

BTW, I am using today's mornning GIT.

                  Thanks, __Yehavi:
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From clive18 at webmail.co.za  Wed Nov 21 17:00:27 2012
From: clive18 at webmail.co.za (clive engelberg)
Date: Wed, 21 Nov 2012 16:00:27 +0200
Subject: [Freeswitch-users] NOTIFY gets a 481 error response
Message-ID: <589ee2d85117015931c187ff908e428c@www.webmail.co.za>

Hi guys.

Has anyone had this issue where an endpoint does not like the NOTIFY request.
?

I am posting the sip trace below.

Thanks in advance
Clive

(NOTIFY sent from Freeswitch)
U 2012/11/21 15:39:13.562784 196.41.2.211:5060 -> 41.150.16.236:5060
NOTIFY sip:253118877988 at 41.150.16.236:5060 SIP/2.0.
Via: SIP/2.0/UDP 196.41.2.211;rport;branch=z9hG4bK2KN80mQ12DFZQ.
Max-Forwards: 70.
From: <sip:253118877988 at 196.41.2.211>;tag=1N243vyr38aUS.
To: <sip:253118877988 at 196.41.2.211>.
Call-ID: adabb26c-ae83-1230-adb4-001aa01d9e0c.
CSeq: 36416512 NOTIFY.
Contact: <sip:mod_sofia at 196.41.2.211:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-073f9c3 2010-05-03 13-46-57
-0500.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: message-summary.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=timeout.
Content-Type: application/simple-message-summary.
Content-Length: 72.
.
Messages-Waiting: no.
Message-Account: sip:253118877988 at 196.41.2.211.
.

(Response)
U 2012/11/21 15:39:13.740207 41.150.16.236:5060 -> 196.41.2.211:5060
SIP/2.0 481 Call/Transaction Does Not Exist.
Via: SIP/2.0/UDP 196.41.2.211;rport;branch=z9hG4bK2KN80mQ12DFZQ.
From: <sip:253118877988 at 196.41.2.211>;tag=1N243vyr38aUS.
To: <sip:253118877988 at 196.41.2.211>.
Call-ID: adabb26c-ae83-1230-adb4-001aa01d9e0c.
CSeq: 36416512 NOTIFY.
Server: KAREL-Sip-Trunk v. AAE_16 (ms48).
Supported: timer,replaces.
Contact: <sip:KAREL_trunk at 41.150.16.236:5060>.
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,UPDATE,REFER,NOTIFY.
Content-Length: 0.


____________________________________________________________
South Africas premier free email service - www.webmail.co.za 

DHL Express. Simple, secure and affordable.
http://www.postnet.co.za/index.php?option=com_content&view=article&id=74&Itemid=72




From chetan.khatri at panamaxil.com  Wed Nov 21 17:48:36 2012
From: chetan.khatri at panamaxil.com (Chetan Khatri)
Date: Wed, 21 Nov 2012 20:18:36 +0530
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
Message-ID: <50ACE9C4.60803@panamaxil.com>

Dear All Experts,

Need help to solve freeswitch's mod_rad_auth capacity issue.
I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The 
Module is working fine till the concurrent request reaches till 245 
after that Accounting is failing.
Below is the Configuration.

FreeSwitch : 1.0.6
RADIUS Server : FreeRADIUS2
Database server : MySQL 5.5

We are stress testing the with freeswitch to handle about 2000 
concurrent calls with RADIUS AAA. without using RADIUS modules 
freeswitch is able to handle 2000 concurrent calls without any issues.
For analysis we analysed from RADIUS server replies access-accept 
request, but in mod_rad_auth error says *"freeswitch : Invalid Digest 
Request from RADIUS server"* only if the concurrent session receives 
245, till that all the requests are been working fine.

Please guide me how to resolve this capacity issue, I can provide all 
the logs and configuration files if required.

-- 
/Regards,
Chetan Khatri/
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From brian at freeswitch.org  Wed Nov 21 17:55:08 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Nov 2012 08:55:08 -0600
Subject: [Freeswitch-users] SRTP - how to enable on the other leg?
In-Reply-To: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
References: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
Message-ID: <E3BEF350-174D-4BF0-932B-38E77333D08A@freeswitch.org>

you make sure sip_secure_media=true is set inside {} on the originate or export the variable.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 21, 2012, at 7:08 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:

> Hi,
>  
>   I am trying to use SRTP without TLS. The initiating side asks and get SRTP (two way) while the other side is not offered SRTP at all.
>  
> I've noted that ${sip_has_crypto} is null. I also tried setting unconditionaly sip_secure_rtp (both set and export), but it doesn't help.
>  
> Any idea what I am doing wrong?
>  
> BTW, I am using today's mornning GIT.
>  
>                   Thanks, __Yehavi:



From brian at freeswitch.org  Wed Nov 21 17:59:10 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Nov 2012 08:59:10 -0600
Subject: [Freeswitch-users] Which one of the force-<>-domain params?
In-Reply-To: <CA+Y7Lf6+jC6ZSLt+GhTTWjcmpcMkzNst9Gy4itdippSCmZuxsQ@mail.gmail.com>
References: <CA+Y7Lf6+jC6ZSLt+GhTTWjcmpcMkzNst9Gy4itdippSCmZuxsQ@mail.gmail.com>
Message-ID: <542AACB8-147B-4D9F-9E18-56937F859CA9@freeswitch.org>

You shouldn't, because it doesn't matter.  If you're doing this for aesthetic then you're failing.   You're only going to add additional delays in all transactions doing SRV lookups for that vanity.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 20, 2012, at 5:33 PM, Scott <8f27e956 at gmail.com> wrote:

> Rolling forward, we'd prefer the mydomain.ca flavor.



From shaheryarkh at gmail.com  Wed Nov 21 18:14:44 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Wed, 21 Nov 2012 16:14:44 +0100
Subject: [Freeswitch-users] Freeswitch Multilingual Setup
Message-ID: <CAFZQphz-0b3h7Y67rLBgzkcQre=z+g74RkopnYjL+cPicB=iww@mail.gmail.com>

Hi,

I am test Freeswitch for Multilingual IVR system. I observed that if I set
default_language parameter before "voicemail" or "say" applications, it
correctly play specified application in given language. However it has no
effect on playback application, it always play the IVR in system default
(i.e. English) language IVR.

Here is an example extension,

    <extension name="catchall">
      <condition field="destination_number" expression="^(.*)$">
        <action application="export" data="default_language=fr"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback"
data="voicemail/vm-that_was_an_invalid_ext.wav"/>
        <action application="hangup"/>
      </condition>
    </extension>

Many thanks in advance.

Thank you.
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From sertys at gmail.com  Wed Nov 21 18:37:51 2012
From: sertys at gmail.com (Daniel Ivanov)
Date: Wed, 21 Nov 2012 16:37:51 +0100
Subject: [Freeswitch-users] SRTP - how to enable on the other leg?
In-Reply-To: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
References: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
Message-ID: <CAMeZp62MCMzWKSkSeyDeTYFUvMS=S_sH7b0dYOnTXQAddQ8q=Q@mail.gmail.com>

Try to set nolocal:sip_secure_media to true. Or however the var was called.
Im going from memory here.

On Nov 21, 2012 4:06 PM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
wrote:
>
> Hi,
>
>   I am trying to use SRTP without TLS. The initiating side asks and get
SRTP (two way) while the other side is not offered SRTP at all.
>
> I've noted that ${sip_has_crypto} is null. I also tried setting
unconditionaly sip_secure_rtp (both set and export), but it doesn't help.
>
> Any idea what I am doing wrong?
>
> BTW, I am using today's mornning GIT.
>
>                   Thanks, __Yehavi:
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Nov 21 19:15:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Nov 2012 08:15:34 -0800
Subject: [Freeswitch-users] changing vm default announcement
In-Reply-To: <50AC27B2.9080905@gmail.com>
References: <mailman.1957.1353348870.576.freeswitch-users@lists.freeswitch.org>
	<50AC27B2.9080905@gmail.com>
Message-ID: <CAKzWOxVoa+G_CpgBOrMY74pzjNL+g+ow6XoZ9Tn4kjp53VORrA@mail.gmail.com>

Yes, that makes sense. After you test it and verify it works would you mind
adding that tidbit to the wiki?
-MC

On Tue, Nov 20, 2012 at 5:00 PM, andy <packetandy at gmail.com> wrote:

>  hi All,
>
> thanks for the input, in the end I figured out the problem I have been
> having -
>
> The following works as intended:
>
> <action application="set" data="voicemail_greeting_number=1"/>
> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
>
>
> However, in a lot of the of the example code,for instance default.xml for ext. 1000-1019, this is how voicemail is reached:
>
> <action application="bridge" data="loopback/app=voicemail:default
> ${domain_name} ${dialed_extension}"/>
>
> In the second case,  <action application="set"
> data="voicemail_greeting_number=1"/>  will not work, probably because the
> channel variable is not set for the correct channel. I am guessing that
> "export" instead of "set" may fix this but I did not have a chance to try.
>
> I would suggest that some clarification in the wiki between the two ways
> of calling voicemail would be very useful - when do you "bridge" and when
> do you just call the voicemail app?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From fs-list at communicatefreely.net  Wed Nov 21 19:22:03 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 21 Nov 2012 11:22:03 -0500
Subject: [Freeswitch-users] Freeswitch Multilingual Setup
In-Reply-To: <CAFZQphz-0b3h7Y67rLBgzkcQre=z+g74RkopnYjL+cPicB=iww@mail.gmail.com>
References: <CAFZQphz-0b3h7Y67rLBgzkcQre=z+g74RkopnYjL+cPicB=iww@mail.gmail.com>
Message-ID: <50ACFFAB.6010401@communicatefreely.net>

Hi Muhammed,

That's because the way you are calling the playback application below is calling a 
specific audio file based on path.

In vars.xml (usually) there is a global variable that sets the sound files prefix.  This 
will be added to whatever you put as the argument to the playback application.  I'm 
willing to bet that whatever you have set includes a language path.

There are a few ways around this -

You can make some phrase macros for each language, and define which prompts in which 
language should be used.  Any time you want to play something, use phrase:your_macro instead.

Likewise, you could also change the sounds prefix not to include the language, and then 
call playback like this:

data="${default_language}/voicemail/prompt.wav"

I would recommend the phrase macro method, as that gives you a lot of flexibility with 
what prompts are used for a given language, and it also lets you account for things like 
quantity, gender, and number ordering if that applies (ie. if you are playing an error for 
a wrong number).

Hope that helps.

-Tim

Muhammad Shahzad wrote:
> Hi,
> 
> I am test Freeswitch for Multilingual IVR system. I observed that if I 
> set default_language parameter before "voicemail" or "say" applications, 
> it correctly play specified application in given language. However it 
> has no effect on playback application, it always play the IVR in system 
> default (i.e. English) language IVR.
> 
> Here is an example extension,
> 
>     <extension name="catchall">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="export" data="default_language=fr"/>
>         <action application="answer"/>
>         <action application="sleep" data="1000"/>
>         <action application="playback" 
> data="voicemail/vm-that_was_an_invalid_ext.wav"/>
>         <action application="hangup"/>
>       </condition>
>     </extension>
> 
> Many thanks in advance.
> 
> Thank you.
> 
> 
> ------------------------------------------------------------------------
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Wed Nov 21 19:42:53 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Nov 2012 08:42:53 -0800
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxU=eiWfZE=WqvG3d09stvwq98R65_DAJ64ts7C=7OAkXA@mail.gmail.com>

Hello all,

Today's agenda page is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21

We are going to do a wiki tutorial, specifically how to add a channel
variable to the wiki. We'll also go over script I wrote that produces an
HTML page that lists all the channel variables.

Talk to you soon!


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Nov 21 20:10:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Nov 2012 09:10:09 -0800
Subject: [Freeswitch-users] directory information pulled from db table
 rather then xml files
In-Reply-To: <CAAtWKeHRf2n-qy17pOP+euiA6nMpmTLJ0MbDuZgY-Xvz0rPG-Q@mail.gmail.com>
References: <B834F68C90C940369BBDD5E8B6A10722@bob>
	<CAAtWKeHRf2n-qy17pOP+euiA6nMpmTLJ0MbDuZgY-Xvz0rPG-Q@mail.gmail.com>
Message-ID: <CAKzWOxV_u_k5XSQ-Ux9G0wovOhZde9mc1PU+aqaEdN7bWecJew@mail.gmail.com>

On Tue, Nov 20, 2012 at 9:05 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> You can't do that directly but need some agent to do it. See mod_curl_xml
> in the WIKI. It is quite simple.
>

Well, it's simple if you get the name of the module correct: mod_xml_curl
;)
-MC
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From philq at qsystemsengineering.com  Wed Nov 21 21:25:13 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Wed, 21 Nov 2012 10:25:13 -0800 (PST)
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
In-Reply-To: <E66DCAE9702E4F869A4A2AE4431F8265@bob>
References: <6626166B66164AB4B5C0E344762D7E4A@bob>
	<472138AA0F3C4B01A08CBF5CB2CFFB2D@bob>
	<CAKzWOxVk-tx7NEX2=3JyhP8w-9XqmcpLNL69vs6N2n8bsRr0uA@mail.gmail.com>
	<E66DCAE9702E4F869A4A2AE4431F8265@bob>
Message-ID: <1353522313292-7584814.post@n2.nabble.com>

I'm going to go out on a limb and make a "scientific-wild-assed-guess" since
I can't see your pastebin, but since you're seeing external IP addresses I
have an idea of what might be going on.

Your client/user agent is probably using STUN and is reporting it's external
IP address to FS which then uses that address for replies to the client. 
With most firewall/NAT devices, those replies won't make it back to the
device unless you have a port forwarding rule set since the device treats it
like it's coming from the outside.  If that is indeed the case, then there
are two things you can try:

1.  Disable STUN on your endpoint (probably not the best solution,
especially if you're on a dynamic IP)  OR

2.  Set a port forwarding rule in your firewall/router/nat device so those
repllies make it back.

Without the benefit of the pastebin this is just a wild guess so please
don't sue me if this is not applicable.

Hope everyone has a happy and safe Thanksgiving holiday!

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/retreiving-voicemail-dropping-after-30-seconds-tp7584653p7584814.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed Nov 21 22:04:36 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Nov 2012 13:04:36 -0600
Subject: [Freeswitch-users] Log Rollover issue
In-Reply-To: <CAPDWOehb7p9JbO8DLqU=sYqSjx_v7jPumuR2zbQ47Wp_fggSSg@mail.gmail.com>
References: <CAPDWOejFaGi6OmavH2Q9nKYRzn6zpz1gDbiSVqEcaCV678+bbg@mail.gmail.com>
	<CAKbxfG8krEgxHPWeWi263ubeiducGnsvPUBJhKs=K-JxtX-9Gg@mail.gmail.com>
	<CAPDWOehb7p9JbO8DLqU=sYqSjx_v7jPumuR2zbQ47Wp_fggSSg@mail.gmail.com>
Message-ID: <CAKbxfG_enpxKsbvLxt7ffJ5_DcivMyO68sQWEbW1LQohe3dQ4w@mail.gmail.com>

Are you deleting the old ones?
It was probably already out of disk space when it was refusing to rename
the files.

Looks like you have 40 of them, you may need to auto delete some.

You are not providing much other data to go on so all I can do is guess.
The system call to move the file is failing which could be due to
permissions or disk full or whatever.

If you want you can update to latest and I added a call to strerror(errno)
so you can see the exact reason.




On Tue, Nov 20, 2012 at 10:23 PM, Nageshwara Rao Moova
<mnrao2001 at gmail.com>wrote:

> Hi Anthony,
>
> I doubt any permission issue as the failure happened not initially but
> after a while. The user we are using has sufficient permissions for doing
> rwx operations. Below is the log snippet from freeswitch log.
>
> If it?s any help, the freeswitch log had entries like this every few
> seconds:
>
> 2012-10-26 11:53:16.653473 [CRIT] mod_logfile.c:164 Error renaming log
> from /usr/local/freeswitch/log/freeswitch.log.40 to
> /usr/local/freeswitch/log/freeswitch.log.41
>
> ??
>
> 2012-10-27 00:19:30.313497 [CRIT] mod_logfile.c:164 Error renaming log
> from /usr/local/freeswitch/log/freeswitch.log.40 to
> /usr/local/freeswitch/log/freeswitch.log.41
>
>
> Until it ran out of disk space and crashed. No reason why it couldn?t
> rename the file, though.
>
>
>
>
> On Tue, Nov 20, 2012 at 1:50 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> When it says it cannot, is that because of file permissions? Dos FS have
>> permission to write to the directory?
>> Can you reproduce this with logs and if so can you report it to Jira
>> http://jira.freeswitch.org
>>
>>
>>
>> On Mon, Nov 19, 2012 at 6:45 AM, Nageshwara Rao Moova <
>> mnrao2001 at gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> I have modified my default logconf file for rollover to be restricted to
>>> 100. But sometimes freeswitch is unable to rename rollover say ?cannot
>>> rename freeswitch.log.41? and fails. But the fail follows with serious
>>> issue of writing all the logs to "freeswitch.log" and ends up filling the
>>> disk space.
>>>
>>> I have not change the default file size i.e 10MB.
>>>
>>> How does 10MB size overridden by freeswitch?
>>>
>>> --
>>> regards & thanks
>>> --
>>> m nageshwara rao
>>> 99891 86280
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 & thanks
> --
> m nageshwara rao
> 99891 86280
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From victor.chukalovskiy at gmail.com  Wed Nov 21 20:30:37 2012
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Wed, 21 Nov 2012 12:30:37 -0500
Subject: [Freeswitch-users] Freeswitch retries same call 4 times for no
 reason - incorrect treatment of timers by mod sofia?
Message-ID: <50AD0FBD.9010601@gmail.com>

Hello,

Looking for help or comments on this. Reproducible on the FS system that 
was updated about a month ago. Call scenario:

SIP call comes in on "a" leg, SIP call is attempted on the "b" leg. "b" 
leg replies "404".

Description of the problem:
FS tries same SIP destination 4 times on the "b" leg instead of doing it 
once. It does not respond "404" to the "a" leg until it finishes all 4 
attempts on the "b" leg. This occurs when "b" leg takes longer than 
usual to return "404".  For example, if "b" leg takes 1.3 seconds until 
we get "404", FS tries the call on "b" leg 3 more times. When this 
happens each consequent call attempt on the "b" leg is exactly 2 seconds 
apart from when the previous call attempt ends.

For comparison, in the same scenario and using exact same config, if "b' 
leg replies "404" sooner (e.g. within 0.3 seconds) FS does not retry 
this call on the "b" leg anymore. "404" is returned to the "a" leg 
instantaneously as it should be.

I suspect this is a bug in Sofia such that is somehow related to 500ms 
timer.

Is this known behavior? Do you need any more details to be able to 
verify or comment?

Thank you,
Victor


From philq at qsystemsengineering.com  Wed Nov 21 22:48:25 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Wed, 21 Nov 2012 11:48:25 -0800 (PST)
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
	contact IP/port
In-Reply-To: <CAFVUaV_mgPH8bSYkXaEETft8eYf+kLzBxWrjgoxj-=Rh+_QSCg@mail.gmail.com>
References: <054001cdc664$92486660$b6d93320$@com>
	<CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
	<1353443727657-7584773.post@n2.nabble.com>
	<CAFVUaV_mgPH8bSYkXaEETft8eYf+kLzBxWrjgoxj-=Rh+_QSCg@mail.gmail.com>
Message-ID: <1353527305527-7584818.post@n2.nabble.com>

Thanks for the recommendation.  I've been looking at OpenSIPS and Kamailio
and was sort of leaning toward OpenSIPS myself.  I seem to recall someone on
the list here (one of the developers, I think) saying that Kamailio had a
better network stack and was better for TLS (or something to that effect)
but the two projects look very similar.

If the person who mentioned that or anyone else would like to weigh-in on
this, I'm sure that many would love to hear what they had to say, I know I
would!

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Rewriting-media-address-in-SDP-as-well-as-contact-IP-port-tp7584712p7584818.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From philq at qsystemsengineering.com  Wed Nov 21 23:11:09 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Wed, 21 Nov 2012 12:11:09 -0800 (PST)
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
	contact IP/port
In-Reply-To: <CAKDfjgcmcA8tp-7WVEzxxHYtUXkyPxXmm3d0dSQqV54CNBwsRQ@mail.gmail.com>
References: <054001cdc664$92486660$b6d93320$@com>
	<CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
	<1353443727657-7584773.post@n2.nabble.com>
	<CAKDfjgcmcA8tp-7WVEzxxHYtUXkyPxXmm3d0dSQqV54CNBwsRQ@mail.gmail.com>
Message-ID: <1353528669121-7584820.post@n2.nabble.com>

I'm trying to approach the problem from the 'far-end' side if at all
possible.  So far, in order to get bypass media to work, I've had to take a
combination of the two approaches which includes rport on both the server
and client-side, configuring endpoints to use a particular port for RTP and
then forwarding a range of ports for each endpoint, etc.  I'd like to get
the server-side of the fence doing as much as possible to minimize the work
needed on the client end.  And yes... I realize that folks in Hell would
really like some ice-water too.

There seems to be almost as much art as science involved and despite many
years of advanced networking experience I feel like I'm only recently
starting to get the big picture when it comes to VoIP... there's a lot going
on here.

So while I'm not the first to do so, I'd like to express a big thank you to
everyone on here who is so willing to help everyone out and of course to the
FS developers and other contributors, helping us in our attempts at
"avoiding deadlock".   :)

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Rewriting-media-address-in-SDP-as-well-as-contact-IP-port-tp7584712p7584820.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From william.king at quentustech.com  Wed Nov 21 23:55:49 2012
From: william.king at quentustech.com (William King)
Date: Wed, 21 Nov 2012 12:55:49 -0800
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <50ACE9C4.60803@panamaxil.com>
References: <50ACE9C4.60803@panamaxil.com>
Message-ID: <50AD3FD5.4070606@quentustech.com>

Try using xml_mod_radius instead.

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

On 11/21/2012 06:48 AM, Chetan Khatri wrote:
> Dear All Experts,
> 
> Need help to solve freeswitch's mod_rad_auth capacity issue.
> I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The
> Module is working fine till the concurrent request reaches till 245
> after that Accounting is failing.
> Below is the Configuration.
> 
> FreeSwitch : 1.0.6
> RADIUS Server : FreeRADIUS2
> Database server : MySQL 5.5
> 
> We are stress testing the with freeswitch to handle about 2000
> concurrent calls with RADIUS AAA. without using RADIUS modules
> freeswitch is able to handle 2000 concurrent calls without any issues.
> For analysis we analysed from RADIUS server replies access-accept
> request, but in mod_rad_auth error says *"freeswitch : Invalid Digest
> Request from RADIUS server"* only if the concurrent session receives
> 245, till that all the requests are been working fine.
> 
> Please guide me how to resolve this capacity issue, I can provide all
> the logs and configuration files if required.
> 
> -- 
> /Regards,
> Chetan Khatri/
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From william.king at quentustech.com  Wed Nov 21 23:57:10 2012
From: william.king at quentustech.com (William King)
Date: Wed, 21 Nov 2012 12:57:10 -0800
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <50ACE9C4.60803@panamaxil.com>
References: <50ACE9C4.60803@panamaxil.com>
Message-ID: <50AD4026.2020602@quentustech.com>

Also, the problem you are likely hitting is that your FreeRADIUS server
is throwing an error and returning an invalid response. Check the logs
for that server and see if there is anything reported.

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

On 11/21/2012 06:48 AM, Chetan Khatri wrote:
> Dear All Experts,
> 
> Need help to solve freeswitch's mod_rad_auth capacity issue.
> I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The
> Module is working fine till the concurrent request reaches till 245
> after that Accounting is failing.
> Below is the Configuration.
> 
> FreeSwitch : 1.0.6
> RADIUS Server : FreeRADIUS2
> Database server : MySQL 5.5
> 
> We are stress testing the with freeswitch to handle about 2000
> concurrent calls with RADIUS AAA. without using RADIUS modules
> freeswitch is able to handle 2000 concurrent calls without any issues.
> For analysis we analysed from RADIUS server replies access-accept
> request, but in mod_rad_auth error says *"freeswitch : Invalid Digest
> Request from RADIUS server"* only if the concurrent session receives
> 245, till that all the requests are been working fine.
> 
> Please guide me how to resolve this capacity issue, I can provide all
> the logs and configuration files if required.
> 
> -- 
> /Regards,
> Chetan Khatri/
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jason.holden at start.ca  Thu Nov 22 00:04:38 2012
From: jason.holden at start.ca (Jason Holden)
Date: Wed, 21 Nov 2012 16:04:38 -0500
Subject: [Freeswitch-users] retreiving voicemail dropping after 30
	seconds
In-Reply-To: <1353522313292-7584814.post@n2.nabble.com>
References: <6626166B66164AB4B5C0E344762D7E4A@bob><472138AA0F3C4B01A08CBF5CB2CFFB2D@bob><CAKzWOxVk-tx7NEX2=3JyhP8w-9XqmcpLNL69vs6N2n8bsRr0uA@mail.gmail.com><E66DCAE9702E4F869A4A2AE4431F8265@bob>
	<1353522313292-7584814.post@n2.nabble.com>
Message-ID: <842A55B801B744F6B999F60124580597@bob>

All.
I did find the issue just had not had a chance to reply until now.
In my internal profile for what ever reason a var was set to reflect my
external ip for sip and rtp.
Once changed to my LAN information all is fine.


-----Original Message-----
From: PhilQ [mailto:philq at qsystemsengineering.com] 
Sent: Wednesday, November 21, 2012 1:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] retreiving voicemail dropping after 30
seconds

I'm going to go out on a limb and make a "scientific-wild-assed-guess" since
I can't see your pastebin, but since you're seeing external IP addresses I
have an idea of what might be going on.

Your client/user agent is probably using STUN and is reporting it's external
IP address to FS which then uses that address for replies to the client. 
With most firewall/NAT devices, those replies won't make it back to the
device unless you have a port forwarding rule set since the device treats it
like it's coming from the outside.  If that is indeed the case, then there
are two things you can try:

1.  Disable STUN on your endpoint (probably not the best solution,
especially if you're on a dynamic IP)  OR

2.  Set a port forwarding rule in your firewall/router/nat device so those
repllies make it back.

Without the benefit of the pastebin this is just a wild guess so please
don't sue me if this is not applicable.

Hope everyone has a happy and safe Thanksgiving holiday!

- Phil



--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/retreiving-voicemail-dropping-
after-30-seconds-tp7584653p7584814.html
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From shaheryarkh at gmail.com  Thu Nov 22 00:13:45 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Wed, 21 Nov 2012 22:13:45 +0100
Subject: [Freeswitch-users] Freeswitch Multilingual Setup
In-Reply-To: <50ACFFAB.6010401@communicatefreely.net>
References: <CAFZQphz-0b3h7Y67rLBgzkcQre=z+g74RkopnYjL+cPicB=iww@mail.gmail.com>
	<50ACFFAB.6010401@communicatefreely.net>
Message-ID: <CAFZQphwe15y1MgxaJ5+XBehpZkOJayg9TTTYHqvEoRq-Gu91og@mail.gmail.com>

Thank you very much for your help. I will give it a try.

Thank you.


On Wed, Nov 21, 2012 at 5:22 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hi Muhammed,
>
> That's because the way you are calling the playback application below is
> calling a
> specific audio file based on path.
>
> In vars.xml (usually) there is a global variable that sets the sound files
> prefix.  This
> will be added to whatever you put as the argument to the playback
> application.  I'm
> willing to bet that whatever you have set includes a language path.
>
> There are a few ways around this -
>
> You can make some phrase macros for each language, and define which
> prompts in which
> language should be used.  Any time you want to play something, use
> phrase:your_macro instead.
>
> Likewise, you could also change the sounds prefix not to include the
> language, and then
> call playback like this:
>
> data="${default_language}/voicemail/prompt.wav"
>
> I would recommend the phrase macro method, as that gives you a lot of
> flexibility with
> what prompts are used for a given language, and it also lets you account
> for things like
> quantity, gender, and number ordering if that applies (ie. if you are
> playing an error for
> a wrong number).
>
> Hope that helps.
>
> -Tim
>
> Muhammad Shahzad wrote:
> > Hi,
> >
> > I am test Freeswitch for Multilingual IVR system. I observed that if I
> > set default_language parameter before "voicemail" or "say" applications,
> > it correctly play specified application in given language. However it
> > has no effect on playback application, it always play the IVR in system
> > default (i.e. English) language IVR.
> >
> > Here is an example extension,
> >
> >     <extension name="catchall">
> >       <condition field="destination_number" expression="^(.*)$">
> >         <action application="export" data="default_language=fr"/>
> >         <action application="answer"/>
> >         <action application="sleep" data="1000"/>
> >         <action application="playback"
> > data="voicemail/vm-that_was_an_invalid_ext.wav"/>
> >         <action application="hangup"/>
> >       </condition>
> >     </extension>
> >
> > Many thanks in advance.
> >
> > Thank you.
> >
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From daniel at pocock.com.au  Thu Nov 22 01:06:22 2012
From: daniel at pocock.com.au (Daniel Pocock)
Date: Wed, 21 Nov 2012 23:06:22 +0100
Subject: [Freeswitch-users] Paris - mini-DebConf - VoIP - 24 November
Message-ID: <50AD505E.1090805@pocock.com.au>



For those using Debian/Ubuntu (and anybody else is welcome of course),
there is a mini-DebConf in Paris this weekend:

   http://fr2012.mini.debconf.org/

There is a presentation at 16:00 about Debian's role in establishing an
alternative to Skype, this will look at some of the packages available
on the upcoming Debian 7 (wheezy), and strategic ways of deploying them
to build a genuinely free and open cloud for real-time communications.

There is no registration fee - all welcome


From ahmed at netelsat.net  Thu Nov 22 01:13:53 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Thu, 22 Nov 2012 02:13:53 +0400
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <50AD3FD5.4070606@quentustech.com>
References: <50ACE9C4.60803@panamaxil.com> <50AD3FD5.4070606@quentustech.com>
Message-ID: <CALFFxt3__=MPVF5a1o6eT+XXdHBDXQMt-EvHFYw+EMrqAwDrdw@mail.gmail.com>

yes we are using xml_mod_radius easily for more then 2000 calls.
regards
Ahmed


On Thu, Nov 22, 2012 at 12:55 AM, William King <william.king at quentustech.com
> wrote:

> Try using xml_mod_radius instead.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 11/21/2012 06:48 AM, Chetan Khatri wrote:
> > Dear All Experts,
> >
> > Need help to solve freeswitch's mod_rad_auth capacity issue.
> > I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The
> > Module is working fine till the concurrent request reaches till 245
> > after that Accounting is failing.
> > Below is the Configuration.
> >
> > FreeSwitch : 1.0.6
> > RADIUS Server : FreeRADIUS2
> > Database server : MySQL 5.5
> >
> > We are stress testing the with freeswitch to handle about 2000
> > concurrent calls with RADIUS AAA. without using RADIUS modules
> > freeswitch is able to handle 2000 concurrent calls without any issues.
> > For analysis we analysed from RADIUS server replies access-accept
> > request, but in mod_rad_auth error says *"freeswitch : Invalid Digest
> > Request from RADIUS server"* only if the concurrent session receives
> > 245, till that all the requests are been working fine.
> >
> > Please guide me how to resolve this capacity issue, I can provide all
> > the logs and configuration files if required.
> >
> > --
> > /Regards,
> > Chetan Khatri/
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Thu Nov 22 01:55:02 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 21 Nov 2012 17:55:02 -0500
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
 contact IP/port
In-Reply-To: <1353528669121-7584820.post@n2.nabble.com>
References: <054001cdc664$92486660$b6d93320$@com>
	<CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
	<1353443727657-7584773.post@n2.nabble.com>
	<CAKDfjgcmcA8tp-7WVEzxxHYtUXkyPxXmm3d0dSQqV54CNBwsRQ@mail.gmail.com>
	<1353528669121-7584820.post@n2.nabble.com>
Message-ID: <CAKDfjgefiw2o-PtnwQUBE9-zjrEv5z2-EHZWE0P+T2URqXPT5A@mail.gmail.com>

On Wed, Nov 21, 2012 at 3:11 PM, PhilQ <philq at qsystemsengineering.com> wrote:
> I'm trying to approach the problem from the 'far-end' side if at all
> possible.  So far, in order to get bypass media to work, I've had to take a
> combination of the two approaches which includes rport on both the server
> and client-side, configuring endpoints to use a particular port for RTP and
> then forwarding a range of ports for each endpoint, etc.  I'd like to get
> the server-side of the fence doing as much as possible to minimize the work
> needed on the client end.  And yes... I realize that folks in Hell would
> really like some ice-water too.

  Woah.  Sounds like an interesting environment...  If you're doing
that much work you shouldn't have an issue using the SER family
nathelper module to rewrite what you need.

-- 
Kristian Kielhofner


From jeff at jefflenk.com  Thu Nov 22 01:59:16 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 21 Nov 2012 14:59:16 -0800 (PST)
Subject: [Freeswitch-users] E-mail voicemails - msmtp in Windows
In-Reply-To: <1352818521085-7584543.post@n2.nabble.com>
References: <CAG0Fp+_kQf1pZyZpCDWuqAVMpYsCfA3p1gUyrSzWK24VNi9Yjg@mail.gmail.com>
	<1352818521085-7584543.post@n2.nabble.com>
Message-ID: <1353538756575-7584824.post@n2.nabble.com>

The original problem that was reported to Jira has been resolved in
http://jira.freeswitch.org/browse/FS-4847



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/E-mail-voicemails-msmtp-in-Windows-tp7584525p7584824.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From shaheryarkh at gmail.com  Thu Nov 22 02:08:35 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Thu, 22 Nov 2012 00:08:35 +0100
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <50AD4026.2020602@quentustech.com>
References: <50ACE9C4.60803@panamaxil.com> <50AD4026.2020602@quentustech.com>
Message-ID: <CAFZQphwbjJ7z4TcWO55tPsY5CO7Nm2=zU-1T+-jpne0augxu9Q@mail.gmail.com>

Also make sure to increase number of open files on radius server. We faced
a simpler problem some while ago and it turned out be too many open files
problem.

ulimit -n 1024000

Thank you.


On Wed, Nov 21, 2012 at 9:57 PM, William King
<william.king at quentustech.com>wrote:

> Also, the problem you are likely hitting is that your FreeRADIUS server
> is throwing an error and returning an invalid response. Check the logs
> for that server and see if there is anything reported.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 11/21/2012 06:48 AM, Chetan Khatri wrote:
> > Dear All Experts,
> >
> > Need help to solve freeswitch's mod_rad_auth capacity issue.
> > I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The
> > Module is working fine till the concurrent request reaches till 245
> > after that Accounting is failing.
> > Below is the Configuration.
> >
> > FreeSwitch : 1.0.6
> > RADIUS Server : FreeRADIUS2
> > Database server : MySQL 5.5
> >
> > We are stress testing the with freeswitch to handle about 2000
> > concurrent calls with RADIUS AAA. without using RADIUS modules
> > freeswitch is able to handle 2000 concurrent calls without any issues.
> > For analysis we analysed from RADIUS server replies access-accept
> > request, but in mod_rad_auth error says *"freeswitch : Invalid Digest
> > Request from RADIUS server"* only if the concurrent session receives
> > 245, till that all the requests are been working fine.
> >
> > Please guide me how to resolve this capacity issue, I can provide all
> > the logs and configuration files if required.
> >
> > --
> > /Regards,
> > Chetan Khatri/
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Nov 22 04:09:52 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Nov 2012 19:09:52 -0600
Subject: [Freeswitch-users] Freeswitch retries same call 4 times for no
 reason - incorrect treatment of timers by mod sofia?
In-Reply-To: <50AD0FBD.9010601@gmail.com>
References: <50AD0FBD.9010601@gmail.com>
Message-ID: <CAKbxfG9ChXy2rfmCEaEpASpaqhfwQpYaZvdR8WDJaK1tS_fujQ@mail.gmail.com>

A complete console log attached to a new jira issue.

http://jira.freeswitch.org

sofia global siptrace on
console loglevel debug



On Wed, Nov 21, 2012 at 11:30 AM, Victor Chukalovskiy <
victor.chukalovskiy at gmail.com> wrote:

> Hello,
>
> Looking for help or comments on this. Reproducible on the FS system that
> was updated about a month ago. Call scenario:
>
> SIP call comes in on "a" leg, SIP call is attempted on the "b" leg. "b"
> leg replies "404".
>
> Description of the problem:
> FS tries same SIP destination 4 times on the "b" leg instead of doing it
> once. It does not respond "404" to the "a" leg until it finishes all 4
> attempts on the "b" leg. This occurs when "b" leg takes longer than
> usual to return "404".  For example, if "b" leg takes 1.3 seconds until
> we get "404", FS tries the call on "b" leg 3 more times. When this
> happens each consequent call attempt on the "b" leg is exactly 2 seconds
> apart from when the previous call attempt ends.
>
> For comparison, in the same scenario and using exact same config, if "b'
> leg replies "404" sooner (e.g. within 0.3 seconds) FS does not retry
> this call on the "b" leg anymore. "404" is returned to the "a" leg
> instantaneously as it should be.
>
> I suspect this is a bug in Sofia such that is somehow related to 500ms
> timer.
>
> Is this known behavior? Do you need any more details to be able to
> verify or comment?
>
> Thank you,
> Victor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From krice at freeswitch.org  Thu Nov 22 04:33:01 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 21 Nov 2012 19:33:01 -0600
Subject: [Freeswitch-users] Freeswitch retries same call 4 times for no
 reason - incorrect treatment of timers by mod sofia?
In-Reply-To: <CAKbxfG9ChXy2rfmCEaEpASpaqhfwQpYaZvdR8WDJaK1tS_fujQ@mail.gmail.com>
Message-ID: <CCD2DCED.60FF6%krice@freeswitch.org>

Sounds more like someone is using a stacked dial with continue_on_fail=true


On 11/21/12 7:09 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> A complete console log attached to a new jira issue.
> 
> http://jira.freeswitch.org
> 
> sofia global siptrace on
> console loglevel debug
> 
> 
> 
> On Wed, Nov 21, 2012 at 11:30 AM, Victor Chukalovskiy
> <victor.chukalovskiy at gmail.com> wrote:
>> Hello,
>> 
>> Looking for help or comments on this. Reproducible on the FS system that
>> was updated about a month ago. Call scenario:
>> 
>> SIP call comes in on "a" leg, SIP call is attempted on the "b" leg. "b"
>> leg replies "404".
>> 
>> Description of the problem:
>> FS tries same SIP destination 4 times on the "b" leg instead of doing it
>> once. It does not respond "404" to the "a" leg until it finishes all 4
>> attempts on the "b" leg. This occurs when "b" leg takes longer than
>> usual to return "404". ?For example, if "b" leg takes 1.3 seconds until
>> we get "404", FS tries the call on "b" leg 3 more times. When this
>> happens each consequent call attempt on the "b" leg is exactly 2 seconds
>> apart from when the previous call attempt ends.
>> 
>> For comparison, in the same scenario and using exact same config, if "b'
>> leg replies "404" sooner (e.g. within 0.3 seconds) FS does not retry
>> this call on the "b" leg anymore. "404" is returned to the "a" leg
>> instantaneously as it should be.
>> 
>> I suspect this is a bug in Sofia such that is somehow related to 500ms
>> timer.
>> 
>> Is this known behavior? Do you need any more details to be able to
>> verify or comment?
>> 
>> Thank you,
>> Victor
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 

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

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From brian at freeswitch.org  Thu Nov 22 05:55:27 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Nov 2012 20:55:27 -0600
Subject: [Freeswitch-users] iCall Problems?
Message-ID: <EF35F13C-3FB2-4081-A9B7-AC9158934C27@freeswitch.org>

http://www.webhostingtalk.com/showthread.php?t=1209956



--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543







From talk2ram at gmail.com  Thu Nov 22 10:24:55 2012
From: talk2ram at gmail.com (ram)
Date: Thu, 22 Nov 2012 12:54:55 +0530
Subject: [Freeswitch-users] digits not able to announce in IVR
Message-ID: <CA+XvL-MZRbCd5a7AvyZ8Ft-eA6RedGKAweg9suALwhgWQxuiug@mail.gmail.com>

Hi all

Iam trying to read the number entered and announce back using lua script

but i get following error..any suggestions


   1. 2012-11-22 12:26:50.850983 [DEBUG] switch_ivr_play_say.c:1309 Codec
   Activated L16 at 8000hz 1 channels 20ms
   2. 2012-11-22 12:26:54.090983 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   8:800
   3. 2012-11-22 12:26:54.690982 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   1:800
   4. 2012-11-22 12:26:55.110992 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   4:800
   5. 2012-11-22 12:26:55.890982 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   3:800
   6. 2012-11-22 12:26:56.550979 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   7:640
   7. 2012-11-22 12:26:57.190981 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   5:640
   8. 2012-11-22 12:26:57.830980 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   0:640
   9. 2012-11-22 12:26:58.330979 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   7:800
   10. 2012-11-22 12:26:58.630979 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   4:640
   11. 2012-11-22 12:26:59.190980 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   6:640
   12. 2012-11-22 12:26:59.750982 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
   #:640
   13. 2012-11-22 12:26:59.750982 [DEBUG] switch_ivr_play_say.c:2034 Test
   Regex [8143750746][\d+]
   14. 2012-11-22 12:26:59.750982 [INFO] switch_cpp.cpp:1227 Announcing
   number: 8143750746
   15. 2012-11-22 12:26:59.750982 [DEBUG] switch_ivr.c:2883 No language
   specified - Using [en]
   16. 2012-11-22 12:26:59.750982 [ERR] mod_say_en.c:130 Parse Error!
   17.
   18.
   my LUA script.

   19.
   20. local announce_number = function(number)
   21.     log("Announcing number: " .. number)
   22.     s:say(number, "en", "number", "pronounced")
   23.     s:streamFile("phone-confirm-re-enter.wav")
   24. end
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From curriegrad2004 at gmail.com  Thu Nov 22 10:25:35 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 21 Nov 2012 23:25:35 -0800
Subject: [Freeswitch-users] iCall Problems?
In-Reply-To: <EF35F13C-3FB2-4081-A9B7-AC9158934C27@freeswitch.org>
References: <EF35F13C-3FB2-4081-A9B7-AC9158934C27@freeswitch.org>
Message-ID: <CAMWiMy-y-ReAapsveyWwf-FeYbVmM05FKvzuPs-BWU9-+MXGew@mail.gmail.com>

We did discuss about this a few conference calls back anyways.

Sadly they'll be missed

On Wed, Nov 21, 2012 at 6:55 PM, Brian West <brian at freeswitch.org> wrote:
> http://www.webhostingtalk.com/showthread.php?t=1209956
>
>
>
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From enp at itx.ru  Thu Nov 22 10:33:40 2012
From: enp at itx.ru (Eugene Prokopiev)
Date: Thu, 22 Nov 2012 10:33:40 +0300
Subject: [Freeswitch-users] LDAP Integration
Message-ID: <CANM4Rui42maROz1u+Y-Z3DeTYvSs4mMQpJvqSiwGw=LPokZCAA@mail.gmail.com>

Hi,

I need to authenticate users from LDAP. Only mod_xml_ldap has example (
http://git.freeswitch.org/git/freeswitch/tree/src/mod/xml_int/mod_xml_ldapand
http://article.gmane.org/gmane.comp.telephony.freeswitch.user/6974), so I
tried to configure it:

<configuration name="xml_ldap.conf">
  <bindings>
    <binding name="directory">
      <param name="basedn" value="dc=home"/>
      <param name="filter"
value="(&(objectClass=person)(telephoneNumber=%s))" bindings="directory"/>
      <param name="url" value="ldap://localhost"/>
      <param name="binddn" value="cn=root,dc=home"/>
      <param name="bindpass" value="rootpwd"/>
      <trans>
        <tran name="id" mapfrom="telephoneNumber"/>
        <tran name="password" mapfrom="telephoneNumber"/>
      </trans>
    </binding>
  </bindings>
</configuration>

Next I tried to authenticate user with login 123 and password 123. Ldif
looks like:

dn:cn=user0,ou=users,dc=home
sn:user0
telephoneNumber: 123
objectClass: person

Authentication succeeded, but authentication for user with login 123 and
password 321 succeeded too. Authentication for user with login 321 failed.

So, my filter configuration works fine, but I can't use password in it. I
need to use <trans>. What is wrong in my configuration?

--
Regards,
Eugene Prokopiev
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From chetan.khatri at panamaxil.com  Thu Nov 22 13:56:51 2012
From: chetan.khatri at panamaxil.com (Chetan Khatri)
Date: Thu, 22 Nov 2012 16:26:51 +0530
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <50AD3FD5.4070606@quentustech.com>
References: <50ACE9C4.60803@panamaxil.com> <50AD3FD5.4070606@quentustech.com>
Message-ID: <50AE04F3.5000606@panamaxil.com>

Hello William,

First of all thanks for your prompt response, I am beginner level user 
of freeswitch so dont know where to get the module  xml_mod_radius and 
it will be great if some documentation on installation and configuration 
of the module is there.

Thanks & Regards,
Chetan

On 11/22/2012 02:25 AM, William King wrote:
> Try using xml_mod_radius instead.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 11/21/2012 06:48 AM, Chetan Khatri wrote:
>> Dear All Experts,
>>
>> Need help to solve freeswitch's mod_rad_auth capacity issue.
>> I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The
>> Module is working fine till the concurrent request reaches till 245
>> after that Accounting is failing.
>> Below is the Configuration.
>>
>> FreeSwitch : 1.0.6
>> RADIUS Server : FreeRADIUS2
>> Database server : MySQL 5.5
>>
>> We are stress testing the with freeswitch to handle about 2000
>> concurrent calls with RADIUS AAA. without using RADIUS modules
>> freeswitch is able to handle 2000 concurrent calls without any issues.
>> For analysis we analysed from RADIUS server replies access-accept
>> request, but in mod_rad_auth error says *"freeswitch : Invalid Digest
>> Request from RADIUS server"* only if the concurrent session receives
>> 245, till that all the requests are been working fine.
>>
>> Please guide me how to resolve this capacity issue, I can provide all
>> the logs and configuration files if required.
>>
>> -- 
>> /Regards,
>> Chetan Khatri/
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From enp at itx.ru  Thu Nov 22 12:26:01 2012
From: enp at itx.ru (Eugene Prokopiev)
Date: Thu, 22 Nov 2012 12:26:01 +0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <CANM4Rui42maROz1u+Y-Z3DeTYvSs4mMQpJvqSiwGw=LPokZCAA@mail.gmail.com>
References: <CANM4Rui42maROz1u+Y-Z3DeTYvSs4mMQpJvqSiwGw=LPokZCAA@mail.gmail.com>
Message-ID: <CANM4RuipzyPL=ON8w9Z1LWbh+Ca=zsB2Gu71aqMvze=CDrgdzA@mail.gmail.com>

I tried to read mod_xml_ldap.c (function xml_ldap_directory_result), but I
can't see something like "trans". So, how can I define attributes in ldif
and how result xml must be looks like?

--
Regards,
Eugene Prokopiev
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From enp at itx.ru  Thu Nov 22 13:59:59 2012
From: enp at itx.ru (Eugene Prokopiev)
Date: Thu, 22 Nov 2012 13:59:59 +0300
Subject: [Freeswitch-users] mod_perl vs mod_xml_curl
Message-ID: <CANM4Ruj+5f9HOdaVz+iA6A2Hx=fnC+Zsf77sRfN_C4+whjboPQ@mail.gmail.com>

Hi,

I tried to to authenticate users from external source with mod_perl or
mod_xml_curl. Configuration and code looks like:

mod_perl:

<configuration name="perl.conf">
  <settings>
    <param name="xml-handler-script" value="/var/lib/freeswitch/scripts/
directory.pl"/>
    <param name="xml-handler-bindings" value="directory"/>
  </settings>
</configuration>

$XML_STRING = '
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="$${local_address}">
      <params>
        <param name="dial-string" value="${sofia_contact(${dialed_user}@
${dialed_domain})}"/>
      </params>
      <groups>
        <group name="local">
          <users>
            <user id="123"><params><param name="password"
value="123"/></params></user>
           </users>
        </group>
      </groups>
    </domain>
  </section>
</document>
';

mod_xml_curl:

<configuration name="xml_curl.conf">
  <bindings>
    <binding name="directory">
      <param name="gateway-url" value="http://localhost:5000"
bindings="directory"/>
    </binding>
  </bindings>
</configuration>

use Mojolicious::Lite;
post '/' => 'index';
app->start;
__DATA__
@@ index.html.ep
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="$${local_address}">
      <params>
        <param name="dial-string" value="${sofia_contact(${dialed_user}@
${dialed_domain})}"/>
      </params>
      <groups>
        <group name="local">
          <users>
            <user id="123"><params><param name="password"
value="123"/></params></user>
           </users>
        </group>
      </groups>
    </domain>
  </section>
</document>

Authentication via mod_xml_curl was succeeded, authentication via mod_perl
was failed. What is wring with it?

--
Regards,
Eugene Prokopiev
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From a.venugopan at mundio.com  Thu Nov 22 14:52:31 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 22 Nov 2012 11:52:31 +0000
Subject: [Freeswitch-users] mod_voicemail.c file change
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>

Hi,

Currently while dialling voicemail from mod_voicemail.c file it picks up the name recorded in system greetings and says like

'<Name> person you are trying to reach is currently not available.Record you message after the tone press any key or stop talking to end the recording'

I require something like this
'You have reached the mailbox of <Name> Record you message after the tone press any key or stop talking to end the recording'.

I knew I have to change in mod_voicemail.c file. But since am not familiar with C code not sure how to get this.

if (!skip_greeting) {
                memset(buf, 0, sizeof(buf));
                args.input_callback = cancel_on_dtmf;
                args.buf = buf;
                args.buflen = sizeof(buf);

                switch_ivr_sleep(session, 100, SWITCH_TRUE, NULL);

                if (switch_file_exists(greet_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                        memset(buf, 0, sizeof(buf));
                        TRY_CODE(switch_ivr_play_file(session, NULL, greet_path, &args));
                } else {
                        if (switch_file_exists(cbt.name_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_play_file(session, NULL, cbt.name_path, &args));
                        }
                        if (*buf == '\0') {
                                if (!read_id) {
                                        if (!(read_id = switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {
                                                read_id = id;
                                        }
                                }
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO, read_id, NULL, &args));
                        }

Can anyone please help me out. Thanks

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From andrew at cassidywebservices.co.uk  Thu Nov 22 16:23:52 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 22 Nov 2012 13:23:52 +0000
Subject: [Freeswitch-users] Cisco-signed SSL Certificates/Alternative phone
	suggestions
Message-ID: <CAMbW9oL=AjAJCDn9D-tggWqPoy8pitDjJT0waZgH27HWJyOx0Q@mail.gmail.com>

Hi guys, just looking for a little help. I've been using Cisco SPA50x
phones for a little while now, they have some pretty nice features for the
price, including the ability to write custom applications in XML. However,
to run them over HTTPS instead of HTTP, you need a certificate signed by
Cisco, which I'm having difficulty obtaining.

So my questions are:

   1. Can anyone help me obtain such a certificate? I have the CSR already
   to go.
   2. Are there any similarly priced and featured phones available that you
   guys have experience with?

Thanks guys,

-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From chetan.khatri at panamaxil.com  Thu Nov 22 17:02:22 2012
From: chetan.khatri at panamaxil.com (Chetan Khatri)
Date: Thu, 22 Nov 2012 19:32:22 +0530
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <CALFFxt3__=MPVF5a1o6eT+XXdHBDXQMt-EvHFYw+EMrqAwDrdw@mail.gmail.com>
References: <50ACE9C4.60803@panamaxil.com> <50AD3FD5.4070606@quentustech.com>
	<CALFFxt3__=MPVF5a1o6eT+XXdHBDXQMt-EvHFYw+EMrqAwDrdw@mail.gmail.com>
Message-ID: <50AE306E.7040909@panamaxil.com>

Hello Ahmed,

Could please provide some guidance to configure xml_mod_radius i have 
deployed it.

/Regards,
Chetan Khatri/

On 11/22/2012 03:43 AM, Ahmed Sboor wrote:
> yes we are using xml_mod_radius easily for more then 2000 calls.
> regards
> Ahmed
>
>
> On Thu, Nov 22, 2012 at 12:55 AM, William King 
> <william.king at quentustech.com <mailto:william.king at quentustech.com>> 
> wrote:
>
>     Try using xml_mod_radius instead.
>
>     William King
>     Senior Engineer
>     Quentus Technologies, INC
>     1037 NE 65th St Suite 273
>     Seattle, WA 98115
>     Main:   (877) 211-9337
>     Office: (206) 388-4772
>     Cell:   (253) 686-5518
>     william.king at quentustech.com <mailto:william.king at quentustech.com>
>
>     On 11/21/2012 06:48 AM, Chetan Khatri wrote:
>     > Dear All Experts,
>     >
>     > Need help to solve freeswitch's mod_rad_auth capacity issue.
>     > I have deployed freeswtich 1.0.6 and enabled mod_rad_auth
>     module. The
>     > Module is working fine till the concurrent request reaches till 245
>     > after that Accounting is failing.
>     > Below is the Configuration.
>     >
>     > FreeSwitch : 1.0.6
>     > RADIUS Server : FreeRADIUS2
>     > Database server : MySQL 5.5
>     >
>     > We are stress testing the with freeswitch to handle about 2000
>     > concurrent calls with RADIUS AAA. without using RADIUS modules
>     > freeswitch is able to handle 2000 concurrent calls without any
>     issues.
>     > For analysis we analysed from RADIUS server replies access-accept
>     > request, but in mod_rad_auth error says *"freeswitch : Invalid
>     Digest
>     > Request from RADIUS server"* only if the concurrent session receives
>     > 245, till that all the requests are been working fine.
>     >
>     > Please guide me how to resolve this capacity issue, I can
>     provide all
>     > the logs and configuration files if required.
>     >
>     > --
>     > /Regards,
>     > Chetan Khatri/
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > FreeSWITCH-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
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From lists at kavun.ch  Thu Nov 22 17:34:38 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 22 Nov 2012 09:34:38 -0500
Subject: [Freeswitch-users] Ringing instead of MOH for valet parking
In-Reply-To: <71943DD5C22943448A24B7C5CDC2380730707DE1@CH1PRD0411MB430.namprd04.prod.outlook.com>
References: <BC2DB117-1A46-47B6-B88F-3C2E8F48E57F@kavun.ch>
	<C7DD95104E584A1EB5458250F1E212D9@iMac>
	<71943DD5C22943448A24B7C5CDC2380730707DE1@CH1PRD0411MB430.namprd04.prod.outlook.com>
Message-ID: <1D2255F3-C827-46E5-8068-B5A4A27481DA@kavun.ch>

You guys are great, thanks for all the suggestions. 
Just saw this email and will experiment.

E
On Nov 20, 2012, at 11:44 AM, Robert Hadley <robert.hadley at teotech.com> wrote:

>  
> Also check out ring_ready application: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready
>  
>  
> From: Alex M?ller [mailto:slickwilly2000 at gmx.de] 
> Sent: Tuesday, November 20, 2012 4:24 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Ringing instead of MOH for valet parking
>  
> Have you already seen the channel-variable ?valet_hold_music??
>  
> This only affects the valet_hold, not the MOH globally in FreeSWITCH...
>  
> From: Emrah
> Sent: Tuesday, November 20, 2012 12:37 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Ringing instead of MOH for valet parking
>  
> Hi all,
> 
> Is there a built-in way to have callers hear ringback instead of MOH when they are valet-parked?
> I do not mean to replace the MOH string with a ringtone, which would ultimately cause the caller to hear ringing whenever put on hold?. I am looking for an actual option.
> 
> It is probably trivial to change in the code, but I thought I would ask anyway before I start messing around? 
> 
> Thanks,
> 
> All the best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mnrao2001 at gmail.com  Thu Nov 22 18:00:31 2012
From: mnrao2001 at gmail.com (Nageshwara Rao Moova)
Date: Thu, 22 Nov 2012 20:30:31 +0530
Subject: [Freeswitch-users] Log Rollover issue
In-Reply-To: <CAKbxfG_enpxKsbvLxt7ffJ5_DcivMyO68sQWEbW1LQohe3dQ4w@mail.gmail.com>
References: <CAPDWOejFaGi6OmavH2Q9nKYRzn6zpz1gDbiSVqEcaCV678+bbg@mail.gmail.com>
	<CAKbxfG8krEgxHPWeWi263ubeiducGnsvPUBJhKs=K-JxtX-9Gg@mail.gmail.com>
	<CAPDWOehb7p9JbO8DLqU=sYqSjx_v7jPumuR2zbQ47Wp_fggSSg@mail.gmail.com>
	<CAKbxfG_enpxKsbvLxt7ffJ5_DcivMyO68sQWEbW1LQohe3dQ4w@mail.gmail.com>
Message-ID: <CAPDWOegbNTe3PMYwDkpRd3EpxQiAj8Ow6yxB2+OdUshgWAMiVw@mail.gmail.com>

Thanks,
Will verify the issue with latest code.


On Thu, Nov 22, 2012 at 12:34 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Are you deleting the old ones?
> It was probably already out of disk space when it was refusing to rename
> the files.
>
> Looks like you have 40 of them, you may need to auto delete some.
>
> You are not providing much other data to go on so all I can do is guess.
> The system call to move the file is failing which could be due to
> permissions or disk full or whatever.
>
> If you want you can update to latest and I added a call to strerror(errno)
> so you can see the exact reason.
>
>
>
>
> On Tue, Nov 20, 2012 at 10:23 PM, Nageshwara Rao Moova <
> mnrao2001 at gmail.com> wrote:
>
>> Hi Anthony,
>>
>> I doubt any permission issue as the failure happened not initially but
>> after a while. The user we are using has sufficient permissions for doing
>> rwx operations. Below is the log snippet from freeswitch log.
>>
>> If it?s any help, the freeswitch log had entries like this every few
>> seconds:
>>
>> 2012-10-26 11:53:16.653473 [CRIT] mod_logfile.c:164 Error renaming log
>> from /usr/local/freeswitch/log/freeswitch.log.40 to
>> /usr/local/freeswitch/log/freeswitch.log.41
>>
>> ??
>>
>> 2012-10-27 00:19:30.313497 [CRIT] mod_logfile.c:164 Error renaming log
>> from /usr/local/freeswitch/log/freeswitch.log.40 to
>> /usr/local/freeswitch/log/freeswitch.log.41
>>
>>
>> Until it ran out of disk space and crashed. No reason why it couldn?t
>> rename the file, though.
>>
>>
>>
>>
>> On Tue, Nov 20, 2012 at 1:50 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> When it says it cannot, is that because of file permissions? Dos FS have
>>> permission to write to the directory?
>>> Can you reproduce this with logs and if so can you report it to Jira
>>> http://jira.freeswitch.org
>>>
>>>
>>>
>>> On Mon, Nov 19, 2012 at 6:45 AM, Nageshwara Rao Moova <
>>> mnrao2001 at gmail.com> wrote:
>>>
>>>> Hi all,
>>>>
>>>> I have modified my default logconf file for rollover to be restricted
>>>> to 100. But sometimes freeswitch is unable to rename rollover say ?cannot
>>>> rename freeswitch.log.41? and fails. But the fail follows with serious
>>>> issue of writing all the logs to "freeswitch.log" and ends up filling the
>>>> disk space.
>>>>
>>>> I have not change the default file size i.e 10MB.
>>>>
>>>> How does 10MB size overridden by freeswitch?
>>>>
>>>> --
>>>> regards & thanks
>>>> --
>>>> m nageshwara rao
>>>> 99891 86280
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 & thanks
>> --
>> m nageshwara rao
>> 99891 86280
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 & thanks
--
m nageshwara rao
99891 86280
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From jnvines at gmail.com  Thu Nov 22 18:10:33 2012
From: jnvines at gmail.com (Nick Vines)
Date: Thu, 22 Nov 2012 10:10:33 -0500
Subject: [Freeswitch-users] Cisco-signed SSL Certificates/Alternative
 phone suggestions
In-Reply-To: <CAMbW9oL=AjAJCDn9D-tggWqPoy8pitDjJT0waZgH27HWJyOx0Q@mail.gmail.com>
References: <CAMbW9oL=AjAJCDn9D-tggWqPoy8pitDjJT0waZgH27HWJyOx0Q@mail.gmail.com>
Message-ID: <CA+NjZ2uUmyw_eVAubva3vP1QgpsKKY0ymHFkEt4kkpgRnSYhsQ@mail.gmail.com>

You have to email them. This page has a walkthrough that should help.

https://supportforums.cisco.com/docs/DOC-9852




On Thu, Nov 22, 2012 at 8:23 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Hi guys, just looking for a little help. I've been using Cisco SPA50x
> phones for a little while now, they have some pretty nice features for the
> price, including the ability to write custom applications in XML. However,
> to run them over HTTPS instead of HTTP, you need a certificate signed by
> Cisco, which I'm having difficulty obtaining.
>
> So my questions are:
>
>    1. Can anyone help me obtain such a certificate? I have the CSR
>    already to go.
>    2. Are there any similarly priced and featured phones available that
>    you guys have experience with?
>
> Thanks guys,
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
> 
> 
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
>
>
> http://wiki.freeswitch.org
>
>
> http://www.cluecon.com
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
> http://www.freeswitch.org
>
>
>
>
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From haloha201 at gmail.com  Thu Nov 22 18:39:19 2012
From: haloha201 at gmail.com (haloha)
Date: Thu, 22 Nov 2012 22:39:19 +0700
Subject: [Freeswitch-users] build freeswitch with lastest version problem
Message-ID: <CAMBhhBLmLxvX__zpZXZNUi2zXBhdo8gqa+2WFiwzrRYEYDgyHQ@mail.gmail.com>

hi
i am trying to install freeswitch with lastest version from git
my OS : centos version 6.3

and i get error when do command : ./configure
./configure: line 28276: unexpected EOF while looking for matching `"'
./configure: line 28281: syntax error: unexpected end of file
configure: error: ./configure.gnu failed for libs/libsndfile

how to fix it

thank you


From andrew at cassidywebservices.co.uk  Thu Nov 22 19:21:10 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 22 Nov 2012 16:21:10 +0000
Subject: [Freeswitch-users] Cisco-signed SSL Certificates/Alternative
 phone suggestions
In-Reply-To: <CA+NjZ2uUmyw_eVAubva3vP1QgpsKKY0ymHFkEt4kkpgRnSYhsQ@mail.gmail.com>
References: <CAMbW9oL=AjAJCDn9D-tggWqPoy8pitDjJT0waZgH27HWJyOx0Q@mail.gmail.com>
	<CA+NjZ2uUmyw_eVAubva3vP1QgpsKKY0ymHFkEt4kkpgRnSYhsQ@mail.gmail.com>
Message-ID: <CAMbW9oKXfAXF4sdQovffXEK4UyvK63pcGK7gQjyj=QrbaEBMmw@mail.gmail.com>

Followed that. Also:

*Note: A certificate will only be generated if a Cisco sales representative
sends the CSR to the email alias.*
*
*
The problem is I don't have a Cisco sales rep to deal with. I've tried
emailing that address myself, it's just a black hole for us normal folk.

On 22 November 2012 15:10, Nick Vines <jnvines at gmail.com> wrote:

> You have to email them. This page has a walkthrough that should help.
>
> https://supportforums.cisco.com/docs/DOC-9852
>
>
>
>
> On Thu, Nov 22, 2012 at 8:23 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> Hi guys, just looking for a little help. I've been using Cisco SPA50x
>> phones for a little while now, they have some pretty nice features for the
>> price, including the ability to write custom applications in XML. However,
>> to run them over HTTPS instead of HTTP, you need a certificate signed by
>> Cisco, which I'm having difficulty obtaining.
>>
>> So my questions are:
>>
>>    1. Can anyone help me obtain such a certificate? I have the CSR
>>    already to go.
>>    2. Are there any similarly priced and featured phones available that
>>    you guys have experience with?
>>
>> Thanks guys,
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>> 
>> 
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>>
>>
>> http://wiki.freeswitch.org
>>
>>
>> http://www.cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From ahmed at netelsat.net  Thu Nov 22 19:59:25 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Thu, 22 Nov 2012 21:59:25 +0500
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <50AE306E.7040909@panamaxil.com>
References: <50ACE9C4.60803@panamaxil.com> <50AD3FD5.4070606@quentustech.com>
	<CALFFxt3__=MPVF5a1o6eT+XXdHBDXQMt-EvHFYw+EMrqAwDrdw@mail.gmail.com>
	<50AE306E.7040909@panamaxil.com>
Message-ID: <CALFFxt1omcKDN7-n8-VbYE0mYuHBz1v8HW5RhbZO0N6Jdm=7Hw@mail.gmail.com>

Dear Chetan
i will be Glad to assist you.
there are series of steps you have to do .

1. first compile the xml-mod-radius , it will download Free-radius client
library itself which is buggy so you have a patch inside the xml-mod-radius
src folder. patch the free-radius client library and recompile the Module.

2. Use 1.2.stable but download xml-mod-radius module from master branch.
3. there is a xml-conf file inside the source which will help to deploy
either you want AAA for Class 5 or Class 4 traffic. We are doing on IP
invites .
4. sample configs are working fine with Jerasoft Radius server but
i believe it must work with any other radius server.
5. there is also a dialplan sample file to show how to force incoming call
to be passed via Radius authentication.
6. If you need any further help i am always there , But i am not expert
just a "happy user" of this module . Mr. William is the person who wrote
this module so he can definitely help you in details.
7. About Call Volume , we have tested it even on 5000+ calls , works
Perfect.

With warm regards
Ahmed Sboor
Netelsat Fze





On Thu, Nov 22, 2012 at 7:02 PM, Chetan Khatri
<chetan.khatri at panamaxil.com>wrote:

> **
> Hello Ahmed,
>
> Could please provide some guidance to configure xml_mod_radius i have
> deployed it.
>
> *Regards,
> Chetan Khatri*
>
> On 11/22/2012 03:43 AM, Ahmed Sboor wrote:
>
> yes we are using xml_mod_radius easily for more then 2000 calls.
> regards
> Ahmed
>
>
>
> On Thu, Nov 22, 2012 at 12:55 AM, William King <
> william.king at quentustech.com> wrote:
>
>> Try using xml_mod_radius instead.
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> On 11/21/2012 06:48 AM, Chetan Khatri wrote:
>> > Dear All Experts,
>> >
>> > Need help to solve freeswitch's mod_rad_auth capacity issue.
>> > I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The
>> > Module is working fine till the concurrent request reaches till 245
>> > after that Accounting is failing.
>> > Below is the Configuration.
>> >
>> > FreeSwitch : 1.0.6
>> > RADIUS Server : FreeRADIUS2
>> > Database server : MySQL 5.5
>> >
>> > We are stress testing the with freeswitch to handle about 2000
>> > concurrent calls with RADIUS AAA. without using RADIUS modules
>> > freeswitch is able to handle 2000 concurrent calls without any issues.
>> > For analysis we analysed from RADIUS server replies access-accept
>> > request, but in mod_rad_auth error says *"freeswitch : Invalid Digest
>> > Request from RADIUS server"* only if the concurrent session receives
>> > 245, till that all the requests are been working fine.
>> >
>> > Please guide me how to resolve this capacity issue, I can provide all
>> > the logs and configuration files if required.
>> >
>> > --
>> > /Regards,
>> > Chetan Khatri/
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From acrow at integrafin.co.uk  Thu Nov 22 20:12:48 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Thu, 22 Nov 2012 17:12:48 +0000
Subject: [Freeswitch-users] Patton M-ATA and T.38
In-Reply-To: <7C0F9004-CFDE-4834-971F-FAAFAE0F23FA@ipeva.fr>
References: <8A1257D8-1943-4B9B-A506-988391854760@ipeva.fr>
	<CALNZuwm=_hxXd4iy8mu4WQAbmjTQFnykgEEz2wMMxmVAfL9T6w@mail.gmail.com>
	<7C0F9004-CFDE-4834-971F-FAAFAE0F23FA@ipeva.fr>
Message-ID: <50AE5D10.5040001@integrafin.co.uk>

Hi David,

Sorry to cc you off list but I saw your post from May 2010 in which you 
said you got a Patton M-ATA working for T.38. Can you please provide 
details of your config (ie did you use t38_passthru or some other way)?

I've just taken delivery of a couple of them, wanting to use them to 
connect my fax machines via Freeswitch to a T.38 capable provider, but 
am having no luck at all (all outbound faxes fail - don't need inbound).

Any hints you can provide would be very gratefully received.

Regards

Alex


From acrow at integrafin.co.uk  Thu Nov 22 20:21:09 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Thu, 22 Nov 2012 17:21:09 +0000
Subject: [Freeswitch-users] SIP to TDM t38 gateway
In-Reply-To: <5BA9AADC-BB63-4FF5-B1A6-1DEC1BE3E931@5ninesolutions.com>
References: <5BA9AADC-BB63-4FF5-B1A6-1DEC1BE3E931@5ninesolutions.com>
Message-ID: <50AE5F05.1010402@integrafin.co.uk>

Hi Spencer,

I use this (gatewaying via a Mitel 3300 but there is no reason why it 
would not work for FreeTDM, just change the bridge at the end).


       <action application="set" data="absolute_codec_string=PCMU"/>
       <action application="set" data="fax_enable_t38=true"/>
<action application="set" data="sip_execute_on_image=t38_gateway self 
nocng"/>
       <action application="bridge" data="sofia/gateway/10.10.0.2/$1"/>


Hope this helps if a bit late.

Cheers

Alex

On 05/11/12 21:32, Spencer Thomason wrote:
> Hello,
> I'm trying to use Freeswitch as a SIP to TDM gateway.  I'd like to use t38_gateway to detect fax tones and send a ReINVITE to t38.
>
> Have a very minimal config with one profile that simply relays to FreeTDM
>
> My dialplan is:
> <include>
>    <context name="sip">
>
>      <extension name="dial-trunk">
>        <condition field="destination_number" expression="^(.*)$">
>          <action application="set" data="fax_enable_t38=true"/>
>          <action application="set" data="fax_enable_t38_request=true"/>
>          <action application="set" data="execute_on_answer=t38_gateway self"/>
>          <action application="bridge" data="freetdm/1/a/$1"/>
>        </condition>
>      </extension>
>
>    </context>
> </include>
>
> The problem is a media bug is created on the channel but almost immediately destroyed so fax tones are never detected.
>
> See:
> http://pastebin.freeswitch.org/20162
>
> Thanks for any assistance,
> Spencer
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From djbinter at gmail.com  Thu Nov 22 21:10:16 2012
From: djbinter at gmail.com (Dorn DJBinter)
Date: Thu, 22 Nov 2012 10:10:16 -0800
Subject: [Freeswitch-users] mod_voicemail.c file change
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>
Message-ID: <7874178664831298079@unknownmsgid>

Look at voicemail_play_greeting macro.

Sent from my iPad

On Nov 22, 2012, at 5:18 AM, Archana Venugopan <a.venugopan at mundio.com>
wrote:

  Hi,



Currently while dialling voicemail from mod_voicemail.c file it picks up
the name recorded in system greetings and says like



?<Name> person you are trying to reach is currently not available.Record
you message after the tone press any key or stop talking to end the
recording?



I require something like this

?You have reached the mailbox of <Name> Record you message after the tone
press any key or stop talking to end the recording?.



I knew I have to change in mod_voicemail.c file. But since am not familiar
with C code not sure how to get this.



if (!skip_greeting) {

                memset(buf, 0, sizeof(buf));

                args.input_callback = cancel_on_dtmf;

                args.buf = buf;

                args.buflen = sizeof(buf);



                switch_ivr_sleep(session, 100, SWITCH_TRUE, NULL);



                if (switch_file_exists(greet_path,
switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {

                        memset(buf, 0, sizeof(buf));

                        TRY_CODE(switch_ivr_play_file(session, NULL,
greet_path, &args));

                } else {

                        if (switch_file_exists(cbt.name_path,
switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {

                                memset(buf, 0, sizeof(buf));

                                TRY_CODE(switch_ivr_play_file(session,
NULL, cbt.name_path, &args));

                        }

                        if (*buf == '\0') {

                                if (!read_id) {

                                        if (!(read_id =
switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {

                                                read_id = id;

                                        }

                                }

                                memset(buf, 0, sizeof(buf));

                                TRY_CODE(switch_ivr_phrase_macro(session,
VM_PLAY_GREETING_MACRO, read_id, NULL, &args));

                        }



Can anyone please help me out. Thanks



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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From chetan.khatri at panamaxil.com  Thu Nov 22 22:00:32 2012
From: chetan.khatri at panamaxil.com (Chetan Khatri)
Date: Fri, 23 Nov 2012 00:30:32 +0530
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <CALFFxt1omcKDN7-n8-VbYE0mYuHBz1v8HW5RhbZO0N6Jdm=7Hw@mail.gmail.com>
References: <50ACE9C4.60803@panamaxil.com>
	<50AD3FD5.4070606@quentustech.com>	<CALFFxt3__=MPVF5a1o6eT+XXdHBDXQMt-EvHFYw+EMrqAwDrdw@mail.gmail.com>	<50AE306E.7040909@panamaxil.com>
	<CALFFxt1omcKDN7-n8-VbYE0mYuHBz1v8HW5RhbZO0N6Jdm=7Hw@mail.gmail.com>
Message-ID: <50AE7650.7010306@panamaxil.com>

Dear Ahmed,

First of all thanks a lot , for your assistance i really appreciate it.

I have some queries if you can please resolve it.

-    Is mod_xml_radius the same you are talking about. I found that in 
source  of freeswitch at location src/mod/xml_int.
-    I am unable to find any other module like xml-mod-radius. if the 
above one is not the same can you please provide from what site location 
i can download the source and compile it.


/Regards,
Chetan Khatri/

On 11/22/2012 10:29 PM, Ahmed Sboor wrote:
> Dear Chetan
> i will be Glad to assist you.
> there are series of steps you have to do .
>
> 1. first compile the xml-mod-radius , it will download Free-radius 
> client library itself which is buggy so you have a patch inside the 
> xml-mod-radius src folder. patch the free-radius client library and 
> recompile the Module.
>
> 2. Use 1.2.stable but download xml-mod-radius module from master branch.
> 3. there is a xml-conf file inside the source which will help to 
> deploy either you want AAA for Class 5 or Class 4 traffic. We are 
> doing on IP invites .
> 4. sample configs are working fine with Jerasoft Radius server but 
> i believe it must work with any other radius server.
> 5. there is also a dialplan sample file to show how to force incoming 
> call to be passed via Radius authentication.
> 6. If you need any further help i am always there , But i am not 
> expert just a "happy user" of this module . Mr. William is the person 
> who wrote this module so he can definitely help you in details.
> 7. About Call Volume , we have tested it even on 5000+ calls , works 
> Perfect.
>
> With warm regards
> Ahmed Sboor
> Netelsat Fze
>
>
>
>
>
> On Thu, Nov 22, 2012 at 7:02 PM, Chetan Khatri 
> <chetan.khatri at panamaxil.com <mailto:chetan.khatri at panamaxil.com>> wrote:
>
>     Hello Ahmed,
>
>     Could please provide some guidance to configure xml_mod_radius i
>     have deployed it.
>
>     /Regards,
>     Chetan Khatri/
>
>     On 11/22/2012 03:43 AM, Ahmed Sboor wrote:
>>     yes we are using xml_mod_radius easily for more then 2000 calls.
>>     regards
>>     Ahmed
>>
>>
>>
>>     On Thu, Nov 22, 2012 at 12:55 AM, William King
>>     <william.king at quentustech.com
>>     <mailto:william.king at quentustech.com>> wrote:
>>
>>         Try using xml_mod_radius instead.
>>
>>         William King
>>         Senior Engineer
>>         Quentus Technologies, INC
>>         1037 NE 65th St Suite 273
>>         Seattle, WA 98115
>>         Main:   (877) 211-9337
>>         Office: (206) 388-4772
>>         Cell:   (253) 686-5518
>>         william.king at quentustech.com
>>         <mailto:william.king at quentustech.com>
>>
>>         On 11/21/2012 06:48 AM, Chetan Khatri wrote:
>>         > Dear All Experts,
>>         >
>>         > Need help to solve freeswitch's mod_rad_auth capacity issue.
>>         > I have deployed freeswtich 1.0.6 and enabled mod_rad_auth
>>         module. The
>>         > Module is working fine till the concurrent request reaches
>>         till 245
>>         > after that Accounting is failing.
>>         > Below is the Configuration.
>>         >
>>         > FreeSwitch : 1.0.6
>>         > RADIUS Server : FreeRADIUS2
>>         > Database server : MySQL 5.5
>>         >
>>         > We are stress testing the with freeswitch to handle about 2000
>>         > concurrent calls with RADIUS AAA. without using RADIUS modules
>>         > freeswitch is able to handle 2000 concurrent calls without
>>         any issues.
>>         > For analysis we analysed from RADIUS server replies
>>         access-accept
>>         > request, but in mod_rad_auth error says *"freeswitch :
>>         Invalid Digest
>>         > Request from RADIUS server"* only if the concurrent session
>>         receives
>>         > 245, till that all the requests are been working fine.
>>         >
>>         > Please guide me how to resolve this capacity issue, I can
>>         provide all
>>         > the logs and configuration files if required.
>>         >
>>         > --
>>         > /Regards,
>>         > Chetan Khatri/
>>         >
>>         >
>>         >
>>         _________________________________________________________________________
>>         > Professional FreeSWITCH Consulting Services:
>>         > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         > http://www.freeswitchsolutions.com
>>         >
>>         > 
>>         > 
>>         >
>>         > Official FreeSWITCH Sites
>>         > http://www.freeswitch.org
>>         > http://wiki.freeswitch.org
>>         > http://www.cluecon.com
>>         >
>>         > FreeSWITCH-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
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-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
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-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
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From fs-list at communicatefreely.net  Thu Nov 22 22:31:40 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 22 Nov 2012 14:31:40 -0500
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting again
Message-ID: <50AE7D9C.8040604@communicatefreely.net>

Hello,

I'm having a bit of an odd problem.

Intermittently, often every 2-3 days or so, Freeswitch stops replying to SIP for about 5 
minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to be pretty close.

During this time, no new registrations or invites can happen, but existing calls stay 
connected for at least a minute or two.  In the logs, you can see calls slowly hanging up 
with "NORMAL_CLEARING".  In 5 minutes, everything starts up again with no word about it at 
all in the logs.

When calls resume, I notice that the number of sessions returned by the status command is 
one higher than the actual number sessions returned by show channels, or by looking in the 
database.  Every time this happens, the discrepancy increases by one.

The interruption happens on all SIP profiles, but calls originated from the socket API 
still work, insofar as they return with PROGRESS_TIMEOUT since the profiles are still 
running, but stuck.

We are using ODBC/MySQL for the core database, and the database server only runs this 
database and some basic PHP/xml-curl stuff.

We have 416 endpoints registered, and usually sit at about 30 sessions during the day.

This never happens at night, only during busier times, but not necessarily busy hour.

I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)

Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)

Yes, I know it's old and I'm trying to upgrade, but I'm still having some problems getting 
all my phones to work properly with 1.2 stable.  This is a production system, so I can't 
just blindly put out the newest release.  Mostly, I need to buy myself some time so that I 
can get the kinks worked out of the latest version and then upgrade the production box.

I'm grateful for any insights as to what could be happening, even if a solution is just a 
temporary workaround.

Thanks!

-Tim


From ahmed at netelsat.net  Thu Nov 22 23:03:10 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Fri, 23 Nov 2012 01:03:10 +0500
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <50AE7650.7010306@panamaxil.com>
References: <50ACE9C4.60803@panamaxil.com> <50AD3FD5.4070606@quentustech.com>
	<CALFFxt3__=MPVF5a1o6eT+XXdHBDXQMt-EvHFYw+EMrqAwDrdw@mail.gmail.com>
	<50AE306E.7040909@panamaxil.com>
	<CALFFxt1omcKDN7-n8-VbYE0mYuHBz1v8HW5RhbZO0N6Jdm=7Hw@mail.gmail.com>
	<50AE7650.7010306@panamaxil.com>
Message-ID: <CALFFxt0KBH869A0m0y=OaGpwi-JTR_jVNbsiNmykJ4cO-XukeQ@mail.gmail.com>

off course i was talking for the same you found
its /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_radius Normally.


On Fri, Nov 23, 2012 at 12:00 AM, Chetan Khatri <chetan.khatri at panamaxil.com
> wrote:

> **
> Dear Ahmed,
>
> First of all thanks a lot , for your assistance i really appreciate it.
>
> I have some queries if you can please resolve it.
>
> -    Is mod_xml_radius the same you are talking about. I found that in
> source  of freeswitch at location src/mod/xml_int.
> -    I am unable to find any other module like xml-mod-radius. if the
> above one is not the same can you please provide from what site location i
> can download the source and compile it.
>
>
> *Regards,
> Chetan Khatri*
>
> On 11/22/2012 10:29 PM, Ahmed Sboor wrote:
>
> Dear Chetan
> i will be Glad to assist you.
> there are series of steps you have to do .
>
>  1. first compile the xml-mod-radius , it will download Free-radius
> client library itself which is buggy so you have a patch inside the
> xml-mod-radius src folder. patch the free-radius client library and
> recompile the Module.
>
>  2. Use 1.2.stable but download xml-mod-radius module from master branch.
> 3. there is a xml-conf file inside the source which will help to deploy
> either you want AAA for Class 5 or Class 4 traffic. We are doing on IP
> invites .
> 4. sample configs are working fine with Jerasoft Radius server but
> i believe it must work with any other radius server.
> 5. there is also a dialplan sample file to show how to force incoming call
> to be passed via Radius authentication.
> 6. If you need any further help i am always there , But i am not expert
> just a "happy user" of this module . Mr. William is the person who wrote
> this module so he can definitely help you in details.
>  7. About Call Volume , we have tested it even on 5000+ calls , works
> Perfect.
>
>  With warm regards
> Ahmed Sboor
> Netelsat Fze
>
>
>
>
>
> On Thu, Nov 22, 2012 at 7:02 PM, Chetan Khatri <
> chetan.khatri at panamaxil.com> wrote:
>
>>  Hello Ahmed,
>>
>> Could please provide some guidance to configure xml_mod_radius i have
>> deployed it.
>>
>> *Regards,
>> Chetan Khatri*
>>
>> On 11/22/2012 03:43 AM, Ahmed Sboor wrote:
>>
>> yes we are using xml_mod_radius easily for more then 2000 calls.
>> regards
>> Ahmed
>>
>>
>>
>> On Thu, Nov 22, 2012 at 12:55 AM, William King <
>> william.king at quentustech.com> wrote:
>>
>>> Try using xml_mod_radius instead.
>>>
>>> William King
>>> Senior Engineer
>>> Quentus Technologies, INC
>>> 1037 NE 65th St Suite 273
>>> Seattle, WA 98115
>>> Main:   (877) 211-9337
>>> Office: (206) 388-4772
>>> Cell:   (253) 686-5518
>>> william.king at quentustech.com
>>>
>>> On 11/21/2012 06:48 AM, Chetan Khatri wrote:
>>> > Dear All Experts,
>>> >
>>> > Need help to solve freeswitch's mod_rad_auth capacity issue.
>>> > I have deployed freeswtich 1.0.6 and enabled mod_rad_auth module. The
>>> > Module is working fine till the concurrent request reaches till 245
>>> > after that Accounting is failing.
>>> > Below is the Configuration.
>>> >
>>> > FreeSwitch : 1.0.6
>>> > RADIUS Server : FreeRADIUS2
>>> > Database server : MySQL 5.5
>>> >
>>> > We are stress testing the with freeswitch to handle about 2000
>>> > concurrent calls with RADIUS AAA. without using RADIUS modules
>>> > freeswitch is able to handle 2000 concurrent calls without any issues.
>>> > For analysis we analysed from RADIUS server replies access-accept
>>> > request, but in mod_rad_auth error says *"freeswitch : Invalid Digest
>>> > Request from RADIUS server"* only if the concurrent session receives
>>> > 245, till that all the requests are been working fine.
>>> >
>>> > Please guide me how to resolve this capacity issue, I can provide all
>>> > the logs and configuration files if required.
>>> >
>>> > --
>>> > /Regards,
>>> > Chetan Khatri/
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From packetandy at gmail.com  Fri Nov 23 03:34:45 2012
From: packetandy at gmail.com (andy)
Date: Thu, 22 Nov 2012 16:34:45 -0800
Subject: [Freeswitch-users] changing vm default announcement
Message-ID: <50AEC4A5.50907@gmail.com>

Michael,

I verified that "export" works with loopback. I will add to the wiki as 
soon as I get a login

cheers
andy


From luis.daniel.lucio at gmail.com  Fri Nov 23 06:21:29 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Thu, 22 Nov 2012 22:21:29 -0500
Subject: [Freeswitch-users] global dial plan for *SUPPORT/*7877678
Message-ID: <CAFLo2QzmivvcUA=_+_LBmthzL6-5U48ne=kGNBauvJ4yG3Rt7w@mail.gmail.com>

Helo,

Well i have FS with multidomain tenant installed and working.

I wonder if there is an easy way to place in default.xml file a dial
plan to let any extension, from any tenant, that when he/she dials
*7877678 it always ring in extension 001 at mytenant

is that possible?
LD


From blee at gocentrix.com  Fri Nov 23 07:46:24 2012
From: blee at gocentrix.com (Bryant Lee)
Date: Thu, 22 Nov 2012 23:46:24 -0500
Subject: [Freeswitch-users] global dial plan for *SUPPORT/*7877678
In-Reply-To: <CAFLo2QzmivvcUA=_+_LBmthzL6-5U48ne=kGNBauvJ4yG3Rt7w@mail.gmail.com>
References: <CAFLo2QzmivvcUA=_+_LBmthzL6-5U48ne=kGNBauvJ4yG3Rt7w@mail.gmail.com>
Message-ID: <CAGodFm_5SkD9+md7RcJRaG2F3xDpCQiYSxoNmNotYrLf+FPb4g@mail.gmail.com>

My suggestion is to create a new file with this extension, and then do a
XML include to this file in all your dialplans.

Best regards,




On Thu, Nov 22, 2012 at 10:21 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Helo,
>
> Well i have FS with multidomain tenant installed and working.
>
> I wonder if there is an easy way to place in default.xml file a dial
> plan to let any extension, from any tenant, that when he/she dials
> *7877678 it always ring in extension 001 at mytenant
>
> is that possible?
> LD
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Fri Nov 23 07:59:12 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 22 Nov 2012 20:59:12 -0800
Subject: [Freeswitch-users] global dial plan for *SUPPORT/*7877678
In-Reply-To: <CAFLo2QzmivvcUA=_+_LBmthzL6-5U48ne=kGNBauvJ4yG3Rt7w@mail.gmail.com>
References: <CAFLo2QzmivvcUA=_+_LBmthzL6-5U48ne=kGNBauvJ4yG3Rt7w@mail.gmail.com>
Message-ID: <CAMWiMy9k9CnOSz5n9gaKhHUA-XRvpdTfLMjJHdG48GsxkfmrRA@mail.gmail.com>

Yes, more than possible. You'll just need to get the regex to escape
the asterisk and have it to transfer/bridge to your extension. That's
all.

Hint: Look in the default.xml of the default config to see an example
of how you escape the asterisk

On Thu, Nov 22, 2012 at 7:21 PM, Luis Daniel Lucio Quiroz
<luis.daniel.lucio at gmail.com> wrote:
> Helo,
>
> Well i have FS with multidomain tenant installed and working.
>
> I wonder if there is an easy way to place in default.xml file a dial
> plan to let any extension, from any tenant, that when he/she dials
> *7877678 it always ring in extension 001 at mytenant
>
> is that possible?
> LD
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From a.venugopan at mundio.com  Fri Nov 23 12:08:05 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 23 Nov 2012 09:08:05 +0000
Subject: [Freeswitch-users] mod_voicemail.c file change
In-Reply-To: <7874178664831298079@unknownmsgid>
References: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>
	<7874178664831298079@unknownmsgid>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF23378D5@Mail-Kilo.squay.com>

Hi,

Yes currently that plays 'person you are trying to reach is currently not available'. But I require this sentence first 'You have reached the mailbox of' followed by <Name> instead of Name reading at first.

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dorn DJBinter
Sent: 22 November 2012 18:10
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail.c file change

Look at voicemail_play_greeting macro.

Sent from my iPad

On Nov 22, 2012, at 5:18 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently while dialling voicemail from mod_voicemail.c file it picks up the name recorded in system greetings and says like

'<Name> person you are trying to reach is currently not available.Record you message after the tone press any key or stop talking to end the recording'

I require something like this
'You have reached the mailbox of <Name> Record you message after the tone press any key or stop talking to end the recording'.

I knew I have to change in mod_voicemail.c file. But since am not familiar with C code not sure how to get this.

if (!skip_greeting) {
                memset(buf, 0, sizeof(buf));
                args.input_callback = cancel_on_dtmf;
                args.buf = buf;
                args.buflen = sizeof(buf);

                switch_ivr_sleep(session, 100, SWITCH_TRUE, NULL);

                if (switch_file_exists(greet_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                        memset(buf, 0, sizeof(buf));
                        TRY_CODE(switch_ivr_play_file(session, NULL, greet_path, &args));
                } else {
                        if (switch_file_exists(cbt.name_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_play_file(session, NULL, cbt.name_path, &args));
                        }
                        if (*buf == '\0') {
                                if (!read_id) {
                                        if (!(read_id = switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {
                                                read_id = id;
                                        }
                                }
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO, read_id, NULL, &args));
                        }

Can anyone please help me out. Thanks

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




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

FreeSWITCH-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://lists.freeswitch.org/mailman/options/freeswitch-users>
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From a.venugopan at mundio.com  Fri Nov 23 13:03:07 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 23 Nov 2012 10:03:07 +0000
Subject: [Freeswitch-users] mod_voicemail compilation error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337901@Mail-Kilo.squay.com>

Hi,

I tried putting debug statements in mod_voicemail.c file and tried to re-compile it but I got this msg and now its not going to voicemail. Earlier it was working properly but after this voicemail was not working. Please help.

[root at squay-laptop-1 freeswitch]# make mod_voicemail-install
 /bin/sh /usr/local/src/freeswitch/quiet_libtool   --mode=install /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai /usr/local/freeswitch/lib/libfreeswitch.la
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: finish: PATH="/usr/lib/qt-3.3/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/sbin" ldconfig -n /usr/local/freeswitch/lib
----------------------------------------------------------------------
Libraries have been installed in:
   /usr/local/freeswitch/lib

If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
     during execution
   - add LIBDIR to the `LD_RUN_PATH' environment variable
     during linking
   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
   - have your system administrator add LIBDIR to `/etc/ld.so.conf'

See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
----------------------------------------------------------------------

making install mod_voicemail
Compiling /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c...
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c  -fPIC -DPIC -o .libs/mod_voicemail.o
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c -o mod_voicemail.o >/dev/null 2>&1
Creating mod_voicemail.la...
installing mod_voicemail.la
quiet_libtool: install: warning: relinking `mod_voicemail.la'


freeswitch at internal> reload mod_voicemail
+OK module unloaded
+OK Reloading XML
-ERR loading module [module load file routine returned an error]

2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:901 Deleting Application 'voicemail'
freeswitch at internal> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:903 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_boxcount'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_boxcount' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_prefs'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_prefs' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_read'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_read' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_auth_login'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_auth_login' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_count'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_count' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_get'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_get' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_undelete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_undelete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_purge'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_purge' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_save'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_save' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_forward'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_forward' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_greeting_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_greeting_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_recname_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_recname_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_password_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_password_set' to wait for existing references.
2012-11-23 09:58:31.448059 [CONSOLE] switch_loadable_module.c:1765 Stopping: mod_voicemail
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:442 Subclass reservation deleted for /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:vm::maintenance
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:1889 Event Binding deleted for mod_voicemail:MESSAGE_QUERY
2012-11-23 09:58:31.518046 [CONSOLE] mod_voicemail.c:3840 Event Thread Ended
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5759 Waiting for write lock (Profile default)
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5762 Destroying Profile default
2012-11-23 09:58:31.518046 [CONSOLE] switch_loadable_module.c:1785 mod_voicemail unloaded.
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 1
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 44
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 49
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_enum.c:775 ENUM Reloaded
2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so
**/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol: switch_channel_expand_variables_check**
2012-11-23 09:58:32.778219 [INFO] switch_time.c:1035 Timezone reloaded 530 definitions



Regards,
Archana

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From peter.olsson at visionutveckling.se  Fri Nov 23 13:46:13 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 23 Nov 2012 10:46:13 +0000
Subject: [Freeswitch-users] mod_voicemail compilation error
Message-ID: <1FFF97C269757C458224B7C895F35F151CAFE9@cantor.std.visionutv.se>

Rebuild the entire project instead. It seems that the current libfreeswitch.so doesn't support switch_channel_expand_variables_check().

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Archana Venugopan
Skickat: den 23 november 2012 11:03
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] mod_voicemail compilation error

Hi,

I tried putting debug statements in mod_voicemail.c file and tried to re-compile it but I got this msg and now its not going to voicemail. Earlier it was working properly but after this voicemail was not working. Please help.

[root at squay-laptop-1 freeswitch]# make mod_voicemail-install
 /bin/sh /usr/local/src/freeswitch/quiet_libtool   --mode=install /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai /usr/local/freeswitch/lib/libfreeswitch.la
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: finish: PATH="/usr/lib/qt-3.3/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/sbin" ldconfig -n /usr/local/freeswitch/lib
----------------------------------------------------------------------
Libraries have been installed in:
   /usr/local/freeswitch/lib

If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
     during execution
   - add LIBDIR to the `LD_RUN_PATH' environment variable
     during linking
   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
   - have your system administrator add LIBDIR to `/etc/ld.so.conf'

See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
----------------------------------------------------------------------

making install mod_voicemail
Compiling /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c...
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c  -fPIC -DPIC -o .libs/mod_voicemail.o
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c -o mod_voicemail.o >/dev/null 2>&1
Creating mod_voicemail.la...
installing mod_voicemail.la
quiet_libtool: install: warning: relinking `mod_voicemail.la'


freeswitch at internal> reload mod_voicemail
+OK module unloaded
+OK Reloading XML
-ERR loading module [module load file routine returned an error]

2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:901 Deleting Application 'voicemail'
freeswitch at internal> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:903 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_boxcount'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_boxcount' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_prefs'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_prefs' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_read'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_read' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_auth_login'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_auth_login' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_count'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_count' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_get'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_get' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_undelete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_undelete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_purge'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_purge' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_save'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_save' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_forward'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_forward' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_greeting_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_greeting_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_recname_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_recname_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_password_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_password_set' to wait for existing references.
2012-11-23 09:58:31.448059 [CONSOLE] switch_loadable_module.c:1765 Stopping: mod_voicemail
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:442 Subclass reservation deleted for /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:vm::maintenance
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:1889 Event Binding deleted for mod_voicemail:MESSAGE_QUERY
2012-11-23 09:58:31.518046 [CONSOLE] mod_voicemail.c:3840 Event Thread Ended
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5759 Waiting for write lock (Profile default)
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5762 Destroying Profile default
2012-11-23 09:58:31.518046 [CONSOLE] switch_loadable_module.c:1785 mod_voicemail unloaded.
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 1
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 44
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 49
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_enum.c:775 ENUM Reloaded
2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so
**/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol: switch_channel_expand_variables_check**
2012-11-23 09:58:32.778219 [INFO] switch_time.c:1035 Timezone reloaded 530 definitions



Regards,
Archana

!DSPAM:50af4e1232761560677559!
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From steveayre at gmail.com  Fri Nov 23 14:12:19 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 11:12:19 +0000
Subject: [Freeswitch-users] mod_voicemail.c file change
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23378D5@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>
	<7874178664831298079@unknownmsgid>
	<592A9CF93E12394E8472A6CC66E66BF23378D5@Mail-Kilo.squay.com>
Message-ID: <CAFiqYu=qtBWUSxsqj=OeHM6k-LDbH-YT4=biHYRouvh85Vk43w@mail.gmail.com>

Leave the C code, you don't need to know or change anything in there.

The voicemail_play_greeting macro is a configuration option.

Look in your configuration at lang/en/vm/sounds.xml, the macro is defined
in there as a combination of play-file and say to produce the message
that's heard.

Changing that macro will change what's heard.

-Steve




On 23 November 2012 09:08, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi,****
>
> ** **
>
> Yes currently that plays ?person you are trying to reach is currently not
> available?. But I require this sentence first ?You have reached the mailbox
> of? followed by <Name> instead of Name reading at first.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Dorn
> DJBinter
> *Sent:* 22 November 2012 18:10
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_voicemail.c file change****
>
> ** **
>
> Look at voicemail_play_greeting macro.
>
> Sent from my iPad****
>
>
> On Nov 22, 2012, at 5:18 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
>  Hi,****
>
>  ****
>
> Currently while dialling voicemail from mod_voicemail.c file it picks up
> the name recorded in system greetings and says like****
>
>  ****
>
> ?<Name> person you are trying to reach is currently not available.Record
> you message after the tone press any key or stop talking to end the
> recording?****
>
>  ****
>
> I require something like this****
>
> ?You have reached the mailbox of <Name> Record you message after the tone
> press any key or stop talking to end the recording?.****
>
>  ****
>
> I knew I have to change in mod_voicemail.c file. But since am not familiar
> with C code not sure how to get this.****
>
>  ****
>
> if (!skip_greeting) {****
>
>                 memset(buf, 0, sizeof(buf));****
>
>                 args.input_callback = cancel_on_dtmf;****
>
>                 args.buf = buf;****
>
>                 args.buflen = sizeof(buf);****
>
>  ****
>
>                 switch_ivr_sleep(session, 100, SWITCH_TRUE, NULL);****
>
>  ****
>
>                 if (switch_file_exists(greet_path,
> switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {****
>
>                         memset(buf, 0, sizeof(buf));****
>
>                         TRY_CODE(switch_ivr_play_file(session, NULL,
> greet_path, &args));****
>
>                 } else {****
>
>                         if (switch_file_exists(cbt.name_path,
> switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {****
>
>                                 memset(buf, 0, sizeof(buf));****
>
>                                 TRY_CODE(switch_ivr_play_file(session,
> NULL, cbt.name_path, &args));****
>
>                         }****
>
>                         if (*buf == '\0') {****
>
>                                 if (!read_id) {****
>
>                                         if (!(read_id =
> switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {**
> **
>
>                                                 read_id = id;****
>
>                                         }****
>
>                                 }****
>
>                                 memset(buf, 0, sizeof(buf));****
>
>                                 TRY_CODE(switch_ivr_phrase_macro(session,
> VM_PLAY_GREETING_MACRO, read_id, NULL, &args));****
>
>                         }****
>
>  ****
>
> Can anyone please help me out. Thanks****
>
>  ****
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Nov 23 15:08:53 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 12:08:53 +0000
Subject: [Freeswitch-users] mod_voicemail compilation error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337901@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337901@Mail-Kilo.squay.com>
Message-ID: <CAFiqYum73QwhygFLD4fziFEx9xxFCBYjbrM6c6BZYHKJTQ6nog@mail.gmail.com>

2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error
Loading module /usr/local/freeswitch/mod/mod_voicemail.so****

**/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol:
switch_channel_expand_variables_check******

Module hasn't compiled properly so cannot load.

You could try another 'make mod_voicemail' then the install, or rebuilding
the entire source again.

Why are you trying to insert debug messages? Is this related to your
greeting prompt query, which I answered in the other thread isn't something
that needs any changes to the code, just your configuration files modified?

-Steve



On 23 November 2012 10:03, Archana Venugopan <a.venugopan at mundio.com> wrote:

> 2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error
> Loading module /usr/local/freeswitch/mod/mod_voicemail.so****
>
> **/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol:
> switch_channel_expand_variables_check******
>
>
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From steveayre at gmail.com  Fri Nov 23 15:16:41 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 12:16:41 +0000
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
	again
In-Reply-To: <50AE7D9C.8040604@communicatefreely.net>
References: <50AE7D9C.8040604@communicatefreely.net>
Message-ID: <CAFiqYu=WYMb2aVqMh-POy-V+fbZMq_QOsn_5a2P=eg_EAtEKow@mail.gmail.com>

>
> Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)


As you've already acknowledged it's a very old version.

It's possible that your issue has already been found and fixed, but if it
hasn't then the code will have changed significantly since then and you'd
really need to reproduce it on the latest code for it to be investigated.


As some general thoughts though, are you able to spot it happening while
it's happening or only afterwards?

If you're able to get on the system during one of those times look at what
else is happening. Is the load average/cpu usage/io high? Perhaps
something's running that's blocking all access or causing very high IO.

What DB backend are you using for Sofia? Is it possible that that's hanging
for a moment? For example if you're running a backup on the DB that blocks
all writes to the DB while Sofia is trying to update the DB that perhaps
would cause this.

Try running a SIP OPTIONS ping your your sofia profile from the localhost
during that time, which should exclude it being any issue on the ethernet.

-Steve




On 22 November 2012 19:31, Tim St. Pierre <fs-list at communicatefreely.net>wrote:

> Hello,
>
> I'm having a bit of an odd problem.
>
> Intermittently, often every 2-3 days or so, Freeswitch stops replying to
> SIP for about 5
> minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to be
> pretty close.
>
> During this time, no new registrations or invites can happen, but existing
> calls stay
> connected for at least a minute or two.  In the logs, you can see calls
> slowly hanging up
> with "NORMAL_CLEARING".  In 5 minutes, everything starts up again with no
> word about it at
> all in the logs.
>
> When calls resume, I notice that the number of sessions returned by the
> status command is
> one higher than the actual number sessions returned by show channels, or
> by looking in the
> database.  Every time this happens, the discrepancy increases by one.
>
> The interruption happens on all SIP profiles, but calls originated from
> the socket API
> still work, insofar as they return with PROGRESS_TIMEOUT since the
> profiles are still
> running, but stuck.
>
> We are using ODBC/MySQL for the core database, and the database server
> only runs this
> database and some basic PHP/xml-curl stuff.
>
> We have 416 endpoints registered, and usually sit at about 30 sessions
> during the day.
>
> This never happens at night, only during busier times, but not necessarily
> busy hour.
>
> I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
>
> Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>
> Yes, I know it's old and I'm trying to upgrade, but I'm still having some
> problems getting
> all my phones to work properly with 1.2 stable.  This is a production
> system, so I can't
> just blindly put out the newest release.  Mostly, I need to buy myself
> some time so that I
> can get the kinks worked out of the latest version and then upgrade the
> production box.
>
> I'm grateful for any insights as to what could be happening, even if a
> solution is just a
> temporary workaround.
>
> Thanks!
>
> -Tim
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Nov 23 15:23:17 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 12:23:17 +0000
Subject: [Freeswitch-users] Rewriting media address in SDP as well as
 contact IP/port
In-Reply-To: <1353528669121-7584820.post@n2.nabble.com>
References: <054001cdc664$92486660$b6d93320$@com>
	<CAFiqYukz4GtOYJz2NwT7SKuhLyB1BewJYcujB80-o_NpKNEPig@mail.gmail.com>
	<1353443727657-7584773.post@n2.nabble.com>
	<CAKDfjgcmcA8tp-7WVEzxxHYtUXkyPxXmm3d0dSQqV54CNBwsRQ@mail.gmail.com>
	<1353528669121-7584820.post@n2.nabble.com>
Message-ID: <CAFiqYu=khJJ-0wOW6YD35M90i9Y1xoEN_dQmS3XxvmzmLUq8cg@mail.gmail.com>

>
> configuring endpoints to use a particular port for RTP and
> then forwarding a range of ports for each endpoint


Guessing this is how you're working around the fact the external RTP port
won't be the same as the internal one through most NAT implementations.

Seems an awful lot of work to go to when it'd be easier to configure a
client correctly (assuming they support NAT correctly). A lot of work for
you to setup and maintain, and that won't scale well.

If the clients can't support NAT correctly you could try using the SIP ALG
on the NAT router (if available) which should rewrite the RTP IP/port
automatically. Usually the advice is to disable that and let FS & the
client handle the NAT (because generally it just causes more problems if
you're handling the NAT twice). But in the bypass_media case it might help.
Does mean you can't use TLS though (since the router can't see inside the
encrypted packets), and it'll probably only spot SIP on port 5060.

-Steve





On 21 November 2012 20:11, PhilQ <philq at qsystemsengineering.com> wrote:

> configuring endpoints to use a particular port for RTP and
> then forwarding a range of ports for each endpoint
>
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From a.venugopan at mundio.com  Fri Nov 23 15:25:09 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 23 Nov 2012 12:25:09 +0000
Subject: [Freeswitch-users] mod_voicemail compilation error
In-Reply-To: <1FFF97C269757C458224B7C895F35F151CAFE9@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151CAFE9@cantor.std.visionutv.se>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337941@Mail-Kilo.squay.com>

Hi,
Thanks. But it worked when the freeswitch was build. After making changes and do a compile this is not supporting and giving this error.
The same libfreeswitch.so might have supported switch_channel_expand_variables_check() during initial build as well right?

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 23 November 2012 10:46
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail compilation error

Rebuild the entire project instead. It seems that the current libfreeswitch.so doesn't support switch_channel_expand_variables_check().

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Archana Venugopan
Skickat: den 23 november 2012 11:03
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] mod_voicemail compilation error

Hi,

I tried putting debug statements in mod_voicemail.c file and tried to re-compile it but I got this msg and now its not going to voicemail. Earlier it was working properly but after this voicemail was not working. Please help.

[root at squay-laptop-1 freeswitch]# make mod_voicemail-install
 /bin/sh /usr/local/src/freeswitch/quiet_libtool   --mode=install /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai /usr/local/freeswitch/lib/libfreeswitch.la
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: finish: PATH="/usr/lib/qt-3.3/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/sbin" ldconfig -n /usr/local/freeswitch/lib
----------------------------------------------------------------------
Libraries have been installed in:
   /usr/local/freeswitch/lib

If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
     during execution
   - add LIBDIR to the `LD_RUN_PATH' environment variable
     during linking
   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
   - have your system administrator add LIBDIR to `/etc/ld.so.conf'

See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
----------------------------------------------------------------------

making install mod_voicemail
Compiling /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c...
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c  -fPIC -DPIC -o .libs/mod_voicemail.o
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c -o mod_voicemail.o >/dev/null 2>&1
Creating mod_voicemail.la...
installing mod_voicemail.la
quiet_libtool: install: warning: relinking `mod_voicemail.la'


freeswitch at internal> reload mod_voicemail
+OK module unloaded
+OK Reloading XML
-ERR loading module [module load file routine returned an error]

2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:901 Deleting Application 'voicemail'
freeswitch at internal> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:903 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_boxcount'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_boxcount' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_prefs'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_prefs' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_read'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_read' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_auth_login'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_auth_login' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_count'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_count' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_get'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_get' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_undelete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_undelete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_purge'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_purge' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_save'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_save' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_forward'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_forward' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_greeting_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_greeting_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_recname_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_recname_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_password_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_password_set' to wait for existing references.
2012-11-23 09:58:31.448059 [CONSOLE] switch_loadable_module.c:1765 Stopping: mod_voicemail
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:442 Subclass reservation deleted for /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:vm::maintenance
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:1889 Event Binding deleted for mod_voicemail:MESSAGE_QUERY
2012-11-23 09:58:31.518046 [CONSOLE] mod_voicemail.c:3840 Event Thread Ended
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5759 Waiting for write lock (Profile default)
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5762 Destroying Profile default
2012-11-23 09:58:31.518046 [CONSOLE] switch_loadable_module.c:1785 mod_voicemail unloaded.
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 1
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 44
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 49
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_enum.c:775 ENUM Reloaded
2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so
**/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol: switch_channel_expand_variables_check**
2012-11-23 09:58:32.778219 [INFO] switch_time.c:1035 Timezone reloaded 530 definitions



Regards,
Archana

!DSPAM:50af4e1232761560677559!
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From steveayre at gmail.com  Fri Nov 23 15:27:31 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 12:27:31 +0000
Subject: [Freeswitch-users] SRTP - how to enable on the other leg?
In-Reply-To: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
References: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
Message-ID: <CAFiqYu=P077q6HwaOA47k=EYeMUpTdmShFHAPdLgb5sJMg6CGg@mail.gmail.com>

This is completely pointless.

SRTP keys are transmitted plaintext within SIP, you need to use TLS to
encrypt them.

Anyone intercepting your SRTP stream would also intercept your SIP packets,
which means if you're not using TLS they'll have the encryption key to
decrypt the SRTP packets. So it's no more secure than RTP.

-Steve




On 21 November 2012 13:08, Yehavi Bourvine <yehavi.bourvine at gmail.com>wrote:

>   I am trying to use SRTP without TLS
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From vbvbrj at gmail.com  Fri Nov 23 15:29:33 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Fri, 23 Nov 2012 14:29:33 +0200
Subject: [Freeswitch-users] FreeSwitch is blocking when doing "shutdown"
Message-ID: <50AF6C2D.2040105@gmail.com>

Hi.

I have a start-up script in lua.conf.xml for capture and log events:
<configuration name="lua.conf" description="LUA Configuration">
   <settings>
	...
     <param name="startup-script" value="callcenter-log-events-mysql.lua"/>
   </settings>
</configuration>


callcenter-log-events-mysql.lua contains a loop:

while true do
   local dbh = freeswitch.Dbh("freeswitch_data","freeswitch","freeswitch")
   while dbh:connected() do
     freeswitch.consoleLog("debug","Connected to DB.\n")
     while dbh:connected() do
       for e in (function() return conlocal:pop(1,1000) end) do
       freeswitch.consoleLog("debug","Event: " .. e:serialize("xml") .. 
"\n")
       if not dbh:connected() then
         freeswitch.consoleLog("debug",argv[0] .. ": Database 
disconnected\n")
         break
       end
       ccAction = e:getHeader("headername")
     end
   end
   dbh:release()
end


No, the problem is that when doing a shutdown command from FS_CLI, or 
service stop, FS does not terminate. ps xua | grep freeswitch shows the 
running process. I have to kill with -9 signal.

I tracked that this is related to the outermost while in my script.  If 
this script is not started at start-up, FS is not blocking on shutdown.

Is there a parameter in lua.conf.xml to instruct FS to terminate any 
running lua scripts? Or is a way to detect this in the script and break 
outermost while loop?

Thank you.

-- 
Mimiko desu.


From luis.daniel.lucio at gmail.com  Fri Nov 23 15:55:23 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Fri, 23 Nov 2012 07:55:23 -0500
Subject: [Freeswitch-users] global dial plan for *SUPPORT/*7877678
In-Reply-To: <CAMWiMy9k9CnOSz5n9gaKhHUA-XRvpdTfLMjJHdG48GsxkfmrRA@mail.gmail.com>
References: <CAFLo2QzmivvcUA=_+_LBmthzL6-5U48ne=kGNBauvJ4yG3Rt7w@mail.gmail.com>
	<CAMWiMy9k9CnOSz5n9gaKhHUA-XRvpdTfLMjJHdG48GsxkfmrRA@mail.gmail.com>
Message-ID: <CAFLo2QzwaOTSvtZdq+378SRZXvzCn2+dyOaT6ahq4xJk4qLVNw@mail.gmail.com>

Thanks,

My question was more on how to use the "any" context,

i did by adding the dial plan to all tenants, but i wonder to do it
faster as  global dialplan

LD

2012/11/22 curriegrad2004 <curriegrad2004 at gmail.com>:
> Yes, more than possible. You'll just need to get the regex to escape
> the asterisk and have it to transfer/bridge to your extension. That's
> all.
>
> Hint: Look in the default.xml of the default config to see an example
> of how you escape the asterisk
>
> On Thu, Nov 22, 2012 at 7:21 PM, Luis Daniel Lucio Quiroz
> <luis.daniel.lucio at gmail.com> wrote:
>> Helo,
>>
>> Well i have FS with multidomain tenant installed and working.
>>
>> I wonder if there is an easy way to place in default.xml file a dial
>> plan to let any extension, from any tenant, that when he/she dials
>> *7877678 it always ring in extension 001 at mytenant
>>
>> is that possible?
>> LD
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From darcy at Vex.Net  Fri Nov 23 16:56:14 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Fri, 23 Nov 2012 08:56:14 -0500
Subject: [Freeswitch-users] global dial plan for *SUPPORT/*7877678
In-Reply-To: <CAFLo2QzwaOTSvtZdq+378SRZXvzCn2+dyOaT6ahq4xJk4qLVNw@mail.gmail.com>
References: <CAFLo2QzmivvcUA=_+_LBmthzL6-5U48ne=kGNBauvJ4yG3Rt7w@mail.gmail.com>
	<CAMWiMy9k9CnOSz5n9gaKhHUA-XRvpdTfLMjJHdG48GsxkfmrRA@mail.gmail.com>
	<CAFLo2QzwaOTSvtZdq+378SRZXvzCn2+dyOaT6ahq4xJk4qLVNw@mail.gmail.com>
Message-ID: <20121123085614.4ba3f090@dilbert>

On Fri, 23 Nov 2012 07:55:23 -0500
Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com> wrote:
> i did by adding the dial plan to all tenants, but i wonder to do it
> faster as  global dialplan

I must be missing something.  What's wrong with just adding an
extension?

<extension name="support">
  <condition field="destination_number"
          expression="^*(SUPPORT|7877678)$">
    <action application="transfer" data="001 at mytenant"/>
  </condition>
</extension>

Are there users on the switch that are not tenants?  If so then how do
you identify them?  You may need another condition.

Side question - are you already using "611" for something else?

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From a.venugopan at mundio.com  Fri Nov 23 16:56:34 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 23 Nov 2012 13:56:34 +0000
Subject: [Freeswitch-users] mod_voicemail.c file change
In-Reply-To: <CAFiqYu=qtBWUSxsqj=OeHM6k-LDbH-YT4=biHYRouvh85Vk43w@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>
	<7874178664831298079@unknownmsgid>
	<592A9CF93E12394E8472A6CC66E66BF23378D5@Mail-Kilo.squay.com>
	<CAFiqYu=qtBWUSxsqj=OeHM6k-LDbH-YT4=biHYRouvh85Vk43w@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337983@Mail-Kilo.squay.com>

Thanks but sorry I am still not clear on <Name> part. <Name> variable which is being picked up from database in name_path was heard first. How am I to bring it in middle instead?
Even if I change voicemail_play_greeting the first word we hear would be our <Name> still followed by voicemail_play_greeting message.
Whether my understanding is correct?


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 23 November 2012 11:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail.c file change

Leave the C code, you don't need to know or change anything in there.

The voicemail_play_greeting macro is a configuration option.

Look in your configuration at lang/en/vm/sounds.xml, the macro is defined in there as a combination of play-file and say to produce the message that's heard.

Changing that macro will change what's heard.

-Steve



On 23 November 2012 09:08, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Yes currently that plays 'person you are trying to reach is currently not available'. But I require this sentence first 'You have reached the mailbox of' followed by <Name> instead of Name reading at first.

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Dorn DJBinter
Sent: 22 November 2012 18:10
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail.c file change

Look at voicemail_play_greeting macro.

Sent from my iPad

On Nov 22, 2012, at 5:18 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently while dialling voicemail from mod_voicemail.c file it picks up the name recorded in system greetings and says like

'<Name> person you are trying to reach is currently not available.Record you message after the tone press any key or stop talking to end the recording'

I require something like this
'You have reached the mailbox of <Name> Record you message after the tone press any key or stop talking to end the recording'.

I knew I have to change in mod_voicemail.c file. But since am not familiar with C code not sure how to get this.

if (!skip_greeting) {
                memset(buf, 0, sizeof(buf));
                args.input_callback = cancel_on_dtmf;
                args.buf = buf;
                args.buflen = sizeof(buf);

                switch_ivr_sleep(session, 100, SWITCH_TRUE, NULL);

                if (switch_file_exists(greet_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                        memset(buf, 0, sizeof(buf));
                        TRY_CODE(switch_ivr_play_file(session, NULL, greet_path, &args));
                } else {
                        if (switch_file_exists(cbt.name_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_play_file(session, NULL, cbt.name_path, &args));
                        }
                        if (*buf == '\0') {
                                if (!read_id) {
                                        if (!(read_id = switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {
                                                read_id = id;
                                        }
                                }
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO, read_id, NULL, &args));
                        }

Can anyone please help me out. Thanks

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




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

FreeSWITCH-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://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From a.venugopan at mundio.com  Fri Nov 23 17:40:54 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 23 Nov 2012 14:40:54 +0000
Subject: [Freeswitch-users] mod_voicemail compilation error
In-Reply-To: <CAFiqYum73QwhygFLD4fziFEx9xxFCBYjbrM6c6BZYHKJTQ6nog@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337901@Mail-Kilo.squay.com>
	<CAFiqYum73QwhygFLD4fziFEx9xxFCBYjbrM6c6BZYHKJTQ6nog@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF23379B7@Mail-Kilo.squay.com>

Yes for the same. But only in c file the name_path was looked up and I assume that if  I need to move say <Name> after voicemail_play_greeting macro I can do only in C file.
Correct me if am wrong please

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 23 November 2012 12:09
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail compilation error

2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so
**/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol: switch_channel_expand_variables_check**

Module hasn't compiled properly so cannot load.

You could try another 'make mod_voicemail' then the install, or rebuilding the entire source again.

Why are you trying to insert debug messages? Is this related to your greeting prompt query, which I answered in the other thread isn't something that needs any changes to the code, just your configuration files modified?

-Steve



On 23 November 2012 10:03, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so
**/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol: switch_channel_expand_variables_check**

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From a.venugopan at mundio.com  Fri Nov 23 17:52:48 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 23 Nov 2012 14:52:48 +0000
Subject: [Freeswitch-users] mod_voicemail.c file change
References: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>
	<7874178664831298079@unknownmsgid>
	<592A9CF93E12394E8472A6CC66E66BF23378D5@Mail-Kilo.squay.com>
	<CAFiqYu=qtBWUSxsqj=OeHM6k-LDbH-YT4=biHYRouvh85Vk43w@mail.gmail.com> 
Message-ID: <592A9CF93E12394E8472A6CC66E66BF23379C9@Mail-Kilo.squay.com>

In this part of the code

greet:
        if (!skip_greeting) {
                memset(buf, 0, sizeof(buf));
                args.input_callback = cancel_on_dtmf;
                args.buf = buf;
                args.buflen = sizeof(buf);

                switch_ivr_sleep(session, 100, SWITCH_TRUE, NULL);

                if (switch_file_exists(greet_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                        memset(buf, 0, sizeof(buf));
                        TRY_CODE(switch_ivr_play_file(session, NULL, greet_path, &args));
                } else {
                        if (switch_file_exists(cbt.name_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_play_file(session, NULL, cbt.name_path, &args));
                        }
                        if (*buf == '\0') {
                                if (!read_id) {
                                        if (!(read_id = switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {
                                                read_id = id;
                                        }
                                }
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO, read_id, NULL, &args));


The bolded first line plays the name stored in name_path and the 2nd bold line reads the greeting message. So I guess only in this C code we need to make change such a way that name_path should be somewhere in middle. Am I right?


Regards,
Archana

From: Archana Venugopan
Sent: 23 November 2012 13:57
To: 'FreeSWITCH Users Help'
Subject: RE: [Freeswitch-users] mod_voicemail.c file change

Thanks but sorry I am still not clear on <Name> part. <Name> variable which is being picked up from database in name_path was heard first. How am I to bring it in middle instead?
Even if I change voicemail_play_greeting the first word we hear would be our <Name> still followed by voicemail_play_greeting message.
Whether my understanding is correct?


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 23 November 2012 11:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail.c file change

Leave the C code, you don't need to know or change anything in there.

The voicemail_play_greeting macro is a configuration option.

Look in your configuration at lang/en/vm/sounds.xml, the macro is defined in there as a combination of play-file and say to produce the message that's heard.

Changing that macro will change what's heard.

-Steve



On 23 November 2012 09:08, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Yes currently that plays 'person you are trying to reach is currently not available'. But I require this sentence first 'You have reached the mailbox of' followed by <Name> instead of Name reading at first.

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Dorn DJBinter
Sent: 22 November 2012 18:10
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail.c file change

Look at voicemail_play_greeting macro.

Sent from my iPad

On Nov 22, 2012, at 5:18 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently while dialling voicemail from mod_voicemail.c file it picks up the name recorded in system greetings and says like

'<Name> person you are trying to reach is currently not available.Record you message after the tone press any key or stop talking to end the recording'

I require something like this
'You have reached the mailbox of <Name> Record you message after the tone press any key or stop talking to end the recording'.

I knew I have to change in mod_voicemail.c file. But since am not familiar with C code not sure how to get this.

if (!skip_greeting) {
                memset(buf, 0, sizeof(buf));
                args.input_callback = cancel_on_dtmf;
                args.buf = buf;
                args.buflen = sizeof(buf);

                switch_ivr_sleep(session, 100, SWITCH_TRUE, NULL);

                if (switch_file_exists(greet_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                        memset(buf, 0, sizeof(buf));
                        TRY_CODE(switch_ivr_play_file(session, NULL, greet_path, &args));
                } else {
                        if (switch_file_exists(cbt.name_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_play_file(session, NULL, cbt.name_path, &args));
                        }
                        if (*buf == '\0') {
                                if (!read_id) {
                                        if (!(read_id = switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {
                                                read_id = id;
                                        }
                                }
                                memset(buf, 0, sizeof(buf));
                                TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO, read_id, NULL, &args));
                        }

Can anyone please help me out. Thanks

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




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

FreeSWITCH-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://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From yehavi.bourvine at gmail.com  Fri Nov 23 17:58:32 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 23 Nov 2012 16:58:32 +0200
Subject: [Freeswitch-users] SRTP - how to enable on the other leg?
In-Reply-To: <CAFiqYu=P077q6HwaOA47k=EYeMUpTdmShFHAPdLgb5sJMg6CGg@mail.gmail.com>
References: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
	<CAFiqYu=P077q6HwaOA47k=EYeMUpTdmShFHAPdLgb5sJMg6CGg@mail.gmail.com>
Message-ID: <CAAtWKeGZwCm1yQvHKEPdnP3b4CuG=uMZYN3RXw4o3Cr7H8BdcQ@mail.gmail.com>

You are right; TLS is my next step... Using non-TLS helps debugging these
issues easier.

I'll try the suggestions raised above on Sunday.

                      Thanks, __Yehavi:


2012/11/23 Steven Ayre <steveayre at gmail.com>

> This is completely pointless.
>
> SRTP keys are transmitted plaintext within SIP, you need to use TLS to
> encrypt them.
>
> Anyone intercepting your SRTP stream would also intercept your SIP
> packets, which means if you're not using TLS they'll have the encryption
> key to decrypt the SRTP packets. So it's no more secure than RTP.
>
> -Steve
>
>
>
>
> On 21 November 2012 13:08, Yehavi Bourvine <yehavi.bourvine at gmail.com>wrote:
>
>>   I am trying to use SRTP without TLS
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fs-list at communicatefreely.net  Fri Nov 23 18:32:04 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 23 Nov 2012 10:32:04 -0500
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
 again
In-Reply-To: <CAFiqYu=WYMb2aVqMh-POy-V+fbZMq_QOsn_5a2P=eg_EAtEKow@mail.gmail.com>
References: <50AE7D9C.8040604@communicatefreely.net>
	<CAFiqYu=WYMb2aVqMh-POy-V+fbZMq_QOsn_5a2P=eg_EAtEKow@mail.gmail.com>
Message-ID: <50AF96F4.2080005@communicatefreely.net>

Hi Steven,

Thanks for the suggestions.  I'm hoping once I get the upgrade done it will all go away. 
I have watched it happen at least once.  I was on the phone at the time.  Console activity 
more or less stopped, except for a few calls hanging up.  The console remains responsive, 
and my call wasn't dropped for at least a minute or two (media timeout?).  I was able to 
run sofia status and other commands that use the database, so I'm assuming that the 
connection was still working.  All our media is runs through the box, so I think things 
are fine on the Ethernet level.  I do see higher load averages - maybe 3-4, but that's the 
only obvious indication.  It's not taking CPU beyond 10% or so.

We are using MySQL as the core DB and also as the DB backend for each sofia profile.  This 
is connecting through ODBC of course.

If I can get the other kinks worked out, then I will try 1.2 stable in production and 
we'll see how it goes.

-Tim

Steven Ayre wrote:
>     Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
> 
> 
> As you've already acknowledged it's a very old version.
> 
> It's possible that your issue has already been found and fixed, but if 
> it hasn't then the code will have changed significantly since then and 
> you'd really need to reproduce it on the latest code for it to be 
> investigated.
> 
> 
> As some general thoughts though, are you able to spot it happening while 
> it's happening or only afterwards?
> 
> If you're able to get on the system during one of those times look at 
> what else is happening. Is the load average/cpu usage/io high? Perhaps 
> something's running that's blocking all access or causing very high IO.
> 
> What DB backend are you using for Sofia? Is it possible that that's 
> hanging for a moment? For example if you're running a backup on the DB 
> that blocks all writes to the DB while Sofia is trying to update the DB 
> that perhaps would cause this.
> 
> Try running a SIP OPTIONS ping your your sofia profile from the 
> localhost during that time, which should exclude it being any issue on 
> the ethernet.
> 
> -Steve
> 
> 
> 
> 
> On 22 November 2012 19:31, Tim St. Pierre <fs-list at communicatefreely.net 
> <mailto:fs-list at communicatefreely.net>> wrote:
> 
>     Hello,
> 
>     I'm having a bit of an odd problem.
> 
>     Intermittently, often every 2-3 days or so, Freeswitch stops
>     replying to SIP for about 5
>     minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to
>     be pretty close.
> 
>     During this time, no new registrations or invites can happen, but
>     existing calls stay
>     connected for at least a minute or two.  In the logs, you can see
>     calls slowly hanging up
>     with "NORMAL_CLEARING".  In 5 minutes, everything starts up again
>     with no word about it at
>     all in the logs.
> 
>     When calls resume, I notice that the number of sessions returned by
>     the status command is
>     one higher than the actual number sessions returned by show
>     channels, or by looking in the
>     database.  Every time this happens, the discrepancy increases by one.
> 
>     The interruption happens on all SIP profiles, but calls originated
>     from the socket API
>     still work, insofar as they return with PROGRESS_TIMEOUT since the
>     profiles are still
>     running, but stuck.
> 
>     We are using ODBC/MySQL for the core database, and the database
>     server only runs this
>     database and some basic PHP/xml-curl stuff.
> 
>     We have 416 endpoints registered, and usually sit at about 30
>     sessions during the day.
> 
>     This never happens at night, only during busier times, but not
>     necessarily busy hour.
> 
>     I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
> 
>     Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
> 
>     Yes, I know it's old and I'm trying to upgrade, but I'm still having
>     some problems getting
>     all my phones to work properly with 1.2 stable.  This is a
>     production system, so I can't
>     just blindly put out the newest release.  Mostly, I need to buy
>     myself some time so that I
>     can get the kinks worked out of the latest version and then upgrade
>     the production box.
> 
>     I'm grateful for any insights as to what could be happening, even if
>     a solution is just a
>     temporary workaround.
> 
>     Thanks!
> 
>     -Tim
> 
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
> 
>     
>     
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
> 
>     FreeSWITCH-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
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveayre at gmail.com  Fri Nov 23 21:03:32 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 18:03:32 +0000
Subject: [Freeswitch-users] mod_voicemail.c file change
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23379C9@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337778@Mail-Kilo.squay.com>
	<7874178664831298079@unknownmsgid>
	<592A9CF93E12394E8472A6CC66E66BF23378D5@Mail-Kilo.squay.com>
	<CAFiqYu=qtBWUSxsqj=OeHM6k-LDbH-YT4=biHYRouvh85Vk43w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF23379C9@Mail-Kilo.squay.com>
Message-ID: <CAFiqYunoGHp+4AB1WO-oQgRa3b0GVENZa3ZucCqo-NPoq_KwyA@mail.gmail.com>

Ok, I see your issue now... Sorry I'd thought you mean the name was
the ID read out not the name_path.

Try commenting this section out:
if (switch_file_exists(cbt.name_path,
switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
   memset(buf, 0, sizeof(buf));
   TRY_CODE(switch_ivr_play_file(session, NULL, cbt.name_path, &args));
}

Add this above "if (!read_id) {"
   char *data = NULL;

And change
   TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO,
read_id, NULL, &args));
to
   data = switch_core_session_sprintf(session, "%s %s", read_id, cbt.name_path);
   TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO,
data, NULL, &args));


The macro should then have the name_path in $2 and you can use
play-file there to play it when you want to...
I haven't tried this myself but I think it'll work - try it and see.

Doing this by modifying the code is awkward though. For a start it'll
be a pain to maintain, reapplying it every time you upgrade especially
if the upstream code changes.

Not sure why name was implemented as always getting played first.
Fixing it in Git would seem preferable, but it'd be a change of
behaviour so might catch people out if they're upgrading an existing
system.

-Steve



On 23 November 2012 14:52, Archana Venugopan <a.venugopan at mundio.com> wrote:
>
> if (switch_file_exists(cbt.name_path, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
>
>                                 memset(buf, 0, sizeof(buf));
>
>                                 TRY_CODE(switch_ivr_play_file(session, NULL, cbt.name_path, &args));
>
>                         }
>
>                         if (*buf == '\0') {
>
>                                 if (!read_id) {
>
>                                         if (!(read_id = switch_channel_get_variable(channel, "voicemail_alternate_greet_id"))) {
>
>                                                 read_id = id;
>
>                                         }
>
>                                 }
>
>                                 memset(buf, 0, sizeof(buf));
>
>                                 TRY_CODE(switch_ivr_phrase_macro(session, VM_PLAY_GREETING_MACRO, read_id, NULL, &args));


From steveayre at gmail.com  Fri Nov 23 21:13:08 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 18:13:08 +0000
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
	again
In-Reply-To: <50AF96F4.2080005@communicatefreely.net>
References: <50AE7D9C.8040604@communicatefreely.net>
	<CAFiqYu=WYMb2aVqMh-POy-V+fbZMq_QOsn_5a2P=eg_EAtEKow@mail.gmail.com>
	<50AF96F4.2080005@communicatefreely.net>
Message-ID: <CAFiqYumcCNB9zmv94er5KgSuG0VrWViqjy7phw1b=-8qNobCUw@mail.gmail.com>

Any kind of DB backup running? Or any long-running queries (innotop is
great for highlighting queries that've been running a while, including
on non-innodb tables).

A global read lock, or queries waiting for a lock could block a db
update from the sofia profile thread but still allow read-only queries
(sofia status) to run.

-Steve



On 23 November 2012 15:32, Tim St. Pierre <fs-list at communicatefreely.net> wrote:
> Hi Steven,
>
> Thanks for the suggestions.  I'm hoping once I get the upgrade done it will all go away.
> I have watched it happen at least once.  I was on the phone at the time.  Console activity
> more or less stopped, except for a few calls hanging up.  The console remains responsive,
> and my call wasn't dropped for at least a minute or two (media timeout?).  I was able to
> run sofia status and other commands that use the database, so I'm assuming that the
> connection was still working.  All our media is runs through the box, so I think things
> are fine on the Ethernet level.  I do see higher load averages - maybe 3-4, but that's the
> only obvious indication.  It's not taking CPU beyond 10% or so.
>
> We are using MySQL as the core DB and also as the DB backend for each sofia profile.  This
> is connecting through ODBC of course.
>
> If I can get the other kinks worked out, then I will try 1.2 stable in production and
> we'll see how it goes.
>
> -Tim
>
> Steven Ayre wrote:
>>     Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>
>>
>> As you've already acknowledged it's a very old version.
>>
>> It's possible that your issue has already been found and fixed, but if
>> it hasn't then the code will have changed significantly since then and
>> you'd really need to reproduce it on the latest code for it to be
>> investigated.
>>
>>
>> As some general thoughts though, are you able to spot it happening while
>> it's happening or only afterwards?
>>
>> If you're able to get on the system during one of those times look at
>> what else is happening. Is the load average/cpu usage/io high? Perhaps
>> something's running that's blocking all access or causing very high IO.
>>
>> What DB backend are you using for Sofia? Is it possible that that's
>> hanging for a moment? For example if you're running a backup on the DB
>> that blocks all writes to the DB while Sofia is trying to update the DB
>> that perhaps would cause this.
>>
>> Try running a SIP OPTIONS ping your your sofia profile from the
>> localhost during that time, which should exclude it being any issue on
>> the ethernet.
>>
>> -Steve
>>
>>
>>
>>
>> On 22 November 2012 19:31, Tim St. Pierre <fs-list at communicatefreely.net
>> <mailto:fs-list at communicatefreely.net>> wrote:
>>
>>     Hello,
>>
>>     I'm having a bit of an odd problem.
>>
>>     Intermittently, often every 2-3 days or so, Freeswitch stops
>>     replying to SIP for about 5
>>     minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to
>>     be pretty close.
>>
>>     During this time, no new registrations or invites can happen, but
>>     existing calls stay
>>     connected for at least a minute or two.  In the logs, you can see
>>     calls slowly hanging up
>>     with "NORMAL_CLEARING".  In 5 minutes, everything starts up again
>>     with no word about it at
>>     all in the logs.
>>
>>     When calls resume, I notice that the number of sessions returned by
>>     the status command is
>>     one higher than the actual number sessions returned by show
>>     channels, or by looking in the
>>     database.  Every time this happens, the discrepancy increases by one.
>>
>>     The interruption happens on all SIP profiles, but calls originated
>>     from the socket API
>>     still work, insofar as they return with PROGRESS_TIMEOUT since the
>>     profiles are still
>>     running, but stuck.
>>
>>     We are using ODBC/MySQL for the core database, and the database
>>     server only runs this
>>     database and some basic PHP/xml-curl stuff.
>>
>>     We have 416 endpoints registered, and usually sit at about 30
>>     sessions during the day.
>>
>>     This never happens at night, only during busier times, but not
>>     necessarily busy hour.
>>
>>     I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
>>
>>     Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>
>>     Yes, I know it's old and I'm trying to upgrade, but I'm still having
>>     some problems getting
>>     all my phones to work properly with 1.2 stable.  This is a
>>     production system, so I can't
>>     just blindly put out the newest release.  Mostly, I need to buy
>>     myself some time so that I
>>     can get the kinks worked out of the latest version and then upgrade
>>     the production box.
>>
>>     I'm grateful for any insights as to what could be happening, even if
>>     a solution is just a
>>     temporary workaround.
>>
>>     Thanks!
>>
>>     -Tim
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-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
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Fri Nov 23 21:16:36 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 18:16:36 +0000
Subject: [Freeswitch-users] FreeSwitch is blocking when doing "shutdown"
In-Reply-To: <50AF6C2D.2040105@gmail.com>
References: <50AF6C2D.2040105@gmail.com>
Message-ID: <CAFiqYu=aoN6s9dkf5znnpmA_+i5DfO7MXUsiZYTHxfaDJbJ_bg@mail.gmail.com>

In a call it'd be generally be advisable to use 'while session:ready'
instead of 'while true'

Guess you need something equivalent in a startup script.

I don't think you'll have a session, but just in case try
session:ready anyway to see if that works.

-Steve




On 23 November 2012 12:29, Mimiko <vbvbrj at gmail.com> wrote:
> Hi.
>
> I have a start-up script in lua.conf.xml for capture and log events:
> <configuration name="lua.conf" description="LUA Configuration">
>    <settings>
>         ...
>      <param name="startup-script" value="callcenter-log-events-mysql.lua"/>
>    </settings>
> </configuration>
>
>
> callcenter-log-events-mysql.lua contains a loop:
>
> while true do
>    local dbh = freeswitch.Dbh("freeswitch_data","freeswitch","freeswitch")
>    while dbh:connected() do
>      freeswitch.consoleLog("debug","Connected to DB.\n")
>      while dbh:connected() do
>        for e in (function() return conlocal:pop(1,1000) end) do
>        freeswitch.consoleLog("debug","Event: " .. e:serialize("xml") ..
> "\n")
>        if not dbh:connected() then
>          freeswitch.consoleLog("debug",argv[0] .. ": Database
> disconnected\n")
>          break
>        end
>        ccAction = e:getHeader("headername")
>      end
>    end
>    dbh:release()
> end
>
>
> No, the problem is that when doing a shutdown command from FS_CLI, or
> service stop, FS does not terminate. ps xua | grep freeswitch shows the
> running process. I have to kill with -9 signal.
>
> I tracked that this is related to the outermost while in my script.  If
> this script is not started at start-up, FS is not blocking on shutdown.
>
> Is there a parameter in lua.conf.xml to instruct FS to terminate any
> running lua scripts? Or is a way to detect this in the script and break
> outermost while loop?
>
> Thank you.
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Fri Nov 23 21:30:12 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 18:30:12 +0000
Subject: [Freeswitch-users] FreeSwitch is blocking when doing "shutdown"
In-Reply-To: <CAFiqYu=aoN6s9dkf5znnpmA_+i5DfO7MXUsiZYTHxfaDJbJ_bg@mail.gmail.com>
References: <50AF6C2D.2040105@gmail.com>
	<CAFiqYu=aoN6s9dkf5znnpmA_+i5DfO7MXUsiZYTHxfaDJbJ_bg@mail.gmail.com>
Message-ID: <CAFiqYu=BsV9_OKBgsczCL6bUXHo8bseCk===_hLP7nH2iQV8vg@mail.gmail.com>

contrib has a few startup script examples
git clone git://git.freeswitch.org/freeswitch-contrib.git
seven/lua/gateway_report.lua
trixter/scheduled_event.lua

Looks like they both wrap the outermost layer in:
for e in (function() return con:pop(1) end) do

I guess this is what you'll need to do.

-Steve




On 23 November 2012 18:16, Steven Ayre <steveayre at gmail.com> wrote:
> In a call it'd be generally be advisable to use 'while session:ready'
> instead of 'while true'
>
> Guess you need something equivalent in a startup script.
>
> I don't think you'll have a session, but just in case try
> session:ready anyway to see if that works.
>
> -Steve
>
>
>
>
> On 23 November 2012 12:29, Mimiko <vbvbrj at gmail.com> wrote:
>> Hi.
>>
>> I have a start-up script in lua.conf.xml for capture and log events:
>> <configuration name="lua.conf" description="LUA Configuration">
>>    <settings>
>>         ...
>>      <param name="startup-script" value="callcenter-log-events-mysql.lua"/>
>>    </settings>
>> </configuration>
>>
>>
>> callcenter-log-events-mysql.lua contains a loop:
>>
>> while true do
>>    local dbh = freeswitch.Dbh("freeswitch_data","freeswitch","freeswitch")
>>    while dbh:connected() do
>>      freeswitch.consoleLog("debug","Connected to DB.\n")
>>      while dbh:connected() do
>>        for e in (function() return conlocal:pop(1,1000) end) do
>>        freeswitch.consoleLog("debug","Event: " .. e:serialize("xml") ..
>> "\n")
>>        if not dbh:connected() then
>>          freeswitch.consoleLog("debug",argv[0] .. ": Database
>> disconnected\n")
>>          break
>>        end
>>        ccAction = e:getHeader("headername")
>>      end
>>    end
>>    dbh:release()
>> end
>>
>>
>> No, the problem is that when doing a shutdown command from FS_CLI, or
>> service stop, FS does not terminate. ps xua | grep freeswitch shows the
>> running process. I have to kill with -9 signal.
>>
>> I tracked that this is related to the outermost while in my script.  If
>> this script is not started at start-up, FS is not blocking on shutdown.
>>
>> Is there a parameter in lua.conf.xml to instruct FS to terminate any
>> running lua scripts? Or is a way to detect this in the script and break
>> outermost while loop?
>>
>> Thank you.
>>
>> --
>> Mimiko desu.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From chetan.khatri at panamaxil.com  Fri Nov 23 21:31:57 2012
From: chetan.khatri at panamaxil.com (Chetan Khatri)
Date: Sat, 24 Nov 2012 00:01:57 +0530
Subject: [Freeswitch-users] Mod_rad_auth Capacity Issue
In-Reply-To: <CALFFxt0KBH869A0m0y=OaGpwi-JTR_jVNbsiNmykJ4cO-XukeQ@mail.gmail.com>
References: <50ACE9C4.60803@panamaxil.com>
	<50AD3FD5.4070606@quentustech.com>	<CALFFxt3__=MPVF5a1o6eT+XXdHBDXQMt-EvHFYw+EMrqAwDrdw@mail.gmail.com>	<50AE306E.7040909@panamaxil.com>	<CALFFxt1omcKDN7-n8-VbYE0mYuHBz1v8HW5RhbZO0N6Jdm=7Hw@mail.gmail.com>	<50AE7650.7010306@panamaxil.com>
	<CALFFxt0KBH869A0m0y=OaGpwi-JTR_jVNbsiNmykJ4cO-XukeQ@mail.gmail.com>
Message-ID: <50AFC11D.2090406@panamaxil.com>

Hello Ahmed,

I have deployed as your steps provided but am getting error while calling.

2012-11-23 23:58:20.801082 [ERR] mod_xml_radius.c:452 mod_xml_radius: 
failed to add option with val 'src-gw-ip=192.168.14.199' to handle
2012-11-23 23:58:20.801082 [ERR] mod_xml_radius.c:881 Failed to add 
params to rc_handle

Can you help me with this.

/Regards,
Chetan Khatri/

On 11/23/2012 01:33 AM, Ahmed Sboor wrote:
> off course i was talking for the same you found
> its /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_radius Normally.
>
>
> On Fri, Nov 23, 2012 at 12:00 AM, Chetan Khatri 
> <chetan.khatri at panamaxil.com <mailto:chetan.khatri at panamaxil.com>> wrote:
>
>     Dear Ahmed,
>
>     First of all thanks a lot , for your assistance i really
>     appreciate it.
>
>     I have some queries if you can please resolve it.
>
>     -    Is mod_xml_radius the same you are talking about. I found
>     that in source  of freeswitch at location src/mod/xml_int.
>     -    I am unable to find any other module like xml-mod-radius. if
>     the above one is not the same can you please provide from what
>     site location i can download the source and compile it.
>
>
>     /Regards,
>     Chetan Khatri/
>
>     On 11/22/2012 10:29 PM, Ahmed Sboor wrote:
>>     Dear Chetan
>>     i will be Glad to assist you.
>>     there are series of steps you have to do .
>>
>>     1. first compile the xml-mod-radius , it will download
>>     Free-radius client library itself which is buggy so you have a
>>     patch inside the xml-mod-radius src folder. patch the free-radius
>>     client library and recompile the Module.
>>
>>     2. Use 1.2.stable but download xml-mod-radius module from master
>>     branch.
>>     3. there is a xml-conf file inside the source which will help to
>>     deploy either you want AAA for Class 5 or Class 4 traffic. We are
>>     doing on IP invites .
>>     4. sample configs are working fine with Jerasoft Radius server
>>     but i believe it must work with any other radius server.
>>     5. there is also a dialplan sample file to show how to force
>>     incoming call to be passed via Radius authentication.
>>     6. If you need any further help i am always there , But i am not
>>     expert just a "happy user" of this module . Mr. William is the
>>     person who wrote this module so he can definitely help you in
>>     details.
>>     7. About Call Volume , we have tested it even on 5000+ calls ,
>>     works Perfect.
>>
>>     With warm regards
>>     Ahmed Sboor
>>     Netelsat Fze
>>
>>
>>
>>
>>
>>     On Thu, Nov 22, 2012 at 7:02 PM, Chetan Khatri
>>     <chetan.khatri at panamaxil.com
>>     <mailto:chetan.khatri at panamaxil.com>> wrote:
>>
>>         Hello Ahmed,
>>
>>         Could please provide some guidance to configure
>>         xml_mod_radius i have deployed it.
>>
>>         /Regards,
>>         Chetan Khatri/
>>
>>         On 11/22/2012 03:43 AM, Ahmed Sboor wrote:
>>>         yes we are using xml_mod_radius easily for more then 2000 calls.
>>>         regards
>>>         Ahmed
>>>
>>>
>>>
>>>         On Thu, Nov 22, 2012 at 12:55 AM, William King
>>>         <william.king at quentustech.com
>>>         <mailto:william.king at quentustech.com>> wrote:
>>>
>>>             Try using xml_mod_radius instead.
>>>
>>>             William King
>>>             Senior Engineer
>>>             Quentus Technologies, INC
>>>             1037 NE 65th St Suite 273
>>>             Seattle, WA 98115
>>>             Main:   (877) 211-9337
>>>             Office: (206) 388-4772
>>>             Cell:   (253) 686-5518
>>>             william.king at quentustech.com
>>>             <mailto:william.king at quentustech.com>
>>>
>>>             On 11/21/2012 06:48 AM, Chetan Khatri wrote:
>>>             > Dear All Experts,
>>>             >
>>>             > Need help to solve freeswitch's mod_rad_auth capacity
>>>             issue.
>>>             > I have deployed freeswtich 1.0.6 and enabled
>>>             mod_rad_auth module. The
>>>             > Module is working fine till the concurrent request
>>>             reaches till 245
>>>             > after that Accounting is failing.
>>>             > Below is the Configuration.
>>>             >
>>>             > FreeSwitch : 1.0.6
>>>             > RADIUS Server : FreeRADIUS2
>>>             > Database server : MySQL 5.5
>>>             >
>>>             > We are stress testing the with freeswitch to handle
>>>             about 2000
>>>             > concurrent calls with RADIUS AAA. without using RADIUS
>>>             modules
>>>             > freeswitch is able to handle 2000 concurrent calls
>>>             without any issues.
>>>             > For analysis we analysed from RADIUS server replies
>>>             access-accept
>>>             > request, but in mod_rad_auth error says *"freeswitch :
>>>             Invalid Digest
>>>             > Request from RADIUS server"* only if the concurrent
>>>             session receives
>>>             > 245, till that all the requests are been working fine.
>>>             >
>>>             > Please guide me how to resolve this capacity issue, I
>>>             can provide all
>>>             > the logs and configuration files if required.
>>>             >
>>>             > --
>>>             > /Regards,
>>>             > Chetan Khatri/
>>>             >
>>>             >
>>>             >
>>>             _________________________________________________________________________
>>>             > Professional FreeSWITCH Consulting Services:
>>>             > consulting at freeswitch.org
>>>             <mailto:consulting at freeswitch.org>
>>>             > http://www.freeswitchsolutions.com
>>>             >
>>>             > FreeSWITCH-powered IP PBX: The CudaTel Communication
>>>             Server
>>>             > 
>>>             >
>>>             > Official FreeSWITCH Sites
>>>             > http://www.freeswitch.org
>>>             > http://wiki.freeswitch.org
>>>             > http://www.cluecon.com
>>>             >
>>>             > FreeSWITCH-users mailing list
>>>             > FreeSWITCH-users at lists.freeswitch.org
>>>             <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
>>>
>>>             _________________________________________________________________________
>>>             Professional FreeSWITCH Consulting Services:
>>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>             http://www.freeswitchsolutions.com
>>>
>>>             
>>>             
>>>
>>>             Official FreeSWITCH Sites
>>>             http://www.freeswitch.org
>>>             http://wiki.freeswitch.org
>>>             http://www.cluecon.com
>>>
>>>             FreeSWITCH-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
>>>
>>>
>>>
>>>         _________________________________________________________________________
>>>         Professional FreeSWITCH Consulting Services:
>>>         consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>>         http://www.freeswitchsolutions.com
>>>
>>>         
>>>         
>>>
>>>         Official FreeSWITCH Sites
>>>         http://www.freeswitch.org
>>>         http://wiki.freeswitch.org
>>>         http://www.cluecon.com
>>>
>>>         FreeSWITCH-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
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-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
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-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
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vbvbrj at gmail.com  Fri Nov 23 23:20:15 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Fri, 23 Nov 2012 22:20:15 +0200
Subject: [Freeswitch-users] FreeSwitch is blocking when doing "shutdown"
In-Reply-To: <CAFiqYu=aoN6s9dkf5znnpmA_+i5DfO7MXUsiZYTHxfaDJbJ_bg@mail.gmail.com>
References: <50AF6C2D.2040105@gmail.com>
	<CAFiqYu=aoN6s9dkf5znnpmA_+i5DfO7MXUsiZYTHxfaDJbJ_bg@mail.gmail.com>
Message-ID: <50AFDA7F.20700@gmail.com>

On 23.11.2012 20:16, Steven Ayre wrote:
> In a call it'd be generally be advisable to use 'while session:ready'
> instead of 'while true'

Nope this does not work:
"attempt to index global 'session' (a nil value)"

 > Looks like they both wrap the outermost layer in:
 > for e in (function() return con:pop(1) end) do

Yes I know about this method, but this does not track DB going down. Or 
isn't using freeswitch.Dbh() connection initiation too slow? If a lot of 
events will come, then connection and disconnect from server will be at 
high rates. Why not use persistent connection?

-- 
Mimiko desu.


From steveayre at gmail.com  Sat Nov 24 02:04:56 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Nov 2012 23:04:56 +0000
Subject: [Freeswitch-users] FreeSwitch is blocking when doing "shutdown"
In-Reply-To: <50AFDA7F.20700@gmail.com>
References: <50AF6C2D.2040105@gmail.com>
	<CAFiqYu=aoN6s9dkf5znnpmA_+i5DfO7MXUsiZYTHxfaDJbJ_bg@mail.gmail.com>
	<50AFDA7F.20700@gmail.com>
Message-ID: <C8C5A2DD-84D9-4BF8-9D40-C77CBA40F0AE@gmail.com>

freeswitch.Dbh uses the internal caching, so it does reuse persistent connections. First connect'll be a full connect but after that it'll be instant.

I believe it checks the handle is still connected before returning it.

Steve on iPhone



On 23 Nov 2012, at 20:20, Mimiko <vbvbrj at gmail.com> wrote:

> On 23.11.2012 20:16, Steven Ayre wrote:
>> In a call it'd be generally be advisable to use 'while session:ready'
>> instead of 'while true'
> 
> Nope this does not work:
> "attempt to index global 'session' (a nil value)"
> 
>> Looks like they both wrap the outermost layer in:
>> for e in (function() return con:pop(1) end) do
> 
> Yes I know about this method, but this does not track DB going down. Or 
> isn't using freeswitch.Dbh() connection initiation too slow? If a lot of 
> events will come, then connection and disconnect from server will be at 
> high rates. Why not use persistent connection?
> 
> -- 
> Mimiko desu.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From luis.daniel.lucio at gmail.com  Sat Nov 24 06:22:02 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Fri, 23 Nov 2012 22:22:02 -0500
Subject: [Freeswitch-users] How to specify in compilling time a specific
	/storage location
Message-ID: <CAFLo2QzUwj-hJa1asM5pP=4Ty36Yp4Gsjk08OyBKE0W2PPWzBQ@mail.gmail.com>

Is it a way to specify in the ./configure where to create storage
directory?  By default it is creating in /usr/storage but i need in
other place.

Im polishing a SRPM,

LD


From krice at freeswitch.org  Sat Nov 24 06:41:34 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 23 Nov 2012 21:41:34 -0600
Subject: [Freeswitch-users] How to specify in compilling time a specific
 /storage location
In-Reply-To: <CAFLo2QzUwj-hJa1asM5pP=4Ty36Yp4Gsjk08OyBKE0W2PPWzBQ@mail.gmail.com>
Message-ID: <CCD59E0E.6109A%krice@freeswitch.org>

Theres already a spec file in tree.. Why not submit the patches so theres no
27 different competing RPM sets?


On 11/23/12 9:22 PM, "Luis Daniel Lucio Quiroz"
<luis.daniel.lucio at gmail.com> wrote:

> Is it a way to specify in the ./configure where to create storage
> directory?  By default it is creating in /usr/storage but i need in
> other place.
> 
> Im polishing a SRPM,
> 
> LD
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From chaiyawut.so at gmail.com  Sat Nov 24 06:53:03 2012
From: chaiyawut.so at gmail.com (chaiyawut.so)
Date: Fri, 23 Nov 2012 19:53:03 -0800 (PST)
Subject: [Freeswitch-users] Can not use Python ESL with Django and
	apache web server
In-Reply-To: <CAMbW9oJpkmfrsZAh8QOLaRydZibP+YjhHEJYJrnfDS9SPbqp9Q@mail.gmail.com>
References: <1353487101708-7584800.post@n2.nabble.com>
	<CAMbW9oJpkmfrsZAh8QOLaRydZibP+YjhHEJYJrnfDS9SPbqp9Q@mail.gmail.com>
Message-ID: <1353729183075-7584885.post@n2.nabble.com>

That solved the problem, thank you 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-not-use-Python-ESL-with-Django-and-apache-web-server-tp7584800p7584885.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kheimerl at cs.berkeley.edu  Sat Nov 24 15:26:08 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sat, 24 Nov 2012 04:26:08 -0800
Subject: [Freeswitch-users] Calls not ending
Message-ID: <CACyT-3mkxEkwC+jWZt4xpMnz=cceCJfezMJ8uCCkKZptCvWUxA@mail.gmail.com>

Hello users,

I'm writing a fairly simple python IVR script. It's available here:
https://github.com/kheimerl/papua-vbts-apps/blob/master/village_idol/scripts/village_idol_record.py

My dialplan is also very simple:
<!-- village idol -->
<extension name="bok_idol">
  <condition field="${destination_number}" expression="^111$">
    <action application="answer"/>
    <action application="python" data="village_idol_record"/>
    <action application="hangup"/>
  </condition>
</extension>


However, every time I call the service, the calls seem to stick around in
FS.

Here's a call record for a call that ended minutes ago:
freeswitch at internal> show calls
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch
e0bc5d12-362e-11e2-8ee4-d16392532c11,inbound,2012-11-24
21:02:54,1353758574,sofia/openbts/IMSI510555550000000 at 192.168.1.200<javascript:_e({},
'cvml', 'IMSI510555550000000 at 192.168.1.200');>
,CS_EXECUTE,IMSI510555550000000,IMSI510555550000000,192.168.1.200,111,
IMSI510555550000000 at 192.168.1.200 <javascript:_e({}, 'cvml',
'IMSI510555550000000 at 192.168.1.200');>
,,ACTIVE,,,,,UCBTelco,,,,,,,,,,,,,,,,,,,,,

1 total.

In the logs, I see my output ("Exiting...") meaning that the python script
ended. I see the call hung up, and the state changes to CS_DESTROY (
http://pastebin.freeswitch.org/20256) but the calls remain in the table...

Anyone know what I'm doing wrong here?

Thanks!
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From kheimerl at cs.berkeley.edu  Sat Nov 24 15:29:12 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sat, 24 Nov 2012 04:29:12 -0800
Subject: [Freeswitch-users] Calls not ending
In-Reply-To: <CACyT-3mkxEkwC+jWZt4xpMnz=cceCJfezMJ8uCCkKZptCvWUxA@mail.gmail.com>
References: <CACyT-3mkxEkwC+jWZt4xpMnz=cceCJfezMJ8uCCkKZptCvWUxA@mail.gmail.com>
Message-ID: <CACyT-3=8qOG5bi8XW9KA0B+b-vUyVWxghcWoaZNWkn2fZFxNSw@mail.gmail.com>

Actually, I think I just fixed it by adding an "unsetInputCallback()" call
to the end of the script. That should probably go to Jira, huh?

On Saturday, November 24, 2012, Kurtis Heimerl wrote:

> Hello users,
>
> I'm writing a fairly simple python IVR script. It's available here:
> https://github.com/kheimerl/papua-vbts-apps/blob/master/village_idol/scripts/village_idol_record.py
>
> My dialplan is also very simple:
> <!-- village idol -->
> <extension name="bok_idol">
>   <condition field="${destination_number}" expression="^111$">
>     <action application="answer"/>
>     <action application="python" data="village_idol_record"/>
>     <action application="hangup"/>
>    </condition>
> </extension>
>
>
> However, every time I call the service, the calls seem to stick around in
> FS.
>
> Here's a call record for a call that ended minutes ago:
> freeswitch at internal> show calls
>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch
> e0bc5d12-362e-11e2-8ee4-d16392532c11,inbound,2012-11-24
> 21:02:54,1353758574,sofia/openbts/IMSI510555550000000 at 192.168.1.200
> ,CS_EXECUTE,IMSI510555550000000,IMSI510555550000000,192.168.1.200,111,
> IMSI510555550000000 at 192.168.1.200
> ,,ACTIVE,,,,,UCBTelco,,,,,,,,,,,,,,,,,,,,,
>
> 1 total.
>
> In the logs, I see my output ("Exiting...") meaning that the python script
> ended. I see the call hung up, and the state changes to CS_DESTROY (
> http://pastebin.freeswitch.org/20256) but the calls remain in the table...
>
> Anyone know what I'm doing wrong here?
>
> Thanks!
>
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From talk2ram at gmail.com  Sat Nov 24 19:43:17 2012
From: talk2ram at gmail.com (ram)
Date: Sat, 24 Nov 2012 22:13:17 +0530
Subject: [Freeswitch-users] build freeswitch with lastest version problem
In-Reply-To: <CAMBhhBLmLxvX__zpZXZNUi2zXBhdo8gqa+2WFiwzrRYEYDgyHQ@mail.gmail.com>
References: <CAMBhhBLmLxvX__zpZXZNUi2zXBhdo8gqa+2WFiwzrRYEYDgyHQ@mail.gmail.com>
Message-ID: <CA+XvL-O3E9C_q8qr_Gcs75XfAfw=Ch6NexxQDQOXLQ9SwDgEkw@mail.gmail.com>

Hi

check the following URL

http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS

install required pre-requisites


On Thu, Nov 22, 2012 at 9:09 PM, haloha <haloha201 at gmail.com> wrote:

> hi
> i am trying to install freeswitch with lastest version from git
> my OS : centos version 6.3
>
> and i get error when do command : ./configure
> ./configure: line 28276: unexpected EOF while looking for matching `"'
> ./configure: line 28281: syntax error: unexpected end of file
> configure: error: ./configure.gnu failed for libs/libsndfile
>
> how to fix it
>
> thank you
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From luis.daniel.lucio at gmail.com  Sat Nov 24 20:03:49 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 24 Nov 2012 12:03:49 -0500
Subject: [Freeswitch-users] How to specify in compilling time a specific
 /storage location
In-Reply-To: <CCD59E0E.6109A%krice@freeswitch.org>
References: <CAFLo2QzUwj-hJa1asM5pP=4Ty36Yp4Gsjk08OyBKE0W2PPWzBQ@mail.gmail.com>
	<CCD59E0E.6109A%krice@freeswitch.org>
Message-ID: <CAFLo2QyBk1d+R=NrLZH8wwjnM2zdmRXX7e_vpG1tsVhU0VjThw@mail.gmail.com>

I took taht SPEC as a template and it doesnt have a reference for storage dir

Mageia has policies on all SPEC must fit perfectly to it, i will keep
SRPM in mageia repo uptodate. dont worry about that.

2012/11/23 Ken Rice <krice at freeswitch.org>:
> Theres already a spec file in tree.. Why not submit the patches so theres no
> 27 different competing RPM sets?
>
>
> On 11/23/12 9:22 PM, "Luis Daniel Lucio Quiroz"
> <luis.daniel.lucio at gmail.com> wrote:
>
>> Is it a way to specify in the ./configure where to create storage
>> directory?  By default it is creating in /usr/storage but i need in
>> other place.
>>
>> Im polishing a SRPM,
>>
>> LD
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From chaiyawut.so at mail.kmutt.ac.th  Sat Nov 24 18:41:05 2012
From: chaiyawut.so at mail.kmutt.ac.th (Chaiyawut Sookplang)
Date: Sat, 24 Nov 2012 22:41:05 +0700
Subject: [Freeswitch-users] Had a problem with X100P 1 port FXO card and
	freeswitch
Message-ID: <CAFUAJ8mnBtvQJ9tX+32yjrmWuW914yrJ252wKqteVLLpsL_NCw@mail.gmail.com>

Hi. I currently using Ubuntu 12.04 server.  I use FreeTDM with freeswitch
and I already  installed Dahdi driver version 2.6 for X100P card. The
driver is loaded and FreeTDM is able to start but I cannot make or receive
 any calls. I condition is like no response from card, freeswitch not
detect any incoming calls and no sign of POTS signalling or anything.

This is a result from "ftdm dump"
freeswitch at internal> ftdm dump 1 1
span_id: 1
chan_id: 1
physical_span_id: 1
physical_chan_id: 1
physical_status: ok
physical_status_red: 0
physical_status_yellow: 0
physical_status_rai: 0
physical_status_blue: 0
physical_status_ais: 0
physical_status_general: 0
signaling_status: UP
type: FXO
state: DOWN
last_state: DOWN
txgain: 0.00
rxgain: 0.00
cid_date:
cid_name:
cid_num:
ani:
aniII:
dnis:
rdnis:
cause: NONE
session: (none)
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From krice at freeswitch.org  Sat Nov 24 20:20:39 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 24 Nov 2012 11:20:39 -0600
Subject: [Freeswitch-users] FreeSWITCH 1.2.5.1 Released
Message-ID: <CCD65E07.610BA%krice@freeswitch.org>

Hey Guys,

Tony found a few problems with the 1.2.5 release that could cause some
performance issues down in the core DB code, so we decided to go ahead and
do a micro point release to address this problem...

So this morning I have released FreeSWITCH 1.2.5.1,
Source Tarball is at http://files.freeswitch.org/freeswitch-1.2.5.1.tar.bz2
as usual 
RPMs are in the FS Yum repo...

And as always open a jira if you find a problem...

Have a great weekend guys! And Join us next Wed for the Weekly FS conference
Call.

K



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

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From gabe at gundy.org  Sat Nov 24 23:14:52 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 24 Nov 2012 13:14:52 -0700
Subject: [Freeswitch-users] FreeSWITCH 1.2.5.1 Released
In-Reply-To: <CCD65E07.610BA%krice@freeswitch.org>
References: <CCD65E07.610BA%krice@freeswitch.org>
Message-ID: <CAMwZ-tL_FpRfETGh9_iiBOmZAnAsGxRQpfsG6X6VJaY36rcmVw@mail.gmail.com>

On Sat, Nov 24, 2012 at 10:20 AM, Ken Rice <krice at freeswitch.org> wrote:
> Tony found a few problems with the 1.2.5 release that could cause some
> performance issues down in the core DB code, so we decided to go ahead and
> do a micro point release to address this problem.

Thanks for hard work of keeping a stable version release updated. I
really enjoy using the latest and greatest in git, but sometimes you
need to go with tied-and-true. I love the 1.2.x stuffs! Keep it up!

Thanks,
Gabe (and everyone else who loves uptime)


From krice at freeswitch.org  Sun Nov 25 01:41:44 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 24 Nov 2012 16:41:44 -0600
Subject: [Freeswitch-users] FreeSWITCH 1.2.5.1 Released
In-Reply-To: <CAMwZ-tL_FpRfETGh9_iiBOmZAnAsGxRQpfsG6X6VJaY36rcmVw@mail.gmail.com>
Message-ID: <CCD6A948.610CC%krice@freeswitch.org>

Hey Gabe,

Thanks! But keep in mind, I cant do it all alone, there's whole pile of
people behind me making this possible...

Don't forget Anthony and his team of coders making FreeSWITCH a reality, and
don't forget the rest of users like you that help out tremendously by
actually filing good bug reports, with good detailed info and logs so we can
actually fix the problems.

K





On 11/24/12 2:14 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:

> On Sat, Nov 24, 2012 at 10:20 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Tony found a few problems with the 1.2.5 release that could cause some
>> performance issues down in the core DB code, so we decided to go ahead and
>> do a micro point release to address this problem.
> 
> Thanks for hard work of keeping a stable version release updated. I
> really enjoy using the latest and greatest in git, but sometimes you
> need to go with tied-and-true. I love the 1.2.x stuffs! Keep it up!
> 
> Thanks,
> Gabe (and everyone else who loves uptime)
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From jmesquita at freeswitch.org  Sun Nov 25 01:48:23 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Sat, 24 Nov 2012 19:48:23 -0300
Subject: [Freeswitch-users] FreeSWITCH 1.2.5.1 Released
In-Reply-To: <CAMwZ-tL_FpRfETGh9_iiBOmZAnAsGxRQpfsG6X6VJaY36rcmVw@mail.gmail.com>
References: <CCD65E07.610BA%krice@freeswitch.org>
	<CAMwZ-tL_FpRfETGh9_iiBOmZAnAsGxRQpfsG6X6VJaY36rcmVw@mail.gmail.com>
Message-ID: <ACB5C82F-5FE0-4584-B971-945F0AADE1D3@freeswitch.org>

+1

Thanks for the hard work guys. 

Sent from my iPhone

On Nov 24, 2012, at 5:14 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sat, Nov 24, 2012 at 10:20 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Tony found a few problems with the 1.2.5 release that could cause some
>> performance issues down in the core DB code, so we decided to go ahead and
>> do a micro point release to address this problem.
> 
> Thanks for hard work of keeping a stable version release updated. I
> really enjoy using the latest and greatest in git, but sometimes you
> need to go with tied-and-true. I love the 1.2.x stuffs! Keep it up!
> 
> Thanks,
> Gabe (and everyone else who loves uptime)
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From sdevoy at bizfocused.com  Sun Nov 25 08:21:06 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sun, 25 Nov 2012 00:21:06 -0500
Subject: [Freeswitch-users] Tricky rollover Dial Plan Question
Message-ID: <0c3d01cdcacc$abedcd00$03c96700$@bizfocused.com>

Hi All,

 

Thanks for your help as always.

 

I have a CISCO 504G four line phone.  I don't think the model is
particularly relevant, just that it is a multi-line phone.

 

Issue one:

If line 1 is busy, I want new calls to be directed to line 2. If 1 and 2 are
in use, ring on 3 and then line 4.  I think this means call waiting will
need to be disabled so that each will ring busy, enabling rollover.  Also, I
don't want lines 2, 3 and 4 to all ring if 1 is busy.

 

I think I just use continue_on_fail=true and hangup_after_bridge=true, then
route to Line 1, then 2, 3 and 4.

 

Issue two:

I have 6 phones in a call group and I need the call group to ring each phone
like explained in issue one. I cannot for the life of me figure out how to
"layer" devices to failover independently when dialed in a group.

 

That is to say, when a call comes in it should be bridged to all 6 phones on
the lowest available line button.  Whoever answers first gets to call and
all other lines return to ready state.

 

Any thoughts would be greatly appreciated.

 

Sean

 

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From kheimerl at cs.berkeley.edu  Sun Nov 25 09:00:08 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sat, 24 Nov 2012 22:00:08 -0800
Subject: [Freeswitch-users] Return code from ESL Message Sending
In-Reply-To: <1388879B-963F-4127-B884-965671CD720D@edge-net.net>
References: <mailman.1329.1352570404.576.freeswitch-users@lists.freeswitch.org>
	<3A82F56B-1332-485E-9F6C-D2126A859CC8@edge-net.net>
	<CACyT-3mMQ8RJfD7J=X8ZZubgR5kKbUeRu_Lj_wYa=T3v+mQmJA@mail.gmail.com>
	<1388879B-963F-4127-B884-965671CD720D@edge-net.net>
Message-ID: <CACyT-3n+zuwsKF+KSeXoGTTi5H6XfQHzXwfsXk3_ZnPSu-3Dzw@mail.gmail.com>

Hi Eli,

I recently updated my installation, and now my old scripts for sending
messages through event creation aren't working. Basically, FS is sending
tens to hundreds of MESSAGEs a second after I send a SMS::SEND_MESSAGE
event. This was working before your patch (and I'm sure a bunch of other
patches), so I'm hoping you have some intuition as to what might be broken.
I'll start tracking it down myself here in a little bit.

On Wednesday, November 21, 2012, Eli Burke wrote:

> Kurtis,
>
> We've been working with FreeSWITCH Consulting to address some issues with
> MESSAGE delivery. A couple of patches were committed on Nov 13 and Nov 14
> to trunk and they may help with your problem. These patches affect the
> following behavior:
> * MESSAGEs fed through the chatplan are correctly delivered or ignored by
> sofia
> * when blocking=False, "Delivery-Failure" is replaced with
> "Nonblocking-Delivery: true"
> * when blocking=True, "Delivery-Failure" is correctly set to true or false
> * when blocking=True, "Delivery-Result-Code" is added to the event
>
> Some background explanation: MESSAGEs are normally delivered in
> non-blocking mode, which means FreeSWITCH makes no attempt to determine if
> they were successfully received. There is a variable that can be set
> ("blocking: true") to force FreeSWITCH to wait for a response. You can
> already see this in action using the chat command in fs_cli-- it will
> report success or failure.
>
> Unfortunately, "blocking" is not set by default. RIght now, the only way
> to get this behavior is to set it manually. For example, a chatplan rule to
> add the header to all inbound MESSAGEs:
>     <extension name="add_blocking" continue="true">
>       <condition>
>           <action application="set"
> data="is_reg=${sofia_contact(${to_user}" inline="true"/>
>           <action application="set" data="blocking=true"/>
>       </condition>
>     </extension>
>
> There is a potential (and untested!) downside to forcing blocking to be
> always-on. The MESSAGE delivery queue is currently handled by a single
> thread. Even if all MESSAGE objects are delivered successfully to the local
> switch, some amount of latency may be introduced. In a real-world
> high-throughput scenario, it's possible that this could cause noticeable
> delays in the time it takes to delivery a MESSAGE, creating an ever-growing
> backlog.
>
> The "is_reg" variable in the rule above could be used to short circuit
> failed attempts by shunting MESSAGEs to a database, or dropping them on the
> floor, but this would not necessarily fix things. The good news is that if
> a high-volume user can demonstrate that there is a problem, it's fixable
> within FreeSWITCH by moving to a multi-threaded message delivery queue.
>
> -Eli
>
>
> On Nov 10, 2012, at 1:00 PM, freeswitch-users-request at lists.freeswitch.orgwrote:
>
> *From: *Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> *Subject: **[Freeswitch-users] Return code from ESL Message Sending*
>> *Date: *November 9, 2012 11:42:19 PM EST
>> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>
>>
>> Hello Freeswitch Users:
>>
>> We're currently trying to get the return code from a MESSAGE we send
>> using ESL. The closest we've found is this jira:
>> http://jira.freeswitch.org/browse/FS-4453 which seems to provide similar
>> functionality for the chat command, but nothing for ESL.
>>
>> Here's a pastebin of our current code:
>> http://pastebin.freeswitch.org/20201
>>
>> The server we are hitting is returning a "415 Unsupported Content Type"
>> (which is correct) and we're trying to discover that in freeswitch, instead
>> of assuming the message was received correctly. Right now, we get that the
>> recvEventTimed is returning None. This is all done on the a pull of FS from
>> yesterday.
>>
>>  Any suggestions?
>>
>>
>
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From yehavi.bourvine at gmail.com  Sun Nov 25 11:03:57 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 25 Nov 2012 10:03:57 +0200
Subject: [Freeswitch-users] SRTP - how to enable on the other leg?
In-Reply-To: <E3BEF350-174D-4BF0-932B-38E77333D08A@freeswitch.org>
References: <CAAtWKeEnSFL8sdiJXwd9vbboUZ9O+hF68-5hGXy04di6HpnPxw@mail.gmail.com>
	<E3BEF350-174D-4BF0-932B-38E77333D08A@freeswitch.org>
Message-ID: <CAAtWKeGONaOE82AqmMA-ow-+5GGr=H93v8=jwbxVLfpA1LRuYA@mail.gmail.com>

Thanks to all who have replied - the problem has been found (my fault...).

The variables set above (with set/export) work as intended. I simply had a
clause in the dial string that sets SRTP only if TLS is active as well.

                    __Yehavi:


2012/11/21 Brian West <brian at freeswitch.org>

> you make sure sip_secure_media=true is set inside {} on the originate or
> export the variable.
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
> On Nov 21, 2012, at 7:08 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>
> > Hi,
> >
> >   I am trying to use SRTP without TLS. The initiating side asks and get
> SRTP (two way) while the other side is not offered SRTP at all.
> >
> > I've noted that ${sip_has_crypto} is null. I also tried setting
> unconditionaly sip_secure_rtp (both set and export), but it doesn't help.
> >
> > Any idea what I am doing wrong?
> >
> > BTW, I am using today's mornning GIT.
> >
> >                   Thanks, __Yehavi:
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From oseslija at gmail.com  Sun Nov 25 12:58:49 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 25 Nov 2012 10:58:49 +0100
Subject: [Freeswitch-users] Tricky rollover Dial Plan Question
In-Reply-To: <0c3d01cdcacc$abedcd00$03c96700$@bizfocused.com>
References: <0c3d01cdcacc$abedcd00$03c96700$@bizfocused.com>
Message-ID: <CAM21H-Mc_MJKn4WYdMEtXcioNrR5TyejYXs1yydf3SWhUj8Q+A@mail.gmail.com>

1) I guess one person uses that phone. With that in mind, one extension
(both on FS and on the phone) will do. If you need to route different DIDs
you can do that on that some extension (not map DIDs to exts 1-1). You can
just set Line Key 2,3,4 to Ext 1 and they will blink and phone will ring
 if another call comes in, during first call. In this case call waiting
needs to be enabled.
No complicated dialplan is needed, default one will do. If you disable CW,
phone will send 486 response immediately if on call, so you use
continue_on_fail=USER_BUSY.
2) You can use something like

  <action application="bridge" data="${group_call(${group_name}@
${domain_name}+A)}"/>

in the dialplan and

     <group name="finance">
        <users>
          <user id="111" type="pointer"/>
          <user id="184" type="pointer"/>
          <user id="119" type="pointer"/>
        </users>
      </group>

in the directory.
This is scalable, you can easily set ring/hunt groups this way, just create
and edit them in the directory.
With regards to the buttons, that's really just configuration of the phone.

Join irc channel (#freeswitch on irc.freenode.net) if you have additional
questions.

On Sun, Nov 25, 2012 at 6:21 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi All,****
>
> ** **
>
> Thanks for your help as always.****
>
> ** **
>
> I have a CISCO 504G four line phone.  I don?t think the model is
> particularly relevant, just that it is a multi-line phone.****
>
> ** **
>
> *Issue one:*
>
> If line 1 is busy, I want new calls to be directed to line 2. If 1 and 2
> are in use, ring on 3 and then line 4.  I think this means call waiting
> will need to be disabled so that each will ring busy, enabling rollover.
> Also, I don?t want lines 2, 3 and 4 to all ring if 1 is busy.****
>
> ** **
>
> I think I just use continue_on_fail=true and hangup_after_bridge=true,
> then route to Line 1, then 2, 3 and 4.****
>
> ** **
>
> *Issue two:*
>
> I have 6 phones in a call group and I need the call group to ring each
> phone like explained in issue one. I cannot for the life of me figure out
> how to ?layer? devices to failover independently when dialed in a group.**
> **
>
> ** **
>
> That is to say, when a call comes in it should be bridged to all 6 phones
> on the lowest available line button.  Whoever answers first gets to call
> and all other lines return to ready state.****
>
> ** **
>
> Any thoughts would be greatly appreciated.****
>
> ** **
>
> Sean****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vbvbrj at gmail.com  Sun Nov 25 13:24:57 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Sun, 25 Nov 2012 12:24:57 +0200
Subject: [Freeswitch-users] max-retries when using freeswitch:Dbh()
Message-ID: <50B1F1F9.4090108@gmail.com>

How to specify max retries to DB with odbc when using freeswitch:Dbh() 
command to connect to some DB?

Default max-retries are set in switch_odbc.h:
#define DEFAULT_ODBC_RETRIES 120

-- 
Mimiko desu.


From shaheryarkh at gmail.com  Sun Nov 25 15:33:10 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Sun, 25 Nov 2012 13:33:10 +0100
Subject: [Freeswitch-users] Had a problem with X100P 1 port FXO card and
	freeswitch
In-Reply-To: <CAFUAJ8mnBtvQJ9tX+32yjrmWuW914yrJ252wKqteVLLpsL_NCw@mail.gmail.com>
References: <CAFUAJ8mnBtvQJ9tX+32yjrmWuW914yrJ252wKqteVLLpsL_NCw@mail.gmail.com>
Message-ID: <CAFZQphx6hUjgPK_TnrOLm6U=VKBjTMj1B2=Q3nLbU5C=YY31SQ@mail.gmail.com>

Well, the State: DOWN indicates, either a physical connectivity problem
e.g. line is dead or configuration parameters are incorrect.

Thank you.


On Sat, Nov 24, 2012 at 4:41 PM, Chaiyawut Sookplang <
chaiyawut.so at mail.kmutt.ac.th> wrote:

> Hi. I currently using Ubuntu 12.04 server.  I use FreeTDM with freeswitch
> and I already  installed Dahdi driver version 2.6 for X100P card. The
> driver is loaded and FreeTDM is able to start but I cannot make or receive
>  any calls. I condition is like no response from card, freeswitch not
> detect any incoming calls and no sign of POTS signalling or anything.
>
> This is a result from "ftdm dump"
> freeswitch at internal> ftdm dump 1 1
> span_id: 1
> chan_id: 1
> physical_span_id: 1
> physical_chan_id: 1
> physical_status: ok
> physical_status_red: 0
> physical_status_yellow: 0
> physical_status_rai: 0
> physical_status_blue: 0
> physical_status_ais: 0
> physical_status_general: 0
> signaling_status: UP
> type: FXO
> state: DOWN
> last_state: DOWN
> txgain: 0.00
> rxgain: 0.00
> cid_date:
> cid_name:
> cid_num:
> ani:
> aniII:
> dnis:
> rdnis:
> cause: NONE
> session: (none)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at peely.com  Sun Nov 25 15:54:36 2012
From: freeswitch at peely.com (peely)
Date: Sun, 25 Nov 2012 04:54:36 -0800 (PST)
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <CCB96C43.5EDD5%krice@freeswitch.org>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
Message-ID: <1353848076648-7584902.post@n2.nabble.com>

It would be interesting to see some support for databases which have no
native unixodbc support, like NoSQL solutions.

Of particular interest in Apache Cassandra, the reason being it supports
SQL-Like statements but has efficient and easy to configure support for
multi-master replication, so a FreeSWITCH Cluster could use it to share
registration and recovery data. It's also extremely fast and efficient.

I currently use MySQL for this, as it has OK support for multi-master, but
the overheads in binary logs on a busy box are pretty heavy.

I'm not sure how easy it would be to implement Apache Cassandra support, but
from my experiences in using it for other projects it would be very useful
in FreeSWITCH for sharing data across multiple FreeSWITCH boxes.


Neil.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Changes-to-how-ODBC-SQL-etc-works-tp7584221p7584902.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From abaci64 at gmail.com  Sun Nov 25 18:51:56 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 25 Nov 2012 10:51:56 -0500
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
 again
In-Reply-To: <CAFiqYumcCNB9zmv94er5KgSuG0VrWViqjy7phw1b=-8qNobCUw@mail.gmail.com>
References: <50AE7D9C.8040604@communicatefreely.net>
	<CAFiqYu=WYMb2aVqMh-POy-V+fbZMq_QOsn_5a2P=eg_EAtEKow@mail.gmail.com>
	<50AF96F4.2080005@communicatefreely.net>
	<CAFiqYumcCNB9zmv94er5KgSuG0VrWViqjy7phw1b=-8qNobCUw@mail.gmail.com>
Message-ID: <50B23E9C.3050103@gmail.com>

you mentioned that you use xml_curl, if your web server hangs it may 
hang sofia, iirc sofia is running a single thread and it will wait for 
the xml_curl response before continuing to the next request.

On 11/23/2012 1:13 PM, Steven Ayre wrote:
> Any kind of DB backup running? Or any long-running queries (innotop is
> great for highlighting queries that've been running a while, including
> on non-innodb tables).
>
> A global read lock, or queries waiting for a lock could block a db
> update from the sofia profile thread but still allow read-only queries
> (sofia status) to run.
>
> -Steve
>
>
>
> On 23 November 2012 15:32, Tim St. Pierre <fs-list at communicatefreely.net> wrote:
>> Hi Steven,
>>
>> Thanks for the suggestions.  I'm hoping once I get the upgrade done it will all go away.
>> I have watched it happen at least once.  I was on the phone at the time.  Console activity
>> more or less stopped, except for a few calls hanging up.  The console remains responsive,
>> and my call wasn't dropped for at least a minute or two (media timeout?).  I was able to
>> run sofia status and other commands that use the database, so I'm assuming that the
>> connection was still working.  All our media is runs through the box, so I think things
>> are fine on the Ethernet level.  I do see higher load averages - maybe 3-4, but that's the
>> only obvious indication.  It's not taking CPU beyond 10% or so.
>>
>> We are using MySQL as the core DB and also as the DB backend for each sofia profile.  This
>> is connecting through ODBC of course.
>>
>> If I can get the other kinks worked out, then I will try 1.2 stable in production and
>> we'll see how it goes.
>>
>> -Tim
>>
>> Steven Ayre wrote:
>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>
>>>
>>> As you've already acknowledged it's a very old version.
>>>
>>> It's possible that your issue has already been found and fixed, but if
>>> it hasn't then the code will have changed significantly since then and
>>> you'd really need to reproduce it on the latest code for it to be
>>> investigated.
>>>
>>>
>>> As some general thoughts though, are you able to spot it happening while
>>> it's happening or only afterwards?
>>>
>>> If you're able to get on the system during one of those times look at
>>> what else is happening. Is the load average/cpu usage/io high? Perhaps
>>> something's running that's blocking all access or causing very high IO.
>>>
>>> What DB backend are you using for Sofia? Is it possible that that's
>>> hanging for a moment? For example if you're running a backup on the DB
>>> that blocks all writes to the DB while Sofia is trying to update the DB
>>> that perhaps would cause this.
>>>
>>> Try running a SIP OPTIONS ping your your sofia profile from the
>>> localhost during that time, which should exclude it being any issue on
>>> the ethernet.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>> On 22 November 2012 19:31, Tim St. Pierre <fs-list at communicatefreely.net
>>> <mailto:fs-list at communicatefreely.net>> wrote:
>>>
>>>      Hello,
>>>
>>>      I'm having a bit of an odd problem.
>>>
>>>      Intermittently, often every 2-3 days or so, Freeswitch stops
>>>      replying to SIP for about 5
>>>      minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to
>>>      be pretty close.
>>>
>>>      During this time, no new registrations or invites can happen, but
>>>      existing calls stay
>>>      connected for at least a minute or two.  In the logs, you can see
>>>      calls slowly hanging up
>>>      with "NORMAL_CLEARING".  In 5 minutes, everything starts up again
>>>      with no word about it at
>>>      all in the logs.
>>>
>>>      When calls resume, I notice that the number of sessions returned by
>>>      the status command is
>>>      one higher than the actual number sessions returned by show
>>>      channels, or by looking in the
>>>      database.  Every time this happens, the discrepancy increases by one.
>>>
>>>      The interruption happens on all SIP profiles, but calls originated
>>>      from the socket API
>>>      still work, insofar as they return with PROGRESS_TIMEOUT since the
>>>      profiles are still
>>>      running, but stuck.
>>>
>>>      We are using ODBC/MySQL for the core database, and the database
>>>      server only runs this
>>>      database and some basic PHP/xml-curl stuff.
>>>
>>>      We have 416 endpoints registered, and usually sit at about 30
>>>      sessions during the day.
>>>
>>>      This never happens at night, only during busier times, but not
>>>      necessarily busy hour.
>>>
>>>      I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
>>>
>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>
>>>      Yes, I know it's old and I'm trying to upgrade, but I'm still having
>>>      some problems getting
>>>      all my phones to work properly with 1.2 stable.  This is a
>>>      production system, so I can't
>>>      just blindly put out the newest release.  Mostly, I need to buy
>>>      myself some time so that I
>>>      can get the kinks worked out of the latest version and then upgrade
>>>      the production box.
>>>
>>>      I'm grateful for any insights as to what could be happening, even if
>>>      a solution is just a
>>>      temporary workaround.
>>>
>>>      Thanks!
>>>
>>>      -Tim
>>>
>>>      _________________________________________________________________________
>>>      Professional FreeSWITCH Consulting Services:
>>>      consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>      http://www.freeswitchsolutions.com
>>>
>>>      
>>>      
>>>
>>>      Official FreeSWITCH Sites
>>>      http://www.freeswitch.org
>>>      http://wiki.freeswitch.org
>>>      http://www.cluecon.com
>>>
>>>      FreeSWITCH-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
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kbdfck at gmail.com  Sun Nov 25 19:16:42 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Sun, 25 Nov 2012 20:16:42 +0400
Subject: [Freeswitch-users] Sofia listens only on 127.0.0.1 after server VM
	reboot?
Message-ID: <CA+GmxbaZf+JnnUQMpv1H27xY+NGt3mcKLAaCCvjGrjpL5HOVEw@mail.gmail.com>

Hi All!

We faced strange problem after FS migration from hardware server to Debian
6.0.6 in VMWare ESXi.
After every VM reboot, FS starts normally, but doesn't receive any SIP
traffic.
netstat -ln shows that freeswitch listens on *:5060, but if we do 'sofia
profile local restart', it says than it restarts on 127.0.0.1
If we do /etc/init.d/freeswitch restart, FS starts normally and work as
expected.

My only idea is that probably FS starts earlier than network interfaces
become configured with IP, although there is $network in # Required-Start
section of init script. Nothing changes if I set $all in #Required-Start to
make FS start at the very end of boot.

I tried to configure static IP in config instead of $local_ipv4, but with
no luck. Probably if the problem is in unconfigured IP interface, we also
need enable_nonlocal_bind proc settings to make this work, but it is a
generally poor and inacceptable workaround :(
Any ideas how can I trace what is the exact problem leading to such
behaviour?


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From krice at freeswitch.org  Sun Nov 25 19:32:44 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 25 Nov 2012 10:32:44 -0600
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
 again
In-Reply-To: <50B23E9C.3050103@gmail.com>
Message-ID: <CCD7A44C.610F0%krice@freeswitch.org>

Sofia is not single threaded except for in one spot deep in libsofia,
>From there, messages are handed off to a number of message queues for FS
core to handle as needed...

Check to see if anything that fs is depending on is blocking on info
retrieval like the databases or other areas...

K

On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com> wrote:

> you mentioned that you use xml_curl, if your web server hangs it may
> hang sofia, iirc sofia is running a single thread and it will wait for
> the xml_curl response before continuing to the next request.
> 
> On 11/23/2012 1:13 PM, Steven Ayre wrote:
>> Any kind of DB backup running? Or any long-running queries (innotop is
>> great for highlighting queries that've been running a while, including
>> on non-innodb tables).
>> 
>> A global read lock, or queries waiting for a lock could block a db
>> update from the sofia profile thread but still allow read-only queries
>> (sofia status) to run.
>> 
>> -Steve
>> 
>> 
>> 
>> On 23 November 2012 15:32, Tim St. Pierre <fs-list at communicatefreely.net>
>> wrote:
>>> Hi Steven,
>>> 
>>> Thanks for the suggestions.  I'm hoping once I get the upgrade done it will
>>> all go away.
>>> I have watched it happen at least once.  I was on the phone at the time.
>>> Console activity
>>> more or less stopped, except for a few calls hanging up.  The console
>>> remains responsive,
>>> and my call wasn't dropped for at least a minute or two (media timeout?).  I
>>> was able to
>>> run sofia status and other commands that use the database, so I'm assuming
>>> that the
>>> connection was still working.  All our media is runs through the box, so I
>>> think things
>>> are fine on the Ethernet level.  I do see higher load averages - maybe 3-4,
>>> but that's the
>>> only obvious indication.  It's not taking CPU beyond 10% or so.
>>> 
>>> We are using MySQL as the core DB and also as the DB backend for each sofia
>>> profile.  This
>>> is connecting through ODBC of course.
>>> 
>>> If I can get the other kinks worked out, then I will try 1.2 stable in
>>> production and
>>> we'll see how it goes.
>>> 
>>> -Tim
>>> 
>>> Steven Ayre wrote:
>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>> 
>>>> 
>>>> As you've already acknowledged it's a very old version.
>>>> 
>>>> It's possible that your issue has already been found and fixed, but if
>>>> it hasn't then the code will have changed significantly since then and
>>>> you'd really need to reproduce it on the latest code for it to be
>>>> investigated.
>>>> 
>>>> 
>>>> As some general thoughts though, are you able to spot it happening while
>>>> it's happening or only afterwards?
>>>> 
>>>> If you're able to get on the system during one of those times look at
>>>> what else is happening. Is the load average/cpu usage/io high? Perhaps
>>>> something's running that's blocking all access or causing very high IO.
>>>> 
>>>> What DB backend are you using for Sofia? Is it possible that that's
>>>> hanging for a moment? For example if you're running a backup on the DB
>>>> that blocks all writes to the DB while Sofia is trying to update the DB
>>>> that perhaps would cause this.
>>>> 
>>>> Try running a SIP OPTIONS ping your your sofia profile from the
>>>> localhost during that time, which should exclude it being any issue on
>>>> the ethernet.
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On 22 November 2012 19:31, Tim St. Pierre <fs-list at communicatefreely.net
>>>> <mailto:fs-list at communicatefreely.net>> wrote:
>>>> 
>>>>      Hello,
>>>> 
>>>>      I'm having a bit of an odd problem.
>>>> 
>>>>      Intermittently, often every 2-3 days or so, Freeswitch stops
>>>>      replying to SIP for about 5
>>>>      minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to
>>>>      be pretty close.
>>>> 
>>>>      During this time, no new registrations or invites can happen, but
>>>>      existing calls stay
>>>>      connected for at least a minute or two.  In the logs, you can see
>>>>      calls slowly hanging up
>>>>      with "NORMAL_CLEARING".  In 5 minutes, everything starts up again
>>>>      with no word about it at
>>>>      all in the logs.
>>>> 
>>>>      When calls resume, I notice that the number of sessions returned by
>>>>      the status command is
>>>>      one higher than the actual number sessions returned by show
>>>>      channels, or by looking in the
>>>>      database.  Every time this happens, the discrepancy increases by one.
>>>> 
>>>>      The interruption happens on all SIP profiles, but calls originated
>>>>      from the socket API
>>>>      still work, insofar as they return with PROGRESS_TIMEOUT since the
>>>>      profiles are still
>>>>      running, but stuck.
>>>> 
>>>>      We are using ODBC/MySQL for the core database, and the database
>>>>      server only runs this
>>>>      database and some basic PHP/xml-curl stuff.
>>>> 
>>>>      We have 416 endpoints registered, and usually sit at about 30
>>>>      sessions during the day.
>>>> 
>>>>      This never happens at night, only during busier times, but not
>>>>      necessarily busy hour.
>>>> 
>>>>      I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
>>>> 
>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>> 
>>>>      Yes, I know it's old and I'm trying to upgrade, but I'm still having
>>>>      some problems getting
>>>>      all my phones to work properly with 1.2 stable.  This is a
>>>>      production system, so I can't
>>>>      just blindly put out the newest release.  Mostly, I need to buy
>>>>      myself some time so that I
>>>>      can get the kinks worked out of the latest version and then upgrade
>>>>      the production box.
>>>> 
>>>>      I'm grateful for any insights as to what could be happening, even if
>>>>      a solution is just a
>>>>      temporary workaround.
>>>> 
>>>>      Thanks!
>>>> 
>>>>      -Tim
>>>> 
>>>>      
>>>> _________________________________________________________________________
>>>>      Professional FreeSWITCH Consulting Services:
>>>>      consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>      http://www.freeswitchsolutions.com
>>>> 
>>>>      
>>>>      
>>>> 
>>>>      Official FreeSWITCH Sites
>>>>      http://www.freeswitch.org
>>>>      http://wiki.freeswitch.org
>>>>      http://www.cluecon.com
>>>> 
>>>>      FreeSWITCH-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
>>>> 
>>>> 
>>>> 
>>>> ------------------------------------------------------------------------
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From krice at freeswitch.org  Sun Nov 25 19:42:28 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 25 Nov 2012 10:42:28 -0600
Subject: [Freeswitch-users] Sofia listens only on 127.0.0.1 after server
 VM reboot?
In-Reply-To: <CA+GmxbaZf+JnnUQMpv1H27xY+NGt3mcKLAaCCvjGrjpL5HOVEw@mail.gmail.com>
Message-ID: <CCD7A694.610F3%krice@freeswitch.org>

Hard set an ip for sip profiles and never use 0.0.0.0 as the bind ip, it
does not work....


On 11/25/12 10:16 AM, "Dmitry Sytchev" <kbdfck at gmail.com> wrote:

> Hi All!
> 
> We faced strange problem after FS migration from hardware server to Debian
> 6.0.6 in VMWare ESXi.?
> After every VM reboot, FS starts normally, but doesn't receive any SIP
> traffic.?
> netstat -ln shows that freeswitch listens on *:5060, but if we do 'sofia
> profile local restart', it says than it restarts on 127.0.0.1?
> If we do /etc/init.d/freeswitch restart, FS starts normally and work as
> expected.
> 
> My only idea is that probably FS starts earlier than network interfaces become
> configured with IP, although there is $network in?# Required-Start section of
> init script. Nothing changes if I set $all in #Required-Start to make FS start
> at the very end of boot.
> 
> I tried to configure static IP in config instead of $local_ipv4, but with no
> luck. Probably if the problem is in unconfigured IP interface, we also need
> enable_nonlocal_bind proc settings to make this work, but it is a generally
> poor and inacceptable workaround :(
> Any ideas how can I trace what is the exact problem leading to such
> behaviour??
> 

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

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From enp at itx.ru  Sun Nov 25 19:25:50 2012
From: enp at itx.ru (Eugene Prokopiev)
Date: Sun, 25 Nov 2012 19:25:50 +0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <CANM4RuipzyPL=ON8w9Z1LWbh+Ca=zsB2Gu71aqMvze=CDrgdzA@mail.gmail.com>
References: <CANM4Rui42maROz1u+Y-Z3DeTYvSs4mMQpJvqSiwGw=LPokZCAA@mail.gmail.com>
	<CANM4RuipzyPL=ON8w9Z1LWbh+Ca=zsB2Gu71aqMvze=CDrgdzA@mail.gmail.com>
Message-ID: <CANM4Ruh7+7BB9GejFAnaZ7LHWfSHWemLmoa+KFgZcRufwSxo6Q@mail.gmail.com>

Current mod_xml_ldap.c looks like broken, but I see mod_xml_ldapv2.c in the
same folder, which is more suitable for me with some fixes - see
http://jira.freeswitch.org/browse/FS-4870

--
Regards,
Eugene Prokopiev
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From development.milos at gmail.com  Sun Nov 25 20:14:31 2012
From: development.milos at gmail.com (Milos Jovanovic)
Date: Sun, 25 Nov 2012 18:14:31 +0100
Subject: [Freeswitch-users] Got HTTP Error 0 posting to web server
Message-ID: <CAPvcmJhwURqS85hs15Ds8hsgRRDam4JiY=FuW-tT12ApWt0L4Q@mail.gmail.com>

Hello,

I am getting these errors:

2012-11-25 01:19:51.857988 [ERR] mod_xml_cdr.c:367 Got error [0] posting to
web server [http://192.168.1.5/interface/fs_cdr]

It happens ONLY when there are a lot of simultaneous calls. Any ideas how
to deal with it?
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From abaci64 at gmail.com  Sun Nov 25 21:46:46 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 25 Nov 2012 13:46:46 -0500
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
 again
In-Reply-To: <CCD7A44C.610F0%krice@freeswitch.org>
References: <CCD7A44C.610F0%krice@freeswitch.org>
Message-ID: <50B26796.8050500@gmail.com>

I just remember reading somewhere on a blog something about a similar 
issue and this was the cause.
Just found that post 
http://blog.godson.in/2011/06/pitfalls-to-avoid-while-using.html not 
sure that's the issue. either way it's a good idea to set the timeout 
option.

On 11/25/2012 11:32 AM, Ken Rice wrote:
> Sofia is not single threaded except for in one spot deep in libsofia,
> >From there, messages are handed off to a number of message queues for FS
> core to handle as needed...
>
> Check to see if anything that fs is depending on is blocking on info
> retrieval like the databases or other areas...
>
> K
>
> On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com> wrote:
>
>> you mentioned that you use xml_curl, if your web server hangs it may
>> hang sofia, iirc sofia is running a single thread and it will wait for
>> the xml_curl response before continuing to the next request.
>>
>> On 11/23/2012 1:13 PM, Steven Ayre wrote:
>>> Any kind of DB backup running? Or any long-running queries (innotop is
>>> great for highlighting queries that've been running a while, including
>>> on non-innodb tables).
>>>
>>> A global read lock, or queries waiting for a lock could block a db
>>> update from the sofia profile thread but still allow read-only queries
>>> (sofia status) to run.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 23 November 2012 15:32, Tim St. Pierre <fs-list at communicatefreely.net>
>>> wrote:
>>>> Hi Steven,
>>>>
>>>> Thanks for the suggestions.  I'm hoping once I get the upgrade done it will
>>>> all go away.
>>>> I have watched it happen at least once.  I was on the phone at the time.
>>>> Console activity
>>>> more or less stopped, except for a few calls hanging up.  The console
>>>> remains responsive,
>>>> and my call wasn't dropped for at least a minute or two (media timeout?).  I
>>>> was able to
>>>> run sofia status and other commands that use the database, so I'm assuming
>>>> that the
>>>> connection was still working.  All our media is runs through the box, so I
>>>> think things
>>>> are fine on the Ethernet level.  I do see higher load averages - maybe 3-4,
>>>> but that's the
>>>> only obvious indication.  It's not taking CPU beyond 10% or so.
>>>>
>>>> We are using MySQL as the core DB and also as the DB backend for each sofia
>>>> profile.  This
>>>> is connecting through ODBC of course.
>>>>
>>>> If I can get the other kinks worked out, then I will try 1.2 stable in
>>>> production and
>>>> we'll see how it goes.
>>>>
>>>> -Tim
>>>>
>>>> Steven Ayre wrote:
>>>>>       Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>>>
>>>>>
>>>>> As you've already acknowledged it's a very old version.
>>>>>
>>>>> It's possible that your issue has already been found and fixed, but if
>>>>> it hasn't then the code will have changed significantly since then and
>>>>> you'd really need to reproduce it on the latest code for it to be
>>>>> investigated.
>>>>>
>>>>>
>>>>> As some general thoughts though, are you able to spot it happening while
>>>>> it's happening or only afterwards?
>>>>>
>>>>> If you're able to get on the system during one of those times look at
>>>>> what else is happening. Is the load average/cpu usage/io high? Perhaps
>>>>> something's running that's blocking all access or causing very high IO.
>>>>>
>>>>> What DB backend are you using for Sofia? Is it possible that that's
>>>>> hanging for a moment? For example if you're running a backup on the DB
>>>>> that blocks all writes to the DB while Sofia is trying to update the DB
>>>>> that perhaps would cause this.
>>>>>
>>>>> Try running a SIP OPTIONS ping your your sofia profile from the
>>>>> localhost during that time, which should exclude it being any issue on
>>>>> the ethernet.
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 22 November 2012 19:31, Tim St. Pierre <fs-list at communicatefreely.net
>>>>> <mailto:fs-list at communicatefreely.net>> wrote:
>>>>>
>>>>>       Hello,
>>>>>
>>>>>       I'm having a bit of an odd problem.
>>>>>
>>>>>       Intermittently, often every 2-3 days or so, Freeswitch stops
>>>>>       replying to SIP for about 5
>>>>>       minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to
>>>>>       be pretty close.
>>>>>
>>>>>       During this time, no new registrations or invites can happen, but
>>>>>       existing calls stay
>>>>>       connected for at least a minute or two.  In the logs, you can see
>>>>>       calls slowly hanging up
>>>>>       with "NORMAL_CLEARING".  In 5 minutes, everything starts up again
>>>>>       with no word about it at
>>>>>       all in the logs.
>>>>>
>>>>>       When calls resume, I notice that the number of sessions returned by
>>>>>       the status command is
>>>>>       one higher than the actual number sessions returned by show
>>>>>       channels, or by looking in the
>>>>>       database.  Every time this happens, the discrepancy increases by one.
>>>>>
>>>>>       The interruption happens on all SIP profiles, but calls originated
>>>>>       from the socket API
>>>>>       still work, insofar as they return with PROGRESS_TIMEOUT since the
>>>>>       profiles are still
>>>>>       running, but stuck.
>>>>>
>>>>>       We are using ODBC/MySQL for the core database, and the database
>>>>>       server only runs this
>>>>>       database and some basic PHP/xml-curl stuff.
>>>>>
>>>>>       We have 416 endpoints registered, and usually sit at about 30
>>>>>       sessions during the day.
>>>>>
>>>>>       This never happens at night, only during busier times, but not
>>>>>       necessarily busy hour.
>>>>>
>>>>>       I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
>>>>>
>>>>>       Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>>>
>>>>>       Yes, I know it's old and I'm trying to upgrade, but I'm still having
>>>>>       some problems getting
>>>>>       all my phones to work properly with 1.2 stable.  This is a
>>>>>       production system, so I can't
>>>>>       just blindly put out the newest release.  Mostly, I need to buy
>>>>>       myself some time so that I
>>>>>       can get the kinks worked out of the latest version and then upgrade
>>>>>       the production box.
>>>>>
>>>>>       I'm grateful for any insights as to what could be happening, even if
>>>>>       a solution is just a
>>>>>       temporary workaround.
>>>>>
>>>>>       Thanks!
>>>>>
>>>>>       -Tim
>>>>>
>>>>>       
>>>>> _________________________________________________________________________
>>>>>       Professional FreeSWITCH Consulting Services:
>>>>>       consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>       http://www.freeswitchsolutions.com
>>>>>
>>>>>       
>>>>>       
>>>>>
>>>>>       Official FreeSWITCH Sites
>>>>>       http://www.freeswitch.org
>>>>>       http://wiki.freeswitch.org
>>>>>       http://www.cluecon.com
>>>>>
>>>>>       FreeSWITCH-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
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org



From anthony.minessale at gmail.com  Sun Nov 25 21:51:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 25 Nov 2012 12:51:44 -0600
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
	again
In-Reply-To: <CCD7A44C.610F0%krice@freeswitch.org>
References: <50B23E9C.3050103@gmail.com> <CCD7A44C.610F0%krice@freeswitch.org>
Message-ID: <CAKbxfG8cwXz=gWYP3P6Qr7mLf9_UArMXnUV=ec4pPRv51GGsdA@mail.gmail.com>

Gcore it when its stuck.  Then you can bt. There will be no real help if
the problem is that its older code.  We don't have the resources to support
old versions.  You may want to test out the stable branch....
On Nov 25, 2012 12:31 PM, "Ken Rice" <krice at freeswitch.org> wrote:

> Sofia is not single threaded except for in one spot deep in libsofia,
> >From there, messages are handed off to a number of message queues for FS
> core to handle as needed...
>
> Check to see if anything that fs is depending on is blocking on info
> retrieval like the databases or other areas...
>
> K
>
> On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com> wrote:
>
> > you mentioned that you use xml_curl, if your web server hangs it may
> > hang sofia, iirc sofia is running a single thread and it will wait for
> > the xml_curl response before continuing to the next request.
> >
> > On 11/23/2012 1:13 PM, Steven Ayre wrote:
> >> Any kind of DB backup running? Or any long-running queries (innotop is
> >> great for highlighting queries that've been running a while, including
> >> on non-innodb tables).
> >>
> >> A global read lock, or queries waiting for a lock could block a db
> >> update from the sofia profile thread but still allow read-only queries
> >> (sofia status) to run.
> >>
> >> -Steve
> >>
> >>
> >>
> >> On 23 November 2012 15:32, Tim St. Pierre <
> fs-list at communicatefreely.net>
> >> wrote:
> >>> Hi Steven,
> >>>
> >>> Thanks for the suggestions.  I'm hoping once I get the upgrade done it
> will
> >>> all go away.
> >>> I have watched it happen at least once.  I was on the phone at the
> time.
> >>> Console activity
> >>> more or less stopped, except for a few calls hanging up.  The console
> >>> remains responsive,
> >>> and my call wasn't dropped for at least a minute or two (media
> timeout?).  I
> >>> was able to
> >>> run sofia status and other commands that use the database, so I'm
> assuming
> >>> that the
> >>> connection was still working.  All our media is runs through the box,
> so I
> >>> think things
> >>> are fine on the Ethernet level.  I do see higher load averages - maybe
> 3-4,
> >>> but that's the
> >>> only obvious indication.  It's not taking CPU beyond 10% or so.
> >>>
> >>> We are using MySQL as the core DB and also as the DB backend for each
> sofia
> >>> profile.  This
> >>> is connecting through ODBC of course.
> >>>
> >>> If I can get the other kinks worked out, then I will try 1.2 stable in
> >>> production and
> >>> we'll see how it goes.
> >>>
> >>> -Tim
> >>>
> >>> Steven Ayre wrote:
> >>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
> >>>>
> >>>>
> >>>> As you've already acknowledged it's a very old version.
> >>>>
> >>>> It's possible that your issue has already been found and fixed, but if
> >>>> it hasn't then the code will have changed significantly since then and
> >>>> you'd really need to reproduce it on the latest code for it to be
> >>>> investigated.
> >>>>
> >>>>
> >>>> As some general thoughts though, are you able to spot it happening
> while
> >>>> it's happening or only afterwards?
> >>>>
> >>>> If you're able to get on the system during one of those times look at
> >>>> what else is happening. Is the load average/cpu usage/io high? Perhaps
> >>>> something's running that's blocking all access or causing very high
> IO.
> >>>>
> >>>> What DB backend are you using for Sofia? Is it possible that that's
> >>>> hanging for a moment? For example if you're running a backup on the DB
> >>>> that blocks all writes to the DB while Sofia is trying to update the
> DB
> >>>> that perhaps would cause this.
> >>>>
> >>>> Try running a SIP OPTIONS ping your your sofia profile from the
> >>>> localhost during that time, which should exclude it being any issue on
> >>>> the ethernet.
> >>>>
> >>>> -Steve
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On 22 November 2012 19:31, Tim St. Pierre <
> fs-list at communicatefreely.net
> >>>> <mailto:fs-list at communicatefreely.net>> wrote:
> >>>>
> >>>>      Hello,
> >>>>
> >>>>      I'm having a bit of an odd problem.
> >>>>
> >>>>      Intermittently, often every 2-3 days or so, Freeswitch stops
> >>>>      replying to SIP for about 5
> >>>>      minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems
> to
> >>>>      be pretty close.
> >>>>
> >>>>      During this time, no new registrations or invites can happen, but
> >>>>      existing calls stay
> >>>>      connected for at least a minute or two.  In the logs, you can see
> >>>>      calls slowly hanging up
> >>>>      with "NORMAL_CLEARING".  In 5 minutes, everything starts up again
> >>>>      with no word about it at
> >>>>      all in the logs.
> >>>>
> >>>>      When calls resume, I notice that the number of sessions returned
> by
> >>>>      the status command is
> >>>>      one higher than the actual number sessions returned by show
> >>>>      channels, or by looking in the
> >>>>      database.  Every time this happens, the discrepancy increases by
> one.
> >>>>
> >>>>      The interruption happens on all SIP profiles, but calls
> originated
> >>>>      from the socket API
> >>>>      still work, insofar as they return with PROGRESS_TIMEOUT since
> the
> >>>>      profiles are still
> >>>>      running, but stuck.
> >>>>
> >>>>      We are using ODBC/MySQL for the core database, and the database
> >>>>      server only runs this
> >>>>      database and some basic PHP/xml-curl stuff.
> >>>>
> >>>>      We have 416 endpoints registered, and usually sit at about 30
> >>>>      sessions during the day.
> >>>>
> >>>>      This never happens at night, only during busier times, but not
> >>>>      necessarily busy hour.
> >>>>
> >>>>      I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
> >>>>
> >>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
> >>>>
> >>>>      Yes, I know it's old and I'm trying to upgrade, but I'm still
> having
> >>>>      some problems getting
> >>>>      all my phones to work properly with 1.2 stable.  This is a
> >>>>      production system, so I can't
> >>>>      just blindly put out the newest release.  Mostly, I need to buy
> >>>>      myself some time so that I
> >>>>      can get the kinks worked out of the latest version and then
> upgrade
> >>>>      the production box.
> >>>>
> >>>>      I'm grateful for any insights as to what could be happening,
> even if
> >>>>      a solution is just a
> >>>>      temporary workaround.
> >>>>
> >>>>      Thanks!
> >>>>
> >>>>      -Tim
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>>      Professional FreeSWITCH Consulting Services:
> >>>>      consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>>>      http://www.freeswitchsolutions.com
> >>>>
> >>>>      
> >>>>      
> >>>>
> >>>>      Official FreeSWITCH Sites
> >>>>      http://www.freeswitch.org
> >>>>      http://wiki.freeswitch.org
> >>>>      http://www.cluecon.com
> >>>>
> >>>>      FreeSWITCH-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
> >>>>
> >>>>
> >>>>
> >>>>
> ------------------------------------------------------------------------
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kbdfck at gmail.com  Sun Nov 25 21:55:28 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Sun, 25 Nov 2012 22:55:28 +0400
Subject: [Freeswitch-users] Sofia listens only on 127.0.0.1 after server
 VM reboot?
In-Reply-To: <CCD7A694.610F3%krice@freeswitch.org>
References: <CA+GmxbaZf+JnnUQMpv1H27xY+NGt3mcKLAaCCvjGrjpL5HOVEw@mail.gmail.com>
	<CCD7A694.610F3%krice@freeswitch.org>
Message-ID: <CA+GmxbbXeQo+NdCp7eH079Sxw0Q7X6WorpfjXMRXrh=+ohet+A@mail.gmail.com>

I tried to set static IP, that doesn't work :(


2012/11/25 Ken Rice <krice at freeswitch.org>

>  Hard set an ip for sip profiles and never use 0.0.0.0 as the bind ip, it
> does not work....
>
>
>
> On 11/25/12 10:16 AM, "Dmitry Sytchev" <kbdfck at gmail.com> wrote:
>
> Hi All!
>
> We faced strange problem after FS migration from hardware server to Debian
> 6.0.6 in VMWare ESXi.
> After every VM reboot, FS starts normally, but doesn't receive any SIP
> traffic.
> netstat -ln shows that freeswitch listens on *:5060, but if we do 'sofia
> profile local restart', it says than it restarts on 127.0.0.1
> If we do /etc/init.d/freeswitch restart, FS starts normally and work as
> expected.
>
> My only idea is that probably FS starts earlier than network interfaces
> become configured with IP, although there is $network in # Required-Start
> section of init script. Nothing changes if I set $all in #Required-Start to
> make FS start at the very end of boot.
>
> I tried to configure static IP in config instead of $local_ipv4, but with
> no luck. Probably if the problem is in unconfigured IP interface, we also
> need enable_nonlocal_bind proc settings to make this work, but it is a
> generally poor and inacceptable workaround :(
> Any ideas how can I trace what is the exact problem leading to such
> behaviour?
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From abaci64 at gmail.com  Sun Nov 25 22:26:54 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 25 Nov 2012 14:26:54 -0500
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
 again
In-Reply-To: <50B26796.8050500@gmail.com>
References: <CCD7A44C.610F0%krice@freeswitch.org> <50B26796.8050500@gmail.com>
Message-ID: <50B270FE.7030405@gmail.com>

See also http://jira.freeswitch.org/browse/FS-3328

On 11/25/2012 1:46 PM, Abaci wrote:
> I just remember reading somewhere on a blog something about a similar 
> issue and this was the cause.
> Just found that post 
> http://blog.godson.in/2011/06/pitfalls-to-avoid-while-using.html not 
> sure that's the issue. either way it's a good idea to set the timeout 
> option.
>
> On 11/25/2012 11:32 AM, Ken Rice wrote:
>> Sofia is not single threaded except for in one spot deep in libsofia,
>> >From there, messages are handed off to a number of message queues 
>> for FS
>> core to handle as needed...
>>
>> Check to see if anything that fs is depending on is blocking on info
>> retrieval like the databases or other areas...
>>
>> K
>>
>> On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com> wrote:
>>
>>> you mentioned that you use xml_curl, if your web server hangs it may
>>> hang sofia, iirc sofia is running a single thread and it will wait for
>>> the xml_curl response before continuing to the next request.
>>>
>>> On 11/23/2012 1:13 PM, Steven Ayre wrote:
>>>> Any kind of DB backup running? Or any long-running queries (innotop is
>>>> great for highlighting queries that've been running a while, including
>>>> on non-innodb tables).
>>>>
>>>> A global read lock, or queries waiting for a lock could block a db
>>>> update from the sofia profile thread but still allow read-only queries
>>>> (sofia status) to run.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 23 November 2012 15:32, Tim St. Pierre 
>>>> <fs-list at communicatefreely.net>
>>>> wrote:
>>>>> Hi Steven,
>>>>>
>>>>> Thanks for the suggestions.  I'm hoping once I get the upgrade 
>>>>> done it will
>>>>> all go away.
>>>>> I have watched it happen at least once.  I was on the phone at the 
>>>>> time.
>>>>> Console activity
>>>>> more or less stopped, except for a few calls hanging up. The console
>>>>> remains responsive,
>>>>> and my call wasn't dropped for at least a minute or two (media 
>>>>> timeout?).  I
>>>>> was able to
>>>>> run sofia status and other commands that use the database, so I'm 
>>>>> assuming
>>>>> that the
>>>>> connection was still working.  All our media is runs through the 
>>>>> box, so I
>>>>> think things
>>>>> are fine on the Ethernet level.  I do see higher load averages - 
>>>>> maybe 3-4,
>>>>> but that's the
>>>>> only obvious indication.  It's not taking CPU beyond 10% or so.
>>>>>
>>>>> We are using MySQL as the core DB and also as the DB backend for 
>>>>> each sofia
>>>>> profile.  This
>>>>> is connecting through ODBC of course.
>>>>>
>>>>> If I can get the other kinks worked out, then I will try 1.2 
>>>>> stable in
>>>>> production and
>>>>> we'll see how it goes.
>>>>>
>>>>> -Tim
>>>>>
>>>>> Steven Ayre wrote:
>>>>>>       Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>>>>
>>>>>>
>>>>>> As you've already acknowledged it's a very old version.
>>>>>>
>>>>>> It's possible that your issue has already been found and fixed, 
>>>>>> but if
>>>>>> it hasn't then the code will have changed significantly since 
>>>>>> then and
>>>>>> you'd really need to reproduce it on the latest code for it to be
>>>>>> investigated.
>>>>>>
>>>>>>
>>>>>> As some general thoughts though, are you able to spot it 
>>>>>> happening while
>>>>>> it's happening or only afterwards?
>>>>>>
>>>>>> If you're able to get on the system during one of those times 
>>>>>> look at
>>>>>> what else is happening. Is the load average/cpu usage/io high? 
>>>>>> Perhaps
>>>>>> something's running that's blocking all access or causing very 
>>>>>> high IO.
>>>>>>
>>>>>> What DB backend are you using for Sofia? Is it possible that that's
>>>>>> hanging for a moment? For example if you're running a backup on 
>>>>>> the DB
>>>>>> that blocks all writes to the DB while Sofia is trying to update 
>>>>>> the DB
>>>>>> that perhaps would cause this.
>>>>>>
>>>>>> Try running a SIP OPTIONS ping your your sofia profile from the
>>>>>> localhost during that time, which should exclude it being any 
>>>>>> issue on
>>>>>> the ethernet.
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 22 November 2012 19:31, Tim St. Pierre 
>>>>>> <fs-list at communicatefreely.net
>>>>>> <mailto:fs-list at communicatefreely.net>> wrote:
>>>>>>
>>>>>>       Hello,
>>>>>>
>>>>>>       I'm having a bit of an odd problem.
>>>>>>
>>>>>>       Intermittently, often every 2-3 days or so, Freeswitch stops
>>>>>>       replying to SIP for about 5
>>>>>>       minutes.  I can't verify if it's EXACTLY 5 minutes, but it 
>>>>>> seems to
>>>>>>       be pretty close.
>>>>>>
>>>>>>       During this time, no new registrations or invites can 
>>>>>> happen, but
>>>>>>       existing calls stay
>>>>>>       connected for at least a minute or two.  In the logs, you 
>>>>>> can see
>>>>>>       calls slowly hanging up
>>>>>>       with "NORMAL_CLEARING".  In 5 minutes, everything starts up 
>>>>>> again
>>>>>>       with no word about it at
>>>>>>       all in the logs.
>>>>>>
>>>>>>       When calls resume, I notice that the number of sessions 
>>>>>> returned by
>>>>>>       the status command is
>>>>>>       one higher than the actual number sessions returned by show
>>>>>>       channels, or by looking in the
>>>>>>       database.  Every time this happens, the discrepancy 
>>>>>> increases by one.
>>>>>>
>>>>>>       The interruption happens on all SIP profiles, but calls 
>>>>>> originated
>>>>>>       from the socket API
>>>>>>       still work, insofar as they return with PROGRESS_TIMEOUT 
>>>>>> since the
>>>>>>       profiles are still
>>>>>>       running, but stuck.
>>>>>>
>>>>>>       We are using ODBC/MySQL for the core database, and the 
>>>>>> database
>>>>>>       server only runs this
>>>>>>       database and some basic PHP/xml-curl stuff.
>>>>>>
>>>>>>       We have 416 endpoints registered, and usually sit at about 30
>>>>>>       sessions during the day.
>>>>>>
>>>>>>       This never happens at night, only during busier times, but not
>>>>>>       necessarily busy hour.
>>>>>>
>>>>>>       I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G 
>>>>>> ram)
>>>>>>
>>>>>>       Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>>>>
>>>>>>       Yes, I know it's old and I'm trying to upgrade, but I'm 
>>>>>> still having
>>>>>>       some problems getting
>>>>>>       all my phones to work properly with 1.2 stable. This is a
>>>>>>       production system, so I can't
>>>>>>       just blindly put out the newest release.  Mostly, I need to 
>>>>>> buy
>>>>>>       myself some time so that I
>>>>>>       can get the kinks worked out of the latest version and then 
>>>>>> upgrade
>>>>>>       the production box.
>>>>>>
>>>>>>       I'm grateful for any insights as to what could be 
>>>>>> happening, even if
>>>>>>       a solution is just a
>>>>>>       temporary workaround.
>>>>>>
>>>>>>       Thanks!
>>>>>>
>>>>>>       -Tim
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>       Professional FreeSWITCH Consulting Services:
>>>>>>       consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>>       http://www.freeswitchsolutions.com
>>>>>>
>>>>>>       
>>>>>>       
>>>>>>
>>>>>>       Official FreeSWITCH Sites
>>>>>>       http://www.freeswitch.org
>>>>>>       http://wiki.freeswitch.org
>>>>>>       http://www.cluecon.com
>>>>>>
>>>>>>       FreeSWITCH-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
>>>>>>
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------ 
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________ 
>>>>>>
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>> _________________________________________________________________________ 
>>>>>
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>>>>
>>>>> http://www.freeswitch.org
>>>> _________________________________________________________________________ 
>>>>
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>>>
>>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________ 
>>>
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>>
>>> http://www.freeswitch.org
>



From william.king at quentustech.com  Mon Nov 26 05:08:25 2012
From: william.king at quentustech.com (William King)
Date: Sun, 25 Nov 2012 18:08:25 -0800
Subject: [Freeswitch-users] Return code from ESL Message Sending
In-Reply-To: <CACyT-3n+zuwsKF+KSeXoGTTi5H6XfQHzXwfsXk3_ZnPSu-3Dzw@mail.gmail.com>
References: <mailman.1329.1352570404.576.freeswitch-users@lists.freeswitch.org>
	<3A82F56B-1332-485E-9F6C-D2126A859CC8@edge-net.net>
	<CACyT-3mMQ8RJfD7J=X8ZZubgR5kKbUeRu_Lj_wYa=T3v+mQmJA@mail.gmail.com>
	<1388879B-963F-4127-B884-965671CD720D@edge-net.net>
	<CACyT-3n+zuwsKF+KSeXoGTTi5H6XfQHzXwfsXk3_ZnPSu-3Dzw@mail.gmail.com>
Message-ID: <50B2CF19.3070908@quentustech.com>

Issue found and replicated, and patch to be pushed momentarily.

The problem is that FS has two methods for sending SMS messages,
blocking and non-blocking. By default endpoints send in blocking form,
and everything else defaults to non-blocking. In a recent patch the
non-blocking form was missed when confirming that the sms was sent
successfully, thus causing the message to be queued up again as soon as
it's sent.

It appears to have only effected sms messages that were queued in a
fully processed form(skipping the chatplan).

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

On 11/24/2012 10:00 PM, Kurtis Heimerl wrote:
> Hi Eli,
> 
> I recently updated my installation, and now my old scripts for sending
> messages through event creation aren't working. Basically, FS is sending
> tens to hundreds of MESSAGEs a second after I send a SMS::SEND_MESSAGE
> event. This was working before your patch (and I'm sure a bunch of other
> patches), so I'm hoping you have some intuition as to what might be
> broken. I'll start tracking it down myself here in a little bit. 
> 
> On Wednesday, November 21, 2012, Eli Burke wrote:
> 
>     Kurtis,
> 
>     We've been working with FreeSWITCH Consulting to address some issues
>     with MESSAGE delivery. A couple of patches were committed on Nov 13
>     and Nov 14 to trunk and they may help with your problem. These
>     patches affect the following behavior:
>     * MESSAGEs fed through the chatplan are correctly delivered or
>     ignored by sofia
>     * when blocking=False, "Delivery-Failure" is replaced with
>     "Nonblocking-Delivery: true"
>     * when blocking=True, "Delivery-Failure" is correctly set to true or
>     false
>     * when blocking=True, "Delivery-Result-Code" is added to the event
> 
>     Some background explanation: MESSAGEs are normally delivered in
>     non-blocking mode, which means FreeSWITCH makes no attempt to
>     determine if they were successfully received. There is a variable
>     that can be set ("blocking: true") to force FreeSWITCH to wait for a
>     response. You can already see this in action using the chat command
>     in fs_cli-- it will report success or failure. 
> 
>     Unfortunately, "blocking" is not set by default. RIght now, the only
>     way to get this behavior is to set it manually. For example, a
>     chatplan rule to add the header to all inbound MESSAGEs:
>         <extension name="add_blocking" continue="true">
>           <condition>
>               <action application="set"
>     data="is_reg=${sofia_contact(${to_user}" inline="true"/>
>               <action application="set" data="blocking=true"/>   
>           </condition>
>         </extension>
> 
>     There is a potential (and untested!) downside to forcing blocking to
>     be always-on. The MESSAGE delivery queue is currently handled by a
>     single thread. Even if all MESSAGE objects are delivered
>     successfully to the local switch, some amount of latency may be
>     introduced. In a real-world high-throughput scenario, it's possible
>     that this could cause noticeable delays in the time it takes to
>     delivery a MESSAGE, creating an ever-growing backlog.
> 
>     The "is_reg" variable in the rule above could be used to short
>     circuit failed attempts by shunting MESSAGEs to a database, or
>     dropping them on the floor, but this would not necessarily fix
>     things. The good news is that if a high-volume user can demonstrate
>     that there is a problem, it's fixable within FreeSWITCH by moving to
>     a multi-threaded message delivery queue.
> 
>     -Eli
> 
> 
>>     On Nov 10, 2012, at 1:00 PM,
>>     freeswitch-users-request at lists.freeswitch.org wrote:
>>
>>         *From: *Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>>         *Subject: **[Freeswitch-users] Return code from ESL Message
>>         Sending*
>>         *Date: *November 9, 2012 11:42:19 PM EST
>>         *To: *FreeSWITCH Users Help
>>         <freeswitch-users at lists.freeswitch.org>
>>         *Reply-To: *FreeSWITCH Users Help
>>         <freeswitch-users at lists.freeswitch.org>
>>
>>
>>         Hello Freeswitch Users:
>>
>>         We're currently trying to get the return code from a MESSAGE
>>         we send using ESL. The closest we've found is this
>>         jira: http://jira.freeswitch.org/browse/FS-4453 which seems to
>>         provide similar functionality for the chat command, but
>>         nothing for ESL.
>>
>>         Here's a pastebin of our current
>>         code: http://pastebin.freeswitch.org/20201
>>
>>         The server we are hitting is returning a "415 Unsupported
>>         Content Type" (which is correct) and we're trying to discover
>>         that in freeswitch, instead of assuming the message was
>>         received correctly. Right now, we get that the recvEventTimed
>>         is returning None. This is all done on the a pull of FS from
>>         yesterday. 
>>
>>         Any suggestions?
>>
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From kheimerl at cs.berkeley.edu  Mon Nov 26 06:13:28 2012
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sun, 25 Nov 2012 19:13:28 -0800
Subject: [Freeswitch-users] Return code from ESL Message Sending
In-Reply-To: <50B2CF19.3070908@quentustech.com>
References: <mailman.1329.1352570404.576.freeswitch-users@lists.freeswitch.org>
	<3A82F56B-1332-485E-9F6C-D2126A859CC8@edge-net.net>
	<CACyT-3mMQ8RJfD7J=X8ZZubgR5kKbUeRu_Lj_wYa=T3v+mQmJA@mail.gmail.com>
	<1388879B-963F-4127-B884-965671CD720D@edge-net.net>
	<CACyT-3n+zuwsKF+KSeXoGTTi5H6XfQHzXwfsXk3_ZnPSu-3Dzw@mail.gmail.com>
	<50B2CF19.3070908@quentustech.com>
Message-ID: <CACyT-3nGqvZ3LJfL8ZXQ=xsP6fgu6=JqmkhgT4qPif8Pr3j=Ww@mail.gmail.com>

Looks good to me, thanks for your time and effort William!

On Sunday, November 25, 2012, William King wrote:

> Issue found and replicated, and patch to be pushed momentarily.
>
> The problem is that FS has two methods for sending SMS messages,
> blocking and non-blocking. By default endpoints send in blocking form,
> and everything else defaults to non-blocking. In a recent patch the
> non-blocking form was missed when confirming that the sms was sent
> successfully, thus causing the message to be queued up again as soon as
> it's sent.
>
> It appears to have only effected sms messages that were queued in a
> fully processed form(skipping the chatplan).
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com <javascript:;>
>
> On 11/24/2012 10:00 PM, Kurtis Heimerl wrote:
> > Hi Eli,
> >
> > I recently updated my installation, and now my old scripts for sending
> > messages through event creation aren't working. Basically, FS is sending
> > tens to hundreds of MESSAGEs a second after I send a SMS::SEND_MESSAGE
> > event. This was working before your patch (and I'm sure a bunch of other
> > patches), so I'm hoping you have some intuition as to what might be
> > broken. I'll start tracking it down myself here in a little bit.
> >
> > On Wednesday, November 21, 2012, Eli Burke wrote:
> >
> >     Kurtis,
> >
> >     We've been working with FreeSWITCH Consulting to address some issues
> >     with MESSAGE delivery. A couple of patches were committed on Nov 13
> >     and Nov 14 to trunk and they may help with your problem. These
> >     patches affect the following behavior:
> >     * MESSAGEs fed through the chatplan are correctly delivered or
> >     ignored by sofia
> >     * when blocking=False, "Delivery-Failure" is replaced with
> >     "Nonblocking-Delivery: true"
> >     * when blocking=True, "Delivery-Failure" is correctly set to true or
> >     false
> >     * when blocking=True, "Delivery-Result-Code" is added to the event
> >
> >     Some background explanation: MESSAGEs are normally delivered in
> >     non-blocking mode, which means FreeSWITCH makes no attempt to
> >     determine if they were successfully received. There is a variable
> >     that can be set ("blocking: true") to force FreeSWITCH to wait for a
> >     response. You can already see this in action using the chat command
> >     in fs_cli-- it will report success or failure.
> >
> >     Unfortunately, "blocking" is not set by default. RIght now, the only
> >     way to get this behavior is to set it manually. For example, a
> >     chatplan rule to add the header to all inbound MESSAGEs:
> >         <extension name="add_blocking" continue="true">
> >           <condition>
> >               <action application="set"
> >     data="is_reg=${sofia_contact(${to_user}" inline="true"/>
> >               <action application="set" data="blocking=true"/>
> >           </condition>
> >         </extension>
> >
> >     There is a potential (and untested!) downside to forcing blocking to
> >     be always-on. The MESSAGE delivery queue is currently handled by a
> >     single thread. Even if all MESSAGE objects are delivered
> >     successfully to the local switch, some amount of latency may be
> >     introduced. In a real-world high-throughput scenario, it's possible
> >     that this could cause noticeable delays in the time it takes to
> >     delivery a MESSAGE, creating an ever-growing backlog.
> >
> >     The "is_reg" variable in the rule above could be used to short
> >     circuit failed attempts by shunting MESSAGEs to a database, or
> >     dropping them on the floor, but this would not necessarily fix
> >     things. The good news is that if a high-volume user can demonstrate
> >     that there is a problem, it's fixable within FreeSWITCH by moving to
> >     a multi-threaded message delivery queue.
> >
> >     -Eli
> >
> >
> >>     On Nov 10, 2012, at 1:00 PM,
> >>     freeswitch-users-request at lists.freeswitch.org <javascript:;> wrote:
> >>
> >>         *From: *Kurtis Heimerl <kheimerl at cs.berkeley.edu <javascript:;>
> >
> >>         *Subject: **[Freeswitch-users] Return code from ESL Message
> >>         Sending*
> >>         *Date: *November 9, 2012 11:42:19 PM EST
> >>         *To: *FreeSWITCH Users Help
> >>         <freeswitch-users at lists.freeswitch.org <javascript:;>>
> >>         *Reply-To: *FreeSWITCH Users Help
> >>         <freeswitch-users at lists.freeswitch.org <javascript:;>>
> >>
> >>
> >>         Hello Freeswitch Users:
> >>
> >>         We're currently trying to get the return code from a MESSAGE
> >>         we send using ESL. The closest we've found is this
> >>         jira: http://jira.freeswitch.org/browse/FS-4453 which seems to
> >>         provide similar functionality for the chat command, but
> >>         nothing for ESL.
> >>
> >>         Here's a pastebin of our current
> >>         code: http://pastebin.freeswitch.org/20201
> >>
> >>         The server we are hitting is returning a "415 Unsupported
> >>         Content Type" (which is correct) and we're trying to discover
> >>         that in freeswitch, instead of assuming the message was
> >>         received correctly. Right now, we get that the recvEventTimed
> >>         is returning None. This is all done on the a pull of FS from
> >>         yesterday.
> >>
> >>         Any suggestions?
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <javascript:;>
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From a.venugopan at mundio.com  Mon Nov 26 13:28:35 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Mon, 26 Nov 2012 10:28:35 +0000
Subject: [Freeswitch-users] Profile
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337C0C@Mail-Kilo.squay.com>

Hi,
When I make a call from my extension then to another extension within the domain then I get the sip_profile as 'internal'.


Core-UUID: 6fbd06fe-2fdd-11e2-b2f2-3ff7b399e7ed
FreeSWITCH-Hostname: squay-laptop-1.squay.com
FreeSWITCH-Switchname: squay-laptop-1.squay.com
FreeSWITCH-IPv4: 192.168.2.29
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2012-11-26%2010%3A12%3A45
Event-Date-GMT: Mon,%2026%20Nov%202012%2010%3A12%3A45%20GMT
Event-Date-Timestamp: 1353924765298047
Event-Calling-File: sofia_reg.c
Event-Calling-Function: sofia_reg_parse_auth
Event-Calling-Line-Number: 2371
action: sip_auth
sip_profile: internal
sip_user_agent: Aastra%2053i/2.6.0.66
sip_auth_username: 100
sip_auth_realm: fsfailover.uk01.com
sip_auth_nonce: d255b9f6-37b1-11e2-9f22-3ff7b399e7ed
sip_auth_uri: sip%3A07867429523%40fsfailover.uk01.com
sip_contact_user: 100
sip_contact_host: 192.168.2.234
sip_to_user: 07867429523
sip_to_host: fsfailover.uk01.com
sip_from_user: 100
sip_from_host: fsfailover.uk01.com
sip_request_user: 07867429523
sip_request_host: fsfailover.uk01.com
sip_auth_qop: auth
sip_auth_cnonce: 8da7d104
sip_auth_nc: 00000001
sip_auth_response: 9207e9f5ab74da1d6cd0414cd31e9354
sip_auth_method: INVITE
key: id
user: 100
domain: fsfailover.uk01.com
ip: 192.168.2.234

But I am trying to make a call from my extension to offnet where I should get sip_profile 'external'.  T below message from the xml file is getting displayed where I get external.

Dialplan: sofia/internal/100 at fsfailover.uk01.com Action bridge(sofia/external/0${msc_prefix}0786751162649 at 10.20.3.16<mailto:sofia/external/0$%7bmsc_prefix%7d07867429523 at 10.20.3.16>)

In lua script how can I configure external?? I tried giving profile='external' but it throwed an error. Only sip_profile='internal' is working. Please help me out in this. Thanks

Regards,
Archana

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From NuwanW at unifybusiness.co.uk  Mon Nov 26 14:14:33 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 26 Nov 2012 11:14:33 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>

Hello All,

I'm resending this request as I haven't had any reply for my last two emails. If anyone know any information regarding this, please let me know. . Any help would be greatly appreciated

I'm trying to broadcast audio on a bridged call. My requirement is to play audio on both legs at the same time. I used uuid_broadcast in following order,

uuid_broadcast uuid 'path' both

Please note that I'm sending uuid_broadcast through an esl connection. So my actual request to freeswitch is as follows,

eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file' both");

(eslWriteConnection is an object of .Net ESLConnection)

The  issue I'm having is, freeswitch not playing the audio on both channels at the same time. FreeSwitch plays the audio on one leg first, then plays on the second leg (After it finished playing on first leg).

I don't have this issue with FreeSwitch 1.0.6, where it plays audio on both legs at the same time. I'm having this issue with FreeSwtich 1.2.3 and later versions.

Could anyone please suggest any solution.

Thank you,

This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From xyangni at gmail.com  Mon Nov 26 14:37:24 2012
From: xyangni at gmail.com (Yihui Li)
Date: Mon, 26 Nov 2012 11:37:24 +0000
Subject: [Freeswitch-users] NAT thread error and can not reg to external sip
	server
Message-ID: <CAO1DH1-Td8B1t4pAgUH8rTRq8wJMAByCmJX07Qb6Z6U+Ke0Z+A@mail.gmail.com>

Hi,
I have downloaded the most recent git head and build it with VC2010 express
today. But fs start up with some nat error and can not register to external
sip server showing "error 408 request time out". what is wrong with it?
Thanks.

Below is part of the log file
D:\Dev\fsgit\Win32\Debug>FreeSwitchConsole.exe
2012-11-26 11:13:59.312500 [INFO] switch_event.c:589 Activate Eventing
Engine.
2012-11-26 11:13:59.343750 [WARNING] switch_event.c:563 Create additional
event dispa
tch thread 0
2012-11-26 11:13:59.468750 [INFO] switch_nat.c:420 Scanning for NAT
2012-11-26 11:13:59.468750 [ERR] switch_nat.c:201 Error checking for PMP
[init failed
]
2012-11-26 11:13:59.468750 [DEBUG] switch_nat.c:425 Checking for UPnP
2012-11-26 11:14:02.484375 [INFO] switch_nat.c:434 NAT detected type: upnp,
ExtIP: '9
5.172.xxxxxx'
2012-11-26 11:14:02.484375 [ERR] switch_nat.c:258 Bind Error
2012-11-26 11:14:02.484375 [ERR] switch_nat.c:365 Unable to initialize NAT
thread
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From a.venugopan at mundio.com  Mon Nov 26 14:42:39 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Mon, 26 Nov 2012 11:42:39 +0000
Subject: [Freeswitch-users] FW: Profile
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337C1D@Mail-Kilo.squay.com>

Hi,

For the below issue that I have raised I thought I would take sip_request_user and do something like if sip_ request _user is >100 and sip_ request _user is <100. But the issue is sip_ request_user is a string and when I tried to convert the string to number using this condition

local sip_request_user = tonumber("sip_request_user")

and dial I get a error like this
2012-11-26 11:34:50.758062 [CRIT] switch_cpp.cpp:1227 to test2aaa2012-11-26 11:34:50.758062 [ERR] mod_lua.cpp:196 /usr/local/freeswitch/scripts/directory.lua:175: attempt to compare nil with number
stack traceback:

what can be done to change that string to number. Please let me know.

Regards,
Archana

From: Archana Venugopan
Sent: 26 November 2012 10:29
To: 'FreeSWITCH Users Help'
Subject: Profile

Hi,
When I make a call from my extension then to another extension within the domain then I get the sip_profile as 'internal'.


Core-UUID: 6fbd06fe-2fdd-11e2-b2f2-3ff7b399e7ed
FreeSWITCH-Hostname: squay-laptop-1.squay.com
FreeSWITCH-Switchname: squay-laptop-1.squay.com
FreeSWITCH-IPv4: 192.168.2.29
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2012-11-26%2010%3A12%3A45
Event-Date-GMT: Mon,%2026%20Nov%202012%2010%3A12%3A45%20GMT
Event-Date-Timestamp: 1353924765298047
Event-Calling-File: sofia_reg.c
Event-Calling-Function: sofia_reg_parse_auth
Event-Calling-Line-Number: 2371
action: sip_auth
sip_profile: internal
sip_user_agent: Aastra%2053i/2.6.0.66
sip_auth_username: 100
sip_auth_realm: fsfailover.uk01.com
sip_auth_nonce: d255b9f6-37b1-11e2-9f22-3ff7b399e7ed
sip_auth_uri: sip%3A07867429523%40fsfailover.uk01.com
sip_contact_user: 100
sip_contact_host: 192.168.2.234
sip_to_user: 07867429523
sip_to_host: fsfailover.uk01.com
sip_from_user: 100
sip_from_host: fsfailover.uk01.com
sip_request_user: 07867429523
sip_request_host: fsfailover.uk01.com
sip_auth_qop: auth
sip_auth_cnonce: 8da7d104
sip_auth_nc: 00000001
sip_auth_response: 9207e9f5ab74da1d6cd0414cd31e9354
sip_auth_method: INVITE
key: id
user: 100
domain: fsfailover.uk01.com
ip: 192.168.2.234

But I am trying to make a call from my extension to offnet where I should get sip_profile 'external'.  T below message from the xml file is getting displayed where I get external.

Dialplan: sofia/internal/100 at fsfailover.uk01.com Action bridge(sofia/external/0${msc_prefix}0786751162649 at 10.20.3.16<mailto:sofia/external/0$%7bmsc_prefix%7d07867429523 at 10.20.3.16>)

In lua script how can I configure external?? I tried giving profile='external' but it throwed an error. Only sip_profile='internal' is working. Please help me out in this. Thanks

Regards,
Archana

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From gopalakrishnan.an at gmail.com  Mon Nov 26 15:51:12 2012
From: gopalakrishnan.an at gmail.com (Gopalakrishnan N)
Date: Mon, 26 Nov 2012 18:21:12 +0530
Subject: [Freeswitch-users] FreeTDM ftmod pri tap error
Message-ID: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>

Hi,

Am trying to install ftmod pri tap by following this link
http://wiki.freeswitch.org/wiki/FreeTDM#Tapping and enabled --with-pritap
while doing ./configure, and while doing make I get error.

Can someone help me on this. Do I need to update any patch for tap?

My installation log is attached here.

Regards.
Gopal.
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============================ FreeTDM configuration ============================

+ Modules

  Signalling:
        ftmod_analog....................... yes
        ftmod_analog_em.................... yes
        ftmod_isdn......................... no
        ftmod_libpri....................... yes
        ftmod_sangoma_isdn................. no
        ftmod_sangoma_ss7.................. no
        ftmod_r2........................... no
        ftmod_gsm.......................... no
        ftmod_pritap....................... yes
  I/O:
        ftmod_zt........................... yes
        ftmod_wanpipe...................... no
        ftmod_misdn........................ no

===============================================================================

[root at localhost freetdm]# make
if /bin/sh ./libtool --tag=CC --mode=compile gcc -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" -DPACKAGE_VERSION=\"pre-alpha\" -DPACKAGE_STRING=\"freetdm\ pre-alpha\" -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 -DHAVE_GETHOSTBYNAME_R=1 -DHAVE_LIBPRI_BRI=1 -DHAVE_LIBPRI_AOC=1  -I. -I./src/include -I.    -I/usr/src/freeswitch-1.2.3/libs/freetdm/src/include -I/usr/src/freeswitch-1.2.3/libs/freetdm/src/include/private -DFTDM_CONFIG_DIR=\"/usr/local/freeswitch//conf\" -DFTDM_MOD_DIR=\"/usr/local/freeswitch//mod\" -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -O0 -g -ggdb -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" -DPACKAGE_VERSION=\"pre-alpha\" -DPACKAGE_STRING=\"freetdm\ pre-alpha\" -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 -DHAVE_GETHOSTBYNAME_R=1 -DHAVE_LIBPRI_BRI=1 -DHAVE_LIBPRI_AOC=1   -g -O2 -DHAVE_ZLIB -MT ftmod_pritap_la-ftmod_pritap.lo -MD -MP -MF ".deps/ftmod_pritap_la-ftmod_pritap.Tpo" -c -o ftmod_pritap_la-ftmod_pritap.lo `test -f 'src/ftmod/ftmod_pritap/ftmod_pritap.c' || echo './'`src/ftmod/ftmod_pritap/ftmod_pritap.c; \
        then mv -f ".deps/ftmod_pritap_la-ftmod_pritap.Tpo" ".deps/ftmod_pritap_la-ftmod_pritap.Plo"; else rm -f ".deps/ftmod_pritap_la-ftmod_pritap.Tpo"; exit 1; fi
 gcc -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" -DPACKAGE_VERSION=\"pre-alpha\" "-DPACKAGE_STRING=\"freetdm pre-alpha\"" -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 -DHAVE_GETHOSTBYNAME_R=1 -DHAVE_LIBPRI_BRI=1 -DHAVE_LIBPRI_AOC=1 -I. -I./src/include -I. -I/usr/src/freeswitch-1.2.3/libs/freetdm/src/include -I/usr/src/freeswitch-1.2.3/libs/freetdm/src/include/private -DFTDM_CONFIG_DIR=\"/usr/local/freeswitch//conf\" -DFTDM_MOD_DIR=\"/usr/local/freeswitch//mod\" -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -O0 -g -ggdb -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" -DPACKAGE_VERSION=\"pre-alpha\" "-DPACKAGE_STRING=\"freetdm pre-alpha\"" -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 -DHAVE_GETHOSTBYNAME_R=1 -DHAVE_LIBPRI_BRI=1 -DHAVE_LIBPRI_AOC=1 -g -O2 -DHAVE_ZLIB -MT ftmod_pritap_la-ftmod_pritap.lo -MD -MP -MF .deps/ftmod_pritap_la-ftmod_pritap.Tpo -c src/ftmod/ftmod_pritap/ftmod_pritap.c  -fPIC -DPIC -o .libs/ftmod_pritap_la-ftmod_pritap.o
cc1: warnings being treated as errors
src/ftmod/ftmod_pritap/ftmod_pritap.c: In function 'tap_pri_get_pcall':
src/ftmod/ftmod_pritap/ftmod_pritap.c:430: warning: implicit declaration of function 'pri_passive_destroycall'
src/ftmod/ftmod_pritap/ftmod_pritap.c: In function 'handle_pri_passive_event':
src/ftmod/ftmod_pritap/ftmod_pritap.c:582: warning: implicit declaration of function 'pri_get_layer1'
src/ftmod/ftmod_pritap/ftmod_pritap.c:583: warning: implicit declaration of function 'pri_get_transcap'
src/ftmod/ftmod_pritap/ftmod_pritap.c: In function 'ftdm_pritap_run':
src/ftmod/ftmod_pritap/ftmod_pritap.c:712: warning: implicit declaration of function 'pri_read_event'
src/ftmod/ftmod_pritap/ftmod_pritap.c:712: warning: assignment makes pointer from integer without a cast
make: *** [ftmod_pritap_la-ftmod_pritap.lo] Error 1

From gopalakrishnan.an at gmail.com  Mon Nov 26 16:43:45 2012
From: gopalakrishnan.an at gmail.com (Gopalakrishnan N)
Date: Mon, 26 Nov 2012 19:13:45 +0530
Subject: [Freeswitch-users] FreeTDM ftmod pri tap error
In-Reply-To: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
References: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
Message-ID: <CAEymo12bugji+6UbV05FshCLAVBnJbOXpUj4gH1GyA0S_qa=Wg@mail.gmail.com>

Even I tried the moy tap1.4 complete patch with libpri, when i try to
execute with this ./configure --with-libpri --with-pritap
--prefix=/usr/local/freeswitch/ in make I am getting error like,

make: *** [ftmod_libpri_la-ftmod_libpri.lo] Error 1

Regards.


On Mon, Nov 26, 2012 at 6:21 PM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> Hi,
>
> Am trying to install ftmod pri tap by following this link
> http://wiki.freeswitch.org/wiki/FreeTDM#Tapping and enabled --with-pritap
> while doing ./configure, and while doing make I get error.
>
> Can someone help me on this. Do I need to update any patch for tap?
>
> My installation log is attached here.
>
> Regards.
> Gopal.
>
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From blee at gocentrix.com  Mon Nov 26 16:46:09 2012
From: blee at gocentrix.com (Bryant Lee)
Date: Mon, 26 Nov 2012 08:46:09 -0500
Subject: [Freeswitch-users] FW: Profile
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337C1D@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337C1D@Mail-Kilo.squay.com>
Message-ID: <CAGodFm_iWJPk=dQFJXLr-4wCReSPd81zmOkb8TbQ-fKD2gS0Uw@mail.gmail.com>

It looks like you're passing the string "sip_request_user" to the tonumber
function.  I think you need to remove the quotes


On Mon, Nov 26, 2012 at 6:42 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> For the below issue that I have raised I thought I would take
> sip_request_user and do something like if sip_ request _user is >100 and
> sip_ request _user is <100. But the issue is sip_ request_user is a string
> and when I tried to convert the string to number using this condition ****
>
> ** **
>
> local sip_request_user = tonumber("sip_request_user")  ****
>
> ** **
>
> and dial I get a error like this ****
>
> 2012-11-26 11:34:50.758062 [CRIT] switch_cpp.cpp:1227 to
> test2aaa2012-11-26 11:34:50.758062 [ERR] mod_lua.cpp:196
> /usr/local/freeswitch/scripts/directory.lua:175: attempt to compare nil
> with number****
>
> stack traceback:****
>
> ** **
>
> what can be done to change that string to number. Please let me know.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* Archana Venugopan
> *Sent:* 26 November 2012 10:29
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Profile****
>
> ** **
>
> Hi,****
>
> When I make a call from my extension then to another extension within the
> domain then I get the sip_profile as ?internal?.****
>
> ** **
>
> ** **
>
> Core-UUID: 6fbd06fe-2fdd-11e2-b2f2-3ff7b399e7ed****
>
> FreeSWITCH-Hostname: squay-laptop-1.squay.com****
>
> FreeSWITCH-Switchname: squay-laptop-1.squay.com****
>
> FreeSWITCH-IPv4: 192.168.2.29****
>
> FreeSWITCH-IPv6: %3A%3A1****
>
> Event-Date-Local: 2012-11-26%2010%3A12%3A45****
>
> Event-Date-GMT: Mon,%2026%20Nov%202012%2010%3A12%3A45%20GMT****
>
> Event-Date-Timestamp: 1353924765298047****
>
> Event-Calling-File: sofia_reg.c****
>
> Event-Calling-Function: sofia_reg_parse_auth****
>
> Event-Calling-Line-Number: 2371****
>
> action: sip_auth****
>
> sip_profile: internal****
>
> sip_user_agent: Aastra%2053i/2.6.0.66****
>
> sip_auth_username: 100****
>
> sip_auth_realm: fsfailover.uk01.com****
>
> sip_auth_nonce: d255b9f6-37b1-11e2-9f22-3ff7b399e7ed****
>
> sip_auth_uri: sip%3A07867429523%40fsfailover.uk01.com****
>
> sip_contact_user: 100****
>
> sip_contact_host: 192.168.2.234****
>
> sip_to_user: 07867429523****
>
> sip_to_host: fsfailover.uk01.com****
>
> sip_from_user: 100****
>
> sip_from_host: fsfailover.uk01.com****
>
> sip_request_user: 07867429523****
>
> sip_request_host: fsfailover.uk01.com****
>
> sip_auth_qop: auth****
>
> sip_auth_cnonce: 8da7d104****
>
> sip_auth_nc: 00000001****
>
> sip_auth_response: 9207e9f5ab74da1d6cd0414cd31e9354****
>
> sip_auth_method: INVITE****
>
> key: id****
>
> user: 100****
>
> domain: fsfailover.uk01.com****
>
> ip: 192.168.2.234****
>
> ** **
>
> But I am trying to make a call from my extension to offnet where I should
> get sip_profile ?external?.  T below message from the xml file is getting
> displayed where I get external.****
>
> ** **
>
> Dialplan: sofia/internal/100 at fsfailover.uk01.com Action bridge(
> sofia/external/0${msc_prefix}0786751162649 at 10.20.3.16<sofia/external/0$%7bmsc_prefix%7d07867429523 at 10.20.3.16>
> )****
>
> ** **
>
> In lua script how can I configure external?? I tried giving
> profile=?external? but it throwed an error. Only sip_profile=?internal? is
> working. Please help me out in this. Thanks ****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shaheryarkh at gmail.com  Mon Nov 26 17:13:16 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Mon, 26 Nov 2012 15:13:16 +0100
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
Message-ID: <CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>

Seems like a bug in FS. Can you open a JIRA item for this?

As a work around, can try to call uuid_broadcast with playback application,
see if that works, e.g.

uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e
playback!user_busy::hello-kitty.wav both

See this url for more info,

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast

Thank you.


On Mon, Nov 26, 2012 at 12:14 PM, Nuwan Wijerathne <
NuwanW at unifybusiness.co.uk> wrote:

>  Hello All,****
>
> ** **
>
> I?m resending this request as I haven?t had any reply for my last two
> emails. If anyone know any information regarding this, please let me know.
> . Any help would be greatly appreciated****
>
> ** **
>
> I?m trying to broadcast audio on a bridged call. My requirement is to play
> audio on both legs at the same time. I used uuid_broadcast in following
> order,****
>
> ** **
>
> uuid_broadcast uuid ?path? both ****
>
> ** **
>
> Please note that I?m sending uuid_broadcast through an esl connection. So
> my actual request to freeswitch is as follows,****
>
> ** **
>
> eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file'
> both");****
>
> ** **
>
> (eslWriteConnection is an object of .Net ESLConnection)****
>
> ** **
>
> The  issue I?m having is, freeswitch not playing the audio on both
> channels at the same time. FreeSwitch plays the audio on one leg first,
> then plays on the second leg (After it finished playing on first leg). ***
> *
>
> ** **
>
> I don?t have this issue with FreeSwitch 1.0.6, where it plays audio on
> both legs at the same time. I?m having this issue with FreeSwtich 1.2.3 and
> later versions. ****
>
> ** **
>
> Could anyone please suggest any solution.****
>
> ** **
>
> Thank you, ****
>
> ** **
>  This e-mail and any attachments are for the intended addressee(s) only
> and may contain confidential and/or privileged material. If you are not a
> named addressee, do not use, retain or disclose such information. This
> email is not guaranteed to be free from viruses and does not bind Unify in
> any contract or obligation. Unify Business Solutions Ltd. Registered in
> England and Wales. No: 4749638 Registered Office: Ambassador House, 5
> Midland Way, Barlborough, S43 4XA United Kingdom.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Mon Nov 26 17:28:55 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Nov 2012 08:28:55 -0600
Subject: [Freeswitch-users] Sofia listens only on 127.0.0.1 after server
	VM reboot?
In-Reply-To: <CA+GmxbbXeQo+NdCp7eH079Sxw0Q7X6WorpfjXMRXrh=+ohet+A@mail.gmail.com>
References: <CA+GmxbaZf+JnnUQMpv1H27xY+NGt3mcKLAaCCvjGrjpL5HOVEw@mail.gmail.com>
	<CCD7A694.610F3%krice@freeswitch.org>
	<CA+GmxbbXeQo+NdCp7eH079Sxw0Q7X6WorpfjXMRXrh=+ohet+A@mail.gmail.com>
Message-ID: <7B0E64C1-7767-46E4-B881-EC9CB3E707C3@freeswitch.org>

What are you setting in your profile as the IP?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 25, 2012, at 12:55 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> I tried to set static IP, that doesn't work :(



From brian at freeswitch.org  Mon Nov 26 17:30:44 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Nov 2012 08:30:44 -0600
Subject: [Freeswitch-users] Profile
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337C0C@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337C0C@Mail-Kilo.squay.com>
Message-ID: <46A318A8-EFB7-44C8-BBBE-47C32412DA3D@freeswitch.org>

We have no context to help here.  What exactly is the error, what is the script, what is the task you're trying to perform?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 4:28 AM, Archana Venugopan <a.venugopan at mundio.com> wrote:

> In lua script how can I configure external?? I tried giving profile=?external? but it throwed an error. Only sip_profile=?internal? is working. Please help me out in this. Thanks
>  
> Regards,
> Archana



From shaheryarkh at gmail.com  Mon Nov 26 17:31:12 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Mon, 26 Nov 2012 15:31:12 +0100
Subject: [Freeswitch-users] FreeTDM ftmod pri tap error
In-Reply-To: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
References: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
Message-ID: <CAFZQphzqUs910p49LkhA-rFSLcxvX-hPzC_LLS52ebb8KpfjWQ@mail.gmail.com>

These are only warnings, which are treated as error due to -Werror flag in
configuration. Trying installing stable version of FreeSWITCH which has
this flag removed, or search and remove this flag yourself in configure
scripts. FreeSWITCH developers discourage the second approach, so only do
that if you are really desperate to get it working. :-)

Thank you.


On Mon, Nov 26, 2012 at 1:51 PM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> Hi,
>
> Am trying to install ftmod pri tap by following this link
> http://wiki.freeswitch.org/wiki/FreeTDM#Tapping and enabled --with-pritap
> while doing ./configure, and while doing make I get error.
>
> Can someone help me on this. Do I need to update any patch for tap?
>
> My installation log is attached here.
>
> Regards.
> Gopal.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Mon Nov 26 17:33:25 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Nov 2012 08:33:25 -0600
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
Message-ID: <A42B214A-55F4-49DA-A2FF-81DCAC148EA2@freeswitch.org>

Support on 1.0.6 is a thing of the past.  I highly recommend you update to the latest as 1.0.6 is very old and unsupported.

--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 5:14 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:

> Hello All,
>  
> I?m resending this request as I haven?t had any reply for my last two emails. If anyone know any information regarding this, please let me know. . Any help would be greatly appreciated
>  
> I?m trying to broadcast audio on a bridged call. My requirement is to play audio on both legs at the same time. I used uuid_broadcast in following order,
>  
> uuid_broadcast uuid ?path? both
>  
> Please note that I?m sending uuid_broadcast through an esl connection. So my actual request to freeswitch is as follows,
>  
> eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file' both");
>  
> (eslWriteConnection is an object of .Net ESLConnection)
>  
> The  issue I?m having is, freeswitch not playing the audio on both channels at the same time. FreeSwitch plays the audio on one leg first, then plays on the second leg (After it finished playing on first leg).
>  
> I don?t have this issue with FreeSwitch 1.0.6, where it plays audio on both legs at the same time. I?m having this issue with FreeSwtich 1.2.3 and later versions.
>  
> Could anyone please suggest any solution.
>  
> Thank you,
>  
> This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom. _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Nov 26 17:34:36 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Nov 2012 08:34:36 -0600
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <20121105133721.01da8d23@dilbert>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
	<CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
	<CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>
	<CAKbxfG9OYOR9cfjSBmnow76d-0C8d6NgqVzuUPv_3CpWB_4mPQ@mail.gmail.com>
	<CALNZuwmuiufpioV-ktnSDky0YrtZwwi1g_MT5mCws+z5f48GvA@mail.gmail.com>
	<CAKbxfG-VDXx8wbwN4EW8K4nABEWmcdiBhb+hUK6-bp-b=avauA@mail.gmail.com>
	<20121105133721.01da8d23@dilbert>
Message-ID: <26F63AC0-A291-4103-8538-D8DDA094AFFA@freeswitch.org>

Do you want jira access to assist in this job?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 5, 2012, at 12:37 PM, D'Arcy J.M. Cain <darcy at Vex.Net> wrote:

> On Mon, 5 Nov 2012 11:42:56 -0600
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> ROADMAP
>> 
>> 1) Get someone to make a roadmap and organize it for us.
> 
> http://jira.freeswitch.org/browse/FS#selectedTab=com.atlassian.jira.plugin.system.project%3Aroadmap-panel
> 
> That's 1a.  Now 1b is to organize it.
> 
> -- 
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.Net
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Nov 26 17:48:02 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Nov 2012 08:48:02 -0600
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
Message-ID: <ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>

Are you doing this on the latest FreeSWTICH?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 8:13 AM, Muhammad Shahzad <shaheryarkh at gmail.com> wrote:

> Seems like a bug in FS. Can you open a JIRA item for this?
> 
> As a work around, can try to call uuid_broadcast with playback application, see if that works, e.g.
> 
> uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::hello-kitty.wav both
> 
> See this url for more info,
> 
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
> 
> Thank you.



From brian at freeswitch.org  Mon Nov 26 17:51:55 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Nov 2012 08:51:55 -0600
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
Message-ID: <EED5659F-1AAA-4AEF-A082-08FFC8FE7BE0@freeswitch.org>

I just tested this exact thing and it works fine on the latest Git rev.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 8:13 AM, Muhammad Shahzad <shaheryarkh at gmail.com> wrote:

> Seems like a bug in FS. Can you open a JIRA item for this?
> 
> As a work around, can try to call uuid_broadcast with playback application, see if that works, e.g.
> 
> uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::hello-kitty.wav both
> 
> See this url for more info,
> 
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
> 
> Thank you.



From NuwanW at unifybusiness.co.uk  Mon Nov 26 18:07:44 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 26 Nov 2012 15:07:44 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D1B81@BARXB0003.UnifyBusiness.local>

Thank you Muhammad. I?ll open a JIRA for this and in the meantime I?ll try your suggestion.

Thank you,
Nuwan.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad
Sent: 26 November 2012 14:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [Confidential] - uuid_broadcast

Seems like a bug in FS. Can you open a JIRA item for this?

As a work around, can try to call uuid_broadcast with playback application, see if that works, e.g.

uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::hello-kitty.wav both

See this url for more info,

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast

Thank you.

On Mon, Nov 26, 2012 at 12:14 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hello All,

I?m resending this request as I haven?t had any reply for my last two emails. If anyone know any information regarding this, please let me know. . Any help would be greatly appreciated

I?m trying to broadcast audio on a bridged call. My requirement is to play audio on both legs at the same time. I used uuid_broadcast in following order,

uuid_broadcast uuid ?path? both

Please note that I?m sending uuid_broadcast through an esl connection. So my actual request to freeswitch is as follows,

eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file' both");

(eslWriteConnection is an object of .Net ESLConnection)

The  issue I?m having is, freeswitch not playing the audio on both channels at the same time. FreeSwitch plays the audio on one leg first, then plays on the second leg (After it finished playing on first leg).

I don?t have this issue with FreeSwitch 1.0.6, where it plays audio on both legs at the same time. I?m having this issue with FreeSwtich 1.2.3 and later versions.

Could anyone please suggest any solution.

Thank you,

This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.

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




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

FreeSWITCH-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



--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From darcy at Vex.Net  Mon Nov 26 18:22:20 2012
From: darcy at Vex.Net (D'Arcy J.M. Cain)
Date: Mon, 26 Nov 2012 10:22:20 -0500
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <26F63AC0-A291-4103-8538-D8DDA094AFFA@freeswitch.org>
References: <CAFVUaV_R627RxHHqnZ_JYxTu8X1dBt+vN38y-X6gA2S3+f+6xA@mail.gmail.com>
	<CCBBEEF1.5EF42%krice@freeswitch.org>
	<CAFVUaV8SAR0W7r=PLKqXgN_9edDgdsgdFaB4F0NRSMqqdFHL5A@mail.gmail.com>
	<CAMwZ-tKmkEixXYvtQJ2+cq=N9VUqbEJFxpyO786N0dijpN11yQ@mail.gmail.com>
	<CAKbxfG9OYOR9cfjSBmnow76d-0C8d6NgqVzuUPv_3CpWB_4mPQ@mail.gmail.com>
	<CALNZuwmuiufpioV-ktnSDky0YrtZwwi1g_MT5mCws+z5f48GvA@mail.gmail.com>
	<CAKbxfG-VDXx8wbwN4EW8K4nABEWmcdiBhb+hUK6-bp-b=avauA@mail.gmail.com>
	<20121105133721.01da8d23@dilbert>
	<26F63AC0-A291-4103-8538-D8DDA094AFFA@freeswitch.org>
Message-ID: <20121126102220.530e5def@dilbert>

On Mon, 26 Nov 2012 08:34:36 -0600
Brian West <brian at freeswitch.org> wrote:
> Do you want jira access to assist in this job?

I am not sure that I know enough about the project to be the best
person for this but if you want I can make a start and ask lots of
questions.

Note that I do have basic jira access already.  I assume that you are
talking about another layer of access.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net


From NuwanW at unifybusiness.co.uk  Mon Nov 26 18:37:32 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 26 Nov 2012 15:37:32 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
	<ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D1BE2@BARXB0003.UnifyBusiness.local>

Yes. Version 1.3

Thank you,




Unify Business Solutions Ltd
Ambassador House, 5 Midland Way, Barlborough, Chesterfield, S43 4XA
Mobile: 07834 001304  | Tel: 08458717788 | Fax: 08458717799
Website: www.unifybusiness.co.uk

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 26 November 2012 14:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [Confidential] - uuid_broadcast

Are you doing this on the latest FreeSWTICH?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 8:13 AM, Muhammad Shahzad <shaheryarkh at gmail.com> wrote:

> Seems like a bug in FS. Can you open a JIRA item for this?
>
> As a work around, can try to call uuid_broadcast with playback application, see if that works, e.g.
>
> uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::hello-kitty.wav both
>
> See this url for more info,
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
>
> Thank you.


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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.


From a.venugopan at mundio.com  Mon Nov 26 19:29:15 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Mon, 26 Nov 2012 16:29:15 +0000
Subject: [Freeswitch-users] lua script
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>

Hi,

Does anyone know how can we use a variable declared inside if statement outside if block?

if sip_request_user == nil then
sip_request_user =  params:getHeader("sip_request_user")
else
local sip_request_user1 = tonumber(sip_request_user)
end

In my lua I have something like above and I want to use the variable sip_request_user1 outside if. What can be done?

Regards,
Archana

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From spencer at 5ninesolutions.com  Mon Nov 26 19:42:37 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 26 Nov 2012 08:42:37 -0800
Subject: [Freeswitch-users] SIP to TDM t38 gateway
In-Reply-To: <50AE5F05.1010402@integrafin.co.uk>
References: <5BA9AADC-BB63-4FF5-B1A6-1DEC1BE3E931@5ninesolutions.com>
	<50AE5F05.1010402@integrafin.co.uk>
Message-ID: <F35B807F-7D80-4643-80BF-EAA97F9FC74F@5ninesolutions.com>

Hi Alex,
Thanks for your help.

The problem is I need to detect the fax tones in the audio stream and then send a ReINVITE.  I'm attempting to do this like this:

    <extension name="dial-pbx"> 
      <condition field="destination_number" expression="^(.*)$"> 
        <action application="set" data="fax_enable_t38=true"/> 
        <action application="set" data="fax_enable_t38_request=true"/> 
        <action application="set" data="execute_on_answer=t38_gateway self"/> 
        <action application="bridge" data="freetdm/1/a/$1"/> 
      </condition> 
    </extension>

The topology is like this:
Incoming Call:
ITSP ----SIP---> Freeswitch -----ISDN PRI----> Legacy PBX ----> Fax Machine
In this case I need to detect the fax and send a ReINVITE.
The media bug gets created as soon as the channel is answered but immediately gets destroyed. The fax tones are never detected and no reinvite is sent.


Outgoing call
Fax Machine ----> Legacy PBX ---ISDN PRI----> Freeswitch ---SIP---> ITSP
The ITSP sends a ReINVITE and when using sip_execute_on_image t38_gateway is working correctly.

See:
http://jira.freeswitch.org/browse/FS-4789
http://pastebin.freeswitch.org/20162

Were you able to get this working?

Thanks for your help!
Spencer


On Nov 22, 2012, at 9:21 AM, Alex Crow wrote:

> Hi Spencer,
> 
> I use this (gatewaying via a Mitel 3300 but there is no reason why it 
> would not work for FreeTDM, just change the bridge at the end).
> 
> 
>       <action application="set" data="absolute_codec_string=PCMU"/>
>       <action application="set" data="fax_enable_t38=true"/>
> <action application="set" data="sip_execute_on_image=t38_gateway self 
> nocng"/>
>       <action application="bridge" data="sofia/gateway/10.10.0.2/$1"/>
> 
> 
> Hope this helps if a bit late.
> 
> Cheers
> 
> Alex
> 
> On 05/11/12 21:32, Spencer Thomason wrote:
>> Hello,
>> I'm trying to use Freeswitch as a SIP to TDM gateway.  I'd like to use t38_gateway to detect fax tones and send a ReINVITE to t38.
>> 
>> Have a very minimal config with one profile that simply relays to FreeTDM
>> 
>> My dialplan is:
>> <include>
>>   <context name="sip">
>> 
>>     <extension name="dial-trunk">
>>       <condition field="destination_number" expression="^(.*)$">
>>         <action application="set" data="fax_enable_t38=true"/>
>>         <action application="set" data="fax_enable_t38_request=true"/>
>>         <action application="set" data="execute_on_answer=t38_gateway self"/>
>>         <action application="bridge" data="freetdm/1/a/$1"/>
>>       </condition>
>>     </extension>
>> 
>>   </context>
>> </include>
>> 
>> The problem is a media bug is created on the channel but almost immediately destroyed so fax tones are never detected.
>> 
>> See:
>> http://pastebin.freeswitch.org/20162
>> 
>> Thanks for any assistance,
>> Spencer
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From msc at freeswitch.org  Mon Nov 26 19:52:05 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 08:52:05 -0800
Subject: [Freeswitch-users] digits not able to announce in IVR
In-Reply-To: <CA+XvL-MZRbCd5a7AvyZ8Ft-eA6RedGKAweg9suALwhgWQxuiug@mail.gmail.com>
References: <CA+XvL-MZRbCd5a7AvyZ8Ft-eA6RedGKAweg9suALwhgWQxuiug@mail.gmail.com>
Message-ID: <CAKzWOxUKF2+6cUqOZQvtuvX33z9dy_4oCtGR2+Y7bVBwccC4tg@mail.gmail.com>

I've never tried calling the say method of the session object directly. I
personally use this syntax:
session:execute("say","en number pronounced " .. digits)

Try that and let us know if you get a different result. We'll take it from
there.
-MC

On Wed, Nov 21, 2012 at 11:24 PM, ram <talk2ram at gmail.com> wrote:

> Hi all
>
> Iam trying to read the number entered and announce back using lua script
>
> but i get following error..any suggestions
>
>
>    1. 2012-11-22 12:26:50.850983 [DEBUG] switch_ivr_play_say.c:1309 Codec
>    Activated L16 at 8000hz 1 channels 20ms
>    2. 2012-11-22 12:26:54.090983 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    8:800
>    3. 2012-11-22 12:26:54.690982 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    1:800
>    4. 2012-11-22 12:26:55.110992 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    4:800
>    5. 2012-11-22 12:26:55.890982 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    3:800
>    6. 2012-11-22 12:26:56.550979 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    7:640
>    7. 2012-11-22 12:26:57.190981 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    5:640
>    8. 2012-11-22 12:26:57.830980 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    0:640
>    9. 2012-11-22 12:26:58.330979 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    7:800
>    10. 2012-11-22 12:26:58.630979 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    4:640
>    11. 2012-11-22 12:26:59.190980 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    6:640
>    12. 2012-11-22 12:26:59.750982 [DEBUG] switch_rtp.c:3798 RTP RECV DTMF
>    #:640
>    13. 2012-11-22 12:26:59.750982 [DEBUG] switch_ivr_play_say.c:2034 Test
>    Regex [8143750746][\d+]
>    14. 2012-11-22 12:26:59.750982 [INFO] switch_cpp.cpp:1227 Announcing
>    number: 8143750746
>    15. 2012-11-22 12:26:59.750982 [DEBUG] switch_ivr.c:2883 No language
>    specified - Using [en]
>    16. 2012-11-22 12:26:59.750982 [ERR] mod_say_en.c:130 Parse Error!
>    17.
>    18.
>    my LUA script.
>
>    19.
>    20. local announce_number = function(number)
>    21.     log("Announcing number: " .. number)
>    22.     s:say(number, "en", "number", "pronounced")
>    23.     s:streamFile("phone-confirm-re-enter.wav")
>    24. end
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 26 20:00:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 09:00:28 -0800
Subject: [Freeswitch-users] mod_perl vs mod_xml_curl
In-Reply-To: <CANM4Ruj+5f9HOdaVz+iA6A2Hx=fnC+Zsf77sRfN_C4+whjboPQ@mail.gmail.com>
References: <CANM4Ruj+5f9HOdaVz+iA6A2Hx=fnC+Zsf77sRfN_C4+whjboPQ@mail.gmail.com>
Message-ID: <CAKzWOxUP_OU+=75C+EBMTghF8PLR53ZdtGvSb3gzUP6yidMnFw@mail.gmail.com>

Did the console show any errors or warnings when using the mod_perl method?
Were you able to confirm that the directory.pl script actually was run?
-MC

On Thu, Nov 22, 2012 at 2:59 AM, Eugene Prokopiev <enp at itx.ru> wrote:

> Hi,
>
> I tried to to authenticate users from external source with mod_perl or
> mod_xml_curl. Configuration and code looks like:
>
> mod_perl:
>
> <configuration name="perl.conf">
>   <settings>
>     <param name="xml-handler-script" value="/var/lib/freeswitch/scripts/
> directory.pl"/>
>     <param name="xml-handler-bindings" value="directory"/>
>   </settings>
> </configuration>
>
> $XML_STRING = '
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="$${local_address}">
>       <params>
>         <param name="dial-string" value="${sofia_contact(${dialed_user}@
> ${dialed_domain})}"/>
>       </params>
>       <groups>
>         <group name="local">
>           <users>
>             <user id="123"><params><param name="password"
> value="123"/></params></user>
>            </users>
>         </group>
>       </groups>
>     </domain>
>   </section>
> </document>
> ';
>
> mod_xml_curl:
>
> <configuration name="xml_curl.conf">
>   <bindings>
>     <binding name="directory">
>       <param name="gateway-url" value="http://localhost:5000"
> bindings="directory"/>
>     </binding>
>   </bindings>
> </configuration>
>
> use Mojolicious::Lite;
> post '/' => 'index';
> app->start;
> __DATA__
> @@ index.html.ep
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="$${local_address}">
>       <params>
>         <param name="dial-string" value="${sofia_contact(${dialed_user}@
> ${dialed_domain})}"/>
>       </params>
>       <groups>
>         <group name="local">
>           <users>
>             <user id="123"><params><param name="password"
> value="123"/></params></user>
>            </users>
>         </group>
>       </groups>
>     </domain>
>   </section>
> </document>
>
> Authentication via mod_xml_curl was succeeded, authentication via
> mod_perl was failed. What is wring with it?
>
> --
> Regards,
> Eugene Prokopiev
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 26 20:03:56 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 09:03:56 -0800
Subject: [Freeswitch-users] changing vm default announcement
In-Reply-To: <50AEC4A5.50907@gmail.com>
References: <50AEC4A5.50907@gmail.com>
Message-ID: <CAKzWOxWO=GEmdZcjDWJRUYR9Tf52410SHPXfWLFadOR_XtoWqw@mail.gmail.com>

Excellent. I just confirmed a bunch of wiki account requests so you should
now be good to go.
-MC

On Thu, Nov 22, 2012 at 4:34 PM, andy <packetandy at gmail.com> wrote:

> Michael,
>
> I verified that "export" works with loopback. I will add to the wiki as
> soon as I get a login
>
> cheers
> andy
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From NuwanW at unifybusiness.co.uk  Mon Nov 26 20:17:20 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 26 Nov 2012 17:17:20 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
	<ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D1C7C@BARXB0003.UnifyBusiness.local>

Brian,

Thanks a lot for the reply. I have downloaded FreSwitch last weekend binaries and it works fine. Thank you again.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 26 November 2012 14:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [Confidential] - uuid_broadcast

Are you doing this on the latest FreeSWTICH?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 8:13 AM, Muhammad Shahzad <shaheryarkh at gmail.com> wrote:

> Seems like a bug in FS. Can you open a JIRA item for this?
>
> As a work around, can try to call uuid_broadcast with playback application, see if that works, e.g.
>
> uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::hello-kitty.wav both
>
> See this url for more info,
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
>
> Thank you.


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




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.


From xyangni at gmail.com  Mon Nov 26 20:29:34 2012
From: xyangni at gmail.com (Yihui Li)
Date: Mon, 26 Nov 2012 17:29:34 +0000
Subject: [Freeswitch-users] NAT thread error and can not reg to external
	sip server
In-Reply-To: <CAO1DH1-Td8B1t4pAgUH8rTRq8wJMAByCmJX07Qb6Z6U+Ke0Z+A@mail.gmail.com>
References: <CAO1DH1-Td8B1t4pAgUH8rTRq8wJMAByCmJX07Qb6Z6U+Ke0Z+A@mail.gmail.com>
Message-ID: <CAO1DH1_Ajai2zKVs6UyLqJXOa+sit5d3gd3yK41fcnK1ox72uA@mail.gmail.com>

tried the weekly build here, it got the same problem on all my WinXP PC
http://files-sync.freeswitch.org/windows_installer/installer/x86/

My Router is BiPAC 7800N. is this caused this problem?

On Mon, Nov 26, 2012 at 11:37 AM, Yihui Li <xyangni at gmail.com> wrote:

> Hi,
> I have downloaded the most recent git head and build it with VC2010
> express today. But fs start up with some nat error and can not register to
> external sip server showing "error 408 request time out". what is wrong
> with it? Thanks.
>
> Below is part of the log file
> D:\Dev\fsgit\Win32\Debug>FreeSwitchConsole.exe
> 2012-11-26 11:13:59.312500 [INFO] switch_event.c:589 Activate Eventing
> Engine.
> 2012-11-26 11:13:59.343750 [WARNING] switch_event.c:563 Create additional
> event dispa
> tch thread 0
> 2012-11-26 11:13:59.468750 [INFO] switch_nat.c:420 Scanning for NAT
> 2012-11-26 11:13:59.468750 [ERR] switch_nat.c:201 Error checking for PMP
> [init failed
> ]
> 2012-11-26 11:13:59.468750 [DEBUG] switch_nat.c:425 Checking for UPnP
> 2012-11-26 11:14:02.484375 [INFO] switch_nat.c:434 NAT detected type:
> upnp, ExtIP: '9
> 5.172.xxxxxx'
> 2012-11-26 11:14:02.484375 [ERR] switch_nat.c:258 Bind Error
> 2012-11-26 11:14:02.484375 [ERR] switch_nat.c:365 Unable to initialize NAT
> thread
>
>
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From msc at freeswitch.org  Mon Nov 26 20:45:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 09:45:41 -0800
Subject: [Freeswitch-users] Got HTTP Error 0 posting to web server
In-Reply-To: <CAPvcmJhwURqS85hs15Ds8hsgRRDam4JiY=FuW-tT12ApWt0L4Q@mail.gmail.com>
References: <CAPvcmJhwURqS85hs15Ds8hsgRRDam4JiY=FuW-tT12ApWt0L4Q@mail.gmail.com>
Message-ID: <CAKzWOxVXt2WnXYGOXhNg2tTLLUEv2oL4UVtvpfn0c3vU99Zd1w@mail.gmail.com>

Does the web server log anything unusual in this case? Does it even see the
web request when you get an error?
-MC

On Sun, Nov 25, 2012 at 9:14 AM, Milos Jovanovic <
development.milos at gmail.com> wrote:

> Hello,
>
> I am getting these errors:
>
> 2012-11-25 01:19:51.857988 [ERR] mod_xml_cdr.c:367 Got error [0] posting
> to web server [http://192.168.1.5/interface/fs_cdr]
>
> It happens ONLY when there are a lot of simultaneous calls. Any ideas how
> to deal with it?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 26 20:59:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 09:59:17 -0800
Subject: [Freeswitch-users] NAT thread error and can not reg to external
 sip server
In-Reply-To: <CAO1DH1-Td8B1t4pAgUH8rTRq8wJMAByCmJX07Qb6Z6U+Ke0Z+A@mail.gmail.com>
References: <CAO1DH1-Td8B1t4pAgUH8rTRq8wJMAByCmJX07Qb6Z6U+Ke0Z+A@mail.gmail.com>
Message-ID: <CAKzWOxVUKdeO+tzmhkEc6t4QX5hPw0LrqxHUehBZn3D=jNN=Vg@mail.gmail.com>

Do you necessarily require the auto NAT stuff? Perhaps you can try with
FreeSwitchConsole.exe -nonat and then manually configure your router not to
do anything silly with SIP traffic. Make sure you turn off any SIP ALG. The
wiki has a list of some of the more popular routers and how to turn off the
ALG:
http://wiki.freeswitch.org/wiki/Alg

Hope this helps.
-MC

On Mon, Nov 26, 2012 at 3:37 AM, Yihui Li <xyangni at gmail.com> wrote:

> Hi,
> I have downloaded the most recent git head and build it with VC2010
> express today. But fs start up with some nat error and can not register to
> external sip server showing "error 408 request time out". what is wrong
> with it? Thanks.
>
> Below is part of the log file
> D:\Dev\fsgit\Win32\Debug>FreeSwitchConsole.exe
> 2012-11-26 11:13:59.312500 [INFO] switch_event.c:589 Activate Eventing
> Engine.
> 2012-11-26 11:13:59.343750 [WARNING] switch_event.c:563 Create additional
> event dispa
> tch thread 0
> 2012-11-26 11:13:59.468750 [INFO] switch_nat.c:420 Scanning for NAT
> 2012-11-26 11:13:59.468750 [ERR] switch_nat.c:201 Error checking for PMP
> [init failed
> ]
> 2012-11-26 11:13:59.468750 [DEBUG] switch_nat.c:425 Checking for UPnP
> 2012-11-26 11:14:02.484375 [INFO] switch_nat.c:434 NAT detected type:
> upnp, ExtIP: '9
> 5.172.xxxxxx'
> 2012-11-26 11:14:02.484375 [ERR] switch_nat.c:258 Bind Error
> 2012-11-26 11:14:02.484375 [ERR] switch_nat.c:365 Unable to initialize NAT
> thread
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 26 21:08:50 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 10:08:50 -0800
Subject: [Freeswitch-users] lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxXDYpfVXJSK4Wm8j0LjVNtbdyQ7t5of+1XAa38=Me2a-w@mail.gmail.com>

I'm assuming you need to remove the "local" directive there.
-MC

On Mon, Nov 26, 2012 at 8:29 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> Does anyone know how can we use a variable declared inside if statement
> outside if block?****
>
> ** **
>
> if sip_request_user == nil then****
>
> sip_request_user =  params:getHeader("sip_request_user")****
>
> else****
>
> local sip_request_user1 = tonumber(sip_request_user)****
>
> end****
>
> ** **
>
> In my lua I have something like above and I want to use the variable
> sip_request_user1 outside if. What can be done?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Nov 26 21:09:42 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Nov 2012 10:09:42 -0800
Subject: [Freeswitch-users] NAT thread error and can not reg to external
 sip server
In-Reply-To: <CAO1DH1_Ajai2zKVs6UyLqJXOa+sit5d3gd3yK41fcnK1ox72uA@mail.gmail.com>
References: <CAO1DH1-Td8B1t4pAgUH8rTRq8wJMAByCmJX07Qb6Z6U+Ke0Z+A@mail.gmail.com>
	<CAO1DH1_Ajai2zKVs6UyLqJXOa+sit5d3gd3yK41fcnK1ox72uA@mail.gmail.com>
Message-ID: <CAKzWOxV=24J7gO6O+g9MJspHqnfcw2HDKt3kd=DsKB_LKmMOcQ@mail.gmail.com>

It very well could be the problem. The only way to know for sure is to
watch the traffic on both sides of the router, or try a different router.
-MC

On Mon, Nov 26, 2012 at 9:29 AM, Yihui Li <xyangni at gmail.com> wrote:

> tried the weekly build here, it got the same problem on all my WinXP PC
> http://files-sync.freeswitch.org/windows_installer/installer/x86/
>
> My Router is BiPAC 7800N. is this caused this problem?
>
> On Mon, Nov 26, 2012 at 11:37 AM, Yihui Li <xyangni at gmail.com> wrote:
>
>> Hi,
>> I have downloaded the most recent git head and build it with VC2010
>> express today. But fs start up with some nat error and can not register to
>> external sip server showing "error 408 request time out". what is wrong
>> with it? Thanks.
>>
>> Below is part of the log file
>> D:\Dev\fsgit\Win32\Debug>FreeSwitchConsole.exe
>> 2012-11-26 11:13:59.312500 [INFO] switch_event.c:589 Activate Eventing
>> Engine.
>> 2012-11-26 11:13:59.343750 [WARNING] switch_event.c:563 Create additional
>> event dispa
>> tch thread 0
>> 2012-11-26 11:13:59.468750 [INFO] switch_nat.c:420 Scanning for NAT
>> 2012-11-26 11:13:59.468750 [ERR] switch_nat.c:201 Error checking for PMP
>> [init failed
>> ]
>> 2012-11-26 11:13:59.468750 [DEBUG] switch_nat.c:425 Checking for UPnP
>> 2012-11-26 11:14:02.484375 [INFO] switch_nat.c:434 NAT detected type:
>> upnp, ExtIP: '9
>> 5.172.xxxxxx'
>> 2012-11-26 11:14:02.484375 [ERR] switch_nat.c:258 Bind Error
>> 2012-11-26 11:14:02.484375 [ERR] switch_nat.c:365 Unable to initialize
>> NAT thread
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Mon Nov 26 21:53:25 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Nov 2012 18:53:25 +0000
Subject: [Freeswitch-users] lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
Message-ID: <CAFiqYuk-jmKStdmAjtH1COf4q4g3gNeqjLnT_neG11+5cCDEzg@mail.gmail.com>

Try removing local to adjust the scope of the variable, and/or defining it
earlier.

On 26 November 2012 16:29, Archana Venugopan <a.venugopan at mundio.com> wrote:

> local sip_request_user1 = tonumber(sip_request_user)****
>
>
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From blee at gocentrix.com  Mon Nov 26 22:00:11 2012
From: blee at gocentrix.com (Bryant Lee)
Date: Mon, 26 Nov 2012 14:00:11 -0500
Subject: [Freeswitch-users] lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
Message-ID: <CAGodFm9Nmpv1i_cbSk-Y2Xuo=hgCYzkPZW=WAz5_bspnXd4c2w@mail.gmail.com>

Remove the local keyword from the variable declaration on sip_request_user1.

On Mon, Nov 26, 2012 at 11:29 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> Does anyone know how can we use a variable declared inside if statement
> outside if block?****
>
> ** **
>
> if sip_request_user == nil then****
>
> sip_request_user =  params:getHeader("sip_request_user")****
>
> else****
>
> local sip_request_user1 = tonumber(sip_request_user)****
>
> end****
>
> ** **
>
> In my lua I have something like above and I want to use the variable
> sip_request_user1 outside if. What can be done?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chad at apartmentlines.com  Mon Nov 26 22:15:56 2012
From: chad at apartmentlines.com (Chad Phillips)
Date: Mon, 26 Nov 2012 11:15:56 -0800
Subject: [Freeswitch-users] lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
Message-ID: <3D6C6A3F10DC4620B47E90EE1EF41331@gmail.com>

The 'local' keyword makes the variable local to the code block it's in. Simply redeclare that variable in the scope you need it available. 

hunmonk 


On Monday, November 26, 2012 at 8:29 AM, Archana Venugopan wrote:

> Hi,
>  
> Does anyone know how can we use a variable declared inside if statement outside if block?
>  
> if sip_request_user == nil then
> sip_request_user =  params:getHeader("sip_request_user")
> else
> local sip_request_user1 = tonumber(sip_request_user)
> end
>  
> In my lua I have something like above and I want to use the variable sip_request_user1 outside if. What can be done?
>  
> Regards,
> Archana
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-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
> 
> 


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From shaheryarkh at gmail.com  Mon Nov 26 22:33:19 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Mon, 26 Nov 2012 20:33:19 +0100
Subject: [Freeswitch-users] lua script
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
Message-ID: <CAFZQphze4oCvty3+rUSCjJebSqj36qjb9FYEs1rpktgpcoJtbg@mail.gmail.com>

classical variable scope problem. :-)

http://lua-users.org/wiki/ScopeTutorial

Basically, you should declare variable above and outside IF condition and
initialize it say 'nil', then change its value in IF per logic, finally
outside below IF block check if variable value is other then 'nil', if so
then do your operation. This is very basic programming man!

Thank you.


On Mon, Nov 26, 2012 at 5:29 PM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> Does anyone know how can we use a variable declared inside if statement
> outside if block?****
>
> ** **
>
> if sip_request_user == nil then****
>
> sip_request_user =  params:getHeader("sip_request_user")****
>
> else****
>
> local sip_request_user1 = tonumber(sip_request_user)****
>
> end****
>
> ** **
>
> In my lua I have something like above and I want to use the variable
> sip_request_user1 outside if. What can be done?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From marketing at cluecon.com  Mon Nov 26 22:39:32 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 26 Nov 2012 11:39:32 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxW1uQwkvZN=-oQoYCX63oHQiC31ydSuPiey_=2FkkJnMQ@mail.gmail.com>

Hello all!

We are all back to a full week after many of us enjoyed some well-deserved
time off last week. However, even though there was a holiday here in the
US, the intrepid FreeSWITCH development team was working hard on your
behalf. As Ken Rice previously mentioned, Anthony spotted a potential issue
in the recently released 1.2.5 version. Therefore, this past Saturday they
made 1.2.5.1 available<http://files.freeswitch.org/freeswitch-1.2.5.1.tar.bz2>for
us. Many thanks to those who work so hard to make sure that FreeSWITCH
is running smoothly for us all.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_21>we spent
some time getting everyone up to speed on how to edit the FreeSWITCH
wiki <http://wiki.freeswitch.org>, specifically focusing on channel
variables <http://wiki.freeswitch.org/wiki/Channel_Variables> pages.
Updating documentation is one of the least glamorous aspects of maintaining
an open source project. Many thanks to those who've stepped up over the
past weeks and months to help us out. With the end of the year upon us we
are slowing down a bit in our speaking schedule for the weekly community
conference call. We have a few things in the works but nothing yet
scheduled. On this week's
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_28>we will be
doing a community scrum. Be sure to bring your questions and
topics for discussion. If you have a tip or trick that you'd like to share
with the group that would be most welcomed. If time permits we will
crowdsource a few selected questions from the mailing list.

Have a great week and we'll talk to you on Wednesday.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From sdevoy at bizfocused.com  Mon Nov 26 22:50:42 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 26 Nov 2012 14:50:42 -0500
Subject: [Freeswitch-users] Tricky rollover Dial Plan Question
In-Reply-To: <CAM21H-Mc_MJKn4WYdMEtXcioNrR5TyejYXs1yydf3SWhUj8Q+A@mail.gmail.com>
References: <0c3d01cdcacc$abedcd00$03c96700$@bizfocused.com>
	<CAM21H-Mc_MJKn4WYdMEtXcioNrR5TyejYXs1yydf3SWhUj8Q+A@mail.gmail.com>
Message-ID: <16b501cdcc0f$51d742d0$f585c870$@bizfocused.com>

THANK YOU SO MUCH Ognjen.  The solution is so simple and is exactly what I
needed.    With the amount of reading and research I really have done, it
always surprises me to find the SIMPLEST things that I totally missed out
on.  Sometimes I feel like the well driller trying to figure out where to
drill to fix the leaking dam.

 

Of course, your solution to #1 makes #2 a complete non-issue.  It is just
standard group calling.

 

Thank you for your response.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ognjen
Seslija
Sent: Sunday, November 25, 2012 4:59 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tricky rollover Dial Plan Question

 

1) I guess one person uses that phone. With that in mind, one extension
(both on FS and on the phone) will do. If you need to route different DIDs
you can do that on that some extension (not map DIDs to exts 1-1). You can
just set Line Key 2,3,4 to Ext 1 and they will blink and phone will ring  if
another call comes in, during first call. In this case call waiting needs to
be enabled.

No complicated dialplan is needed, default one will do. If you disable CW,
phone will send 486 response immediately if on call, so you use
continue_on_fail=USER_BUSY.

2) You can use something like 

 

  <action application="bridge"
data="${group_call(${group_name}@${domain_name}+A)}"/>

 

in the dialplan and 

 

     <group name="finance">

        <users>

          <user id="111" type="pointer"/>

          <user id="184" type="pointer"/>

          <user id="119" type="pointer"/>

        </users>

      </group>

 

in the directory.

This is scalable, you can easily set ring/hunt groups this way, just create
and edit them in the directory.

With regards to the buttons, that's really just configuration of the phone.

 

Join irc channel (#freeswitch on irc.freenode.net) if you have additional
questions.

 

On Sun, Nov 25, 2012 at 6:21 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Hi All,

 

Thanks for your help as always.

 

I have a CISCO 504G four line phone.  I don't think the model is
particularly relevant, just that it is a multi-line phone.

 

Issue one:

If line 1 is busy, I want new calls to be directed to line 2. If 1 and 2 are
in use, ring on 3 and then line 4.  I think this means call waiting will
need to be disabled so that each will ring busy, enabling rollover.  Also, I
don't want lines 2, 3 and 4 to all ring if 1 is busy.

 

I think I just use continue_on_fail=true and hangup_after_bridge=true, then
route to Line 1, then 2, 3 and 4.

 

Issue two:

I have 6 phones in a call group and I need the call group to ring each phone
like explained in issue one. I cannot for the life of me figure out how to
"layer" devices to failover independently when dialed in a group.

 

That is to say, when a call comes in it should be bridged to all 6 phones on
the lowest available line button.  Whoever answers first gets to call and
all other lines return to ready state.

 

Any thoughts would be greatly appreciated.

 

Sean

 


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




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

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

 

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From brian at freeswitch.org  Tue Nov 27 00:26:12 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Nov 2012 15:26:12 -0600
Subject: [Freeswitch-users] ATTENTION ALL (please read and perform the task
	if it applies to you.)
Message-ID: <AC78D8BC-801A-415F-8C30-8C9B5A32999D@freeswitch.org>

If you currently have something setup to register to conference.freeswitch.org please REMOVE IT or disable registration till further notice. 

Thanks,
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543







From chad at apartmentlines.com  Tue Nov 27 00:58:24 2012
From: chad at apartmentlines.com (Chad Phillips)
Date: Mon, 26 Nov 2012 13:58:24 -0800
Subject: [Freeswitch-users] lua script
In-Reply-To: <CAGodFm9Nmpv1i_cbSk-Y2Xuo=hgCYzkPZW=WAz5_bspnXd4c2w@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2337CA1@Mail-Kilo.squay.com>
	<CAGodFm9Nmpv1i_cbSk-Y2Xuo=hgCYzkPZW=WAz5_bspnXd4c2w@mail.gmail.com>
Message-ID: <F4DDE7DBEEE341AC9C246909D0AB27CA@gmail.com>

I would advise leaving the local keyword. Local variable lookups are faster in Lua, and more importantly it's best to restrict the variable's scope to the smallest scope you'll use it in. Even if you're using the variable at the scope of an entire script, you should declare it local to the script -- global variables are accessed beyond the actual script they are in, as illustrated by the following example: 

test1.lua:

require "test2"

print(string.format("foo in test1 is: %s", tostring(foo)))
print(string.format("baz in test1 is: %s", tostring(baz)))


test2.lua:

foo = "bar"
local baz = "bang"

print(string.format("foo in test2 is: %s", tostring(foo)))
print(string.format("baz in test2 is: %s", tostring(baz)))


Output:

foo in test2 is: bar
baz in test2 is: bang
foo in test1 is: bar
baz in test1 is: nil


Hope this helps.

hunmonk 


On Monday, November 26, 2012 at 11:00 AM, Bryant Lee wrote:

> Remove the local keyword from the variable declaration on sip_request_user1.
> 
> On Mon, Nov 26, 2012 at 11:29 AM, Archana Venugopan <a.venugopan at mundio.com (mailto:a.venugopan at mundio.com)> wrote:
> > Hi, 
> >  
> > Does anyone know how can we use a variable declared inside if statement outside if block?
> >  
> > if sip_request_user == nil then
> > sip_request_user =  params:getHeader("sip_request_user")
> > else
> > local sip_request_user1 = tonumber(sip_request_user)
> > end
> >  
> > In my lua I have something like above and I want to use the variable sip_request_user1 outside if. What can be done?
> >  
> > Regards,
> > Archana
> >  
> > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-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
> > 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-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
> 
> 


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From fs-list at communicatefreely.net  Tue Nov 27 05:25:30 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 26 Nov 2012 21:25:30 -0500
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
 again
In-Reply-To: <CCD7A44C.610F0%krice@freeswitch.org>
References: <CCD7A44C.610F0%krice@freeswitch.org>
Message-ID: <50B4249A.30702@communicatefreely.net>

Thanks for all the input everyone!

I just had it happen again, this time at night when there were only 2 sessions active!

I was in doing some other work, so I managed to see it happen and found a few interesting 
things.

CPU load next to nothing, load average around 0.12.
I used iSQL to test ODBC connectivity, and I could read and write to the freeswitch 
database using the same DSN that freeswitch is using.

I could delete a SIP registration from the API

Since I had things on a higher debug level, I saw this:

freeswitch at stefan> 2012-11-26 21:21:51.914082 [CONSOLE] sofia.c:1144 MSG Thread Started

As soon as that came up, everything started working again.

What's that about?

-Tim

Ken Rice wrote:
> Sofia is not single threaded except for in one spot deep in libsofia,
>>From there, messages are handed off to a number of message queues for FS
> core to handle as needed...
> 
> Check to see if anything that fs is depending on is blocking on info
> retrieval like the databases or other areas...
> 
> K
> 
> On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com> wrote:
> 
>> you mentioned that you use xml_curl, if your web server hangs it may
>> hang sofia, iirc sofia is running a single thread and it will wait for
>> the xml_curl response before continuing to the next request.
>>
>> On 11/23/2012 1:13 PM, Steven Ayre wrote:
>>> Any kind of DB backup running? Or any long-running queries (innotop is
>>> great for highlighting queries that've been running a while, including
>>> on non-innodb tables).
>>>
>>> A global read lock, or queries waiting for a lock could block a db
>>> update from the sofia profile thread but still allow read-only queries
>>> (sofia status) to run.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 23 November 2012 15:32, Tim St. Pierre <fs-list at communicatefreely.net>
>>> wrote:
>>>> Hi Steven,
>>>>
>>>> Thanks for the suggestions.  I'm hoping once I get the upgrade done it will
>>>> all go away.
>>>> I have watched it happen at least once.  I was on the phone at the time.
>>>> Console activity
>>>> more or less stopped, except for a few calls hanging up.  The console
>>>> remains responsive,
>>>> and my call wasn't dropped for at least a minute or two (media timeout?).  I
>>>> was able to
>>>> run sofia status and other commands that use the database, so I'm assuming
>>>> that the
>>>> connection was still working.  All our media is runs through the box, so I
>>>> think things
>>>> are fine on the Ethernet level.  I do see higher load averages - maybe 3-4,
>>>> but that's the
>>>> only obvious indication.  It's not taking CPU beyond 10% or so.
>>>>
>>>> We are using MySQL as the core DB and also as the DB backend for each sofia
>>>> profile.  This
>>>> is connecting through ODBC of course.
>>>>
>>>> If I can get the other kinks worked out, then I will try 1.2 stable in
>>>> production and
>>>> we'll see how it goes.
>>>>
>>>> -Tim
>>>>
>>>> Steven Ayre wrote:
>>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>>>
>>>>>
>>>>> As you've already acknowledged it's a very old version.
>>>>>
>>>>> It's possible that your issue has already been found and fixed, but if
>>>>> it hasn't then the code will have changed significantly since then and
>>>>> you'd really need to reproduce it on the latest code for it to be
>>>>> investigated.
>>>>>
>>>>>
>>>>> As some general thoughts though, are you able to spot it happening while
>>>>> it's happening or only afterwards?
>>>>>
>>>>> If you're able to get on the system during one of those times look at
>>>>> what else is happening. Is the load average/cpu usage/io high? Perhaps
>>>>> something's running that's blocking all access or causing very high IO.
>>>>>
>>>>> What DB backend are you using for Sofia? Is it possible that that's
>>>>> hanging for a moment? For example if you're running a backup on the DB
>>>>> that blocks all writes to the DB while Sofia is trying to update the DB
>>>>> that perhaps would cause this.
>>>>>
>>>>> Try running a SIP OPTIONS ping your your sofia profile from the
>>>>> localhost during that time, which should exclude it being any issue on
>>>>> the ethernet.
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 22 November 2012 19:31, Tim St. Pierre <fs-list at communicatefreely.net
>>>>> <mailto:fs-list at communicatefreely.net>> wrote:
>>>>>
>>>>>      Hello,
>>>>>
>>>>>      I'm having a bit of an odd problem.
>>>>>
>>>>>      Intermittently, often every 2-3 days or so, Freeswitch stops
>>>>>      replying to SIP for about 5
>>>>>      minutes.  I can't verify if it's EXACTLY 5 minutes, but it seems to
>>>>>      be pretty close.
>>>>>
>>>>>      During this time, no new registrations or invites can happen, but
>>>>>      existing calls stay
>>>>>      connected for at least a minute or two.  In the logs, you can see
>>>>>      calls slowly hanging up
>>>>>      with "NORMAL_CLEARING".  In 5 minutes, everything starts up again
>>>>>      with no word about it at
>>>>>      all in the logs.
>>>>>
>>>>>      When calls resume, I notice that the number of sessions returned by
>>>>>      the status command is
>>>>>      one higher than the actual number sessions returned by show
>>>>>      channels, or by looking in the
>>>>>      database.  Every time this happens, the discrepancy increases by one.
>>>>>
>>>>>      The interruption happens on all SIP profiles, but calls originated
>>>>>      from the socket API
>>>>>      still work, insofar as they return with PROGRESS_TIMEOUT since the
>>>>>      profiles are still
>>>>>      running, but stuck.
>>>>>
>>>>>      We are using ODBC/MySQL for the core database, and the database
>>>>>      server only runs this
>>>>>      database and some basic PHP/xml-curl stuff.
>>>>>
>>>>>      We have 416 endpoints registered, and usually sit at about 30
>>>>>      sessions during the day.
>>>>>
>>>>>      This never happens at night, only during busier times, but not
>>>>>      necessarily busy hour.
>>>>>
>>>>>      I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
>>>>>
>>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>>>
>>>>>      Yes, I know it's old and I'm trying to upgrade, but I'm still having
>>>>>      some problems getting
>>>>>      all my phones to work properly with 1.2 stable.  This is a
>>>>>      production system, so I can't
>>>>>      just blindly put out the newest release.  Mostly, I need to buy
>>>>>      myself some time so that I
>>>>>      can get the kinks worked out of the latest version and then upgrade
>>>>>      the production box.
>>>>>
>>>>>      I'm grateful for any insights as to what could be happening, even if
>>>>>      a solution is just a
>>>>>      temporary workaround.
>>>>>
>>>>>      Thanks!
>>>>>
>>>>>      -Tim
>>>>>
>>>>>      
>>>>> _________________________________________________________________________
>>>>>      Professional FreeSWITCH Consulting Services:
>>>>>      consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>      http://www.freeswitchsolutions.com
>>>>>
>>>>>      
>>>>>      
>>>>>
>>>>>      Official FreeSWITCH Sites
>>>>>      http://www.freeswitch.org
>>>>>      http://wiki.freeswitch.org
>>>>>      http://www.cluecon.com
>>>>>
>>>>>      FreeSWITCH-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
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 



From NuwanW at unifybusiness.co.uk  Tue Nov 27 13:59:48 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Tue, 27 Nov 2012 10:59:48 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695EA82D1C7C@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
	<ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>
	<78990CE7CC964442A7C2CA5F4689695EA82D1C7C@BARXB0003.UnifyBusiness.local>
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D1E1A@BARXB0003.UnifyBusiness.local>

Brian,

Thank you for your reply. After further testing I have noticed the following. (FreeSwitch last week binary, Version 1.3.6)

1. About 1% of uuid_broadcast requests, FreeSwitch broadcasts audio on both channels at the same time.
2. Usually the first uuid_broadcase request works fine.
3. Second requests onwards it broadcast audio on one leg, wait till it finish, then other leg.
4. Restarting FreeSwitch sometimes fix the issue. However as usual only the first uuid_broadcast works fine.

Could you please advice as I'm not sure the way forward at this stage.

Thank you,
Nuwan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: 26 November 2012 17:17
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [Confidential] - uuid_broadcast

Brian,

Thanks a lot for the reply. I have downloaded FreSwitch last weekend binaries and it works fine. Thank you again.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 26 November 2012 14:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [Confidential] - uuid_broadcast

Are you doing this on the latest FreeSWTICH?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 8:13 AM, Muhammad Shahzad <shaheryarkh at gmail.com> wrote:

> Seems like a bug in FS. Can you open a JIRA item for this?
>
> As a work around, can try to call uuid_broadcast with playback application, see if that works, e.g.
>
> uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::hello-kitty.wav both
>
> See this url for more info,
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
>
> Thank you.


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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.

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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.


From ben122uk at gmail.com  Tue Nov 27 14:06:18 2012
From: ben122uk at gmail.com (Ben N)
Date: Tue, 27 Nov 2012 11:06:18 +0000
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
Message-ID: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>

Hi All,

I'm wondering if anyone out there is using G729 with 50ms ptime through
Freeswitch, without proxy media?

I am unable to get this to work, even in a very basic lab with just two
clients and an up to date FS server on the same LAN.  I can get G729 with
20ms working, but 50ms causes Freeswitch to hang the call.

It might be that mod_g729 and mod_com_g729 simply don't support it, so if
anyone in the know can tell me that would be great!

Further info can be supplied, I believe I have tried pretty much all
options....

Cheers,

Ben
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From gopalakrishnan.an at gmail.com  Tue Nov 27 19:52:12 2012
From: gopalakrishnan.an at gmail.com (Gopalakrishnan N)
Date: Tue, 27 Nov 2012 22:22:12 +0530
Subject: [Freeswitch-users] FreeTDM ftmod pri tap error
In-Reply-To: <CAEymo12bugji+6UbV05FshCLAVBnJbOXpUj4gH1GyA0S_qa=Wg@mail.gmail.com>
References: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
	<CAEymo12bugji+6UbV05FshCLAVBnJbOXpUj4gH1GyA0S_qa=Wg@mail.gmail.com>
Message-ID: <CAEymo133NELNUgy1jwUbtEQwH_gznq6M2v=cQNaJ9Lmj9woH0w@mail.gmail.com>

Has somebody used pritap module?


On Mon, Nov 26, 2012 at 7:13 PM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> Even I tried the moy tap1.4 complete patch with libpri, when i try to
> execute with this ./configure --with-libpri --with-pritap
> --prefix=/usr/local/freeswitch/ in make I am getting error like,
>
> make: *** [ftmod_libpri_la-ftmod_libpri.lo] Error 1
>
> Regards.
>
>
> On Mon, Nov 26, 2012 at 6:21 PM, Gopalakrishnan N <
> gopalakrishnan.an at gmail.com> wrote:
>
>> Hi,
>>
>> Am trying to install ftmod pri tap by following this link
>> http://wiki.freeswitch.org/wiki/FreeTDM#Tapping and enabled
>> --with-pritap while doing ./configure, and while doing make I get error.
>>
>> Can someone help me on this. Do I need to update any patch for tap?
>>
>> My installation log is attached here.
>>
>> Regards.
>> Gopal.
>>
>
>
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From tshepo.maphutha at gmail.com  Tue Nov 27 17:47:33 2012
From: tshepo.maphutha at gmail.com (Tshepo Maphutha)
Date: Tue, 27 Nov 2012 16:47:33 +0200
Subject: [Freeswitch-users] FusionPBX for freeswitch
Message-ID: <CAC_Ms12WJMwvnXD0fzK=Ps+4My32z08hF+hm3YqOkCpGeVL=jQ@mail.gmail.com>

Hi All,

Where can I download FusionPBX GUI for FreeSWITCH.
Any reliable url's will be highly appreciated or any other GUI for
FreeSWITCH recommended
by any of you guys will be highly appreciated

Thanks :)

-- 
Tshepo
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From brian at freeswitch.org  Wed Nov 28 06:54:23 2012
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Nov 2012 21:54:23 -0600
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695EA82D1E1A@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
	<ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>
	<78990CE7CC964442A7C2CA5F4689695EA82D1C7C@BARXB0003.UnifyBusiness.local>
	<78990CE7CC964442A7C2CA5F4689695EA82D1E1A@BARXB0003.UnifyBusiness.local>
Message-ID: <45FB941D-85A3-47EB-84D4-C7D77F052215@freeswitch.org>

Now that jira is back, it would be prudent to open a jira with a test case included.  
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 27, 2012, at 4:59 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:

> Brian,
> 
> Thank you for your reply. After further testing I have noticed the following. (FreeSwitch last week binary, Version 1.3.6)
> 
> 1. About 1% of uuid_broadcast requests, FreeSwitch broadcasts audio on both channels at the same time.
> 2. Usually the first uuid_broadcase request works fine.
> 3. Second requests onwards it broadcast audio on one leg, wait till it finish, then other leg.
> 4. Restarting FreeSwitch sometimes fix the issue. However as usual only the first uuid_broadcast works fine.
> 
> Could you please advice as I'm not sure the way forward at this stage.
> 
> Thank you,
> Nuwan



From ntomer at newgen.co.in  Tue Nov 27 13:21:41 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Tue, 27 Nov 2012 15:51:41 +0530
Subject: [Freeswitch-users] mod_db help needed
In-Reply-To: <CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>
	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>
Message-ID: <012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>

Hi Yiftach,

 

Thanks for your reply. I have installed and configured ODBC on my server. 

 

/etc/odbc.ini has following entries -

 

[freesiwtch-connector]

Description           = MySQL connection to 'freeswitch' database

Driver                = MySQL

Database              = freeswitch

Server                = localhost

UserName              = root

Password              = system123

Port                  = 3306

Socket                = /var/run/mysqld/mysqld.sock

 

After this I did ./configure and "make && make install"

 

Now what all changes I need to make in Freeswitch so that I can make ODBC
work in it? Please help me out.

 

N.B. Today Freeswitch.org is down and therefore I am not able to look for
any help there

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yiftach
Golan
Sent: Friday, November 16, 2012 3:03 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_db help needed

 

Hi Nitin,
Just install odbc on you system, I used this one
(http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/install
ing_configuring_odbc.html) but you can use FreeSWITCH's explanations as well
(http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core)
Once you are done the configure should already know that you have odbc
support, just hit regular configure or as FreeSWITCH docs says : 


Compile FreeSWITCH with ODBC support


UnixODBC support will be autodetected by ./configure, and if found, will be
compiled into FreeSWITCH. 

As for the keys it is pretty easy just go to
(http://wiki.freeswitch.org/wiki/Mod_db) the query should be something like
: 

<action application="set" data="var=${db(select/realm/key)}"/>

The documentation is actually pretty good so try to read it again it will
make sense
Thanks,
Yiftach.

On Thu, Nov 15, 2012 at 4:51 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

I understand that I will have to make changes in db.conf.xml -

 

<configuration name="db.conf" description="LIMIT DB Configuration">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

  </settings>

</configuration>

 

But the wiki also says this - "ODBC must be configured to use ODBC resources
(configure with --enable-core-odbc-support)." What does this mean? I
couldn't find any more references to it.

 

Wiki says - "Realm and key are arbitrary strings. Consider realm as a
container for keys." Does this mean that I can use a realm for a particular
type of data e.g. "callerdata", use caller number as key and set the entered
data against that key. But how would I be able to access this data from
outside FreeSwitch?

 

Is the realm mapped to a table in the database or something similar?

 

Thanks in advance

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Thursday, November 15, 2012 3:30 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_db help needed

 

Hi,

 

I am trying to enter some data (entered by caller, received by Play and Get
Digits). But I am not able to get much information about it from mod_db
wiki. I want to enter data in an Oracle database. Please tell me what all I
need to do, in order to do this.

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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

 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From msc at freeswitch.org  Wed Nov 28 19:34:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Nov 2012 08:34:14 -0800
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695EA82D1E1A@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695EA82D1AC3@BARXB0003.UnifyBusiness.local>
	<CAFZQphzjaZ=Xzj0N2Kmy-s8ipDuzVcuCKe1gqc7UyCw+HMou9Q@mail.gmail.com>
	<ECAA3656-4710-4358-A3D9-C5CFE5C4C5B8@freeswitch.org>
	<78990CE7CC964442A7C2CA5F4689695EA82D1C7C@BARXB0003.UnifyBusiness.local>
	<78990CE7CC964442A7C2CA5F4689695EA82D1E1A@BARXB0003.UnifyBusiness.local>
Message-ID: <CAKzWOxWgWDZLwqsqd06xRULda46pHUmYrtP_mUvH_+MXQ3E-4w@mail.gmail.com>

Open a jira and report all the details.
-MC

On Tue, Nov 27, 2012 at 2:59 AM, Nuwan Wijerathne <
NuwanW at unifybusiness.co.uk> wrote:

> Brian,
>
> Thank you for your reply. After further testing I have noticed the
> following. (FreeSwitch last week binary, Version 1.3.6)
>
> 1. About 1% of uuid_broadcast requests, FreeSwitch broadcasts audio on
> both channels at the same time.
> 2. Usually the first uuid_broadcase request works fine.
> 3. Second requests onwards it broadcast audio on one leg, wait till it
> finish, then other leg.
> 4. Restarting FreeSwitch sometimes fix the issue. However as usual only
> the first uuid_broadcast works fine.
>
> Could you please advice as I'm not sure the way forward at this stage.
>
> Thank you,
> Nuwan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan
> Wijerathne
> Sent: 26 November 2012 17:17
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] [Confidential] - uuid_broadcast
>
> Brian,
>
> Thanks a lot for the reply. I have downloaded FreSwitch last weekend
> binaries and it works fine. Thank you again.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: 26 November 2012 14:48
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] [Confidential] - uuid_broadcast
>
> Are you doing this on the latest FreeSWTICH?
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
> On Nov 26, 2012, at 8:13 AM, Muhammad Shahzad <shaheryarkh at gmail.com>
> wrote:
>
> > Seems like a bug in FS. Can you open a JIRA item for this?
> >
> > As a work around, can try to call uuid_broadcast with playback
> application, see if that works, e.g.
> >
> > uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e
> playback!user_busy::hello-kitty.wav both
> >
> > See this url for more info,
> >
> > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
> >
> > Thank you.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> This e-mail and any attachments are for the intended addressee(s) only and
> may contain confidential and/or privileged material. If you are not a named
> addressee, do not use, retain or disclose such information. This email is
> not guaranteed to be free from viruses and does not bind Unify in any
> contract or obligation. Unify Business Solutions Ltd. Registered in England
> and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way,
> Barlborough, S43 4XA United Kingdom.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> This e-mail and any attachments are for the intended addressee(s) only and
> may contain confidential and/or privileged material. If you are not a named
> addressee, do not use, retain or disclose such information. This email is
> not guaranteed to be free from viruses and does not bind Unify in any
> contract or obligation. Unify Business Solutions Ltd. Registered in England
> and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way,
> Barlborough, S43 4XA United Kingdom.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From sparklezou at 163.com  Wed Nov 28 12:16:28 2012
From: sparklezou at 163.com (sparklezou)
Date: Wed, 28 Nov 2012 17:16:28 +0800
Subject: [Freeswitch-users] About the "transfer" function
Message-ID: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>

Hi All,

When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321".

And on the caller phone will disply "Outbound Call", also show the number "87654321".

I checked the sip message, the info is got from the 180 or 183 message.

Sometime, don't want to show the real external number "87654321" to the enduser.

How to keep the caller phone displaying "1234".

Thanks in advance!

2012-11-28



sparklezou
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From mickstevens at yahoo.com  Wed Nov 28 14:28:50 2012
From: mickstevens at yahoo.com (Mick Stevens)
Date: Wed, 28 Nov 2012 03:28:50 -0800 (PST)
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML
	CDR's?
Message-ID: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>

Hi Folks,

I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example?

I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there.

I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured.

Any help much appreciated!
?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429

Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens
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From richgration at gmail.com  Wed Nov 28 13:54:34 2012
From: richgration at gmail.com (Richard Gration)
Date: Wed, 28 Nov 2012 10:54:34 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
Message-ID: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>

Hi all,

I'm trying to write something to fit into a two factor auth mechanism.
I have a CGI script which is called with a phone number and a PIN as
parameters. In the script I use the event socket api to place a call.
The PHP code I'm using to write to the socket is straight out of the
FS docs (can't remember exactly, I think I got it from the wiki). This
is my api command:

api originate {originate_timeout=30}sofia/external/${dExt}@MGC
'&play_and_get_digits(4 4 2 30000 # torvalds-says-linux.wav
torvalds-says-linux.wav foo \d+)'

This all works fine, up to a point. The call is made, the wav is
played. But then PAGD is supposed to put the digits into a channel
var, in this case "foo". But I've been having trouble getting the
value of this channel var *reliably*. If the # key is used to
terminate the input, then I can get the channel var value with

api uuid_getvar $uuid foo

However, if the input is terminated by PAGD because 4 digits have been
entered, then the call is hung up (by FS/PAGD) before I have a chance
to get the value of the channel var. Can anyone help here please? I
suppose I can work around this by increasing the digits to 5 and
telling people to use # to terminate the input, but I'd like to
understand the correct way to use PAGD if possible.

If you think I'm going about this the wrong way entirely, then please
suggest an alternative :-)

My code is below. As I say, lines 1-71 are straight out of the docs.

  1 <?php
  2
  3  $password = "xxxxxx";
  4  $port = "xxxx";
  5  $host = "127.0.0.1";
  6
  7  function event_socket_create($host, $port, $password) {
  8       $fp = fsockopen($host, $port, $errno, $errdesc)
  9          or die("Connection to $host failed");
 10       socket_set_blocking($fp,false);
 11
 12       if ($fp) {
 13             while (!feof($fp)) {
 14                 $buffer = fgets($fp, 1024);
 15                 usleep(100); //allow time for reponse
 16                 if (trim($buffer) == "Content-Type: auth/request") {
 17                     fputs($fp, "auth $password\n\n");
 18                     break;
 19                 }
 20             }
 21             return $fp;
 22       }
 23       else {
 24             return false;
 25       }
 26  }
 27
 28  function event_socket_request($fp, $cmd) {
 29
 30       if ($fp) {
 31             fputs($fp, $cmd."\n\n");
 32             usleep(200); //allow time for reponse
 33
 34             $response = "";
 35             $i = 0;
 36             $contentlength = 0;
 37             while (!feof($fp)) {
 38                 $buffer = fgets($fp, 4096);
 39                 if ($contentlength > 0) {
 40                     $response .= $buffer;
 41                 }
 42
 43                 if ($contentlength == 0) { //if contentlenght is
already don't process again
 44                      if (strlen(trim($buffer)) > 0) { //run only
if buffer has content
 45                           $temparray = split(":", trim($buffer));
 46                           if ($temparray[0] == "Content-Length") {
 47                               $contentlength = trim($temparray[1]);
 48                           }
 49                      }
 50                 }
 51
 52                 usleep(200); //allow time for reponse
 53
 54                 //optional because of script timeout //don't let
while loop become endless
 55                 if ($i > 90000) { break; }
 56
 57                 if ($contentlength > 0) { //is contentlength set
 58                      //stop reading if all content has been read.
 59                      if (strlen($response) >= $contentlength) {
 60                          break;
 61                      }
 62                 }
 63                 $i++;
 64             }
 65
 66             return $response;
 67       }
 68       else {
 69          echo "no handle";
 70       }
 71  }
 72
 73  function strip_cr($str) {
 74     return preg_replace("/\n/",'',$str);
 75  }
 76
 77  $dExt = $_GET['dExt'];
 78  if ($dExt) {
 79     $fp = event_socket_create($host, $port, $password);
 80
 81     # Place call and find uuid
 82     $cmd = "api originate
{originate_timeout=30}sofia/external/677${dExt}@194.145.191.131
'&play_and_get_digits(4 4 2 30000 # torvalds-says-linux.wav
torvalds-says-linux.wav foo \d+)'";
 83     $response = event_socket_request($fp, $cmd);
 84     $response = preg_replace('/</','&lt;',$response);
 85     $response = preg_replace('/>/','&gt;',$response);
 86     $response = strip_cr($response);
 87     $bits = preg_split('/\s+/',$response);
 88     $uuid = $bits[1];
 89
 90     echo '<pre>';
 91     echo date('H:i:s') . "\n";
 92     echo "call placed: $cmd\n";
 93     echo "response :$response:\n";
 94     echo "uuid $uuid\n";
 95
 96     # Retrieve channel variable
 97     $cmd = "api uuid_getvar $uuid foo";
 98     $response = '';
 99     while (! $response or $response == '_undef_') {
100         echo "in loop : got _undef_\n";
101         $response = event_socket_request($fp, $cmd);
102         $response = strip_cr($response);
103     }
104     echo "response :$response:\n";
105     echo '</pre>';
106
107     # Close socket
108     fclose($fp);
109  } else {
110     echo 'Request URL in this format: /test.php?dExt=<phone_number>' . "\n";
111  }
112
113  ?>


Cheers,
Rich

-- 
Once our basic material needs are met - in my utopia, anyway - life
becomes a perpetual celebration in which everyone has a talent to
contribute. But we cannot levitate ourselves into that blessed
condition by wishing it. We need to brace ourselves for a struggle
against terrifying obstacles, both of our own making and imposed by
the natural world. And the first step is to recover from the delusion
that is positive thinking.
       -- Barbara Ehrenreich


From shahzad.bhatti at g-r-v.com  Wed Nov 28 13:42:10 2012
From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti)
Date: Wed, 28 Nov 2012 15:42:10 +0500
Subject: [Freeswitch-users] use "myevents" make FreeSwitch Crash!
Message-ID: <CAN+NqUa+o-ERwrWdU0o0zPrp=yO9PCgvBhGe-ofGCdPVz1LX-g@mail.gmail.com>

Hi everyone,
I am trying to use *myevents* to subscribe events but on call hang
FREESWITCH Crash.
here is my script

*test.lua*

require "ESL"

hostname        =
"localhost"--'92.63.208.111';--'94.249.139.66'--'localhost';
port            = '8021';
password        = "ClueCon"--'Pak!stan'--'Pakistan';
sock            = ESL.ESLconnection(hostname, port, password);

uuid = arg[1];

e = sock:sendRecv("myevents "..uuid);
while  sock:connected() do
       e = sock:recvEvent();
       event = e:getType();
       print (event.." EVENT Recieved")
end

*call.lua*
*
*
        api = freeswitch.API()
        uuid = api:executeString("create_uuid")

        os.execute("lua test.lua "..uuid.." &")
        os.execute("sleep 1")

session =
freeswitch.Session("{ignore_early_media=true,origination_uuid="..uuid.."}user/1000");

        session:setAutoHangup(false)
        session:set_tts_parms("flite", "slt")

        session:speak("this is a test call")


please tell me how to avoid crash and any help will be highly appreciated

Regards

Shahzad Bhatti
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From mailing-lists at phoenixinternet.net  Wed Nov 28 20:29:21 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Wed, 28 Nov 2012 10:29:21 -0700
Subject: [Freeswitch-users] Site Down
Message-ID: <50B649F1.4040802@phoenixinternet.net>

Is the FreeSWITCH site down or has it been having problems? I was 
developing a new system to replace my asterisk system. Today I wanted to 
replicate the system, and could not access anything FreeSWITCH.org to do 
my build.

If this email posts to the list, obviously that would mean the mailing 
list email server is not down. But I cannot access the www or git.

Thanks,
Gilbert


From asilva at wirelessmundi.com  Wed Nov 28 19:25:09 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Wed, 28 Nov 2012 17:25:09 +0100
Subject: [Freeswitch-users] sqlite3 and regex
In-Reply-To: <CCB9BB87.5EE38%krice@freeswitch.org>
References: <CCB9BB87.5EE38%krice@freeswitch.org>
Message-ID: <1354119909.13048.56.camel@marces.madrid.commsmundi.com>

Hi,

Coudn't we load just an external extension to sqlite?

for example, to have regular expression support you could load an
external library like "sqlite3-pcre" (available for unbuntu), you can
load from the console interface or directly in sql.

console:
sqlite> .load '/usr/lib/sqlite3/pcre.so

sql in console:
sqlite> select load_extension(''/usr/lib/sqlite3/pcre.so'); 


In fs is not possible, when i try to do it from my lua script i have the
following error:

2012-11-28 17:21:34.196304 [ERR] switch_core_sqldb.c:572 NATIVE SQL ERR
[no such function: load_extension]
select load_extension('/usr/lib/sqlite3/pcre.so');


Since you have already a switch type of db ( core, odbc or pgsql) could
be nice when using core, be able to use a few more functions available
in sqlite?


Thanks,
Ant?nio






On Fri, 2012-11-02 at 18:03 -0500, Ken Rice wrote:

> No and this wont happen anytime soon... The SQL interfaces for
> FreeSwitch are kept generic as we support more then just sqlite from
> common code and if we did it for sqlite we would have to make sure its
> implemented equally well for postgresql and mysql and mssql and any
> other database someone might want to use via ODBC
> 
> K
> 
> 
> On 11/2/12 3:34 PM, "Scott" <8f27e956 at gmail.com> wrote:
> 
> 
>         "LIKE" notwithstanding, sqlite3 does not have a built-in true
>         regex function; it does allow for a a c-language hook to one.
>         Given fs extensive use of the regex engine and of sqlite3,
>         we're wondering if the hook is already written and rolled.  If
>         so, can the rest of us hook it to our sqlite3 uses (e.g. from
>         dial plan lua sqlite3).
>         
>         With thanks,
>         
>         
>         
>         ______________________________________________________________
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From chmp99 at gmail.com  Wed Nov 28 18:46:04 2012
From: chmp99 at gmail.com (=?ISO-8859-1?Q?Germ=E1n_Ruiz?=)
Date: Wed, 28 Nov 2012 12:46:04 -0300
Subject: [Freeswitch-users] Questions about RTMP endpoint
Message-ID: <50B631BC.8060001@gmail.com>

Hi,
I'm adding voice calls to a web site. I'm using RTMP endpoint and Flex
clientprovided. I have the following questions:
- Can flex clients (using RTMP protocol) register their presence and are
visible by others connected using SIP?
- Is it possible to send and receive text messages with RTMP protocol?

Thanks
Germ?n


From lloyd.aloysius at gmail.com  Thu Nov 29 01:07:54 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 28 Nov 2012 17:07:54 -0500
Subject: [Freeswitch-users] freeswitch.org - Not Accessable
Message-ID: <CAGvuWYbrZ5bd4Tg_Qv2RkgeHjkpQht8WM0OWBWkE_OimLNLWOQ@mail.gmail.com>

Hello:

I am trying to access freeswitch.org  from Toronto. Three different ISP's.

freeswitch.org is not available?

Thanks
Lloyd
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From talk2ram at gmail.com  Wed Nov 28 17:37:42 2012
From: talk2ram at gmail.com (ram)
Date: Wed, 28 Nov 2012 20:07:42 +0530
Subject: [Freeswitch-users] Changes to how ODBC, SQL, etc works
In-Reply-To: <1353848076648-7584902.post@n2.nabble.com>
References: <CCB96C43.5EDD5%krice@freeswitch.org>
	<1353848076648-7584902.post@n2.nabble.com>
Message-ID: <CA+XvL-OKr6wBkFQBeOizO7pWJ_dfnHun5gv5a=JkJTcTLdnYog@mail.gmail.com>

Hi

thats good news..

can any one point me PGSQL schema for Freeswitch.




On Sun, Nov 25, 2012 at 6:24 PM, peely <freeswitch at peely.com> wrote:

> It would be interesting to see some support for databases which have no
> native unixodbc support, like NoSQL solutions.
>
> Of particular interest in Apache Cassandra, the reason being it supports
> SQL-Like statements but has efficient and easy to configure support for
> multi-master replication, so a FreeSWITCH Cluster could use it to share
> registration and recovery data. It's also extremely fast and efficient.
>
> I currently use MySQL for this, as it has OK support for multi-master, but
> the overheads in binary logs on a busy box are pretty heavy.
>
> I'm not sure how easy it would be to implement Apache Cassandra support,
> but
> from my experiences in using it for other projects it would be very useful
> in FreeSWITCH for sharing data across multiple FreeSWITCH boxes.
>
>
> Neil.
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Changes-to-how-ODBC-SQL-etc-works-tp7584221p7584902.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bdfoster at endigotech.com  Thu Nov 29 04:36:47 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 28 Nov 2012 20:36:47 -0500
Subject: [Freeswitch-users] About the "transfer" function
In-Reply-To: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>
References: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>
Message-ID: <01A2D7CD-868A-4EBC-91C3-1EFE860F68A0@endigotech.com>

If you have that situation, it would be beneficial to have a dial plan set up in your default context to deal with this issue. I usually have a "custom extensions" folder inside of /.../freeswitch/conf/dialplan/default. From there I immediately bridge to the external number, setting your caller I'd and number along the way. 

-BDF

Sent from my iPhone

On Nov 28, 2012, at 4:16 AM, "sparklezou"<sparklezou at 163.com> wrote:

> Hi All,
>  
> When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321".
>  
> And on the caller phone will disply "Outbound Call", also show the number "87654321".
>  
> I checked the sip message, the info is got from the 180 or 183 message.
>  
> Sometime, don't want to show the real external number "87654321" to the enduser.
>  
> How to keep the caller phone displaying "1234".
>  
> Thanks in advance!
>  
> 2012-11-28
> sparklezou
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From brian at freeswitch.org  Thu Nov 29 04:38:50 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 19:38:50 -0600
Subject: [Freeswitch-users] use "myevents" make FreeSwitch Crash!
In-Reply-To: <CAN+NqUa+o-ERwrWdU0o0zPrp=yO9PCgvBhGe-ofGCdPVz1LX-g@mail.gmail.com>
References: <CAN+NqUa+o-ERwrWdU0o0zPrp=yO9PCgvBhGe-ofGCdPVz1LX-g@mail.gmail.com>
Message-ID: <C3AB73FD-4270-4319-AC8F-A76BE7031209@freeswitch.org>

Now that jira is up that seems to be the right place... as long as you're tried this with git head.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 28, 2012, at 4:42 AM, Shahzad Bhatti <shahzad.bhatti at g-r-v.com> wrote:

> Hi everyone,
> I am trying to use myevents to subscribe events but on call hang FREESWITCH Crash.
> here is my script
> 
> test.lua
> 
> require "ESL"
> 
> hostname        = "localhost"--'92.63.208.111';--'94.249.139.66'--'localhost';
> port            = '8021';
> password        = "ClueCon"--'Pak!stan'--'Pakistan';
> sock            = ESL.ESLconnection(hostname, port, password);
> 
> uuid = arg[1];
> 
> e = sock:sendRecv("myevents "..uuid);
> while  sock:connected() do
>        e = sock:recvEvent();
>        event = e:getType();
>        print (event.." EVENT Recieved")
> end
> 
> call.lua
> 
>         api = freeswitch.API()
>         uuid = api:executeString("create_uuid")
> 
>         os.execute("lua test.lua "..uuid.." &")
>         os.execute("sleep 1")
> 
> session = freeswitch.Session("{ignore_early_media=true,origination_uuid="..uuid.."}user/1000");
> 
>         session:setAutoHangup(false)
>         session:set_tts_parms("flite", "slt")
> 
>         session:speak("this is a test call")
> 
> 
> please tell me how to avoid crash and any help will be highly appreciated
> 
> Regards
> 
> Shahzad Bhatti
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Nov 29 04:39:28 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 19:39:28 -0600
Subject: [Freeswitch-users] Site Down
In-Reply-To: <50B649F1.4040802@phoenixinternet.net>
References: <50B649F1.4040802@phoenixinternet.net>
Message-ID: <2DFED399-15F5-45C7-A347-1745B4A9E4BC@freeswitch.org>

Yep We've noticed.  LOL  The list server was down too... as was EVERYTHING else.  :P
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 28, 2012, at 11:29 AM, "Gilbert T. Gutierrez, Jr." <mailing-lists at phoenixinternet.net> wrote:

> Is the FreeSWITCH site down or has it been having problems? I was 
> developing a new system to replace my asterisk system. Today I wanted to 
> replicate the system, and could not access anything FreeSWITCH.org to do 
> my build.
> 
> If this email posts to the list, obviously that would mean the mailing 
> list email server is not down. But I cannot access the www or git.
> 
> Thanks,
> Gilbert



From brian at freeswitch.org  Thu Nov 29 04:40:55 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 19:40:55 -0600
Subject: [Freeswitch-users] Side note (READ THIS PLEASE)
Message-ID: <F7E9ADB7-9C86-466E-931A-7732187DE121@freeswitch.org>

You should follow us on twitter, if you had done so you could have know what was going on during this wondering event for the past few days.

FOLLOW US NOW ON TWITTER @FreeSWITCH_Wire  (cuz some ass has freeswitch and won't give it up)

--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543







From brian at freeswitch.org  Thu Nov 29 04:41:37 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 19:41:37 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] About the "transfer"
	function
In-Reply-To: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>
References: <f996992.43dc.13b46500e9d.Coremail.sparklezou@163.com>
Message-ID: <B7CEC52B-11E6-4DDF-A395-CF1BCDD63B7E@freeswitch.org>

search wiki for ignore_display_updates. 
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 28, 2012, at 3:16 AM, sparklezou <sparklezou at 163.com> wrote:

> Hi All,
>  
> When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321".
>  
> And on the caller phone will disply "Outbound Call", also show the number "87654321".
>  
> I checked the sip message, the info is got from the 180 or 183 message.
>  
> Sometime, don't want to show the real external number "87654321" to the enduser.
>  
> How to keep the caller phone displaying "1234".
>  
> Thanks in advance!
>  
> 2012-11-28
> sparklezou
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Nov 29 04:59:27 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 19:59:27 -0600
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>
Message-ID: <BEE7B5CC-7CE2-457E-BE2F-7287558618D2@freeswitch.org>

Did you allow G729 at 50i?
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 27, 2012, at 5:06 AM, Ben N <ben122uk at gmail.com> wrote:

> Hi All,
> 
> I'm wondering if anyone out there is using G729 with 50ms ptime through Freeswitch, without proxy media?
> 
> I am unable to get this to work, even in a very basic lab with just two clients and an up to date FS server on the same LAN.  I can get G729 with 20ms working, but 50ms causes Freeswitch to hang the call.
> 
> It might be that mod_g729 and mod_com_g729 simply don't support it, so if anyone in the know can tell me that would be great!
> 
> Further info can be supplied, I believe I have tried pretty much all options....
> 
> Cheers,
> 
> Ben



From brian at freeswitch.org  Thu Nov 29 05:00:39 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 20:00:39 -0600
Subject: [Freeswitch-users] SIP to TDM t38 gateway
In-Reply-To: <F35B807F-7D80-4643-80BF-EAA97F9FC74F@5ninesolutions.com>
References: <5BA9AADC-BB63-4FF5-B1A6-1DEC1BE3E931@5ninesolutions.com>
	<50AE5F05.1010402@integrafin.co.uk>
	<F35B807F-7D80-4643-80BF-EAA97F9FC74F@5ninesolutions.com>
Message-ID: <17A84941-E7D0-4C05-AD01-4F40FF282B77@freeswitch.org>

Actually you should let your endpoint over SIP do it and NOT do it at the gateway.  Just saying you'll have less pain.  :P  Then when the ATA or device sends the re-invite it'll trigger it.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 26, 2012, at 10:42 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:

>    <extension name="dial-pbx"> 
>      <condition field="destination_number" expression="^(.*)$"> 
>        <action application="set" data="fax_enable_t38=true"/> 
>        <action application="set" data="fax_enable_t38_request=true"/> 
>        <action application="set" data="execute_on_answer=t38_gateway self"/> 
>        <action application="bridge" data="freetdm/1/a/$1"/> 
>      </condition> 
>    </extension>



From krice at freeswitch.org  Thu Nov 29 05:09:57 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 28 Nov 2012 20:09:57 -0600
Subject: [Freeswitch-users] Site Down
In-Reply-To: <2DFED399-15F5-45C7-A347-1745B4A9E4BC@freeswitch.org>
Message-ID: <CCDC2015.61243%krice@freeswitch.org>

Wait what? The servers were down?


On 11/28/12 7:39 PM, "Brian West" <brian at freeswitch.org> wrote:

> Yep We've noticed.  LOL  The list server was down too... as was EVERYTHING
> else.  :P
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
> 
> 
> 
> 
> 
> On Nov 28, 2012, at 11:29 AM, "Gilbert T. Gutierrez, Jr."
> <mailing-lists at phoenixinternet.net> wrote:
> 
>> Is the FreeSWITCH site down or has it been having problems? I was
>> developing a new system to replace my asterisk system. Today I wanted to
>> replicate the system, and could not access anything FreeSWITCH.org to do
>> my build.
>> 
>> If this email posts to the list, obviously that would mean the mailing
>> list email server is not down. But I cannot access the www or git.
>> 
>> Thanks,
>> Gilbert
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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




From enp at itx.ru  Tue Nov 27 08:16:43 2012
From: enp at itx.ru (Eugene Prokopiev)
Date: Tue, 27 Nov 2012 08:16:43 +0300
Subject: [Freeswitch-users] mod_perl vs mod_xml_curl
In-Reply-To: <CAKzWOxUP_OU+=75C+EBMTghF8PLR53ZdtGvSb3gzUP6yidMnFw@mail.gmail.com>
References: <CANM4Ruj+5f9HOdaVz+iA6A2Hx=fnC+Zsf77sRfN_C4+whjboPQ@mail.gmail.com>
	<CAKzWOxUP_OU+=75C+EBMTghF8PLR53ZdtGvSb3gzUP6yidMnFw@mail.gmail.com>
Message-ID: <CANM4RuhbnH_8Eg6V4qJ_EF9bVr9vKe_iD8qyQGfpLRZjA2DzrA@mail.gmail.com>

Now it works but I can't find configuration/code difference :(
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From schoch+freeswitch.org at xwin32.com  Thu Nov 29 04:52:42 2012
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Wed, 28 Nov 2012 17:52:42 -0800
Subject: [Freeswitch-users] FreeSwitch on a VM?
Message-ID: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>

In setting up a PBX to run FreeSwitch, I have allocated a machine that is
overpowered, so the best use of this resource is to put a hypervisor such
as Xen on it, and make it available for other office uses.

Is there any reason why running FreeSwitch on a VM is a bad idea?

-- 
Steve
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From enp at itx.ru  Wed Nov 28 10:42:29 2012
From: enp at itx.ru (Eugene Prokopiev)
Date: Wed, 28 Nov 2012 10:42:29 +0300
Subject: [Freeswitch-users] mod_perl vs mod_xml_curl
In-Reply-To: <CAKzWOxUP_OU+=75C+EBMTghF8PLR53ZdtGvSb3gzUP6yidMnFw@mail.gmail.com>
References: <CANM4Ruj+5f9HOdaVz+iA6A2Hx=fnC+Zsf77sRfN_C4+whjboPQ@mail.gmail.com>
	<CAKzWOxUP_OU+=75C+EBMTghF8PLR53ZdtGvSb3gzUP6yidMnFw@mail.gmail.com>
Message-ID: <CANM4RugejWubba=SUeWDyve2jQvnCTWWk7GjFcy8dYZj_uu0cQ@mail.gmail.com>

Now it works but I can't find configuration/code difference :(
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From ntomer at newgen.co.in  Wed Nov 28 07:24:37 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Wed, 28 Nov 2012 09:54:37 +0530
Subject: [Freeswitch-users] mod_db help needed
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>
	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>
Message-ID: <002501cdcd20$4819f280$d84dd780$@co.in>

Hi Yiftach,

 

Thanks for your reply. I have installed and configured ODBC on my server. 

 

/etc/odbc.ini has following entries -

 

[freesiwtch-connector]

Description           = MySQL connection to 'freeswitch' database

Driver                = MySQL

Database              = freeswitch

Server                = localhost

UserName              = root

Password              = system123

Port                  = 3306

Socket                = /var/run/mysqld/mysqld.sock

 

After this I did ./configure and "make && make install"

 

Now what all changes I need to make in Freeswitch so that I can make ODBC
work in it? Please help me out.

 

N.B. Today Freeswitch.org is down and therefore I am not able to look for
any help there

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yiftach
Golan
Sent: Friday, November 16, 2012 3:03 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_db help needed

 

Hi Nitin,
Just install odbc on you system, I used this one
(http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/install
ing_configuring_odbc.html) but you can use FreeSWITCH's explanations as well
(http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core)
Once you are done the configure should already know that you have odbc
support, just hit regular configure or as FreeSWITCH docs says : 


Compile FreeSWITCH with ODBC support


UnixODBC support will be autodetected by ./configure, and if found, will be
compiled into FreeSWITCH. 

As for the keys it is pretty easy just go to
(http://wiki.freeswitch.org/wiki/Mod_db) the query should be something like
: 

<action application="set" data="var=${db(select/realm/key)}"/>

The documentation is actually pretty good so try to read it again it will
make sense
Thanks,
Yiftach.

On Thu, Nov 15, 2012 at 4:51 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

I understand that I will have to make changes in db.conf.xml -

 

<configuration name="db.conf" description="LIMIT DB Configuration">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

  </settings>

</configuration>

 

But the wiki also says this - "ODBC must be configured to use ODBC resources
(configure with --enable-core-odbc-support)." What does this mean? I
couldn't find any more references to it.

 

Wiki says - "Realm and key are arbitrary strings. Consider realm as a
container for keys." Does this mean that I can use a realm for a particular
type of data e.g. "callerdata", use caller number as key and set the entered
data against that key. But how would I be able to access this data from
outside FreeSwitch?

 

Is the realm mapped to a table in the database or something similar?

 

Thanks in advance

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Thursday, November 15, 2012 3:30 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_db help needed

 

Hi,

 

I am trying to enter some data (entered by caller, received by Play and Get
Digits). But I am not able to get much information about it from mod_db
wiki. I want to enter data in an Oracle database. Please tell me what all I
need to do, in order to do this.

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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

 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From sparklezou at 163.com  Thu Nov 29 05:25:49 2012
From: sparklezou at 163.com (sparklezou)
Date: Thu, 29 Nov 2012 10:25:49 +0800
Subject: [Freeswitch-users] [Freeswitch-dev] About the "transfer"
	function
In-Reply-To: <B7CEC52B-11E6-4DDF-A395-CF1BCDD63B7E@freeswitch.org>
References: <B7CEC52B-11E6-4DDF-A395-CF1BCDD63B7E@freeswitch.org>
Message-ID: <2b32dd13.12a6.13b49fe985e.Coremail.sparklezou@163.com>

Hi All,

Thanks! I have fixed it.

The "update" info will be notify from "180", "183", "UPDATE" sip message.

BR,
Zou Yu

2012-11-29



sparklezou



????Brian West
?????2012-11-29 09:41
???Re: [Freeswitch-users] [Freeswitch-dev] About the "transfer" function
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

search wiki for ignore_display_updates.  
-- 
Brian West 
brian at freeswitch.org 
FreeSWITCH Solutions, LLC 
PO BOX PO BOX 2531 
Brookfield, WI 53008-2531 
Twitter: @FreeSWITCH_Wire 
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST 
iNUM: +883 5100 1420 9266 
UK: +44 20 3298 4900 
ISN: 410*543 





On Nov 28, 2012, at 3:16 AM, sparklezou <sparklezou at 163.com> wrote: 

> Hi All, 
>   
> When dial an internal number "1234", it will "bridge" or "transfer" to an external number "87654321". 
>   
> And on the caller phone will disply "Outbound Call", also show the number "87654321". 
>   
> I checked the sip message, the info is got from the 180 or 183 message. 
>   
> Sometime, don't want to show the real external number "87654321" to the enduser. 
>   
> How to keep the caller phone displaying "1234". 
>   
> Thanks in advance! 
>   
> 2012-11-28 
> sparklezou 
> _________________________________________________________________________ 
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org 
> http://www.freeswitchsolutions.com 
>  
>  
>  
>  
> Official FreeSWITCH Sites 
> http://www.freeswitch.org 
> http://wiki.freeswitch.org 
> http://www.cluecon.com 
>  
> FreeSWITCH-dev mailing list 
> FreeSWITCH-dev at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev 
> http://www.freeswitch.org 
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From brian at freeswitch.org  Thu Nov 29 05:29:58 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Nov 2012 20:29:58 -0600
Subject: [Freeswitch-users] FreeSwitch on a VM?
In-Reply-To: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
References: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
Message-ID: <164EF57D-5B27-45E3-97DD-9EDC2B90C6BB@freeswitch.org>

Um we run our entire infrastructure on Proxmox and VM/Containers ... I say go for it.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 28, 2012, at 7:52 PM, Steven Schoch <schoch+freeswitch.org at xwin32.com> wrote:

> In setting up a PBX to run FreeSwitch, I have allocated a machine that is overpowered, so the best use of this resource is to put a hypervisor such as Xen on it, and make it available for other office uses.
> 
> Is there any reason why running FreeSwitch on a VM is a bad idea?
> 
> -- 
> Steve



From william.king at quentustech.com  Thu Nov 29 05:44:28 2012
From: william.king at quentustech.com (William King)
Date: Wed, 28 Nov 2012 18:44:28 -0800
Subject: [Freeswitch-users] FreeSwitch on a VM?
In-Reply-To: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
References: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
Message-ID: <50B6CC0C.2030303@quentustech.com>

Others would probably have additional information, but one of the issues
you have to be aware about is that if your physical machine is over
capacity, the virtual instance may become resource limited.

In other words, if you have another VM that's a CPU or IO hog then your
FS VM may run into call audio issues. That said I've run FS in a KVM
virtual environment for a few years and that has gone very well.

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

On 11/28/2012 05:52 PM, Steven Schoch wrote:
> In setting up a PBX to run FreeSwitch, I have allocated a machine that
> is overpowered, so the best use of this resource is to put a hypervisor
> such as Xen on it, and make it available for other office uses.
> 
> Is there any reason why running FreeSwitch on a VM is a bad idea?
> 
> -- 
> Steve
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From dujinfang at gmail.com  Thu Nov 29 06:16:10 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 29 Nov 2012 11:16:10 +0800
Subject: [Freeswitch-users] Questions about RTMP endpoint
In-Reply-To: <50B631BC.8060001@gmail.com>
References: <50B631BC.8060001@gmail.com>
Message-ID: <1726AEE3708448019DFAD73E4C2F8ACB@gmail.com>

We have text messages via rtmp in our branch. Just need time to find them out.  

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


On Wednesday, November 28, 2012 at 11:46 PM, Germ?n Ruiz wrote:

> Hi,
> I'm adding voice calls to a web site. I'm using RTMP endpoint and Flex
> clientprovided. I have the following questions:
> - Can flex clients (using RTMP protocol) register their presence and are
> visible by others connected using SIP?
> - Is it possible to send and receive text messages with RTMP protocol?
>  
> Thanks
> Germ?n
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-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
>  
>  


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From chris at gonumina.com  Thu Nov 29 06:35:16 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Wed, 28 Nov 2012 22:35:16 -0500
Subject: [Freeswitch-users] FreeSwitch on a VM?
In-Reply-To: <164EF57D-5B27-45E3-97DD-9EDC2B90C6BB@freeswitch.org>
References: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
	<164EF57D-5B27-45E3-97DD-9EDC2B90C6BB@freeswitch.org>
Message-ID: <CAEuuoeO+2aE9HHVM_EGKDihuM1Pwb1eBm0_Tnf9uzEOQBGq7xw@mail.gmail.com>

Hi Brian,


Any considerations when running as a container vs a VM?

Thanks,

-Chris




____________________________
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Founder & CEO
Numina Networks, Inc.
Voice: 201-510-2560
Fax: 866-577-3411
Chris at GoNumina.com

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Twitter]<http://twitter.com/NuminaNetworks>
 <http://www.gonumina.com/>



On Wed, Nov 28, 2012 at 9:29 PM, Brian West <brian at freeswitch.org> wrote:

> Um we run our entire infrastructure on Proxmox and VM/Containers ... I say
> go for it.
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
> On Nov 28, 2012, at 7:52 PM, Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
>
> > In setting up a PBX to run FreeSwitch, I have allocated a machine that
> is overpowered, so the best use of this resource is to put a hypervisor
> such as Xen on it, and make it available for other office uses.
> >
> > Is there any reason why running FreeSwitch on a VM is a bad idea?
> >
> > --
> > Steve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From govoiper at gmail.com  Wed Nov 28 08:13:56 2012
From: govoiper at gmail.com (SamyGo)
Date: Wed, 28 Nov 2012 10:13:56 +0500
Subject: [Freeswitch-users] Cluecon 2012 presentations availability !?
Message-ID: <CAJUJwtj4wHLXSsaiE6Nb8P3gs5BH-Qsw-hWDGuzxoN=0PjGyZw@mail.gmail.com>

Hi,

Is there any way we can get the presentations or videos of sessions in
cluecon 2012 ? Will really be a treat for those around the world who
couldn't be there.

Thanks,
Sammy Go.
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From jkomar at jbox.ca  Thu Nov 29 06:47:05 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Wed, 28 Nov 2012 20:47:05 -0700
Subject: [Freeswitch-users] Trouble with voicemail say digits
Message-ID: <CA+zebiY3-N37JnDNVcgonxT-Qw-+tY64uQw7zCNZSY9joy9Rxw@mail.gmail.com>

Just updated to FS 1.2.5.1 on Gentoo from 1.2.3. I have one extension that
uses the default voicemail greeting. Since I updated, when it gets to the
part where it says the extension number, it cannot find the wav files for
the digits. It is looking for them in the digits folder and not in the
frequency subfolders (i.e. 16000  32000  48000  8000). As soon as it hits
this error, the voicemail app says goodbye and hangs up.

2012-11-28 20:35:17.068585 [DEBUG] switch_ivr_play_say.c:244 Handle
say:[2003] (en:en)
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
[/opt/freeswitch/sounds/en/us/callie/digits/2.wav] does not exist.
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
[/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
[/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
[/opt/freeswitch/sounds/en/us/callie/digits/3.wav] does not exist.

Is there a config option to specify the frequency subfolder, or did
something not build correctly?

Thanks,
Jason Komar
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From ivan.mironov at infra-it.ru  Thu Nov 29 06:54:45 2012
From: ivan.mironov at infra-it.ru (Ivan Mironov)
Date: Thu, 29 Nov 2012 09:54:45 +0600
Subject: [Freeswitch-users] Site Down
In-Reply-To: <50B649F1.4040802@phoenixinternet.net>
References: <50B649F1.4040802@phoenixinternet.net>
Message-ID: <50B6DC85.5030605@infra-it.ru>

28.11.2012 23:29, Gilbert T. Gutierrez, Jr. ?????:
> Is the FreeSWITCH site down or has it been having problems? I was
> developing a new system to replace my asterisk system. Today I wanted to
> replicate the system, and could not access anything FreeSWITCH.org to do
> my build.
>
> If this email posts to the list, obviously that would mean the mailing
> list email server is not down. But I cannot access the www or git.
>
> Thanks,
> Gilbert
>

There is a github mirror: https://github.com/FreeSWITCH

-- 



From ivan.mironov at infra-it.ru  Thu Nov 29 07:13:58 2012
From: ivan.mironov at infra-it.ru (Ivan Mironov)
Date: Thu, 29 Nov 2012 10:13:58 +0600
Subject: [Freeswitch-users] Trouble with voicemail say digits
In-Reply-To: <CA+zebiY3-N37JnDNVcgonxT-Qw-+tY64uQw7zCNZSY9joy9Rxw@mail.gmail.com>
References: <CA+zebiY3-N37JnDNVcgonxT-Qw-+tY64uQw7zCNZSY9joy9Rxw@mail.gmail.com>
Message-ID: <50B6E106.3030608@infra-it.ru>

29.11.2012 09:47, Komar, Jason ?????:
> Just updated to FS 1.2.5.1 on Gentoo from 1.2.3. I have one extension that
> uses the default voicemail greeting. Since I updated, when it gets to the
> part where it says the extension number, it cannot find the wav files for
> the digits. It is looking for them in the digits folder and not in the
> frequency subfolders (i.e. 16000  32000  48000  8000). As soon as it hits
> this error, the voicemail app says goodbye and hangs up.
>
> 2012-11-28 20:35:17.068585 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[2003] (en:en)
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/2.wav] does not exist.
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/3.wav] does not exist.
>
> Is there a config option to specify the frequency subfolder, or did
> something not build correctly?
>
> Thanks,
> Jason Komar
>

It seems that this bug already fixed in git:

commit 0b148a85b94be33fe70b692240e0d449a59f0ef2
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Mon Nov 26 11:59:29 2012 -0600

     this breaks the auto rate hunting code in mod_sndfile

  src/mod/applications/mod_dptools/mod_dptools.c | 5 -----
  1 file changed, 5 deletions(-)

But I didn't try it yet.

-- 



From kbdfck at gmail.com  Thu Nov 29 07:26:01 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 29 Nov 2012 08:26:01 +0400
Subject: [Freeswitch-users] Sofia listens only on 127.0.0.1 after server
 VM reboot?
In-Reply-To: <7B0E64C1-7767-46E4-B881-EC9CB3E707C3@freeswitch.org>
References: <CA+GmxbaZf+JnnUQMpv1H27xY+NGt3mcKLAaCCvjGrjpL5HOVEw@mail.gmail.com>
	<CCD7A694.610F3%krice@freeswitch.org>
	<CA+GmxbbXeQo+NdCp7eH079Sxw0Q7X6WorpfjXMRXrh=+ohet+A@mail.gmail.com>
	<7B0E64C1-7767-46E4-B881-EC9CB3E707C3@freeswitch.org>
Message-ID: <CA+GmxbbbzfY2NfzOpDVr2A8PZcnYW3rq_orPzKdONwXF5q0SDA@mail.gmail.com>

I'm setting public ip address of my FS server
I think problem appears when this IP is not on interface yet when FS starts

2012/11/26 Brian West <brian at freeswitch.org>

> What are you setting in your profile as the IP?
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
> On Nov 25, 2012, at 12:55 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
> > I tried to set static IP, that doesn't work :(
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From ntomer at newgen.co.in  Thu Nov 29 07:40:47 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 29 Nov 2012 10:10:47 +0530
Subject: [Freeswitch-users] mod_db help needed
In-Reply-To: <012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>
	<012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>
Message-ID: <00c101cdcdeb$b38bbaa0$1aa32fe0$@co.in>

Since site was down yesterday, I did some hit and trials myself, made the
following changes in db.conf.xml -

 

<configuration name="db.conf" description="LIMIT DB Configuration">
  <settings>
      <param name="freeswitch-connector"
value="freeswitch:root:system123#"/>
  </settings>
</configuration>

 

And added this entry in dialplan -

 

<action application="db" data="insert/testapp/newcall1/${res}" />

 

When I tried running it, I didn't get any error. But neither any entries
were made in the database.

 

Please tell me what I am doing wrong.

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Tuesday, November 27, 2012 3:52 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] mod_db help needed

 

Hi Yiftach,

 

Thanks for your reply. I have installed and configured ODBC on my server. 

 

/etc/odbc.ini has following entries -

 

[freesiwtch-connector]

Description           = MySQL connection to 'freeswitch' database

Driver                = MySQL

Database              = freeswitch

Server                = localhost

UserName              = root

Password              = system123

Port                  = 3306

Socket                = /var/run/mysqld/mysqld.sock

 

After this I did ./configure and "make && make install"

 

Now what all changes I need to make in Freeswitch so that I can make ODBC
work in it? Please help me out.

 

N.B. Today Freeswitch.org is down and therefore I am not able to look for
any help there

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yiftach
Golan
Sent: Friday, November 16, 2012 3:03 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_db help needed

 

Hi Nitin,
Just install odbc on you system, I used this one
(http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/install
ing_configuring_odbc.html) but you can use FreeSWITCH's explanations as well
(http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core)
Once you are done the configure should already know that you have odbc
support, just hit regular configure or as FreeSWITCH docs says : 


Compile FreeSWITCH with ODBC support


UnixODBC support will be autodetected by ./configure, and if found, will be
compiled into FreeSWITCH. 

As for the keys it is pretty easy just go to
(http://wiki.freeswitch.org/wiki/Mod_db) the query should be something like
: 

<action application="set" data="var=${db(select/realm/key)}"/>

The documentation is actually pretty good so try to read it again it will
make sense
Thanks,
Yiftach.

On Thu, Nov 15, 2012 at 4:51 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

I understand that I will have to make changes in db.conf.xml -

 

<configuration name="db.conf" description="LIMIT DB Configuration">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

  </settings>

</configuration>

 

But the wiki also says this - "ODBC must be configured to use ODBC resources
(configure with --enable-core-odbc-support)." What does this mean? I
couldn't find any more references to it.

 

Wiki says - "Realm and key are arbitrary strings. Consider realm as a
container for keys." Does this mean that I can use a realm for a particular
type of data e.g. "callerdata", use caller number as key and set the entered
data against that key. But how would I be able to access this data from
outside FreeSwitch?

 

Is the realm mapped to a table in the database or something similar?

 

Thanks in advance

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Thursday, November 15, 2012 3:30 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_db help needed

 

Hi,

 

I am trying to enter some data (entered by caller, received by Play and Get
Digits). But I am not able to get much information about it from mod_db
wiki. I want to enter data in an Oracle database. Please tell me what all I
need to do, in order to do this.

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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

 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From drk at drkngs.net  Thu Nov 29 07:42:05 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Wed, 28 Nov 2012 20:42:05 -0800
Subject: [Freeswitch-users] Trouble with voicemail say digits
In-Reply-To: <CA+zebiY3-N37JnDNVcgonxT-Qw-+tY64uQw7zCNZSY9joy9Rxw@mail.gmail.com>
Message-ID: <20121129044205.141c5071@mail.tritonwest.net>

I thought it was me... Did something change whre the /8000/ got removed from the path? Is there a point where you can't leave your old configs in place when you update?  
   
--Dave
      _____  

  From: Komar, Jason [mailto:jkomar at jbox.ca]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Wed, 28 Nov 2012 19:47:05 -0800
Subject: [Freeswitch-users] Trouble with voicemail say digits

  
Just updated to FS 1.2.5.1 on Gentoo from 1.2.3. I have one extension that uses the default voicemail greeting. Since I updated, when it gets to the part where it says the extension number, it cannot find the wav files for the digits. It is looking for them in the digits folder and not in the frequency subfolders (i.e. 16000  32000  48000  8000). As soon as it hits this error, the voicemail app says goodbye and hangs up.  

  
2012-11-28 20:35:17.068585 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2003] (en:en)  
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/2.wav] does not exist.  
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.  
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.  
2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/3.wav] does not exist.  

  
Is there a config option to specify the frequency subfolder, or did something not build correctly?  

  
Thanks,  
Jason Komar      
   
 
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From lpradovera at mojolingo.com  Thu Nov 29 08:13:48 2012
From: lpradovera at mojolingo.com (Luca Pradovera)
Date: Thu, 29 Nov 2012 06:13:48 +0100
Subject: [Freeswitch-users] FreeSwitch on a VM?
In-Reply-To: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
References: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
Message-ID: <23420731-934C-483C-AEBF-8944BD32FF15@mojolingo.com>

I have never run FS outside of a VM and never saw issues, although I only have three production deployments and half a dozen dev instances.
The usual resource caveats apply but it's more about general VM usage than FS in particular.

Inviato da iPhone

Il giorno 29/nov/2012, alle ore 02:52, Steven Schoch <schoch+freeswitch.org at xwin32.com> ha scritto:

> In setting up a PBX to run FreeSwitch, I have allocated a machine that is overpowered, so the best use of this resource is to put a hypervisor such as Xen on it, and make it available for other office uses.
> 
> Is there any reason why running FreeSwitch on a VM is a bad idea?
> 
> -- 
> Steve
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From packetandy at gmail.com  Thu Nov 29 08:43:47 2012
From: packetandy at gmail.com (andy)
Date: Wed, 28 Nov 2012 21:43:47 -0800
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 77, Issue 196
In-Reply-To: <mailman.2512.1354160168.576.freeswitch-users@lists.freeswitch.org>
References: <mailman.2512.1354160168.576.freeswitch-users@lists.freeswitch.org>
Message-ID: <50B6F613.3050904@gmail.com>

with respect to running FS on a VM, one thing that has not been 
mentioned is the need for a real time kernel in the underlying host. 
This can be a big issue when running on a hosted vm in the cloud, as 
typically you will not have a choice of kernel, nor the ability to 
recompile. I have had bad experience with several hosting companies 
claiming to run 10msec kernels, but in reality running 25 or even 100 
msec versions - lot of potential for dropped and out of order speech 
with the later.

packetandy



From gopalakrishnan.an at gmail.com  Thu Nov 29 09:15:54 2012
From: gopalakrishnan.an at gmail.com (Gopalakrishnan N)
Date: Thu, 29 Nov 2012 11:45:54 +0530
Subject: [Freeswitch-users] FreeTDM ftmod pri tap error
In-Reply-To: <CAFZQphzqUs910p49LkhA-rFSLcxvX-hPzC_LLS52ebb8KpfjWQ@mail.gmail.com>
References: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
	<CAFZQphzqUs910p49LkhA-rFSLcxvX-hPzC_LLS52ebb8KpfjWQ@mail.gmail.com>
Message-ID: <CAEymo11fjQivRBmzdxcoak+Je0gvVpvUzOj9ntO+NfVJQLiAhQ@mail.gmail.com>

ok thanks will try that...


On Mon, Nov 26, 2012 at 8:01 PM, Muhammad Shahzad <shaheryarkh at gmail.com>wrote:

> or search and remove this flag yourself in configure scripts
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From 8f27e956 at gmail.com  Thu Nov 29 09:19:41 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Thu, 29 Nov 2012 01:19:41 -0500
Subject: [Freeswitch-users] sqlite3 and regex
In-Reply-To: <1354119909.13048.56.camel@marces.madrid.commsmundi.com>
References: <CCB9BB87.5EE38%krice@freeswitch.org>
	<1354119909.13048.56.camel@marces.madrid.commsmundi.com>
Message-ID: <1413611559183349171@unknownmsgid>

That's EXACTLY what I'm hunting for!  My freeswitch build doesn't have a
pcre.so anywhere on the disk.  The .../freeswitch/libs/pcre/ exists with
many files but no .so.

What does one need to do to get the sqlite3 and the pcre.so rolled (from
the git ingredients)?  make() and stuff not my strong suit.

Thanks,
?????
iThing:  Big thumbs & little keys. Please excuse typo, spelling and grammar
errors ? Last night I played a blank CD at full blast. The Mime next door
went nuts.


On 2012-11-28, at 23:53, Antonio <asilva at wirelessmundi.com> wrote:

 Re: [Freeswitch-users] sqlite3 and regex Hi,

Coudn't we load just an external extension to sqlite?

for example, to have regular expression support you could load an external
library like "sqlite3-pcre" (available for unbuntu), you can load from the
console interface or directly in sql.

console:
sqlite> .load '/usr/lib/sqlite3/pcre.so

sql in console:
sqlite> select load_extension(''/usr/lib/sqlite3/pcre.so');


In fs is not possible, when i try to do it from my lua script i have the
following error:

2012-11-28 17:21:34.196304 [ERR] switch_core_sqldb.c:572 NATIVE SQL ERR [no
such function: load_extension]
select load_extension('/usr/lib/sqlite3/pcre.so');


Since you have already a switch type of db ( core, odbc or pgsql) could be
nice when using core, be able to use a few more functions available in
sqlite?


Thanks,
Ant?nio






On Fri, 2012-11-02 at 18:03 -0500, Ken Rice wrote:

No and this wont happen anytime soon... The SQL interfaces for FreeSwitch
are kept generic as we support more then just sqlite from common code and
if we did it for sqlite we would have to make sure its implemented equally
well for postgresql and mysql and mssql and any other database someone
might want to use via ODBC

K


On 11/2/12 3:34 PM, "Scott" <8f27e956 at gmail.com> wrote:

 "LIKE" notwithstanding, sqlite3 does not have a built-in true regex
function; it does allow for a a c-language hook to one.  Given fs extensive
use of the regex engine and of sqlite3, we're wondering if the hook is
already written and rolled.  If so, can the rest of us hook it to our
sqlite3 uses (e.g. from dial plan lua sqlite3).

With thanks,


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


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




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

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


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

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

FreeSWITCH-powered IP PBX: The CudaTel Communication
Server

Official FreeSWITCH
Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com

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


  --

Un cordial saludo / Best regards,

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com

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




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

FreeSWITCH-users mailing 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 8f27e956 at gmail.com  Thu Nov 29 09:47:54 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Thu, 29 Nov 2012 01:47:54 -0500
Subject: [Freeswitch-users] FreeSwitch on a VM?
In-Reply-To: <23420731-934C-483C-AEBF-8944BD32FF15@mojolingo.com>
References: <CAOkuKdeVf7poefuEkRsqAqrb=zW9rNTg1T03-QjSs587tFj6LQ@mail.gmail.com>
	<23420731-934C-483C-AEBF-8944BD32FF15@mojolingo.com>
Message-ID: <-1594403226918363549@unknownmsgid>

There are some good/better/best Virtual engine config than others.
And your milage may vary, but keys to allow for the highest performing
fs-as-guest ... Config the fs-guest with a CPU affinity such that it
is NEVER without two or more cores,,, if really, really necessary the
vm engine may be config'ed to allowed swap of fs-guest for memory
pressures but NOT cores,,,  plus the network interfaces seen by
fs-guest should be pass-thru and the real hardware NICs be virtual
capable.  (i've seen vm cause issues with non pass thru and 20 ms
framing rates under load.)

On 2012-11-29, at 1:22, Luca Pradovera <lpradovera at mojolingo.com> wrote:

> I have never run FS outside of a VM and never saw issues, although I only have three production deployments and half a dozen dev instances.
> The usual resource caveats apply but it's more about general VM usage than FS in particular.
>
> Inviato da iPhone
>
> Il giorno 29/nov/2012, alle ore 02:52, Steven Schoch <schoch+freeswitch.org at xwin32.com> ha scritto:
>
>> In setting up a PBX to run FreeSwitch, I have allocated a machine that is overpowered, so the best use of this resource is to put a hypervisor such as Xen on it, and make it available for other office uses.
>>
>> Is there any reason why running FreeSwitch on a VM is a bad idea?
>>
>> --
>> Steve
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gopalakrishnan.an at gmail.com  Thu Nov 29 09:59:15 2012
From: gopalakrishnan.an at gmail.com (Gopalakrishnan N)
Date: Thu, 29 Nov 2012 12:29:15 +0530
Subject: [Freeswitch-users] FreeTDM ftmod pri tap error
In-Reply-To: <CAFZQphzqUs910p49LkhA-rFSLcxvX-hPzC_LLS52ebb8KpfjWQ@mail.gmail.com>
References: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
	<CAFZQphzqUs910p49LkhA-rFSLcxvX-hPzC_LLS52ebb8KpfjWQ@mail.gmail.com>
Message-ID: <CAEymo11jxHTCzibCpAmi+E53NR7ONWRTeJyUKec-6S4ctjzNhw@mail.gmail.com>

I have used stable tar ball from 1.2.3. anyways let me try from git
version.

On Mon, Nov 26, 2012 at 8:01 PM, Muhammad Shahzad <shaheryarkh at gmail.com>wrote:

> -Werror flag in configuratio
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From avi at avimarcus.net  Thu Nov 29 10:21:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 29 Nov 2012 09:21:49 +0200
Subject: [Freeswitch-users] Side note (READ THIS PLEASE)
In-Reply-To: <F7E9ADB7-9C86-466E-931A-7732187DE121@freeswitch.org>
References: <F7E9ADB7-9C86-466E-931A-7732187DE121@freeswitch.org>
Message-ID: <CALNZuwkSqA2hcFGeHqPAs2yV=_V-bGM8FcR=+h2qOPtXzzSDsA@mail.gmail.com>

Have you contacted twitter to claim the name?
http://support.twitter.com/articles/18367-trademark-policy

-Avi

On Thu, Nov 29, 2012 at 3:40 AM, Brian West <brian at freeswitch.org> wrote:

> You should follow us on twitter, if you had done so you could have know
> what was going on during this wondering event for the past few days.
>
> FOLLOW US NOW ON TWITTER @FreeSWITCH_Wire  (cuz some ass has freeswitch
> and won't give it up)
>
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
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From avi at avimarcus.net  Thu Nov 29 10:24:01 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 29 Nov 2012 09:24:01 +0200
Subject: [Freeswitch-users] FusionPBX for freeswitch
In-Reply-To: <CAC_Ms12WJMwvnXD0fzK=Ps+4My32z08hF+hm3YqOkCpGeVL=jQ@mail.gmail.com>
References: <CAC_Ms12WJMwvnXD0fzK=Ps+4My32z08hF+hm3YqOkCpGeVL=jQ@mail.gmail.com>
Message-ID: <CALNZuw=1MWp_KSL8T1hvYc2p1S4ruZCRdm9j00cWqXdES+U+Eg@mail.gmail.com>

See: http://wiki.freeswitch.org/wiki/Freeswitch_Gui

-Avi
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From anthony.minessale at gmail.com  Thu Nov 29 10:34:00 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Nov 2012 01:34:00 -0600
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
	again
In-Reply-To: <50B4249A.30702@communicatefreely.net>
References: <CCD7A44C.610F0%krice@freeswitch.org>
	<50B4249A.30702@communicatefreely.net>
Message-ID: <CAKbxfG-Mnm-yhx0D6mp=WM3xCC6bmg=OmSy0ph8YU-ecL31V9Q@mail.gmail.com>

too bad you did not gcore it like I suggested, otherwise who knows...
That's just the message it prints when the message parsing thread goes up.



On Mon, Nov 26, 2012 at 8:25 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Thanks for all the input everyone!
>
> I just had it happen again, this time at night when there were only 2
> sessions active!
>
> I was in doing some other work, so I managed to see it happen and found a
> few interesting
> things.
>
> CPU load next to nothing, load average around 0.12.
> I used iSQL to test ODBC connectivity, and I could read and write to the
> freeswitch
> database using the same DSN that freeswitch is using.
>
> I could delete a SIP registration from the API
>
> Since I had things on a higher debug level, I saw this:
>
> freeswitch at stefan> 2012-11-26 21:21:51.914082 [CONSOLE] sofia.c:1144 MSG
> Thread Started
>
> As soon as that came up, everything started working again.
>
> What's that about?
>
> -Tim
>
> Ken Rice wrote:
> > Sofia is not single threaded except for in one spot deep in libsofia,
> >>From there, messages are handed off to a number of message queues for FS
> > core to handle as needed...
> >
> > Check to see if anything that fs is depending on is blocking on info
> > retrieval like the databases or other areas...
> >
> > K
> >
> > On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com> wrote:
> >
> >> you mentioned that you use xml_curl, if your web server hangs it may
> >> hang sofia, iirc sofia is running a single thread and it will wait for
> >> the xml_curl response before continuing to the next request.
> >>
> >> On 11/23/2012 1:13 PM, Steven Ayre wrote:
> >>> Any kind of DB backup running? Or any long-running queries (innotop is
> >>> great for highlighting queries that've been running a while, including
> >>> on non-innodb tables).
> >>>
> >>> A global read lock, or queries waiting for a lock could block a db
> >>> update from the sofia profile thread but still allow read-only queries
> >>> (sofia status) to run.
> >>>
> >>> -Steve
> >>>
> >>>
> >>>
> >>> On 23 November 2012 15:32, Tim St. Pierre <
> fs-list at communicatefreely.net>
> >>> wrote:
> >>>> Hi Steven,
> >>>>
> >>>> Thanks for the suggestions.  I'm hoping once I get the upgrade done
> it will
> >>>> all go away.
> >>>> I have watched it happen at least once.  I was on the phone at the
> time.
> >>>> Console activity
> >>>> more or less stopped, except for a few calls hanging up.  The console
> >>>> remains responsive,
> >>>> and my call wasn't dropped for at least a minute or two (media
> timeout?).  I
> >>>> was able to
> >>>> run sofia status and other commands that use the database, so I'm
> assuming
> >>>> that the
> >>>> connection was still working.  All our media is runs through the box,
> so I
> >>>> think things
> >>>> are fine on the Ethernet level.  I do see higher load averages -
> maybe 3-4,
> >>>> but that's the
> >>>> only obvious indication.  It's not taking CPU beyond 10% or so.
> >>>>
> >>>> We are using MySQL as the core DB and also as the DB backend for each
> sofia
> >>>> profile.  This
> >>>> is connecting through ODBC of course.
> >>>>
> >>>> If I can get the other kinks worked out, then I will try 1.2 stable in
> >>>> production and
> >>>> we'll see how it goes.
> >>>>
> >>>> -Tim
> >>>>
> >>>> Steven Ayre wrote:
> >>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
> >>>>>
> >>>>>
> >>>>> As you've already acknowledged it's a very old version.
> >>>>>
> >>>>> It's possible that your issue has already been found and fixed, but
> if
> >>>>> it hasn't then the code will have changed significantly since then
> and
> >>>>> you'd really need to reproduce it on the latest code for it to be
> >>>>> investigated.
> >>>>>
> >>>>>
> >>>>> As some general thoughts though, are you able to spot it happening
> while
> >>>>> it's happening or only afterwards?
> >>>>>
> >>>>> If you're able to get on the system during one of those times look at
> >>>>> what else is happening. Is the load average/cpu usage/io high?
> Perhaps
> >>>>> something's running that's blocking all access or causing very high
> IO.
> >>>>>
> >>>>> What DB backend are you using for Sofia? Is it possible that that's
> >>>>> hanging for a moment? For example if you're running a backup on the
> DB
> >>>>> that blocks all writes to the DB while Sofia is trying to update the
> DB
> >>>>> that perhaps would cause this.
> >>>>>
> >>>>> Try running a SIP OPTIONS ping your your sofia profile from the
> >>>>> localhost during that time, which should exclude it being any issue
> on
> >>>>> the ethernet.
> >>>>>
> >>>>> -Steve
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> On 22 November 2012 19:31, Tim St. Pierre <
> fs-list at communicatefreely.net
> >>>>> <mailto:fs-list at communicatefreely.net>> wrote:
> >>>>>
> >>>>>      Hello,
> >>>>>
> >>>>>      I'm having a bit of an odd problem.
> >>>>>
> >>>>>      Intermittently, often every 2-3 days or so, Freeswitch stops
> >>>>>      replying to SIP for about 5
> >>>>>      minutes.  I can't verify if it's EXACTLY 5 minutes, but it
> seems to
> >>>>>      be pretty close.
> >>>>>
> >>>>>      During this time, no new registrations or invites can happen,
> but
> >>>>>      existing calls stay
> >>>>>      connected for at least a minute or two.  In the logs, you can
> see
> >>>>>      calls slowly hanging up
> >>>>>      with "NORMAL_CLEARING".  In 5 minutes, everything starts up
> again
> >>>>>      with no word about it at
> >>>>>      all in the logs.
> >>>>>
> >>>>>      When calls resume, I notice that the number of sessions
> returned by
> >>>>>      the status command is
> >>>>>      one higher than the actual number sessions returned by show
> >>>>>      channels, or by looking in the
> >>>>>      database.  Every time this happens, the discrepancy increases
> by one.
> >>>>>
> >>>>>      The interruption happens on all SIP profiles, but calls
> originated
> >>>>>      from the socket API
> >>>>>      still work, insofar as they return with PROGRESS_TIMEOUT since
> the
> >>>>>      profiles are still
> >>>>>      running, but stuck.
> >>>>>
> >>>>>      We are using ODBC/MySQL for the core database, and the database
> >>>>>      server only runs this
> >>>>>      database and some basic PHP/xml-curl stuff.
> >>>>>
> >>>>>      We have 416 endpoints registered, and usually sit at about 30
> >>>>>      sessions during the day.
> >>>>>
> >>>>>      This never happens at night, only during busier times, but not
> >>>>>      necessarily busy hour.
> >>>>>
> >>>>>      I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores, 4G ram)
> >>>>>
> >>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
> >>>>>
> >>>>>      Yes, I know it's old and I'm trying to upgrade, but I'm still
> having
> >>>>>      some problems getting
> >>>>>      all my phones to work properly with 1.2 stable.  This is a
> >>>>>      production system, so I can't
> >>>>>      just blindly put out the newest release.  Mostly, I need to buy
> >>>>>      myself some time so that I
> >>>>>      can get the kinks worked out of the latest version and then
> upgrade
> >>>>>      the production box.
> >>>>>
> >>>>>      I'm grateful for any insights as to what could be happening,
> even if
> >>>>>      a solution is just a
> >>>>>      temporary workaround.
> >>>>>
> >>>>>      Thanks!
> >>>>>
> >>>>>      -Tim
> >>>>>
> >>>>>
> >>>>>
> _________________________________________________________________________
> >>>>>      Professional FreeSWITCH Consulting Services:
> >>>>>      consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>>>>      http://www.freeswitchsolutions.com
> >>>>>
> >>>>>      
> >>>>>      
> >>>>>
> >>>>>      Official FreeSWITCH Sites
> >>>>>      http://www.freeswitch.org
> >>>>>      http://wiki.freeswitch.org
> >>>>>      http://www.cluecon.com
> >>>>>
> >>>>>      FreeSWITCH-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
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>>
> >>>>>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From brian at freeswitch.org  Thu Nov 29 10:38:08 2012
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Nov 2012 01:38:08 -0600
Subject: [Freeswitch-users] Side note (READ THIS PLEASE)
In-Reply-To: <CALNZuwkSqA2hcFGeHqPAs2yV=_V-bGM8FcR=+h2qOPtXzzSDsA@mail.gmail.com>
References: <F7E9ADB7-9C86-466E-931A-7732187DE121@freeswitch.org>
	<CALNZuwkSqA2hcFGeHqPAs2yV=_V-bGM8FcR=+h2qOPtXzzSDsA@mail.gmail.com>
Message-ID: <A0627BA5-2CD4-435E-8E90-1D4BB9C72626@freeswitch.org>

it didn't go anywhere fast.
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 29, 2012, at 1:21 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Have you contacted twitter to claim the name?
> http://support.twitter.com/articles/18367-trademark-policy
> 
> -Avi
> 
> On Thu, Nov 29, 2012 at 3:40 AM, Brian West <brian at freeswitch.org> wrote:
> You should follow us on twitter, if you had done so you could have know what was going on during this wondering event for the past few days.
> 
> FOLLOW US NOW ON TWITTER @FreeSWITCH_Wire  (cuz some ass has freeswitch and won't give it up)
> 
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
> 



From steveayre at gmail.com  Thu Nov 29 10:43:43 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 07:43:43 +0000
Subject: [Freeswitch-users] Sofia listens only on 127.0.0.1 after server
 VM reboot?
In-Reply-To: <CA+GmxbbbzfY2NfzOpDVr2A8PZcnYW3rq_orPzKdONwXF5q0SDA@mail.gmail.com>
References: <CA+GmxbaZf+JnnUQMpv1H27xY+NGt3mcKLAaCCvjGrjpL5HOVEw@mail.gmail.com>
	<CCD7A694.610F3%krice@freeswitch.org>
	<CA+GmxbbXeQo+NdCp7eH079Sxw0Q7X6WorpfjXMRXrh=+ohet+A@mail.gmail.com>
	<7B0E64C1-7767-46E4-B881-EC9CB3E707C3@freeswitch.org>
	<CA+GmxbbbzfY2NfzOpDVr2A8PZcnYW3rq_orPzKdONwXF5q0SDA@mail.gmail.com>
Message-ID: <CAFiqYunhnVhtAmJrgkYmD-p+Q94WDXd0NBwicZR7zn8DfEL6NA@mail.gmail.com>

How is the public IP assigned to the interface? DHCP? If so can you set it
statically? The network init.d should normally not finish until the
interface is up, which should mean the freeswitch init.d script isn't run
until it's up.


On 29 November 2012 04:26, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> I'm setting public ip address of my FS server
> I think problem appears when this IP is not on interface yet when FS starts
>
>
> 2012/11/26 Brian West <brian at freeswitch.org>
>
>> What are you setting in your profile as the IP?
>> --
>> Brian West
>> brian at freeswitch.org
>> FreeSWITCH Solutions, LLC
>> PO BOX PO BOX 2531
>> Brookfield, WI 53008-2531
>> Twitter: @FreeSWITCH_Wire
>> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
>> iNUM: +883 5100 1420 9266
>> UK: +44 20 3298 4900
>> ISN: 410*543
>>
>>
>>
>>
>>
>> On Nov 25, 2012, at 12:55 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>> > I tried to set static IP, that doesn't work :(
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Nov 29 10:45:54 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 07:45:54 +0000
Subject: [Freeswitch-users] freeswitch.org - Not Accessable
In-Reply-To: <CAGvuWYbrZ5bd4Tg_Qv2RkgeHjkpQht8WM0OWBWkE_OimLNLWOQ@mail.gmail.com>
References: <CAGvuWYbrZ5bd4Tg_Qv2RkgeHjkpQht8WM0OWBWkE_OimLNLWOQ@mail.gmail.com>
Message-ID: <CAFiqYu=+=5jP9ApfUf300KLj9TxTmu4jZCtzc3U_oLfr7dDNYg@mail.gmail.com>

The server's been under a heavy DDOS attack for a few days. It's looking
like it might be back up today...


On 28 November 2012 22:07, Lloyd Aloysius <lloyd.aloysius at gmail.com> wrote:

> Hello:
>
> I am trying to access freeswitch.org  from Toronto. Three different ISP's.
>
> freeswitch.org is not available?
>
>  Thanks
> Lloyd
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Nov 29 10:56:06 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 07:56:06 +0000
Subject: [Freeswitch-users] mod_db help needed
In-Reply-To: <00c101cdcdeb$b38bbaa0$1aa32fe0$@co.in>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>
	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>
	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>
	<012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>
	<00c101cdcdeb$b38bbaa0$1aa32fe0$@co.in>
Message-ID: <CAFiqYuk9cYxDMyxetKHdxixukk3u1TCpr3GhYBm4Ccxvt1ZkRQ@mail.gmail.com>

Server seems back up today. It's been under a DDOS attack for a few days.

Your main problem is that your db.conf setup is wrong:
      <param name="freeswitch-connector" value="freeswitch:root:
system123#"/>
Should be:
      <param name="odbc-dsn" value="freeswitch:root:system123#"/>

Pretty much every module uses the 'odbc-dsn' parameter name. Any unknown
parameters are ignored. This is why you're seeing no error, FS is never
even seeing the odbc-dsn param has been set. It'll be using the sqlite
files by default.

Fix the file then 'reloadxml' and 'reload_db' to pick up the change without
restarting FS.

If still failing after correcting that...

Have you also installed the MyODBC driver? If you're installing unixodbc
from a package on your distro then it'll likely be there but in a separate
package.

You can confirm that your ODBC DSN is working correctly by using the isql
program at the CLI. If that connects then you know that your unixODBC
settings are correct.

-Steve


On 29 November 2012 04:40, Nitin Tomer <ntomer at newgen.co.in> wrote:

>  Since site was down yesterday, I did some hit and trials myself, made
> the following changes in db.conf.xml ?****
>
> ** **
>
> <configuration name="db.conf" description="LIMIT DB Configuration">
>   <settings>
>       <param name="freeswitch-connector"
> value="freeswitch:root:system123#"/>
>   </settings>
> </configuration>****
>
> ** **
>
> And added this entry in dialplan ?****
>
> ** **
>
> <action application="db" data="insert/testapp/newcall1/${res}" />****
>
> ** **
>
> When I tried running it, I didn?t get any error. But neither any entries
> were made in the database.****
>
> ** **
>
> Please tell me what I am doing wrong.****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nitin Tomer
> *Sent:* Tuesday, November 27, 2012 3:52 PM
> *To:* 'FreeSWITCH Users Help'
>
> *Subject:* Re: [Freeswitch-users] mod_db help needed****
>
>  ** **
>
> Hi Yiftach,****
>
> ** **
>
> Thanks for your reply. I have installed and configured ODBC on my server.
> ****
>
> ** **
>
> /etc/odbc.ini has following entries ?****
>
> ** **
>
> [freesiwtch-connector]****
>
> Description           = MySQL connection to 'freeswitch' database****
>
> Driver                = MySQL****
>
> Database              = freeswitch****
>
> Server                = localhost****
>
> UserName              = root****
>
> Password              = system123****
>
> Port                  = 3306****
>
> Socket                = /var/run/mysqld/mysqld.sock****
>
> ** **
>
> After this I did ./configure and ?make && make install?****
>
> ** **
>
> Now what all changes I need to make in Freeswitch so that I can make ODBC
> work in it? Please help me out.****
>
> ** **
>
> N.B. Today Freeswitch.org is down and therefore I am not able to look for
> any help there****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yiftach
> Golan
> *Sent:* Friday, November 16, 2012 3:03 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_db help needed****
>
> ** **
>
> Hi Nitin,
> Just install odbc on you system, I used this one (
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html)
> but you can use FreeSWITCH's explanations as well (
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core)
> Once you are done the configure should already know that you have odbc
> support, just hit regular configure or as FreeSWITCH docs says : ****
> Compile FreeSWITCH with ODBC support****
>
> UnixODBC support will be autodetected by ./configure, and if found, will
> be compiled into FreeSWITCH. ****
>
> As for the keys it is pretty easy just go to (
> http://wiki.freeswitch.org/wiki/Mod_db) the query should be something
> like : ****
>
> <action application="set" data="var=${db(select/realm/key)}"/>****
>
> The documentation is actually pretty good so try to read it again it will
> make sense
> Thanks,
> Yiftach.****
>
> On Thu, Nov 15, 2012 at 4:51 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:*
> ***
>
> I understand that I will have to make changes in db.conf.xml -****
>
>  ****
>
> <configuration name="db.conf" description="LIMIT DB Configuration">****
>
>   <settings>****
>
>     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->****
>
>   </settings>****
>
> </configuration>****
>
>  ****
>
> But the wiki also says this ? ?ODBC must be configured to use ODBC
> resources (configure with --enable-core-odbc-support).? What does this
> mean? I couldn?t find any more references to it.****
>
>  ****
>
> Wiki says ? ?Realm and key are arbitrary strings. Consider realm as a
> container for keys.? Does this mean that I can use a realm for a particular
> type of data e.g. ?callerdata?, use caller number as key and set the
> entered data against that key. But how would I be able to access this data
> from outside FreeSwitch?****
>
>  ****
>
> Is the realm mapped to a table in the database or something similar?****
>
>  ****
>
> Thanks in advance****
>
>  ****
>
> Nitin****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nitin Tomer
> *Sent:* Thursday, November 15, 2012 3:30 PM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* [Freeswitch-users] mod_db help needed****
>
>  ****
>
> Hi,****
>
>  ****
>
> I am trying to enter some data (entered by caller, received by Play and
> Get Digits). But I am not able to get much information about it from mod_db
> wiki. I want to enter data in an Oracle database. Please tell me what all I
> need to do, in order to do this.****
>
>  ****
>
> Regards****
>
>  ****
>
> Nitin****
>
>  ****
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
>  ****
>
> ** **
>  Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> ** **
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
> ** **
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From steveayre at gmail.com  Thu Nov 29 11:01:41 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 08:01:41 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
Message-ID: <CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>

If you listen to events then you should be able to reliably receive a
hangup event containing the variables *if you are still connected at that
time*.

If you want a 100% reliable method use CDRs.

mod_xml_cdr can submit call CDRs to a webserver just after call hangup. The
XML is verbose and will contain a full call history and all channel
variables. Because it's submitted ASAP you get the results in real time and
reliable. If the module can't submit the CDRs it writes them to an error
folder on disk where you can resubmit them.

Real time seems useful for your use-case, as your script would just need to
check your DB whether the CDR had been submitted yet. The channel UUID can
either be captured from events or specified in advance in the originate
(originate_uuid). There's a uuid api call to generate them for you. I'd
suggest looking at this method.


On 28 November 2012 10:54, Richard Gration <richgration at gmail.com> wrote:

> Hi all,
>
> I'm trying to write something to fit into a two factor auth mechanism.
> I have a CGI script which is called with a phone number and a PIN as
> parameters. In the script I use the event socket api to place a call.
> The PHP code I'm using to write to the socket is straight out of the
> FS docs (can't remember exactly, I think I got it from the wiki). This
> is my api command:
>
> api originate {originate_timeout=30}sofia/external/${dExt}@MGC
> '&play_and_get_digits(4 4 2 30000 # torvalds-says-linux.wav
> torvalds-says-linux.wav foo \d+)'
>
> This all works fine, up to a point. The call is made, the wav is
> played. But then PAGD is supposed to put the digits into a channel
> var, in this case "foo". But I've been having trouble getting the
> value of this channel var *reliably*. If the # key is used to
> terminate the input, then I can get the channel var value with
>
> api uuid_getvar $uuid foo
>
> However, if the input is terminated by PAGD because 4 digits have been
> entered, then the call is hung up (by FS/PAGD) before I have a chance
> to get the value of the channel var. Can anyone help here please? I
> suppose I can work around this by increasing the digits to 5 and
> telling people to use # to terminate the input, but I'd like to
> understand the correct way to use PAGD if possible.
>
> If you think I'm going about this the wrong way entirely, then please
> suggest an alternative :-)
>
> My code is below. As I say, lines 1-71 are straight out of the docs.
>
>   1 <?php
>   2
>   3  $password = "xxxxxx";
>   4  $port = "xxxx";
>   5  $host = "127.0.0.1";
>   6
>   7  function event_socket_create($host, $port, $password) {
>   8       $fp = fsockopen($host, $port, $errno, $errdesc)
>   9          or die("Connection to $host failed");
>  10       socket_set_blocking($fp,false);
>  11
>  12       if ($fp) {
>  13             while (!feof($fp)) {
>  14                 $buffer = fgets($fp, 1024);
>  15                 usleep(100); //allow time for reponse
>  16                 if (trim($buffer) == "Content-Type: auth/request") {
>  17                     fputs($fp, "auth $password\n\n");
>  18                     break;
>  19                 }
>  20             }
>  21             return $fp;
>  22       }
>  23       else {
>  24             return false;
>  25       }
>  26  }
>  27
>  28  function event_socket_request($fp, $cmd) {
>  29
>  30       if ($fp) {
>  31             fputs($fp, $cmd."\n\n");
>  32             usleep(200); //allow time for reponse
>  33
>  34             $response = "";
>  35             $i = 0;
>  36             $contentlength = 0;
>  37             while (!feof($fp)) {
>  38                 $buffer = fgets($fp, 4096);
>  39                 if ($contentlength > 0) {
>  40                     $response .= $buffer;
>  41                 }
>  42
>  43                 if ($contentlength == 0) { //if contentlenght is
> already don't process again
>  44                      if (strlen(trim($buffer)) > 0) { //run only
> if buffer has content
>  45                           $temparray = split(":", trim($buffer));
>  46                           if ($temparray[0] == "Content-Length") {
>  47                               $contentlength = trim($temparray[1]);
>  48                           }
>  49                      }
>  50                 }
>  51
>  52                 usleep(200); //allow time for reponse
>  53
>  54                 //optional because of script timeout //don't let
> while loop become endless
>  55                 if ($i > 90000) { break; }
>  56
>  57                 if ($contentlength > 0) { //is contentlength set
>  58                      //stop reading if all content has been read.
>  59                      if (strlen($response) >= $contentlength) {
>  60                          break;
>  61                      }
>  62                 }
>  63                 $i++;
>  64             }
>  65
>  66             return $response;
>  67       }
>  68       else {
>  69          echo "no handle";
>  70       }
>  71  }
>  72
>  73  function strip_cr($str) {
>  74     return preg_replace("/\n/",'',$str);
>  75  }
>  76
>  77  $dExt = $_GET['dExt'];
>  78  if ($dExt) {
>  79     $fp = event_socket_create($host, $port, $password);
>  80
>  81     # Place call and find uuid
>  82     $cmd = "api originate
> {originate_timeout=30}sofia/external/677${dExt}@194.145.191.131
> '&play_and_get_digits(4 4 2 30000 # torvalds-says-linux.wav
> torvalds-says-linux.wav foo \d+)'";
>  83     $response = event_socket_request($fp, $cmd);
>  84     $response = preg_replace('/</','&lt;',$response);
>  85     $response = preg_replace('/>/','&gt;',$response);
>  86     $response = strip_cr($response);
>  87     $bits = preg_split('/\s+/',$response);
>  88     $uuid = $bits[1];
>  89
>  90     echo '<pre>';
>  91     echo date('H:i:s') . "\n";
>  92     echo "call placed: $cmd\n";
>  93     echo "response :$response:\n";
>  94     echo "uuid $uuid\n";
>  95
>  96     # Retrieve channel variable
>  97     $cmd = "api uuid_getvar $uuid foo";
>  98     $response = '';
>  99     while (! $response or $response == '_undef_') {
> 100         echo "in loop : got _undef_\n";
> 101         $response = event_socket_request($fp, $cmd);
> 102         $response = strip_cr($response);
> 103     }
> 104     echo "response :$response:\n";
> 105     echo '</pre>';
> 106
> 107     # Close socket
> 108     fclose($fp);
> 109  } else {
> 110     echo 'Request URL in this format: /test.php?dExt=<phone_number>' .
> "\n";
> 111  }
> 112
> 113  ?>
>
>
> Cheers,
> Rich
>
> --
> Once our basic material needs are met - in my utopia, anyway - life
> becomes a perpetual celebration in which everyone has a talent to
> contribute. But we cannot levitate ourselves into that blessed
> condition by wishing it. We need to brace ourselves for a struggle
> against terrifying obstacles, both of our own making and imposed by
> the natural world. And the first step is to recover from the delusion
> that is positive thinking.
>        -- Barbara Ehrenreich
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From clive18 at webmail.co.za  Thu Nov 29 11:08:12 2012
From: clive18 at webmail.co.za (clive engelberg)
Date: Thu, 29 Nov 2012 10:08:12 +0200
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>
Message-ID: <6036422a0e8b5982ed7af8823015bb38@www.webmail.co.za>

Hi 

 I have had limited success with : 

 Sometimes the sound was garbled, sometimes worked fine, not sure why it was
sometimes garbled, so I dont use it anymore, as complaints are not what I
want :) 

 Hope this helps. 

 regards 

 Clive 

 On Tue, 27 Nov 2012 11:06:18 +0000 Ben N wrote

 Hi All, I'm wondering if anyone out there is using G729 with 50ms ptime
through Freeswitch, without proxy media?
 I am unable to get this to work, even in a very basic lab with just two
clients and an up to date FS server on the same LAN. I can get G729 with 20ms
working, but 50ms causes Freeswitch to hang the call.
 It might be that mod_g729 and mod_com_g729 simply don't support it, so if
anyone in the know can tell me that would be great!
 Further info can be supplied, I believe I have tried pretty much all
options....
 Cheers,
 Ben
 

____________________________________________________________
South Africas premier free email service - www.webmail.co.za 

DHL Express. Simple, secure and affordable.
http://www.postnet.co.za/index.php?option=com_content&view=article&id=74&Itemid=72
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From gregor at infomedia.si  Thu Nov 29 11:36:25 2012
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 29 Nov 2012 09:36:25 +0100
Subject: [Freeswitch-users] Inbound DID trunk without authentication
Message-ID: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>

Hi!

I have DID provider and give me only IP from where call will come. So I
need to configure gateway without authentication. I read several posts and
h ere is what have I done

in acl.conf.xml i add:

<list name="testdid" default="deny">
<node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
</list>

then in dialplan\public\00_inbound_did.xml I add
<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(mydidnumber)$">
      <action application="set" data="domain_name=$${domain}"/>
      <!-- This example maps the DID 5551212 to ring 1000 in the default
context -->
      <action application="transfer" data="1000 XML default"/>
    </condition>
  </extension>
</include>

What I see in console I keep getting rejected by ACL "domains"

If I understand correctly if IP of inbound call is in ACL then call is
transfered to public dialplan?

I need this gateway only for inbound calls.

Please, any suggestions?
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From ntomer at newgen.co.in  Thu Nov 29 11:52:16 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 29 Nov 2012 14:22:16 +0530
Subject: [Freeswitch-users] mod_db help needed
In-Reply-To: <CAFiqYuk9cYxDMyxetKHdxixukk3u1TCpr3GhYBm4Ccxvt1ZkRQ@mail.gmail.com>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>	<012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>	<00c101cdcdeb$b38bbaa0$1aa32fe0$@co.in>
	<CAFiqYuk9cYxDMyxetKHdxixukk3u1TCpr3GhYBm4Ccxvt1ZkRQ@mail.gmail.com>
Message-ID: <019801cdce0e$d5eb6410$81c22c30$@co.in>

Thanks Steven.

 

I changed it to <param name="odbc-dsn"
value="freeswitch-connector:root:system123#"/> and now it's working fine. 

 

Now I will work on call parking and picking. I will be using valet_park for
that.

 

Thanks again

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Thursday, November 29, 2012 1:26 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_db help needed

 

Server seems back up today. It's been under a DDOS attack for a few days.

 

Your main problem is that your db.conf setup is wrong:

      <param name="freeswitch-connector"
value="freeswitch:root:system123#"/>

Should be:

      <param name="odbc-dsn" value="freeswitch:root:system123#"/>

 

Pretty much every module uses the 'odbc-dsn' parameter name. Any unknown
parameters are ignored. This is why you're seeing no error, FS is never even
seeing the odbc-dsn param has been set. It'll be using the sqlite files by
default.

 

Fix the file then 'reloadxml' and 'reload_db' to pick up the change without
restarting FS.

 

If still failing after correcting that...

 

Have you also installed the MyODBC driver? If you're installing unixodbc
from a package on your distro then it'll likely be there but in a separate
package.

 

You can confirm that your ODBC DSN is working correctly by using the isql
program at the CLI. If that connects then you know that your unixODBC
settings are correct.

 

-Steve

 

On 29 November 2012 04:40, Nitin Tomer <ntomer at newgen.co.in> wrote:

Since site was down yesterday, I did some hit and trials myself, made the
following changes in db.conf.xml -

 

<configuration name="db.conf" description="LIMIT DB Configuration">
  <settings>

      <param name="freeswitch-connector"
value="freeswitch:root:system123#"/>
  </settings>
</configuration>

 

And added this entry in dialplan -

 

<action application="db" data="insert/testapp/newcall1/${res}" />

 

When I tried running it, I didn't get any error. But neither any entries
were made in the database.

 

Please tell me what I am doing wrong.

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Tuesday, November 27, 2012 3:52 PM
To: 'FreeSWITCH Users Help'


Subject: Re: [Freeswitch-users] mod_db help needed

 

Hi Yiftach,

 

Thanks for your reply. I have installed and configured ODBC on my server. 

 

/etc/odbc.ini has following entries -

 

[freesiwtch-connector]

Description           = MySQL connection to 'freeswitch' database

Driver                = MySQL

Database              = freeswitch

Server                = localhost

UserName              = root

Password              = system123

Port                  = 3306

Socket                = /var/run/mysqld/mysqld.sock

 

After this I did ./configure and "make && make install"

 

Now what all changes I need to make in Freeswitch so that I can make ODBC
work in it? Please help me out.

 

N.B. Today Freeswitch.org is down and therefore I am not able to look for
any help there

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yiftach
Golan
Sent: Friday, November 16, 2012 3:03 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_db help needed

 

Hi Nitin,
Just install odbc on you system, I used this one
(http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/install
ing_configuring_odbc.html) but you can use FreeSWITCH's explanations as well
(http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core)
Once you are done the configure should already know that you have odbc
support, just hit regular configure or as FreeSWITCH docs says : 


Compile FreeSWITCH with ODBC support


UnixODBC support will be autodetected by ./configure, and if found, will be
compiled into FreeSWITCH. 

As for the keys it is pretty easy just go to
(http://wiki.freeswitch.org/wiki/Mod_db) the query should be something like
: 

<action application="set" data="var=${db(select/realm/key)}"/>

The documentation is actually pretty good so try to read it again it will
make sense
Thanks,
Yiftach.

On Thu, Nov 15, 2012 at 4:51 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

I understand that I will have to make changes in db.conf.xml -

 

<configuration name="db.conf" description="LIMIT DB Configuration">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

  </settings>

</configuration>

 

But the wiki also says this - "ODBC must be configured to use ODBC resources
(configure with --enable-core-odbc-support)." What does this mean? I
couldn't find any more references to it.

 

Wiki says - "Realm and key are arbitrary strings. Consider realm as a
container for keys." Does this mean that I can use a realm for a particular
type of data e.g. "callerdata", use caller number as key and set the entered
data against that key. But how would I be able to access this data from
outside FreeSwitch?

 

Is the realm mapped to a table in the database or something similar?

 

Thanks in advance

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Thursday, November 15, 2012 3:30 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_db help needed

 

Hi,

 

I am trying to enter some data (entered by caller, received by Play and Get
Digits). But I am not able to get much information about it from mod_db
wiki. I want to enter data in an Oracle database. Please tell me what all I
need to do, in order to do this.

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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

 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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

 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From vbvbrj at gmail.com  Thu Nov 29 11:55:33 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 29 Nov 2012 10:55:33 +0200
Subject: [Freeswitch-users] a=sillenceSupp: off
Message-ID: <50B72305.9020503@gmail.com>

Hello.

The provider asks to eliminate this option from SDP packet in order to 
work correctly connection. How to do this? I've found how to set this 
option to "off" but how to eliminate this completely from SDP?

Thank you.

-- 
Mimiko desu.


From steveayre at gmail.com  Thu Nov 29 12:26:10 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 11:26:10 +0200
Subject: [Freeswitch-users] Side note (READ THIS PLEASE)
In-Reply-To: <F7E9ADB7-9C86-466E-931A-7732187DE121@freeswitch.org>
References: <F7E9ADB7-9C86-466E-931A-7732187DE121@freeswitch.org>
Message-ID: <CAFiqYu=twb22bcD=6ADvGvw3gRvW=rsObZ26iZE57MaYxtyTVw@mail.gmail.com>

Did we find out out who was doing the DDOS?

Have they actually gone away, or are we just coping with it better now?


On 29 November 2012 03:40, Brian West <brian at freeswitch.org> wrote:

> You should follow us on twitter, if you had done so you could have know
> what was going on during this wondering event for the past few days.
>
> FOLLOW US NOW ON TWITTER @FreeSWITCH_Wire  (cuz some ass has freeswitch
> and won't give it up)
>
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From asilva at wirelessmundi.com  Thu Nov 29 12:28:15 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Thu, 29 Nov 2012 10:28:15 +0100
Subject: [Freeswitch-users] sqlite3 and regex
In-Reply-To: <1413611559183349171@unknownmsgid>
References: <CCB9BB87.5EE38%krice@freeswitch.org>
	<1354119909.13048.56.camel@marces.madrid.commsmundi.com>
	<1413611559183349171@unknownmsgid>
Message-ID: <1354181295.13048.82.camel@marces.madrid.commsmundi.com>

Hi,

Last night came up with this patch:

http://jira.freeswitch.org/browse/FS-4883

Is working in my fs, this way i can use REGEXP in my lua scripts :)




On Thu, 2012-11-29 at 01:19 -0500, S. Scott wrote: 

> That's EXACTLY what I'm hunting for!  My freeswitch build doesn't have
> a pcre.so anywhere on the disk.  The .../freeswitch/libs/pcre/ exists
> with many files but no .so.  
> 
> 
> What does one need to do to get the sqlite3 and the pcre.so rolled
> (from the git ingredients)?  make() and stuff not my strong suit.
> 
> Thanks,
> ?????
> iThing:  Big thumbs & little keys. Please excuse typo, spelling and
> grammar errors ? Last night I played a blank CD at full blast. The
> Mime next door went nuts.
> 
> 
> 
> On 2012-11-28, at 23:53, Antonio <asilva at wirelessmundi.com> wrote:
> 
> 
> 
> 
> > Hi,
> > 
> > Coudn't we load just an external extension to sqlite?
> > 
> > for example, to have regular expression support you could load an
> > external library like "sqlite3-pcre" (available for unbuntu), you
> > can load from the console interface or directly in sql.
> > 
> > console:
> > sqlite> .load '/usr/lib/sqlite3/pcre.so
> > 
> > sql in console:
> > sqlite> select load_extension(''/usr/lib/sqlite3/pcre.so'); 
> > 
> > 
> > In fs is not possible, when i try to do it from my lua script i have
> > the following error:
> > 
> > 2012-11-28 17:21:34.196304 [ERR] switch_core_sqldb.c:572 NATIVE SQL
> > ERR [no such function: load_extension]
> > select load_extension('/usr/lib/sqlite3/pcre.so');
> > 
> > 
> > Since you have already a switch type of db ( core, odbc or pgsql)
> > could be nice when using core, be able to use a few more functions
> > available in sqlite?
> > 
> > 
> > Thanks,
> > Ant?nio
> > 
> > 
> > 
> > 
> > 
> > 
> > On Fri, 2012-11-02 at 18:03 -0500, Ken Rice wrote:
> > 
> > > No and this wont happen anytime soon... The SQL interfaces for
> > > FreeSwitch are kept generic as we support more then just sqlite
> > > from common code and if we did it for sqlite we would have to make
> > > sure its implemented equally well for postgresql and mysql and
> > > mssql and any other database someone might want to use via ODBC
> > > 
> > > K
> > > 
> > > 
> > > On 11/2/12 3:34 PM, "Scott" <8f27e956 at gmail.com> wrote:
> > > 
> > > 
> > >         "LIKE" notwithstanding, sqlite3 does not have a built-in
> > >         true regex function; it does allow for a a c-language hook
> > >         to one.  Given fs extensive use of the regex engine and of
> > >         sqlite3, we're wondering if the hook is already written
> > >         and rolled.  If so, can the rest of us hook it to our
> > >         sqlite3 uses (e.g. from dial plan lua sqlite3).
> > >         
> > >         With thanks,
> > >         
> > >         
> > >         
> > >         __________________________________________________________
> > >         
> > >         
> > >         
> > >         _________________________________________________________________________
> > >         Professional FreeSWITCH Consulting Services:
> > >         consulting at freeswitch.org
> > >         http://www.freeswitchsolutions.com
> > >         
> > >         FreeSWITCH-powered IP PBX: The CudaTel Communication
> > >         Server
> > >         
> > >         
> > >         Official FreeSWITCH Sites
> > >         http://www.freeswitch.org
> > >         http://wiki.freeswitch.org
> > >         http://www.cluecon.com
> > >         
> > >         FreeSWITCH-users mailing list
> > >         FreeSWITCH-users at lists.freeswitch.org
> > >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >         http://www.freeswitch.org
> > > 
> > > 
> > > -- 
> > > Ken
> > > http://www.FreeSWITCH.org
> > > http://www.ClueCon.com
> > > http://www.OSTAG.org
> > > irc.freenode.net #freeswitch 
> > > 
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > > 
> > > 
> > > 
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > 
> > -- 
> > 
> > Un cordial saludo / Best regards, 
> > 
> >  _________________________
> > 
> > Ant?nio Silva
> > 
> > E-mail:asilva at wirelessmundi.com
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From NuwanW at unifybusiness.co.uk  Thu Nov 29 12:30:42 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Thu, 29 Nov 2012 09:30:42 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D21F7@BARXB0003.UnifyBusiness.local>

Hi All,
*         I have created a JIRA case for this. The case number is FS-4884.
Thank you,
Nuwan
This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From anton.jugatsu at gmail.com  Thu Nov 29 12:48:44 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 29 Nov 2012 13:48:44 +0400
Subject: [Freeswitch-users] Inbound DID trunk without authentication
In-Reply-To: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
References: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
Message-ID: <CANBhyPB9E2-4T1knH+aaXR2-+Hc0w7tJxggmayT8Y6yWQDUL6Q@mail.gmail.com>

What acl do you use @ external profile? Also, don't forget to reloadacl.


2012/11/29 Gregor Nanger <gregor at infomedia.si>

> Hi!
>
> I have DID provider and give me only IP from where call will come. So I
> need to configure gateway without authentication. I read several posts and
> h ere is what have I done
>
> in acl.conf.xml i add:
>
> <list name="testdid" default="deny">
> <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
> </list>
>
> then in dialplan\public\00_inbound_did.xml I add
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(mydidnumber)$">
>       <action application="set" data="domain_name=$${domain}"/>
>       <!-- This example maps the DID 5551212 to ring 1000 in the default
> context -->
>       <action application="transfer" data="1000 XML default"/>
>     </condition>
>   </extension>
> </include>
>
> What I see in console I keep getting rejected by ACL "domains"
>
> If I understand correctly if IP of inbound call is in ACL then call is
> transfered to public dialplan?
>
> I need this gateway only for inbound calls.
>
> Please, any suggestions?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gregor at infomedia.si  Thu Nov 29 13:12:25 2012
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 29 Nov 2012 11:12:25 +0100
Subject: [Freeswitch-users] Inbound DID trunk without authentication
In-Reply-To: <CANBhyPB9E2-4T1knH+aaXR2-+Hc0w7tJxggmayT8Y6yWQDUL6Q@mail.gmail.com>
References: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
	<CANBhyPB9E2-4T1knH+aaXR2-+Hc0w7tJxggmayT8Y6yWQDUL6Q@mail.gmail.com>
Message-ID: <CAC7ttFFiCL0ouqx4sq+8Y6bxotqQ=Gc3pLwdwr7nye74wT5ztw@mail.gmail.com>

Can you please give me more information. Aren't ACLs only in one place for
all profiles?


Best regards, Gregor


2012/11/29 Anton Kvashenkin <anton.jugatsu at gmail.com>

> What acl do you use @ external profile? Also, don't forget to reloadacl.
>
>
> 2012/11/29 Gregor Nanger <gregor at infomedia.si>
>
>> Hi!
>>
>> I have DID provider and give me only IP from where call will come. So I
>> need to configure gateway without authentication. I read several posts and
>> h ere is what have I done
>>
>> in acl.conf.xml i add:
>>
>> <list name="testdid" default="deny">
>> <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>> </list>
>>
>> then in dialplan\public\00_inbound_did.xml I add
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^(mydidnumber)$">
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <!-- This example maps the DID 5551212 to ring 1000 in the default
>> context -->
>>       <action application="transfer" data="1000 XML default"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> What I see in console I keep getting rejected by ACL "domains"
>>
>> If I understand correctly if IP of inbound call is in ACL then call is
>> transfered to public dialplan?
>>
>> I need this gateway only for inbound calls.
>>
>> Please, any suggestions?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ntomer at newgen.co.in  Thu Nov 29 13:15:35 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 29 Nov 2012 15:45:35 +0530
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <CAFiqYuk9cYxDMyxetKHdxixukk3u1TCpr3GhYBm4Ccxvt1ZkRQ@mail.gmail.com>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>	<012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>	<00c101cdcdeb$b38bbaa0$1aa32fe0$@co.in>
	<CAFiqYuk9cYxDMyxetKHdxixukk3u1TCpr3GhYBm4Ccxvt1ZkRQ@mail.gmail.com>
Message-ID: <01f101cdce1a$79309130$6b91b390$@co.in>

Hi,

 

I am using valet_park. I've configure a IVR menu of an extension, based on
user's input call is forwarded to other extensions.

 

Extension on which end-users will call -

 

<extension name="ivr_nitin">
      <condition field="destination_number" expression="^5002$">
        <action application="answer"/>
        <action application="sleep" data="500"/>
        <action application="ivr" data="nitin_ivr"/>
      </condition>
</extension>

 

The IVR configuration XML is -

 

<include>
        <menu name="nitin_ivr"
                greet-long="say:Welcome to Newgen General Insurance Company.
Press 1 for Changing Address, 2 for Changing Nominee or 3 for Close Policy."
                greet-short="say:Welcome to Newgen. Press 1 for Changing
Address, 2 for Changing Nominee or 3 for Close Policy."
                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
                exit-sound="voicemail/vm-goodbye.wav"
                confirm-macro=""
                confirm-key=""
                tts-engine="flite"
                tts-voice="slt"
                confirm-attempts="3"
                timeout="3000"
                inter-digit-timeout="2000"
                max-failures="3"
                max-timeouts="3"
                digit-len="4">
 
                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Change Address -->
                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Change Nominee -->
                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Close Policy -->

        </menu>
</include>



Once user presses "1", call is forwarded to 450, for this extension dialplan
entry is -

 

<extension>
          <condition field="destination_number" expression="^(450)$">
          <action application="play_and_get_digits" data="10 16 3 3000 #
say:'Press your account number, followed by hash key' say:'Wrong Input' res
\d+" />
          <action application="phrase" data="spell,${res}"/>
          <action application="db" data="insert/testapp/newcall1/${res}" />

              <action application="set" data="caller_id_name=Change Address
Request" />
              <action application="set" data="call_timeout=60" />
              <action application="set" data="originate_timeout=60" />
              <action application="valet_park" data="my_lot auto in 8501
8599" />           

          </condition>
</extension>

 

Here, the call is parked at any available extension between 8501 to 8599.

 

Then I've set up an extension to pick up calls -

 

<extension name="park-out">
   <condition field="destination_number" expression="^(85\d\d)$">
     <action application="answer"/>
     <action application="valet_park" data="my_lot $1"/>
   </condition>
 </extension>

 

I have a few questions -

 

1.       Valet_park parks the call on any available extension between 8501
to 8599 (<action application="valet_park" data="my_lot auto in 8501 8599"
/>). Is there any way to let me know on which extension the call have been
parked?

2.       How can I get the number from which call was made in extension 450.
The idea is to use the caller number as key and entered value as value while
making entry in database (<action application="db"
data="insert/testapp/newcall1/${res}" />)?

3.       If two users call on extension 5002 (where IVR menu is played),
what will happen? Will the second user have to wait for first to finish or
whether both will be connected parallel?

 

Please help me out.

 

Regards

 

Nitin


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From paul at cupis.co.uk  Thu Nov 29 13:32:19 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Thu, 29 Nov 2012 10:32:19 +0000
Subject: [Freeswitch-users] Inbound DID trunk without authentication
In-Reply-To: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
References: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
Message-ID: <20121129103218.GA10306@eagle.cupis.co.uk>

On Thu, Nov 29, 2012 at 09:36:25AM +0100, Gregor Nanger wrote:
>    in acl.conf.xml i add:
> 
>    <list name="testdid" default="deny">
>    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>    </list>

>    What I see in console I keep getting rejected by ACL "domains"
> 
>    If I understand correctly if IP of inbound call is in ACL then call is
>    transfered to public dialplan?
> 
>    I need this gateway only for inbound calls.

Try adding:

   <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>

into the existing stanza in acl.conf.xml which starts with:

   <list name="domains" ...

as this is the ACL which your SIP profile seems to be using to authenticate
incoming calls.

Regards,



From ssoni at lifesize.com  Thu Nov 29 13:52:18 2012
From: ssoni at lifesize.com (Sanjay Soni)
Date: Thu, 29 Nov 2012 04:52:18 -0600
Subject: [Freeswitch-users] Inbound DID trunk without authentication
In-Reply-To: <20121129103218.GA10306@eagle.cupis.co.uk>
References: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
	<20121129103218.GA10306@eagle.cupis.co.uk>
Message-ID: <F3C02C5E8650B74FA0C199FA1A01D26E169A3A1A89@ausmsex00.austin.kmvtechnologies.com>

I was getting similar problem and modifying acl.conf didn't work for me. What work for me was modifying sip_profiles/internal.xml and sip_profiles/external.xml with Either 
param name="local-network-acl" value="localnet.auto" - For Private IPs (This is default)
Or
param name="candidate-acl" value="wan.auto"
param name="local-network-acl" value="localnet.auto" - For Public IPs. 



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis
Sent: 29 November 2012 16:02
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Inbound DID trunk without authentication

On Thu, Nov 29, 2012 at 09:36:25AM +0100, Gregor Nanger wrote:
>    in acl.conf.xml i add:
> 
>    <list name="testdid" default="deny">
>    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>    </list>

>    What I see in console I keep getting rejected by ACL "domains"
> 
>    If I understand correctly if IP of inbound call is in ACL then call is
>    transfered to public dialplan?
> 
>    I need this gateway only for inbound calls.

Try adding:

   <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>

into the existing stanza in acl.conf.xml which starts with:

   <list name="domains" ...

as this is the ACL which your SIP profile seems to be using to authenticate
incoming calls.

Regards,


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




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

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


From gregor at infomedia.si  Thu Nov 29 14:15:01 2012
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 29 Nov 2012 12:15:01 +0100
Subject: [Freeswitch-users] Inbound DID trunk without authentication
In-Reply-To: <F3C02C5E8650B74FA0C199FA1A01D26E169A3A1A89@ausmsex00.austin.kmvtechnologies.com>
References: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
	<20121129103218.GA10306@eagle.cupis.co.uk>
	<F3C02C5E8650B74FA0C199FA1A01D26E169A3A1A89@ausmsex00.austin.kmvtechnologies.com>
Message-ID: <CAC7ttFF-qkxB7UKp-Ca3pTF8oQwsvcYPt4=m_wghjF7w85qydQ@mail.gmail.com>

I thought that just set acl and set in /sip_profiles/external.xml <param
name="apply-inbound-acl" value="testdid"/> would solve this, but it doesn't
working.

Trunk DID provider I am using is sending calls to FS. Now I just want to
get this calls into public dialplan based only on IP, not authorization. :-(



2012/11/29 Sanjay Soni <ssoni at lifesize.com>

> I was getting similar problem and modifying acl.conf didn't work for me.
> What work for me was modifying sip_profiles/internal.xml and
> sip_profiles/external.xml with Either
> param name="local-network-acl" value="localnet.auto" - For Private IPs
> (This is default)
> Or
> param name="candidate-acl" value="wan.auto"
> param name="local-network-acl" value="localnet.auto" - For Public IPs.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis
> Sent: 29 November 2012 16:02
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Inbound DID trunk without authentication
>
> On Thu, Nov 29, 2012 at 09:36:25AM +0100, Gregor Nanger wrote:
> >    in acl.conf.xml i add:
> >
> >    <list name="testdid" default="deny">
> >    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
> >    </list>
>
> >    What I see in console I keep getting rejected by ACL "domains"
> >
> >    If I understand correctly if IP of inbound call is in ACL then call is
> >    transfered to public dialplan?
> >
> >    I need this gateway only for inbound calls.
>
> Try adding:
>
>    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>
> into the existing stanza in acl.conf.xml which starts with:
>
>    <list name="domains" ...
>
> as this is the ACL which your SIP profile seems to be using to authenticate
> incoming calls.
>
> Regards,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.venugopan at mundio.com  Thu Nov 29 14:15:50 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 29 Nov 2012 11:15:50 +0000
Subject: [Freeswitch-users] mod_voicemail compilation error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2337941@Mail-Kilo.squay.com>
References: <1FFF97C269757C458224B7C895F35F151CAFE9@cantor.std.visionutv.se>
	<592A9CF93E12394E8472A6CC66E66BF2337941@Mail-Kilo.squay.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2338E30@Mail-Kilo.squay.com>

Even if i rebuild initially it was working, but when I change the c code and compile am getting the same error and voicemail stops working. I have taken the source code which is currently running in our production and did a build.
The version we are using is FreeSWITCH Version 1.0.head (git-a0a77f8 2011-12-15 12-23-53 -0500). Since this is the one that is running in production I can't install the latest version too.

Please help me out in resolving this issue in the version that am using. Thanks


Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan
Sent: 23 November 2012 12:25
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail compilation error

Hi,
Thanks. But it worked when the freeswitch was build. After making changes and do a compile this is not supporting and giving this error.
The same libfreeswitch.so might have supported switch_channel_expand_variables_check() during initial build as well right?

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 23 November 2012 10:46
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_voicemail compilation error

Rebuild the entire project instead. It seems that the current libfreeswitch.so doesn't support switch_channel_expand_variables_check().

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Archana Venugopan
Skickat: den 23 november 2012 11:03
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] mod_voicemail compilation error

Hi,

I tried putting debug statements in mod_voicemail.c file and tried to re-compile it but I got this msg and now its not going to voicemail. Earlier it was working properly but after this voicemail was not working. Please help.

[root at squay-laptop-1 freeswitch]# make mod_voicemail-install
 /bin/sh /usr/local/src/freeswitch/quiet_libtool   --mode=install /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai /usr/local/freeswitch/lib/libfreeswitch.la
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: finish: PATH="/usr/lib/qt-3.3/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/sbin" ldconfig -n /usr/local/freeswitch/lib
----------------------------------------------------------------------
Libraries have been installed in:
   /usr/local/freeswitch/lib

If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
     during execution
   - add LIBDIR to the `LD_RUN_PATH' environment variable
     during linking
   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
   - have your system administrator add LIBDIR to `/etc/ld.so.conf'

See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
----------------------------------------------------------------------

making install mod_voicemail
Compiling /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c...
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c  -fPIC -DPIC -o .libs/mod_voicemail.o
quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c -o mod_voicemail.o >/dev/null 2>&1
Creating mod_voicemail.la...
installing mod_voicemail.la
quiet_libtool: install: warning: relinking `mod_voicemail.la'


freeswitch at internal> reload mod_voicemail
+OK module unloaded
+OK Reloading XML
-ERR loading module [module load file routine returned an error]

2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:901 Deleting Application 'voicemail'
freeswitch at internal> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:903 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'voicemail_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'voicemail_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_inject'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_inject' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_boxcount'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_boxcount' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_prefs'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_prefs' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_read'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_read' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_auth_login'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_auth_login' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_count'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_count' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_list'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_list' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_get'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_get' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_delete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_delete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_undelete'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_undelete' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_purge'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_purge' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_save'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_save' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_msg_forward'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_msg_forward' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_greeting_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_greeting_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_recname_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_recname_set' to wait for existing references.
2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting API Function 'vm_fsdb_pref_password_set'
2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock interface 'vm_fsdb_pref_password_set' to wait for existing references.
2012-11-23 09:58:31.448059 [CONSOLE] switch_loadable_module.c:1765 Stopping: mod_voicemail
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:442 Subclass reservation deleted for /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:vm::maintenance
2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:1889 Event Binding deleted for mod_voicemail:MESSAGE_QUERY
2012-11-23 09:58:31.518046 [CONSOLE] mod_voicemail.c:3840 Event Thread Ended
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5759 Waiting for write lock (Profile default)
2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5762 Destroying Profile default
2012-11-23 09:58:31.518046 [CONSOLE] switch_loadable_module.c:1785 mod_voicemail unloaded.
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 1
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 44
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding tone_descriptor: 49
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: CED_TONE(0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: SIT(1)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-11-23 09:58:32.768049 [INFO] mod_enum.c:775 ENUM Reloaded
2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so
**/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol: switch_channel_expand_variables_check**
2012-11-23 09:58:32.778219 [INFO] switch_time.c:1035 Timezone reloaded 530 definitions



Regards,
Archana

!DSPAM:50af4e1232761560677559!
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From cal.leeming at simplicitymedialtd.co.uk  Thu Nov 29 14:40:30 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 29 Nov 2012 11:40:30 +0000
Subject: [Freeswitch-users] Inbound DID trunk without authentication
In-Reply-To: <CAC7ttFF-qkxB7UKp-Ca3pTF8oQwsvcYPt4=m_wghjF7w85qydQ@mail.gmail.com>
References: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
	<20121129103218.GA10306@eagle.cupis.co.uk>
	<F3C02C5E8650B74FA0C199FA1A01D26E169A3A1A89@ausmsex00.austin.kmvtechnologies.com>
	<CAC7ttFF-qkxB7UKp-Ca3pTF8oQwsvcYPt4=m_wghjF7w85qydQ@mail.gmail.com>
Message-ID: <CAHKQagExjduWDO83PAEV5BZGYGitiEbgFQRwQS8yjkY5iAj2rg@mail.gmail.com>

Hi Gregor,

I have to admit, every time I've tried using ACLs for inbound
authentication (i.e. an inbound DID), it always gave me the same damn error
(rejected by ACL 'domains' etc).

I've searched endlessly for a reason why, but couldn't seem to find a
combination of variables that worked.

I figured that it must be working else everyone would be complaining, and
that my own lack of knowledge/experience with FreeSWITCH meant I was
getting something fundamentally wrong in the config.

So, I removed ACLs, threw up an iptables rule and put a TODO in the project
plan to fix ACLs before release lol.

I'll have another look at this, and see if I can figure out wtf I'm doing
wrong.

This may/may not be the same problem your having, but I'll report back
either way and hopefully it'll help!

Cal


On Thu, Nov 29, 2012 at 11:15 AM, Gregor Nanger <gregor at infomedia.si> wrote:

> I thought that just set acl and set in /sip_profiles/external.xml <param
> name="apply-inbound-acl" value="testdid"/> would solve this, but it
> doesn't working.
>
> Trunk DID provider I am using is sending calls to FS. Now I just want to
> get this calls into public dialplan based only on IP, not authorization. :-(
>
>
>
> 2012/11/29 Sanjay Soni <ssoni at lifesize.com>
>
>> I was getting similar problem and modifying acl.conf didn't work for me.
>> What work for me was modifying sip_profiles/internal.xml and
>> sip_profiles/external.xml with Either
>> param name="local-network-acl" value="localnet.auto" - For Private IPs
>> (This is default)
>> Or
>> param name="candidate-acl" value="wan.auto"
>> param name="local-network-acl" value="localnet.auto" - For Public IPs.
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis
>> Sent: 29 November 2012 16:02
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Inbound DID trunk without authentication
>>
>> On Thu, Nov 29, 2012 at 09:36:25AM +0100, Gregor Nanger wrote:
>> >    in acl.conf.xml i add:
>> >
>> >    <list name="testdid" default="deny">
>> >    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>> >    </list>
>>
>> >    What I see in console I keep getting rejected by ACL "domains"
>> >
>> >    If I understand correctly if IP of inbound call is in ACL then call
>> is
>> >    transfered to public dialplan?
>> >
>> >    I need this gateway only for inbound calls.
>>
>> Try adding:
>>
>>    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>>
>> into the existing stanza in acl.conf.xml which starts with:
>>
>>    <list name="domains" ...
>>
>> as this is the ACL which your SIP profile seems to be using to
>> authenticate
>> incoming calls.
>>
>> Regards,
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tomasz.szuster at gmail.com  Thu Nov 29 14:47:26 2012
From: tomasz.szuster at gmail.com (Tomasz Szuster)
Date: Thu, 29 Nov 2012 12:47:26 +0100
Subject: [Freeswitch-users] predictive dialer.
In-Reply-To: <CAMAX9-0hYx=NuB_aWEZohDKVF=3VYWjd0LU9khmPpJsaOR=1iQ@mail.gmail.com>
References: <CAMAX9-0hYx=NuB_aWEZohDKVF=3VYWjd0LU9khmPpJsaOR=1iQ@mail.gmail.com>
Message-ID: <CAMAX9-3n=Dk9uNcb642A9vc4kpvGv6tJqvfFOKiGW3gy9dYCrA@mail.gmail.com>

Hello,
Do you know any good predictive dialer which you can recomend for the
freeswitch ?

-- 
Regards
Tom
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From richgration at gmail.com  Thu Nov 29 14:58:58 2012
From: richgration at gmail.com (Richard Gration)
Date: Thu, 29 Nov 2012 11:58:58 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
	<CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
Message-ID: <CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>

Hi,

Thanks for your reply. I was wondering ...

> If you listen to events then you should be able to reliably receive a hangup
> event containing the variables if you are still connected at that time.

... how I listen for these events. Is it just a case of polling the
socket for them at frequent enough intervals?

> If you want a 100% reliable method use CDRs.
>
> mod_xml_cdr can submit call CDRs to a webserver just after call hangup. The
> XML is verbose and will contain a full call history and all channel
> variables. Because it's submitted ASAP you get the results in real time and
> reliable. If the module can't submit the CDRs it writes them to an error
> folder on disk where you can resubmit them.
>
> Real time seems useful for your use-case, as your script would just need to
> check your DB whether the CDR had been submitted yet. The channel UUID can
> either be captured from events or specified in advance in the originate
> (originate_uuid). There's a uuid api call to generate them for you. I'd
> suggest looking at this method.

This is all great info, thanks :-) I'm capturing the uuid from the
response from the originate command, that's no problem.

The box I'm doing this on is being a B2BUA for us at the moment. The
CDRs are already being HTTP POSTed to a URL, I can't change that as
our billing depends on it. I can see the channel variable coming
though in the POST content, so maybe I can frob the CDR script to give
me what I want.

Thanks again for the info.

Cheers,
Rich

-- 
Once our basic material needs are met - in my utopia, anyway - life
becomes a perpetual celebration in which everyone has a talent to
contribute. But we cannot levitate ourselves into that blessed
condition by wishing it. We need to brace ourselves for a struggle
against terrifying obstacles, both of our own making and imposed by
the natural world. And the first step is to recover from the delusion
that is positive thinking.
       -- Barbara Ehrenreich


From gregor at infomedia.si  Thu Nov 29 15:39:30 2012
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 29 Nov 2012 13:39:30 +0100
Subject: [Freeswitch-users] Inbound DID trunk without authentication
In-Reply-To: <CAHKQagExjduWDO83PAEV5BZGYGitiEbgFQRwQS8yjkY5iAj2rg@mail.gmail.com>
References: <CAC7ttFE-v_3rKO3AcXu5hL_s8Zoyzbw9cuCbt_bmJszqB1y9CA@mail.gmail.com>
	<20121129103218.GA10306@eagle.cupis.co.uk>
	<F3C02C5E8650B74FA0C199FA1A01D26E169A3A1A89@ausmsex00.austin.kmvtechnologies.com>
	<CAC7ttFF-qkxB7UKp-Ca3pTF8oQwsvcYPt4=m_wghjF7w85qydQ@mail.gmail.com>
	<CAHKQagExjduWDO83PAEV5BZGYGitiEbgFQRwQS8yjkY5iAj2rg@mail.gmail.com>
Message-ID: <CAC7ttFHx8DXvQFau_wOL__=jxeG-WkcVVSJ=e-L1m+7fE_qx9A@mail.gmail.com>

Hi Cal!

I think I got it now and it may help you.

If call is coming to port 5060, then it is automaticly routed as
sip_profiles/internal profile. In internal.xml it is rule to obbey
"domains" list from  acl.conf.xml. So, whatever you do in acl, nothing will
change behaviour.

So what I did? I expanded domains list to add IP node of my provider. Now
incoming call is granted from acl and doesn't require authentication, but
automaticlly goes to public dial plan. I still have to figure it out why
does it go to public diaplan. In the end, this is what I wanted, but just
want to know wha it goes to public, where is this rule set.

Gregor

2012/11/29 Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk>

> Hi Gregor,
>
> I have to admit, every time I've tried using ACLs for inbound
> authentication (i.e. an inbound DID), it always gave me the same damn error
> (rejected by ACL 'domains' etc).
>
> I've searched endlessly for a reason why, but couldn't seem to find a
> combination of variables that worked.
>
> I figured that it must be working else everyone would be complaining, and
> that my own lack of knowledge/experience with FreeSWITCH meant I was
> getting something fundamentally wrong in the config.
>
> So, I removed ACLs, threw up an iptables rule and put a TODO in the
> project plan to fix ACLs before release lol.
>
> I'll have another look at this, and see if I can figure out wtf I'm doing
> wrong.
>
> This may/may not be the same problem your having, but I'll report back
> either way and hopefully it'll help!
>
> Cal
>
>
> On Thu, Nov 29, 2012 at 11:15 AM, Gregor Nanger <gregor at infomedia.si>wrote:
>
>> I thought that just set acl and set in /sip_profiles/external.xml <param
>> name="apply-inbound-acl" value="testdid"/> would solve this, but it
>> doesn't working.
>>
>> Trunk DID provider I am using is sending calls to FS. Now I just want to
>> get this calls into public dialplan based only on IP, not authorization. :-(
>>
>>
>>
>> 2012/11/29 Sanjay Soni <ssoni at lifesize.com>
>>
>>> I was getting similar problem and modifying acl.conf didn't work for me.
>>> What work for me was modifying sip_profiles/internal.xml and
>>> sip_profiles/external.xml with Either
>>> param name="local-network-acl" value="localnet.auto" - For Private IPs
>>> (This is default)
>>> Or
>>> param name="candidate-acl" value="wan.auto"
>>> param name="local-network-acl" value="localnet.auto" - For Public IPs.
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis
>>> Sent: 29 November 2012 16:02
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] Inbound DID trunk without authentication
>>>
>>> On Thu, Nov 29, 2012 at 09:36:25AM +0100, Gregor Nanger wrote:
>>> >    in acl.conf.xml i add:
>>> >
>>> >    <list name="testdid" default="deny">
>>> >    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>>> >    </list>
>>>
>>> >    What I see in console I keep getting rejected by ACL "domains"
>>> >
>>> >    If I understand correctly if IP of inbound call is in ACL then call
>>> is
>>> >    transfered to public dialplan?
>>> >
>>> >    I need this gateway only for inbound calls.
>>>
>>> Try adding:
>>>
>>>    <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>>>
>>> into the existing stanza in acl.conf.xml which starts with:
>>>
>>>    <list name="domains" ...
>>>
>>> as this is the ACL which your SIP profile seems to be using to
>>> authenticate
>>> incoming calls.
>>>
>>> Regards,
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Vladislav.Grishin at vts24.ru  Thu Nov 29 15:56:30 2012
From: Vladislav.Grishin at vts24.ru (=?UTF-8?B?ItCT0YDQuNGI0LjQvSDQki7QoS4i?=)
Date: Thu, 29 Nov 2012 16:56:30 +0400
Subject: [Freeswitch-users] How to add username in SIP packet OPTIONS
Message-ID: <50B75B7E.6090500@vts24.ru>

My gataway profile

<include>
   <gateway name="SIP_M200">
   <param name="username" value="USERNAME_STRING"/>
   <param name="realm" value="172.30.0.2"/>
   <param name="register" value="false"/>
   <param name="caller-id-in-from" value="true"/>
   <param name="ping" value="5"/>
   </gateway>
</include>

SIP packet which is accepted by the remote UA

Thu Nov 29 16:38:24 2012
RECEIVE 15:5060 <-- 172.30.0.1:5080
OPTIONS sip:172.30.0.2;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.30.0.1:5080;rport;branch=z9hG4bK6DKm54N817tUS
Max-Forwards: 70
*From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ
To: <sip:172.30.0.2>*
Call-ID: 79818268-b4c4-1230-60ac-00d0b7ded66a
CSeq: 36758947 OPTIONS
User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20120920T220849Z~f718a5e8e6
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0

but in reply arrives nothing...    the developer points to that that in 
a SIP packet  there is no field username in

*From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ     <----in this string no 
username
To: <sip:172.30.0.2>      <---- and  in this string no username*

it is necessary to make as

From: "USERNAME_STRING" <sip:USERNAME_STRING at 172.30.0.1>;tag=9DcB7gSNZZ41j

What it is necessary to change or add in a gateway profile (or other 
place)  for this purpose?

Vladislav Grishin
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From vbvbrj at gmail.com  Thu Nov 29 16:12:32 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 29 Nov 2012 15:12:32 +0200
Subject: [Freeswitch-users] How to add username in SIP packet OPTIONS
In-Reply-To: <50B75B7E.6090500@vts24.ru>
References: <50B75B7E.6090500@vts24.ru>
Message-ID: <50B75F40.30000@gmail.com>

On 29.11.2012 14:56, "?????? ?.?." wrote:

> *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ     <----in this string no
> username
> To: <sip:172.30.0.2>      <---- and  in this string no username*
>
> it is necessary to make as
>
> From: "USERNAME_STRING" <sip:USERNAME_STRING at 172.30.0.1>;tag=9DcB7gSNZZ41j

You have to use
<action application="set" data="effective_caller_id_name=USERNAME_STRING"/>
<action application="set" 
data="effective_caller_id_number=USERNAME_STRING"/>

in dialplan before bridging, or in users directory profile.
-- 
Mimiko desu.


From msalman212 at gmail.com  Thu Nov 29 13:30:49 2012
From: msalman212 at gmail.com (Salman Zafar)
Date: Thu, 29 Nov 2012 15:30:49 +0500
Subject: [Freeswitch-users] Cluecon 2012 presentations availability !?
In-Reply-To: <CAJUJwtj4wHLXSsaiE6Nb8P3gs5BH-Qsw-hWDGuzxoN=0PjGyZw@mail.gmail.com>
References: <CAJUJwtj4wHLXSsaiE6Nb8P3gs5BH-Qsw-hWDGuzxoN=0PjGyZw@mail.gmail.com>
Message-ID: <CAP2a2YURQGPCXq8Yt5ZZ4KZpn4Nha44_QCfe4kX5Z6szZMrVLQ@mail.gmail.com>

+1 for videos. I think many of us would be interested.

On Wed, Nov 28, 2012 at 10:13 AM, SamyGo <govoiper at gmail.com> wrote:

> Hi,
>
> Is there any way we can get the presentations or videos of sessions in
> cluecon 2012 ? Will really be a treat for those around the world who
> couldn't be there.
>
> Thanks,
> Sammy Go.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards

**************************
Muhammad Salman
***************************
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From steveayre at gmail.com  Thu Nov 29 16:56:44 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 13:56:44 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
	<CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
	<CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>
Message-ID: <CAFiqYukX75FDSpAT9zdvnOBb_boJxQAZKJSk2QhuJKroY_LJng@mail.gmail.com>

>
> ... how I listen for these events. Is it just a case of polling the
> socket for them at frequent enough intervals?


You subscribe to the events you're interested in with 'events'
http://wiki.freeswitch.org/wiki/Event_socket#event

There's no polling involved. FS will send all subscribed events through the
socket, you just need to receive them. Use select()/poll()/equivalent to
detect when data has arrived.

Since you already have xml_cdr working that would be the best place to
collect it, if you can add an extra DB insert in the handler.

-Steve


On 29 November 2012 11:58, Richard Gration <richgration at gmail.com> wrote:

> Hi,
>
> Thanks for your reply. I was wondering ...
>
> > If you listen to events then you should be able to reliably receive a
> hangup
> > event containing the variables if you are still connected at that time.
>
> ... how I listen for these events. Is it just a case of polling the
> socket for them at frequent enough intervals?
>
> > If you want a 100% reliable method use CDRs.
> >
> > mod_xml_cdr can submit call CDRs to a webserver just after call hangup.
> The
> > XML is verbose and will contain a full call history and all channel
> > variables. Because it's submitted ASAP you get the results in real time
> and
> > reliable. If the module can't submit the CDRs it writes them to an error
> > folder on disk where you can resubmit them.
> >
> > Real time seems useful for your use-case, as your script would just need
> to
> > check your DB whether the CDR had been submitted yet. The channel UUID
> can
> > either be captured from events or specified in advance in the originate
> > (originate_uuid). There's a uuid api call to generate them for you. I'd
> > suggest looking at this method.
>
> This is all great info, thanks :-) I'm capturing the uuid from the
> response from the originate command, that's no problem.
>
> The box I'm doing this on is being a B2BUA for us at the moment. The
> CDRs are already being HTTP POSTed to a URL, I can't change that as
> our billing depends on it. I can see the channel variable coming
> though in the POST content, so maybe I can frob the CDR script to give
> me what I want.
>
> Thanks again for the info.
>
> Cheers,
> Rich
>
> --
> Once our basic material needs are met - in my utopia, anyway - life
> becomes a perpetual celebration in which everyone has a talent to
> contribute. But we cannot levitate ourselves into that blessed
> condition by wishing it. We need to brace ourselves for a struggle
> against terrifying obstacles, both of our own making and imposed by
> the natural world. And the first step is to recover from the delusion
> that is positive thinking.
>        -- Barbara Ehrenreich
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Thu Nov 29 17:19:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 29 Nov 2012 16:19:53 +0200
Subject: [Freeswitch-users] predictive dialer.
In-Reply-To: <CAMAX9-3n=Dk9uNcb642A9vc4kpvGv6tJqvfFOKiGW3gy9dYCrA@mail.gmail.com>
References: <CAMAX9-0hYx=NuB_aWEZohDKVF=3VYWjd0LU9khmPpJsaOR=1iQ@mail.gmail.com>
	<CAMAX9-3n=Dk9uNcb642A9vc4kpvGv6tJqvfFOKiGW3gy9dYCrA@mail.gmail.com>
Message-ID: <CALNZuwmeOfMZ04Yd3eDNU=BtMKSHVBSf=qFZWZvCh30R0CdwjQ@mail.gmail.com>

http://www.newfies-dialer.org/ might be what you are looking for.
-Avi
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From Vladislav.Grishin at vts24.ru  Thu Nov 29 17:20:06 2012
From: Vladislav.Grishin at vts24.ru (=?UTF-8?B?ItCT0YDQuNGI0LjQvSDQki7QoS4i?=)
Date: Thu, 29 Nov 2012 18:20:06 +0400
Subject: [Freeswitch-users] How to add username in SIP packet OPTIONS
In-Reply-To: <50B75F40.30000@gmail.com>
References: <50B75B7E.6090500@vts24.ru> <50B75F40.30000@gmail.com>
Message-ID: <50B76F16.6050900@vts24.ru>


FreeSwitch make ping remote site (gateway) by SIP OPTIONS pakets. If 
remote side nothing answers to freeswitch, a gateway state changes  to 
DOWN. After it calls (INVITES) from dialplan won't be sent on this gateway.




29.11.2012 17:12, Mimiko ?????:
> On 29.11.2012 14:56, "?????? ?.?." wrote:
>
>> *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ     <----in this string no
>> username
>> To: <sip:172.30.0.2>      <---- and  in this string no username*
>>
>> it is necessary to make as
>>
>> From: "USERNAME_STRING" <sip:USERNAME_STRING at 172.30.0.1>;tag=9DcB7gSNZZ41j
> You have to use
> <action application="set" data="effective_caller_id_name=USERNAME_STRING"/>
> <action application="set"
> data="effective_caller_id_number=USERNAME_STRING"/>
>
> in dialplan before bridging, or in users directory profile.
Vladislav

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From areski at gmail.com  Thu Nov 29 17:30:55 2012
From: areski at gmail.com (Areski)
Date: Thu, 29 Nov 2012 15:30:55 +0100
Subject: [Freeswitch-users] predictive dialer.
In-Reply-To: <CALNZuwmeOfMZ04Yd3eDNU=BtMKSHVBSf=qFZWZvCh30R0CdwjQ@mail.gmail.com>
References: <CAMAX9-0hYx=NuB_aWEZohDKVF=3VYWjd0LU9khmPpJsaOR=1iQ@mail.gmail.com>
	<CAMAX9-3n=Dk9uNcb642A9vc4kpvGv6tJqvfFOKiGW3gy9dYCrA@mail.gmail.com>
	<CALNZuwmeOfMZ04Yd3eDNU=BtMKSHVBSf=qFZWZvCh30R0CdwjQ@mail.gmail.com>
Message-ID: <CAMf7LBgmGMw7YNMBkW=G1n7cDXq7gsWP2yJG8iXLUMsPsQ9=xw@mail.gmail.com>

Newfies-Dialer is not "yet" a predictive dialer , it's more a scalable
voice broadcasting/ autodialer solution.
Nevertheless it's a good core base to start building one.

On Thu, Nov 29, 2012 at 3:19 PM, Avi Marcus <avi at avimarcus.net> wrote:

> http://www.newfies-dialer.org/ might be what you are looking for.
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Vladislav.Grishin at vts24.ru  Thu Nov 29 17:31:39 2012
From: Vladislav.Grishin at vts24.ru (=?UTF-8?B?ItCT0YDQuNGI0LjQvSDQki7QoS4i?=)
Date: Thu, 29 Nov 2012 18:31:39 +0400
Subject: [Freeswitch-users] How to add username in SIP packet OPTIONS
In-Reply-To: <50B75F40.30000@gmail.com>
References: <50B75B7E.6090500@vts24.ru> <50B75F40.30000@gmail.com>
Message-ID: <50B771CB.7040601@vts24.ru>


freeswitch at default> sofia status gateway SIP_M200
=================================================================================================
Name            SIP_M200
Profile         external
Scheme          Digest
Realm           172.30.0.2
Username        USERNAME_STRING
Password        yes
*From <sip:USERNAME_STRING at 172.30.0.2>*
Contact <sip:gw+SIP_M200 at 172.30.0.1:5080;transport=udp;gw=SIP_M200>
Exten           USERNAME_STRING
*To sip:USERNAME_STRING at 172.30.0.2*
Proxy           sip:172.30.0.2
Context         public
Expires         3600
Freq            3600
Ping            1354199070
PingFreq        5
PingState       -1/-1/1
State           NOREG
*Status          DOWN (ping)*
CallsIN         0
CallsOUT        0
FailedCallsIN   0
FailedCallsOUT  0
=================================================================================================

freeswitch at default>

but the remote UA accepts

Thu Nov 29 18:27:16 2012
RECEIVE 15:5060 <-- 172.30.0.1:5080
OPTIONS sip:172.30.0.2;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.30.0.1:5080;rport;branch=z9hG4bKv39BcX339a2ZK
Max-Forwards: 70
*From: <sip:172.30.0.2>;tag=H7jt6HH1p05cg
To: <sip:172.30.0.2>*
Call-ID: acb53903-b4d3-1230-60ac-00d0b7ded66a
CSeq: 36762086 OPTIONS
User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20120920T220849Z~f718a5e8e6
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0

Vladislav

29.11.2012 17:12, Mimiko ?????:
> On 29.11.2012 14:56, "?????? ?.?." wrote:
>
>> *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ     <----in this string no
>> username
>> To: <sip:172.30.0.2>      <---- and  in this string no username*
>>
>> it is necessary to make as
>>
>> From: "USERNAME_STRING" <sip:USERNAME_STRING at 172.30.0.1>;tag=9DcB7gSNZZ41j
> You have to use
> <action application="set" data="effective_caller_id_name=USERNAME_STRING"/>
> <action application="set"
> data="effective_caller_id_number=USERNAME_STRING"/>
>
> in dialplan before bridging, or in users directory profile.

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From krice at freeswitch.org  Thu Nov 29 17:44:50 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 29 Nov 2012 08:44:50 -0600
Subject: [Freeswitch-users] Side note (READ THIS PLEASE)
In-Reply-To: <CAFiqYu=twb22bcD=6ADvGvw3gRvW=rsObZ26iZE57MaYxtyTVw@mail.gmail.com>
Message-ID: <CCDCD102.612A9%krice@freeswitch.org>

No one has come forward with info as to who... We are taking steps to
mitigate the effects of the DoS attack...

K


On 11/29/12 3:26 AM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Did we find out out who was doing the DDOS?
> 
> Have they actually gone away, or are we just coping with it better now?
> 
> 
> On 29 November 2012 03:40, Brian West <brian at freeswitch.org> wrote:
>> You should follow us on twitter, if you had done so you could have know what
>> was going on during this wondering event for the past few days.
>> 
>> FOLLOW US NOW ON TWITTER @FreeSWITCH_Wire ?(cuz some ass has freeswitch and
>> won't give it up)
>> 
>> --
>> Brian West
>> brian at freeswitch.org
>> FreeSWITCH Solutions, LLC
>> PO BOX PO BOX 2531
>> Brookfield, WI 53008-2531
>> Twitter: @FreeSWITCH_Wire
>> T: +1.918.420.9266 <tel:%2B1.918.420.9266>  ?| ?F: +1.918.420.9267
>> <tel:%2B1.918.420.9267>  ?| ?M: +1.918.424.WEST
>> iNUM: +883 5100 1420 9266 <tel:%2B883%205100%201420%209266>
>> UK: +44 20 3298 4900 <tel:%2B44%2020%203298%204900>
>> ISN: 410*543
>> 
>> 
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

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From steveayre at gmail.com  Thu Nov 29 18:43:50 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 15:43:50 +0000
Subject: [Freeswitch-users] FS debian sid repo broken (hash mismatches)
Message-ID: <CAFiqYumnscOGLC2PLAqCi+PN3Ri1=YTbz=L7mPuKAnj+aKJykQ@mail.gmail.com>

apt-get update fails with 'Hash sum mismatch'.

For example:
W: Failed to fetch
gzip:/var/lib/apt/lists/partial/files.freeswitch.org_repo_deb_debian_dists_sid_main_binary-amd64_Packages
 Hash Sum mismatch

http://files.freeswitch.org/repo/deb/debian/dists/sid/InRelease contains:
3b775b7987d60e128780f1e979132c1a 300910 main/binary-amd64/Packages
b7014356d8a601fc827f37fa556edc9d 41654 main/binary-amd64/Packages.gz

The MD5 hashes for those files are actually:
11e5b078ef6a36ab11d94e9c7770a0aa  Packages
1925eae422f52cdaa27f4d788f7e6276  Packages.gz

I haven't checked the rest of the files, but I expect they're incorrect too.
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From mailing-lists at phoenixinternet.net  Thu Nov 29 19:11:37 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Thu, 29 Nov 2012 09:11:37 -0700
Subject: [Freeswitch-users] Site Down
In-Reply-To: <2DFED399-15F5-45C7-A347-1745B4A9E4BC@freeswitch.org>
References: <50B649F1.4040802@phoenixinternet.net>
	<2DFED399-15F5-45C7-A347-1745B4A9E4BC@freeswitch.org>
Message-ID: <50B78939.3080801@phoenixinternet.net>

I am now following on Twitter... After I tried to post, I found the irc 
channel via wikipedia. I try to make posts a little funny if I may make 
an ass of myself.

I went ahead and added your twitter feed to the wikipedia page.

Gilbert

On 11/28/2012 6:39 PM, Brian West wrote:
> Yep We've noticed.  LOL  The list server was down too... as was EVERYTHING else.  :P
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
> ISN: 410*543
>
>
>
>
>
> On Nov 28, 2012, at 11:29 AM, "Gilbert T. Gutierrez, Jr." <mailing-lists at phoenixinternet.net> wrote:
>
>> Is the FreeSWITCH site down or has it been having problems? I was
>> developing a new system to replace my asterisk system. Today I wanted to
>> replicate the system, and could not access anything FreeSWITCH.org to do
>> my build.
>>
>> If this email posts to the list, obviously that would mean the mailing
>> list email server is not down. But I cannot access the www or git.
>>
>> Thanks,
>> Gilbert
>



From ben122uk at gmail.com  Thu Nov 29 20:43:30 2012
From: ben122uk at gmail.com (Ben N)
Date: Thu, 29 Nov 2012 17:43:30 +0000
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>
	<6036422a0e8b5982ed7af8823015bb38@www.webmail.co.za>
	<CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>
Message-ID: <CADaqzFP6Kck0Sm5Z9i6K1hW6dDoRqyPdoFAU2UEGuDaW_3geYQ@mail.gmail.com>

Hi Brian, yes I've set G729 at 50i on both the external and internal sofia
profiles statically, and confirmed the allowed codecs by running "sofia
status profile *profilename*".

I can also confirm in the FS console that codec comparisons for both the
a-leg, and b-leg match instantly to G729 at 50i.

Clive, thanks for the pointer, although in my case I don't think there's a
mis-match as each side of the call supports sending/receiving G729 50ms
ptimes, and is negotiated correctly in the SDP.  However I'm still willing
to give it a try as I need to try everything!  I'll let you know what
happens.

Cheers,

Ben

On Thu, Nov 29, 2012 at 9:58 AM, Ben N <ben122uk at gmail.com> wrote:

> Hi Brian, yes I've set G729 at 50i on both the external and internal sofia
> profiles statically, and confirmed the allowed codecs by running "sofia
> status profile *profilename*".
>
> I can also confirm in the FS console that codec comparisons for both the
> a-leg, and b-leg match instantly to G729 at 50i.
>
> Clive, thanks for the pointer, although in my case I don't think there's a
> mis-match as each side of the call supports sending/receiving G729 50ms
> ptimes, and is negotiated correctly in the SDP.  However I'm still willing
> to give it a try as I need to try everything!  I'll let you know what
> happens.
>
> Cheers,
>
> Ben
>
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From fs-list at communicatefreely.net  Thu Nov 29 21:09:02 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 29 Nov 2012 13:09:02 -0500
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
 again
In-Reply-To: <CAKbxfG-Mnm-yhx0D6mp=WM3xCC6bmg=OmSy0ph8YU-ecL31V9Q@mail.gmail.com>
References: <CCD7A44C.610F0%krice@freeswitch.org>	<50B4249A.30702@communicatefreely.net>
	<CAKbxfG-Mnm-yhx0D6mp=WM3xCC6bmg=OmSy0ph8YU-ecL31V9Q@mail.gmail.com>
Message-ID: <50B7A4BE.8040108@communicatefreely.net>

It's not a matter of not taking your suggestion, it's an issue of not having had the time 
yet to learn how gcore works and put it in place on a production system.  It's on my 
"to-do" list, but I thought I would at least update the thread, hoping maybe that message 
meant something to someone.



Anthony Minessale wrote:
> too bad you did not gcore it like I suggested, otherwise who knows... 
> That's just the message it prints when the message parsing thread goes up.
> 
> 
> 
> On Mon, Nov 26, 2012 at 8:25 PM, Tim St. Pierre 
> <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>> 
> wrote:
> 
>     Thanks for all the input everyone!
> 
>     I just had it happen again, this time at night when there were only
>     2 sessions active!
> 
>     I was in doing some other work, so I managed to see it happen and
>     found a few interesting
>     things.
> 
>     CPU load next to nothing, load average around 0.12.
>     I used iSQL to test ODBC connectivity, and I could read and write to
>     the freeswitch
>     database using the same DSN that freeswitch is using.
> 
>     I could delete a SIP registration from the API
> 
>     Since I had things on a higher debug level, I saw this:
> 
>     freeswitch at stefan> 2012-11-26 21:21:51.914082 [CONSOLE] sofia.c:1144
>     MSG Thread Started
> 
>     As soon as that came up, everything started working again.
> 
>     What's that about?
> 
>     -Tim
> 
>     Ken Rice wrote:
>      > Sofia is not single threaded except for in one spot deep in libsofia,
>      >>From there, messages are handed off to a number of message queues
>     for FS
>      > core to handle as needed...
>      >
>      > Check to see if anything that fs is depending on is blocking on info
>      > retrieval like the databases or other areas...
>      >
>      > K
>      >
>      > On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com
>     <mailto:abaci64 at gmail.com>> wrote:
>      >
>      >> you mentioned that you use xml_curl, if your web server hangs it may
>      >> hang sofia, iirc sofia is running a single thread and it will
>     wait for
>      >> the xml_curl response before continuing to the next request.
>      >>
>      >> On 11/23/2012 1:13 PM, Steven Ayre wrote:
>      >>> Any kind of DB backup running? Or any long-running queries
>     (innotop is
>      >>> great for highlighting queries that've been running a while,
>     including
>      >>> on non-innodb tables).
>      >>>
>      >>> A global read lock, or queries waiting for a lock could block a db
>      >>> update from the sofia profile thread but still allow read-only
>     queries
>      >>> (sofia status) to run.
>      >>>
>      >>> -Steve
>      >>>
>      >>>
>      >>>
>      >>> On 23 November 2012 15:32, Tim St. Pierre
>     <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
>      >>> wrote:
>      >>>> Hi Steven,
>      >>>>
>      >>>> Thanks for the suggestions.  I'm hoping once I get the upgrade
>     done it will
>      >>>> all go away.
>      >>>> I have watched it happen at least once.  I was on the phone at
>     the time.
>      >>>> Console activity
>      >>>> more or less stopped, except for a few calls hanging up.  The
>     console
>      >>>> remains responsive,
>      >>>> and my call wasn't dropped for at least a minute or two (media
>     timeout?).  I
>      >>>> was able to
>      >>>> run sofia status and other commands that use the database, so
>     I'm assuming
>      >>>> that the
>      >>>> connection was still working.  All our media is runs through
>     the box, so I
>      >>>> think things
>      >>>> are fine on the Ethernet level.  I do see higher load averages
>     - maybe 3-4,
>      >>>> but that's the
>      >>>> only obvious indication.  It's not taking CPU beyond 10% or so.
>      >>>>
>      >>>> We are using MySQL as the core DB and also as the DB backend
>     for each sofia
>      >>>> profile.  This
>      >>>> is connecting through ODBC of course.
>      >>>>
>      >>>> If I can get the other kinks worked out, then I will try 1.2
>     stable in
>      >>>> production and
>      >>>> we'll see how it goes.
>      >>>>
>      >>>> -Tim
>      >>>>
>      >>>> Steven Ayre wrote:
>      >>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20
>     -0500)
>      >>>>>
>      >>>>>
>      >>>>> As you've already acknowledged it's a very old version.
>      >>>>>
>      >>>>> It's possible that your issue has already been found and
>     fixed, but if
>      >>>>> it hasn't then the code will have changed significantly since
>     then and
>      >>>>> you'd really need to reproduce it on the latest code for it to be
>      >>>>> investigated.
>      >>>>>
>      >>>>>
>      >>>>> As some general thoughts though, are you able to spot it
>     happening while
>      >>>>> it's happening or only afterwards?
>      >>>>>
>      >>>>> If you're able to get on the system during one of those times
>     look at
>      >>>>> what else is happening. Is the load average/cpu usage/io
>     high? Perhaps
>      >>>>> something's running that's blocking all access or causing
>     very high IO.
>      >>>>>
>      >>>>> What DB backend are you using for Sofia? Is it possible that
>     that's
>      >>>>> hanging for a moment? For example if you're running a backup
>     on the DB
>      >>>>> that blocks all writes to the DB while Sofia is trying to
>     update the DB
>      >>>>> that perhaps would cause this.
>      >>>>>
>      >>>>> Try running a SIP OPTIONS ping your your sofia profile from the
>      >>>>> localhost during that time, which should exclude it being any
>     issue on
>      >>>>> the ethernet.
>      >>>>>
>      >>>>> -Steve
>      >>>>>
>      >>>>>
>      >>>>>
>      >>>>>
>      >>>>> On 22 November 2012 19:31, Tim St. Pierre
>     <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>
>      >>>>> <mailto:fs-list at communicatefreely.net
>     <mailto:fs-list at communicatefreely.net>>> wrote:
>      >>>>>
>      >>>>>      Hello,
>      >>>>>
>      >>>>>      I'm having a bit of an odd problem.
>      >>>>>
>      >>>>>      Intermittently, often every 2-3 days or so, Freeswitch stops
>      >>>>>      replying to SIP for about 5
>      >>>>>      minutes.  I can't verify if it's EXACTLY 5 minutes, but
>     it seems to
>      >>>>>      be pretty close.
>      >>>>>
>      >>>>>      During this time, no new registrations or invites can
>     happen, but
>      >>>>>      existing calls stay
>      >>>>>      connected for at least a minute or two.  In the logs,
>     you can see
>      >>>>>      calls slowly hanging up
>      >>>>>      with "NORMAL_CLEARING".  In 5 minutes, everything starts
>     up again
>      >>>>>      with no word about it at
>      >>>>>      all in the logs.
>      >>>>>
>      >>>>>      When calls resume, I notice that the number of sessions
>     returned by
>      >>>>>      the status command is
>      >>>>>      one higher than the actual number sessions returned by show
>      >>>>>      channels, or by looking in the
>      >>>>>      database.  Every time this happens, the discrepancy
>     increases by one.
>      >>>>>
>      >>>>>      The interruption happens on all SIP profiles, but calls
>     originated
>      >>>>>      from the socket API
>      >>>>>      still work, insofar as they return with PROGRESS_TIMEOUT
>     since the
>      >>>>>      profiles are still
>      >>>>>      running, but stuck.
>      >>>>>
>      >>>>>      We are using ODBC/MySQL for the core database, and the
>     database
>      >>>>>      server only runs this
>      >>>>>      database and some basic PHP/xml-curl stuff.
>      >>>>>
>      >>>>>      We have 416 endpoints registered, and usually sit at
>     about 30
>      >>>>>      sessions during the day.
>      >>>>>
>      >>>>>      This never happens at night, only during busier times,
>     but not
>      >>>>>      necessarily busy hour.
>      >>>>>
>      >>>>>      I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores,
>     4G ram)
>      >>>>>
>      >>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20
>     -0500)
>      >>>>>
>      >>>>>      Yes, I know it's old and I'm trying to upgrade, but I'm
>     still having
>      >>>>>      some problems getting
>      >>>>>      all my phones to work properly with 1.2 stable.  This is a
>      >>>>>      production system, so I can't
>      >>>>>      just blindly put out the newest release.  Mostly, I need
>     to buy
>      >>>>>      myself some time so that I
>      >>>>>      can get the kinks worked out of the latest version and
>     then upgrade
>      >>>>>      the production box.
>      >>>>>
>      >>>>>      I'm grateful for any insights as to what could be
>     happening, even if
>      >>>>>      a solution is just a
>      >>>>>      temporary workaround.
>      >>>>>
>      >>>>>      Thanks!
>      >>>>>
>      >>>>>      -Tim
>      >>>>>
>      >>>>>
>      >>>>>
>     _________________________________________________________________________
>      >>>>>      Professional FreeSWITCH Consulting Services:
>      >>>>>      consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org> <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>>
>      >>>>>      http://www.freeswitchsolutions.com
>      >>>>>
>      >>>>>      
>      >>>>>      
>      >>>>>
>      >>>>>      Official FreeSWITCH Sites
>      >>>>>      http://www.freeswitch.org
>      >>>>>      http://wiki.freeswitch.org
>      >>>>>      http://www.cluecon.com
>      >>>>>
>      >>>>>      FreeSWITCH-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
>      >>>>>
>      >>>>>
>      >>>>>
>      >>>>>
>     ------------------------------------------------------------------------
>      >>>>>
>      >>>>>
>     _________________________________________________________________________
>      >>>>> Professional FreeSWITCH Consulting Services:
>      >>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>      >>>>> http://www.freeswitchsolutions.com
>      >>>>>
>      >>>>> 
>      >>>>> 
>      >>>>>
>      >>>>> Official FreeSWITCH Sites
>      >>>>> http://www.freeswitch.org
>      >>>>> http://wiki.freeswitch.org
>      >>>>> http://www.cluecon.com
>      >>>>>
>      >>>>> FreeSWITCH-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
>      >>>>
>     _________________________________________________________________________
>      >>>> Professional FreeSWITCH Consulting Services:
>      >>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>      >>>> http://www.freeswitchsolutions.com
>      >>>>
>      >>>> 
>      >>>> 
>      >>>>
>      >>>> Official FreeSWITCH Sites
>      >>>> http://www.freeswitch.org
>      >>>> http://wiki.freeswitch.org
>      >>>> http://www.cluecon.com
>      >>>>
>      >>>> FreeSWITCH-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
>      >>>
>     _________________________________________________________________________
>      >>> Professional FreeSWITCH Consulting Services:
>      >>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>      >>> http://www.freeswitchsolutions.com
>      >>>
>      >>> 
>      >>> 
>      >>>
>      >>> Official FreeSWITCH Sites
>      >>> http://www.freeswitch.org
>      >>> http://wiki.freeswitch.org
>      >>> http://www.cluecon.com
>      >>>
>      >>> FreeSWITCH-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
>      >>
>      >>
>     _________________________________________________________________________
>      >> Professional FreeSWITCH Consulting Services:
>      >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>      >> http://www.freeswitchsolutions.com
>      >>
>      >> 
>      >> 
>      >>
>      >> Official FreeSWITCH Sites
>      >> http://www.freeswitch.org
>      >> http://wiki.freeswitch.org
>      >> http://www.cluecon.com
>      >>
>      >> FreeSWITCH-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
>      >
> 
> 
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
> 
>     
>     
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
> 
>     FreeSWITCH-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
> 
> 
> ------------------------------------------------------------------------
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mailing-lists at phoenixinternet.net  Thu Nov 29 21:32:07 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Thu, 29 Nov 2012 11:32:07 -0700
Subject: [Freeswitch-users] Compilation Issue
Message-ID: <50B7AA27.2000003@phoenixinternet.net>

I am having an issue compiling 1.2 Stable this morning. My notes are all 
below. When I compile a second time, it appears to compile ok. I am not 
sure if I trust that compilation because some of the modules may not be 
complete. What do you guys feel the problem may be?

Gilbert

Centos6 64bit

The procedure I am following:
cd /usr/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
cd freeswitch
./bootstrap.sh -j
./configure --without-libcurl -C
make -j `cat /proc/cpuinfo |grep processor |wc -l`


The make first throws the following error...
"In file included from ./src/include/private/switch_core_pvt.h:44,
                  from src/switch.c:48:
/usr/src/freeswitch/libs/apr/include/apr_pools.h:37:17: error: apr.h: No 
such file or directory"

Everything following is an error terminating in "make: *** [all] Error 2"





From jkomar at jbox.ca  Thu Nov 29 21:38:34 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Thu, 29 Nov 2012 11:38:34 -0700
Subject: [Freeswitch-users] Compilation Issue
In-Reply-To: <50B7AA27.2000003@phoenixinternet.net>
References: <50B7AA27.2000003@phoenixinternet.net>
Message-ID: <CA+zebiYzT75EewBfBfWWvXDNpPcdc+sfToconYKv+PBvoBDnng@mail.gmail.com>

Had a similar thing on Gentoo. Seems to be a race condition. Stefan
Knoblich (Gentoo FreeSWITCH overlay maintainer) did a workaround in the
ebuild to get past it. I'm not sure what it entailed. Maybe if he sees this
message, he could chime in on it.

Jason


On Thu, Nov 29, 2012 at 11:32 AM, Gilbert T. Gutierrez, Jr. <
mailing-lists at phoenixinternet.net> wrote:

> I am having an issue compiling 1.2 Stable this morning. My notes are all
> below. When I compile a second time, it appears to compile ok. I am not
> sure if I trust that compilation because some of the modules may not be
> complete. What do you guys feel the problem may be?
>
> Gilbert
>
> Centos6 64bit
>
> The procedure I am following:
> cd /usr/src
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
> cd freeswitch
> ./bootstrap.sh -j
> ./configure --without-libcurl -C
> make -j `cat /proc/cpuinfo |grep processor |wc -l`
>
>
> The make first throws the following error...
> "In file included from ./src/include/private/switch_core_pvt.h:44,
>                   from src/switch.c:48:
> /usr/src/freeswitch/libs/apr/include/apr_pools.h:37:17: error: apr.h: No
> such file or directory"
>
> Everything following is an error terminating in "make: *** [all] Error 2"
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ksrigo at gmail.com  Thu Nov 29 20:40:14 2012
From: ksrigo at gmail.com (Srigo)
Date: Thu, 29 Nov 2012 09:40:14 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch is crashing when httapi playback
	application is executed
Message-ID: <1354210814465-7585041.post@n2.nabble.com>

Hello,

I'm using Httapi for dynamic application service. The problem, I'm getting
is when im using httapi with the application "playback", freeswitch is
crashing.

Seems the crashed is caused by the parameter of "file" in playback. Thats
mean when i'm getting a file from a remote server
(http://xxxxxx.com/test.wav) its crashing but when using a local wav file
everthing is fine.

here is the httapi xml:

<document type="xml/freeswitch-httapi">
<params/>
<variables>
<sou>someval</sou>
</variables>
<work>
<execute application="flush_dtmf"/>
<execute application="set"
data="${http_prefetch(http://xxxxxxx.com/test.wav)}"></execute>
<execute application="info"/>
<playback name="user_entry" file="http://xxxxxxx.com/test.wav" loops="3"
digit-timeout="1000" input-timeout="1000">
<bind strip="#">~\d+</bind>
</playback>
</work>
</document>


Here the log before the freeswitch crash:

freeswitch: src/switch_event.c:710: switch_event_get_header_ptr: Assertion
`event' failed.


I try to use http_cache to try to resolve the prob:

<playback name="user_entry" file="${http_get(http://xxxxxxx.com/test.wav)}"
loops="3" digit-timeout="1000" input-timeout="1000">
<bind strip="#">~\d+</bind>
</playback>

but i'm getting the following error in the log:

2012-11-23 19:06:10.015936 [ERR] switch_core_file.c:112 Unknown file Format
[$]

Finally i try the following idea:

- set a variable (<execute application="set"
data="play=${http_get(http://xxxxxxx.com/test.wav)}"></execute>

then use in playback parameter the variable ${play} but the variable is not
parsed!

2012-11-23 17:30:09.855933 [ERR] mod_native_file.c:74 Error opening
/usr/share/freeswitch/sounds/en/us/callie/${play}.PCMA


Could anyone please tell me where i did the mistake ?

thx a lot



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-is-crashing-when-httapi-playback-application-is-executed-tp7585041.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Nov 29 21:44:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 10:44:30 -0800
Subject: [Freeswitch-users] Site Down
In-Reply-To: <50B78939.3080801@phoenixinternet.net>
References: <50B649F1.4040802@phoenixinternet.net>
	<2DFED399-15F5-45C7-A347-1745B4A9E4BC@freeswitch.org>
	<50B78939.3080801@phoenixinternet.net>
Message-ID: <CAKzWOxW2dXD+mGEEmnDC7sS6avbPwPz-fd_tPrdoa4E_A49mAw@mail.gmail.com>

On Thu, Nov 29, 2012 at 8:11 AM, Gilbert T. Gutierrez, Jr. <
mailing-lists at phoenixinternet.net> wrote:

> I am now following on Twitter... After I tried to post, I found the irc
> channel via wikipedia. I try to make posts a little funny if I may make
> an ass of myself.
>
> I went ahead and added your twitter feed to the wikipedia page.
>
+1
Thanks for being a good community member and taking the initiative - we
appreciate it when someone spends 10 seconds to fix the problem rather than
spending 20 seconds to whine about it. :)
-MC
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From msc at freeswitch.org  Thu Nov 29 21:57:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 10:57:59 -0800
Subject: [Freeswitch-users] Freeswitch is crashing when httapi playback
 application is executed
In-Reply-To: <1354210814465-7585041.post@n2.nabble.com>
References: <1354210814465-7585041.post@n2.nabble.com>
Message-ID: <CAKzWOxVLtFG2YZCe35X1-5vGBG5BUj9vhd2eE6r4iPAFHtdUtw@mail.gmail.com>

Hi Srigo,

It looks like you're new to the list, so let me welcome you to the
community!

In a case like this, any time there is a crash, we need the user to open a
jira. You've already collected most of the necessary data. Go to
jira.freeswitch.org, open an account, and then add this data to a new jira
ticket. If you need help on how to file a jira then join us in #freeswitch
on irc.freenode.net. Also, our wiki has some good information here:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

Also, make sure you test with the latest git version because our devs work
fast and sometimes they've caught and fixed bugs before you finish opening
your jira ticket. :)

-MC

On Thu, Nov 29, 2012 at 9:40 AM, Srigo <ksrigo at gmail.com> wrote:

> Hello,
>
> I'm using Httapi for dynamic application service. The problem, I'm getting
> is when im using httapi with the application "playback", freeswitch is
> crashing.
>
> Seems the crashed is caused by the parameter of "file" in playback. Thats
> mean when i'm getting a file from a remote server
> (http://xxxxxx.com/test.wav) its crashing but when using a local wav file
> everthing is fine.
>
> here is the httapi xml:
>
> <document type="xml/freeswitch-httapi">
> <params/>
> <variables>
> <sou>someval</sou>
> </variables>
> <work>
> <execute application="flush_dtmf"/>
> <execute application="set"
> data="${http_prefetch(http://xxxxxxx.com/test.wav)}"></execute>
> <execute application="info"/>
> <playback name="user_entry" file="http://xxxxxxx.com/test.wav" loops="3"
> digit-timeout="1000" input-timeout="1000">
> <bind strip="#">~\d+</bind>
> </playback>
> </work>
> </document>
>
>
> Here the log before the freeswitch crash:
>
> freeswitch: src/switch_event.c:710: switch_event_get_header_ptr: Assertion
> `event' failed.
>
>
> I try to use http_cache to try to resolve the prob:
>
> <playback name="user_entry" file="${http_get(http://xxxxxxx.com/test.wav)
> }"
> loops="3" digit-timeout="1000" input-timeout="1000">
> <bind strip="#">~\d+</bind>
> </playback>
>
> but i'm getting the following error in the log:
>
> 2012-11-23 19:06:10.015936 [ERR] switch_core_file.c:112 Unknown file Format
> [$]
>
> Finally i try the following idea:
>
> - set a variable (<execute application="set"
> data="play=${http_get(http://xxxxxxx.com/test.wav)}"></execute>
>
> then use in playback parameter the variable ${play} but the variable is not
> parsed!
>
> 2012-11-23 17:30:09.855933 [ERR] mod_native_file.c:74 Error opening
> /usr/share/freeswitch/sounds/en/us/callie/${play}.PCMA
>
>
> Could anyone please tell me where i did the mistake ?
>
> thx a lot
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-is-crashing-when-httapi-playback-application-is-executed-tp7585041.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From avi at avimarcus.net  Thu Nov 29 22:05:31 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 29 Nov 2012 21:05:31 +0200
Subject: [Freeswitch-users] Compilation Issue
In-Reply-To: <50B7AA27.2000003@phoenixinternet.net>
References: <50B7AA27.2000003@phoenixinternet.net>
Message-ID: <CALNZuwkWz0qJ687cdCNOC9mLDeF--xEdzgFcVqLNMxQ4Kz9fiw@mail.gmail.com>

Please post your compilation issue on http://jira.freeswitch.org to ensure
it gets followed up on.
Thanks,
 -Avi



On Thu, Nov 29, 2012 at 8:32 PM, Gilbert T. Gutierrez, Jr. <
mailing-lists at phoenixinternet.net> wrote:

> I am having an issue compiling 1.2 Stable this morning. My notes are all
> below. When I compile a second time, it appears to compile ok. I am not
> sure if I trust that compilation because some of the modules may not be
> complete. What do you guys feel the problem may be?
>
> Gilbert
>
> Centos6 64bit
>
> The procedure I am following:
> cd /usr/src
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
> cd freeswitch
> ./bootstrap.sh -j
> ./configure --without-libcurl -C
> make -j `cat /proc/cpuinfo |grep processor |wc -l`
>
>
> The make first throws the following error...
> "In file included from ./src/include/private/switch_core_pvt.h:44,
>                   from src/switch.c:48:
> /usr/src/freeswitch/libs/apr/include/apr_pools.h:37:17: error: apr.h: No
> such file or directory"
>
> Everything following is an error terminating in "make: *** [all] Error 2"
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Nov 29 22:14:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 11:14:55 -0800
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in
 XML CDR's?
In-Reply-To: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>
References: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxUcpgaS29KWmSHg8FmMAraOVAa75i71sXdpbBC0S7EC_w@mail.gmail.com>

Mick,

This data is definitely not on the wiki - or anywhere else that I can see.
I think we can crowdsource this to get the info collected and then add it
to the mod_xml_curl wiki page <http://wiki.freeswitch.org/wiki/Mod_xml_cdr>.
For the record, here's a quick dump of the fields that I took from an XML
CDR:

    <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
    <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
    <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
    <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
    <rtp_audio_in_skip_packet_count>18</rtp_audio_in_skip_packet_count>
    <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
    <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
    <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
    <rtp_audio_in_flush_packet_count>11</rtp_audio_in_flush_packet_count>
    <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
    <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
    <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
    <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
    <rtp_audio_out_media_packet_count>67</rtp_audio_out_media_packet_count>
    <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
    <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
    <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>

I think raw_bytes, media_bytes, packet_count, and media_packet_count are
self-explanatory. I think cng_packet_count is probably self-explanatory
too. My question on dtmf_packet_count would be whether it's only for
RFC2833 packets (I suspect yes, but would like confirmation).

If anyone knows these please reply to this email and Mick and I will get
them documented on the wiki (right Mick? ;)

rtp_audio_in_skip_packet_count
rtp_audio_out_skip_packet_count
rtp_audio_in_jb_packet_count
rtp_audio_in_flush_packet_count
rtp_audio_in_largest_jb_size

Thanks all!

-MC

On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com> wrote:

> Hi Folks,
>
> I'm trying to use FS XML CDR's to diagnose historic audio problems. I
> think I can work out some of the rtp_audio_in / out fields (raw bytes &
> media bytes being nearly equal looks like a good sign) but am wondering
> about the skip, cng & flush fields for example?
>
> I have tried Googling this & can find evidence of other people having
> asked this question but not of the answer. I have also checked my FS 106 &
> Cookbook book's without success. The wiki appears to be down at the moment
> so my apologies if the answer lies there.
>
> I know how to do this in real time using wireshark etc but am interested
> in being able to do some analysis on historic problems reported by
> customers that aren't willing/able to replicate the problem in order for a
> protocol trace to be captured.
>
> Any help much appreciated!
>
> Rgds, Mick
> Tel/SMS. +44(0)7967 594432
> Fax. +44(0)7053 452429
> Email/IM. mickstevens at yahoo.com
> Skype: mick_stevens
> www.facebook.com/mickstevens
> www.twitter.com/mickstevens
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Nov 29 22:22:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 11:22:18 -0800
Subject: [Freeswitch-users] Trouble with voicemail say digits
In-Reply-To: <20121129044205.141c5071@mail.tritonwest.net>
References: <CA+zebiY3-N37JnDNVcgonxT-Qw-+tY64uQw7zCNZSY9joy9Rxw@mail.gmail.com>
	<20121129044205.141c5071@mail.tritonwest.net>
Message-ID: <CAKzWOxX+=SDMt24qZ4EF48KLDao8Lv9OuC6-wuXZPuL3QdC-dQ@mail.gmail.com>

This was a regression in mod_dptools that made it into 1.2.5.1. It was
fixed on Monday IIRC but with all the DDoS drama it flew under the radar.
This commit fixes it, so if you need a quick repair you can remove these
lines and recompile mod_dptools:

http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=0b148a85b94be33fe70b692240e0d449a59f0ef2

You should be able to do that without even stopping FS. I'll talk to Ken
and see if we can't get something more elegant in place soon.

-MC

On Wed, Nov 28, 2012 at 8:42 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> I thought it was me... Did something change whre the /8000/ got removed
> from the path? Is there a point where you can't leave your old configs in
> place when you update?
>
> --Dave
>
>  ------------------------------
> *From:* Komar, Jason [mailto:jkomar at jbox.ca]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Wed, 28 Nov 2012 19:47:05 -0800
> *Subject:* [Freeswitch-users] Trouble with voicemail say digits
>
>
> Just updated to FS 1.2.5.1 on Gentoo from 1.2.3. I have one extension that
> uses the default voicemail greeting. Since I updated, when it gets to the
> part where it says the extension number, it cannot find the wav files for
> the digits. It is looking for them in the digits folder and not in the
> frequency subfolders (i.e. 16000  32000  48000  8000). As soon as it hits
> this error, the voicemail app says goodbye and hangs up.
>
> 2012-11-28 20:35:17.068585 [DEBUG] switch_ivr_play_say.c:244 Handle
> say:[2003] (en:en)
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/2.wav] does not exist.
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
> [/opt/freeswitch/sounds/en/us/callie/digits/3.wav] does not exist.
>
> Is there a config option to specify the frequency subfolder, or did
> something not build correctly?
>
> Thanks,
> Jason Komar
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From richgration at gmail.com  Thu Nov 29 22:23:22 2012
From: richgration at gmail.com (Richard Gration)
Date: Thu, 29 Nov 2012 19:23:22 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAFiqYukX75FDSpAT9zdvnOBb_boJxQAZKJSk2QhuJKroY_LJng@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
	<CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
	<CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>
	<CAFiqYukX75FDSpAT9zdvnOBb_boJxQAZKJSk2QhuJKroY_LJng@mail.gmail.com>
Message-ID: <CAAAWEqDAh=0bXG=+3gsneO+R6TmNW4BE2b2G=9P3rZqajY9wxQ@mail.gmail.com>

Hi Steve,

Thanks for all your help, I have working code now :-)

I don't know if this is a local configuration "feature" but the events
are returned synchronously from the "event plain CHANNEL_HANGUP" call,
so I just loop looking for my uuid.

Cheers,
Rich

PS For anyone interested in some alpha code ...

$dExt = $_GET['dExt'];
$pin = $_GET['pin'];

if ($dExt and $pin) {
    $fp = event_socket_create($host, $port, $password);

    # How shall we abuse the callee today?
    $wav1 = 'ivr/32000/ivr-douche_telecom.wav';
    $wav2 = 'ivr/32000/ivr-thank_you_for_calling.wav';

    # Place call and find uuid
    $cmd = "api originate
{ignore_early_media=true,originate_timeout=30}sofia/external/677${dExt}@MGC
'&play_and_get_digits(4 9 1 10000 # $wav1 $wav2 foo \d+)'";
    $response = event_socket_request($fp, $cmd);
    $response = strip_cr($response);
    $bits = preg_split('/\s+/',$response);
    $uuid = $bits[1];

    $continue = 1;
    $cmd = "event plain CHANNEL_HANGUP";
    while ($continue) {
        error_log("running $cmd");
        $response = event_socket_request($fp, $cmd);
        error_log("got $response");
        if (preg_match("/$uuid/",$response)) {
            $continue = 0;
        }
    }

    $dbhRo = DbConnect::GetConnection('db-readonly');
    $res_struct = mysql_query("SELECT strSrcMedia FROM cdr WHERE
strSipSessionId = '$uuid'",$dbhRo);
    $res = mysql_fetch_array($res_struct);
    $pin_from_user = $res[0];
    $result = $pin == $pin_from_user ? 1 : 0 ;

    echo "<pre>";
    echo date('H:i:s') . "\n";
    echo "uuid = $uuid\n";
    echo "$result\n";
    echo "</pre>";

    # Close socket
    fclose($fp);
} else {
    echo -1;
}


On 29 November 2012 13:56, Steven Ayre <steveayre at gmail.com> wrote:
>> ... how I listen for these events. Is it just a case of polling the
>> socket for them at frequent enough intervals?
>
>
> You subscribe to the events you're interested in with 'events'
> http://wiki.freeswitch.org/wiki/Event_socket#event
>
> There's no polling involved. FS will send all subscribed events through the
> socket, you just need to receive them. Use select()/poll()/equivalent to
> detect when data has arrived.
>
> Since you already have xml_cdr working that would be the best place to
> collect it, if you can add an extra DB insert in the handler.
>
> -Steve
>
>
> On 29 November 2012 11:58, Richard Gration <richgration at gmail.com> wrote:
>>
>> Hi,
>>
>> Thanks for your reply. I was wondering ...
>>
>> > If you listen to events then you should be able to reliably receive a
>> > hangup
>> > event containing the variables if you are still connected at that time.
>>
>> ... how I listen for these events. Is it just a case of polling the
>> socket for them at frequent enough intervals?
>>
>> > If you want a 100% reliable method use CDRs.
>> >
>> > mod_xml_cdr can submit call CDRs to a webserver just after call hangup.
>> > The
>> > XML is verbose and will contain a full call history and all channel
>> > variables. Because it's submitted ASAP you get the results in real time
>> > and
>> > reliable. If the module can't submit the CDRs it writes them to an error
>> > folder on disk where you can resubmit them.
>> >
>> > Real time seems useful for your use-case, as your script would just need
>> > to
>> > check your DB whether the CDR had been submitted yet. The channel UUID
>> > can
>> > either be captured from events or specified in advance in the originate
>> > (originate_uuid). There's a uuid api call to generate them for you. I'd
>> > suggest looking at this method.
>>
>> This is all great info, thanks :-) I'm capturing the uuid from the
>> response from the originate command, that's no problem.
>>
>> The box I'm doing this on is being a B2BUA for us at the moment. The
>> CDRs are already being HTTP POSTed to a URL, I can't change that as
>> our billing depends on it. I can see the channel variable coming
>> though in the POST content, so maybe I can frob the CDR script to give
>> me what I want.
>>
>> Thanks again for the info.
>>
>> Cheers,
>> Rich
>>
>> --
>> Once our basic material needs are met - in my utopia, anyway - life
>> becomes a perpetual celebration in which everyone has a talent to
>> contribute. But we cannot levitate ourselves into that blessed
>> condition by wishing it. We need to brace ourselves for a struggle
>> against terrifying obstacles, both of our own making and imposed by
>> the natural world. And the first step is to recover from the delusion
>> that is positive thinking.
>>        -- Barbara Ehrenreich
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Once our basic material needs are met - in my utopia, anyway - life
becomes a perpetual celebration in which everyone has a talent to
contribute. But we cannot levitate ourselves into that blessed
condition by wishing it. We need to brace ourselves for a struggle
against terrifying obstacles, both of our own making and imposed by
the natural world. And the first step is to recover from the delusion
that is positive thinking.
       -- Barbara Ehrenreich


From steveayre at gmail.com  Thu Nov 29 22:43:25 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 19:43:25 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAAAWEqDAh=0bXG=+3gsneO+R6TmNW4BE2b2G=9P3rZqajY9wxQ@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
	<CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
	<CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>
	<CAFiqYukX75FDSpAT9zdvnOBb_boJxQAZKJSk2QhuJKroY_LJng@mail.gmail.com>
	<CAAAWEqDAh=0bXG=+3gsneO+R6TmNW4BE2b2G=9P3rZqajY9wxQ@mail.gmail.com>
Message-ID: <CAFiqYukszAL9z+f8hwf59+7KLoDQtxfiSHSSCv9ZWsNawW_cpQ@mail.gmail.com>

In this case just receiving events until yours arrives is good enough. A
regex for the UUID is

My suggestion of select/poll/etc would be for a script using a single
connection to start multiple calls, where you'd need to multiplex between
detecting start call events (timeout/dbrow/etc) and incoming events. Since
you're not sending anything else just receiving events works fine.

It's be better to parse events and check the Unique-ID header, but a simple
regex would work. It'd fail in cases like where calls are bridged (the uuid
will appear in events of both call legs) but in this case there's only one
leg so that's probably going to be good enough.

As an optimisation, since you already know the UUID and only watch a single
one on the connection you can use the 'myevents <uuid>' command - that'll
limit it to only events for that channel, so you'll no longer waste
processing on events for other channels and no longer need to check the
uuid in the event.

Since you're still always waiting for CHANNEL_HANGUP and assuming you still
have the ESL connection up, does the variable you're looking appear in the
hangup event? If so perhaps CDR isn't needed.

The wait for the CHANNEL_HANGUP event isn't strictly speaking required, you
could also poll the DB. But waiting for CHANNEL_HANGUP will have the
advantage of not checking the DB until it expects the CDR to be available.

Since


On 29 November 2012 19:23, Richard Gration <richgration at gmail.com> wrote:

> Hi Steve,
>
> Thanks for all your help, I have working code now :-)
>
> I don't know if this is a local configuration "feature" but the events
> are returned synchronously from the "event plain CHANNEL_HANGUP" call,
> so I just loop looking for my uuid.
>
> Cheers,
> Rich
>
> PS For anyone interested in some alpha code ...
>
> $dExt = $_GET['dExt'];
> $pin = $_GET['pin'];
>
> if ($dExt and $pin) {
>     $fp = event_socket_create($host, $port, $password);
>
>     # How shall we abuse the callee today?
>     $wav1 = 'ivr/32000/ivr-douche_telecom.wav';
>     $wav2 = 'ivr/32000/ivr-thank_you_for_calling.wav';
>
>     # Place call and find uuid
>     $cmd = "api originate
> {ignore_early_media=true,originate_timeout=30}sofia/external/677${dExt}@MGC
> '&play_and_get_digits(4 9 1 10000 # $wav1 $wav2 foo \d+)'";
>     $response = event_socket_request($fp, $cmd);
>     $response = strip_cr($response);
>     $bits = preg_split('/\s+/',$response);
>     $uuid = $bits[1];
>
>     $continue = 1;
>     $cmd = "event plain CHANNEL_HANGUP";
>     while ($continue) {
>         error_log("running $cmd");
>         $response = event_socket_request($fp, $cmd);
>         error_log("got $response");
>         if (preg_match("/$uuid/",$response)) {
>             $continue = 0;
>         }
>     }
>
>     $dbhRo = DbConnect::GetConnection('db-readonly');
>     $res_struct = mysql_query("SELECT strSrcMedia FROM cdr WHERE
> strSipSessionId = '$uuid'",$dbhRo);
>     $res = mysql_fetch_array($res_struct);
>     $pin_from_user = $res[0];
>     $result = $pin == $pin_from_user ? 1 : 0 ;
>
>     echo "<pre>";
>     echo date('H:i:s') . "\n";
>     echo "uuid = $uuid\n";
>     echo "$result\n";
>     echo "</pre>";
>
>     # Close socket
>     fclose($fp);
> } else {
>     echo -1;
> }
>
>
> On 29 November 2012 13:56, Steven Ayre <steveayre at gmail.com> wrote:
> >> ... how I listen for these events. Is it just a case of polling the
> >> socket for them at frequent enough intervals?
> >
> >
> > You subscribe to the events you're interested in with 'events'
> > http://wiki.freeswitch.org/wiki/Event_socket#event
> >
> > There's no polling involved. FS will send all subscribed events through
> the
> > socket, you just need to receive them. Use select()/poll()/equivalent to
> > detect when data has arrived.
> >
> > Since you already have xml_cdr working that would be the best place to
> > collect it, if you can add an extra DB insert in the handler.
> >
> > -Steve
> >
> >
> > On 29 November 2012 11:58, Richard Gration <richgration at gmail.com>
> wrote:
> >>
> >> Hi,
> >>
> >> Thanks for your reply. I was wondering ...
> >>
> >> > If you listen to events then you should be able to reliably receive a
> >> > hangup
> >> > event containing the variables if you are still connected at that
> time.
> >>
> >> ... how I listen for these events. Is it just a case of polling the
> >> socket for them at frequent enough intervals?
> >>
> >> > If you want a 100% reliable method use CDRs.
> >> >
> >> > mod_xml_cdr can submit call CDRs to a webserver just after call
> hangup.
> >> > The
> >> > XML is verbose and will contain a full call history and all channel
> >> > variables. Because it's submitted ASAP you get the results in real
> time
> >> > and
> >> > reliable. If the module can't submit the CDRs it writes them to an
> error
> >> > folder on disk where you can resubmit them.
> >> >
> >> > Real time seems useful for your use-case, as your script would just
> need
> >> > to
> >> > check your DB whether the CDR had been submitted yet. The channel UUID
> >> > can
> >> > either be captured from events or specified in advance in the
> originate
> >> > (originate_uuid). There's a uuid api call to generate them for you.
> I'd
> >> > suggest looking at this method.
> >>
> >> This is all great info, thanks :-) I'm capturing the uuid from the
> >> response from the originate command, that's no problem.
> >>
> >> The box I'm doing this on is being a B2BUA for us at the moment. The
> >> CDRs are already being HTTP POSTed to a URL, I can't change that as
> >> our billing depends on it. I can see the channel variable coming
> >> though in the POST content, so maybe I can frob the CDR script to give
> >> me what I want.
> >>
> >> Thanks again for the info.
> >>
> >> Cheers,
> >> Rich
> >>
> >> --
> >> Once our basic material needs are met - in my utopia, anyway - life
> >> becomes a perpetual celebration in which everyone has a talent to
> >> contribute. But we cannot levitate ourselves into that blessed
> >> condition by wishing it. We need to brace ourselves for a struggle
> >> against terrifying obstacles, both of our own making and imposed by
> >> the natural world. And the first step is to recover from the delusion
> >> that is positive thinking.
> >>        -- Barbara Ehrenreich
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Once our basic material needs are met - in my utopia, anyway - life
> becomes a perpetual celebration in which everyone has a talent to
> contribute. But we cannot levitate ourselves into that blessed
> condition by wishing it. We need to brace ourselves for a struggle
> against terrifying obstacles, both of our own making and imposed by
> the natural world. And the first step is to recover from the delusion
> that is positive thinking.
>        -- Barbara Ehrenreich
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Nov 29 22:43:34 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 19:43:34 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAAAWEqDAh=0bXG=+3gsneO+R6TmNW4BE2b2G=9P3rZqajY9wxQ@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
	<CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
	<CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>
	<CAFiqYukX75FDSpAT9zdvnOBb_boJxQAZKJSk2QhuJKroY_LJng@mail.gmail.com>
	<CAAAWEqDAh=0bXG=+3gsneO+R6TmNW4BE2b2G=9P3rZqajY9wxQ@mail.gmail.com>
Message-ID: <CAFiqYumcs0r9X06vSkOuujdeB13iYLanqBVjsr2ucLWWR2AaTA@mail.gmail.com>

I would recommend that you adjust your DB check code to run in a loop. CDR
submission occurs *after* CHANNEL_HANGUP - so you have a race condition and
may check the DB before the CDR has been committed. You'd need to handle a)
trying again after a short delay if the CDR isn't there yet and b) a
timeout in case xml_cdr fails and the CDR isn't available (or the script'll
keep running until it is).


On 29 November 2012 19:23, Richard Gration <richgration at gmail.com> wrote:

> Hi Steve,
>
> Thanks for all your help, I have working code now :-)
>
> I don't know if this is a local configuration "feature" but the events
> are returned synchronously from the "event plain CHANNEL_HANGUP" call,
> so I just loop looking for my uuid.
>
> Cheers,
> Rich
>
> PS For anyone interested in some alpha code ...
>
> $dExt = $_GET['dExt'];
> $pin = $_GET['pin'];
>
> if ($dExt and $pin) {
>     $fp = event_socket_create($host, $port, $password);
>
>     # How shall we abuse the callee today?
>     $wav1 = 'ivr/32000/ivr-douche_telecom.wav';
>     $wav2 = 'ivr/32000/ivr-thank_you_for_calling.wav';
>
>     # Place call and find uuid
>     $cmd = "api originate
> {ignore_early_media=true,originate_timeout=30}sofia/external/677${dExt}@MGC
> '&play_and_get_digits(4 9 1 10000 # $wav1 $wav2 foo \d+)'";
>     $response = event_socket_request($fp, $cmd);
>     $response = strip_cr($response);
>     $bits = preg_split('/\s+/',$response);
>     $uuid = $bits[1];
>
>     $continue = 1;
>     $cmd = "event plain CHANNEL_HANGUP";
>     while ($continue) {
>         error_log("running $cmd");
>         $response = event_socket_request($fp, $cmd);
>         error_log("got $response");
>         if (preg_match("/$uuid/",$response)) {
>             $continue = 0;
>         }
>     }
>
>     $dbhRo = DbConnect::GetConnection('db-readonly');
>     $res_struct = mysql_query("SELECT strSrcMedia FROM cdr WHERE
> strSipSessionId = '$uuid'",$dbhRo);
>     $res = mysql_fetch_array($res_struct);
>     $pin_from_user = $res[0];
>     $result = $pin == $pin_from_user ? 1 : 0 ;
>
>     echo "<pre>";
>     echo date('H:i:s') . "\n";
>     echo "uuid = $uuid\n";
>     echo "$result\n";
>     echo "</pre>";
>
>     # Close socket
>     fclose($fp);
> } else {
>     echo -1;
> }
>
>
> On 29 November 2012 13:56, Steven Ayre <steveayre at gmail.com> wrote:
> >> ... how I listen for these events. Is it just a case of polling the
> >> socket for them at frequent enough intervals?
> >
> >
> > You subscribe to the events you're interested in with 'events'
> > http://wiki.freeswitch.org/wiki/Event_socket#event
> >
> > There's no polling involved. FS will send all subscribed events through
> the
> > socket, you just need to receive them. Use select()/poll()/equivalent to
> > detect when data has arrived.
> >
> > Since you already have xml_cdr working that would be the best place to
> > collect it, if you can add an extra DB insert in the handler.
> >
> > -Steve
> >
> >
> > On 29 November 2012 11:58, Richard Gration <richgration at gmail.com>
> wrote:
> >>
> >> Hi,
> >>
> >> Thanks for your reply. I was wondering ...
> >>
> >> > If you listen to events then you should be able to reliably receive a
> >> > hangup
> >> > event containing the variables if you are still connected at that
> time.
> >>
> >> ... how I listen for these events. Is it just a case of polling the
> >> socket for them at frequent enough intervals?
> >>
> >> > If you want a 100% reliable method use CDRs.
> >> >
> >> > mod_xml_cdr can submit call CDRs to a webserver just after call
> hangup.
> >> > The
> >> > XML is verbose and will contain a full call history and all channel
> >> > variables. Because it's submitted ASAP you get the results in real
> time
> >> > and
> >> > reliable. If the module can't submit the CDRs it writes them to an
> error
> >> > folder on disk where you can resubmit them.
> >> >
> >> > Real time seems useful for your use-case, as your script would just
> need
> >> > to
> >> > check your DB whether the CDR had been submitted yet. The channel UUID
> >> > can
> >> > either be captured from events or specified in advance in the
> originate
> >> > (originate_uuid). There's a uuid api call to generate them for you.
> I'd
> >> > suggest looking at this method.
> >>
> >> This is all great info, thanks :-) I'm capturing the uuid from the
> >> response from the originate command, that's no problem.
> >>
> >> The box I'm doing this on is being a B2BUA for us at the moment. The
> >> CDRs are already being HTTP POSTed to a URL, I can't change that as
> >> our billing depends on it. I can see the channel variable coming
> >> though in the POST content, so maybe I can frob the CDR script to give
> >> me what I want.
> >>
> >> Thanks again for the info.
> >>
> >> Cheers,
> >> Rich
> >>
> >> --
> >> Once our basic material needs are met - in my utopia, anyway - life
> >> becomes a perpetual celebration in which everyone has a talent to
> >> contribute. But we cannot levitate ourselves into that blessed
> >> condition by wishing it. We need to brace ourselves for a struggle
> >> against terrifying obstacles, both of our own making and imposed by
> >> the natural world. And the first step is to recover from the delusion
> >> that is positive thinking.
> >>        -- Barbara Ehrenreich
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Once our basic material needs are met - in my utopia, anyway - life
> becomes a perpetual celebration in which everyone has a talent to
> contribute. But we cannot levitate ourselves into that blessed
> condition by wishing it. We need to brace ourselves for a struggle
> against terrifying obstacles, both of our own making and imposed by
> the natural world. And the first step is to recover from the delusion
> that is positive thinking.
>        -- Barbara Ehrenreich
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bdfoster at endigotech.com  Thu Nov 29 23:01:35 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 29 Nov 2012 15:01:35 -0500
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <01f101cdce1a$79309130$6b91b390$@co.in>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>
	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>
	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>
	<012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>
	<00c101cdcdeb$b38bbaa0$1aa32fe0$@co.in>
	<CAFiqYuk9cYxDMyxetKHdxixukk3u1TCpr3GhYBm4Ccxvt1ZkRQ@mail.gmail.com>
	<01f101cdce1a$79309130$6b91b390$@co.in>
Message-ID: <14215987-3200-4B8D-82E9-D019DE605962@endigotech.com>

Please start a new thread instead of hitting 'Reply' and changing the Subject. This can cause issues with those who use services like Nabble and even cause logging issues with the mailing list. 

Sent from my iPhone

On Nov 29, 2012, at 5:15 AM, "Nitin Tomer" <ntomer at newgen.co.in> wrote:

> Hi,
>  
> I am using valet_park. I?ve configure a IVR menu of an extension, based on user?s input call is forwarded to other extensions.
>  
> Extension on which end-users will call ?
>  
> <extension name="ivr_nitin">
>       <condition field="destination_number" expression="^5002$">
>         <action application="answer"/>
>         <action application="sleep" data="500"/>
>         <action application="ivr" data="nitin_ivr"/>
>       </condition>
> </extension>
>  
> The IVR configuration XML is ?
>  
> <include>
>         <menu name="nitin_ivr"
>                 greet-long="say:Welcome to Newgen General Insurance Company. Press 1 for Changing Address, 2 for Changing Nominee or 3 for Close Policy."
>                 greet-short="say:Welcome to Newgen. Press 1 for Changing Address, 2 for Changing Nominee or 3 for Close Policy."
>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>                 exit-sound="voicemail/vm-goodbye.wav"
>                 confirm-macro=""
>                 confirm-key=""
>                 tts-engine="flite"
>                 tts-voice="slt"
>                 confirm-attempts="3"
>                 timeout="3000"
>                 inter-digit-timeout="2000"
>                 max-failures="3"
>                 max-timeouts="3"
>                 digit-len="4">
>  
>                 <entry action="menu-exec-app" digits="1" param="transfer 450 XML default"/>    <!-- Change Address -->
>                 <entry action="menu-exec-app" digits="2" param="transfer 451 XML default"/>    <!-- Change Nominee -->
>                 <entry action="menu-exec-app" digits="3" param="transfer 452 XML default"/>    <!-- Close Policy -->
> 
>         </menu>
> </include>
> 
> Once user presses ?1?, call is forwarded to 450, for this extension dialplan entry is ?
>  
> <extension>
>           <condition field="destination_number" expression="^(450)$">
>           <action application="play_and_get_digits" data="10 16 3 3000 # say:'Press your account number, followed by hash key' say:'Wrong Input' res \d+" />
>           <action application="phrase" data="spell,${res}"/>
>           <action application="db" data="insert/testapp/newcall1/${res}" />    
>               <action application="set" data="caller_id_name=Change Address Request" />
>               <action application="set" data="call_timeout=60" />
>               <action application="set" data="originate_timeout=60" />
>               <action application="valet_park" data="my_lot auto in 8501 8599" />           
>           </condition>
> </extension>
>  
> Here, the call is parked at any available extension between 8501 to 8599.
>  
> Then I?ve set up an extension to pick up calls ?
>  
> <extension name="park-out">
>    <condition field="destination_number" expression="^(85\d\d)$">
>      <action application="answer"/>
>      <action application="valet_park" data="my_lot $1"/>
>    </condition>
>  </extension>
>  
> I have a few questions ?
>  
> 1.       Valet_park parks the call on any available extension between 8501 to 8599 (<action application="valet_park" data="my_lot auto in 8501 8599" />). Is there any way to let me know on which extension the call have been parked?
> 2.       How can I get the number from which call was made in extension 450. The idea is to use the caller number as key and entered value as value while making entry in database (<action application="db" data="insert/testapp/newcall1/${res}" />)?
> 3.       If two users call on extension 5002 (where IVR menu is played), what will happen? Will the second user have to wait for first to finish or whether both will be connected parallel?
>  
> Please help me out.
>  
> Regards
>  
> Nitin
> 
> Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL) accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mickstevens at yahoo.com  Thu Nov 29 23:35:31 2012
From: mickstevens at yahoo.com (Mick Stevens)
Date: Thu, 29 Nov 2012 12:35:31 -0800 (PST)
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in
	XML CDR's?
In-Reply-To: <CAKzWOxUcpgaS29KWmSHg8FmMAraOVAa75i71sXdpbBC0S7EC_w@mail.gmail.com>
References: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>
	<CAKzWOxUcpgaS29KWmSHg8FmMAraOVAa75i71sXdpbBC0S7EC_w@mail.gmail.com>
Message-ID: <1354221331.77886.YahooMailNeo@web160803.mail.bf1.yahoo.com>

Hi Michael & ?Co,

Many thanks for the prompt & positive response! Yes, more than happy to collate feedback from the global team & update the wiki accordingly (I'd like to be able to document what the field values between the > < indicate as well as just explain the field names if possible...) +anything else I can do to to contribute to the project...?

Yes, now the wiki is back up I've managed to work out that cng_packet = comfort noise generation! Also from the wiki, possibly that the jb in?rtp_audio_in_jb_packet_count &?rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!).

To provide the background to my original enquiry, I'm trying to identify if the rtp_audio_in_skip_packet_count in the following XML CDR extract is indicative of packet loss, or perhaps more accurately noticeable audio loss (as the packets haven't been lost, just ignored/"skipped"?: If anybody knows please speak up!?

? ? <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
? ? <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
? ? <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
? ? <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
? ? <rtp_audio_in_skip_packet_count>1609</rtp_audio_in_skip_packet_count>
? ? <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
? ? <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
? ? <rtp_audio_in_cng_packet_count>15</rtp_audio_in_cng_packet_count>
? ? <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>

Thank you in anticipation of enlightenment!

#FreeSWITCH
?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429

Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens



________________________________
 From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, 29 November 2012, 19:14
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
 

Mick,

This data is definitely not on the wiki - or anywhere else that I can see. I think we can crowdsource this to get the info collected and then add it to the mod_xml_curl wiki page. For the record, here's a quick dump of the fields that I took from an XML CDR:

??? <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
??? <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
??? <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
??? <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
??? <rtp_audio_in_skip_packet_count>18</rtp_audio_in_skip_packet_count>
??? <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
??? <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
??? <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
??? <rtp_audio_in_flush_packet_count>11</rtp_audio_in_flush_packet_count>
??? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
??? <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
??? <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
??? <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
??? <rtp_audio_out_media_packet_count>67</rtp_audio_out_media_packet_count>
??? <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
??? <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
??? <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>

I think raw_bytes, media_bytes, packet_count, and media_packet_count are self-explanatory. I think cng_packet_count is probably self-explanatory too. My question on dtmf_packet_count would be whether it's only for RFC2833 packets (I suspect yes, but would like confirmation). 

If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;)

rtp_audio_in_skip_packet_count
rtp_audio_out_skip_packet_count
rtp_audio_in_jb_packet_count
rtp_audio_in_flush_packet_count
rtp_audio_in_largest_jb_size

Thanks all!

-MC


On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com> wrote:

Hi Folks,
>
>
>I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example?
>
>
>I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there.
>
>
>I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured.
>
>
>Any help much appreciated!
>?
>Rgds, Mick
>Tel/SMS. +44(0)7967 594432
>Fax. +44(0)7053 452429
>
>Email/IM. mickstevens at yahoo.com
>Skype: mick_stevens
>www.facebook.com/mickstevens
>www.twitter.com/mickstevens
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



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




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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Thu Nov 29 23:50:42 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Nov 2012 14:50:42 -0600
Subject: [Freeswitch-users] Sofia freezing for 5 minutes then starting
	again
In-Reply-To: <50B7A4BE.8040108@communicatefreely.net>
References: <CCD7A44C.610F0%krice@freeswitch.org>
	<50B4249A.30702@communicatefreely.net>
	<CAKbxfG-Mnm-yhx0D6mp=WM3xCC6bmg=OmSy0ph8YU-ecL31V9Q@mail.gmail.com>
	<50B7A4BE.8040108@communicatefreely.net>
Message-ID: <CAKbxfG-e1oEhrGHAtgsofnP=0Tfv+zQ3kkAOxxbhaNjcD9JsLg@mail.gmail.com>

cliff notes version would be

gcore `cat /usr/local/freeswitch/log/freeswitch.pid`

Take the resulting core file and follow the backtrace instructions at

http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace






On Thu, Nov 29, 2012 at 12:09 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> It's not a matter of not taking your suggestion, it's an issue of not
> having had the time
> yet to learn how gcore works and put it in place on a production system.
>  It's on my
> "to-do" list, but I thought I would at least update the thread, hoping
> maybe that message
> meant something to someone.
>
>
>
> Anthony Minessale wrote:
> > too bad you did not gcore it like I suggested, otherwise who knows...
> > That's just the message it prints when the message parsing thread goes
> up.
> >
> >
> >
> > On Mon, Nov 26, 2012 at 8:25 PM, Tim St. Pierre
> > <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
> > wrote:
> >
> >     Thanks for all the input everyone!
> >
> >     I just had it happen again, this time at night when there were only
> >     2 sessions active!
> >
> >     I was in doing some other work, so I managed to see it happen and
> >     found a few interesting
> >     things.
> >
> >     CPU load next to nothing, load average around 0.12.
> >     I used iSQL to test ODBC connectivity, and I could read and write to
> >     the freeswitch
> >     database using the same DSN that freeswitch is using.
> >
> >     I could delete a SIP registration from the API
> >
> >     Since I had things on a higher debug level, I saw this:
> >
> >     freeswitch at stefan> 2012-11-26 21:21:51.914082 [CONSOLE] sofia.c:1144
> >     MSG Thread Started
> >
> >     As soon as that came up, everything started working again.
> >
> >     What's that about?
> >
> >     -Tim
> >
> >     Ken Rice wrote:
> >      > Sofia is not single threaded except for in one spot deep in
> libsofia,
> >      >>From there, messages are handed off to a number of message queues
> >     for FS
> >      > core to handle as needed...
> >      >
> >      > Check to see if anything that fs is depending on is blocking on
> info
> >      > retrieval like the databases or other areas...
> >      >
> >      > K
> >      >
> >      > On 11/25/12 9:51 AM, "Abaci" <abaci64 at gmail.com
> >     <mailto:abaci64 at gmail.com>> wrote:
> >      >
> >      >> you mentioned that you use xml_curl, if your web server hangs it
> may
> >      >> hang sofia, iirc sofia is running a single thread and it will
> >     wait for
> >      >> the xml_curl response before continuing to the next request.
> >      >>
> >      >> On 11/23/2012 1:13 PM, Steven Ayre wrote:
> >      >>> Any kind of DB backup running? Or any long-running queries
> >     (innotop is
> >      >>> great for highlighting queries that've been running a while,
> >     including
> >      >>> on non-innodb tables).
> >      >>>
> >      >>> A global read lock, or queries waiting for a lock could block a
> db
> >      >>> update from the sofia profile thread but still allow read-only
> >     queries
> >      >>> (sofia status) to run.
> >      >>>
> >      >>> -Steve
> >      >>>
> >      >>>
> >      >>>
> >      >>> On 23 November 2012 15:32, Tim St. Pierre
> >     <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net
> >>
> >      >>> wrote:
> >      >>>> Hi Steven,
> >      >>>>
> >      >>>> Thanks for the suggestions.  I'm hoping once I get the upgrade
> >     done it will
> >      >>>> all go away.
> >      >>>> I have watched it happen at least once.  I was on the phone at
> >     the time.
> >      >>>> Console activity
> >      >>>> more or less stopped, except for a few calls hanging up.  The
> >     console
> >      >>>> remains responsive,
> >      >>>> and my call wasn't dropped for at least a minute or two (media
> >     timeout?).  I
> >      >>>> was able to
> >      >>>> run sofia status and other commands that use the database, so
> >     I'm assuming
> >      >>>> that the
> >      >>>> connection was still working.  All our media is runs through
> >     the box, so I
> >      >>>> think things
> >      >>>> are fine on the Ethernet level.  I do see higher load averages
> >     - maybe 3-4,
> >      >>>> but that's the
> >      >>>> only obvious indication.  It's not taking CPU beyond 10% or so.
> >      >>>>
> >      >>>> We are using MySQL as the core DB and also as the DB backend
> >     for each sofia
> >      >>>> profile.  This
> >      >>>> is connecting through ODBC of course.
> >      >>>>
> >      >>>> If I can get the other kinks worked out, then I will try 1.2
> >     stable in
> >      >>>> production and
> >      >>>> we'll see how it goes.
> >      >>>>
> >      >>>> -Tim
> >      >>>>
> >      >>>> Steven Ayre wrote:
> >      >>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20
> >     -0500)
> >      >>>>>
> >      >>>>>
> >      >>>>> As you've already acknowledged it's a very old version.
> >      >>>>>
> >      >>>>> It's possible that your issue has already been found and
> >     fixed, but if
> >      >>>>> it hasn't then the code will have changed significantly since
> >     then and
> >      >>>>> you'd really need to reproduce it on the latest code for it
> to be
> >      >>>>> investigated.
> >      >>>>>
> >      >>>>>
> >      >>>>> As some general thoughts though, are you able to spot it
> >     happening while
> >      >>>>> it's happening or only afterwards?
> >      >>>>>
> >      >>>>> If you're able to get on the system during one of those times
> >     look at
> >      >>>>> what else is happening. Is the load average/cpu usage/io
> >     high? Perhaps
> >      >>>>> something's running that's blocking all access or causing
> >     very high IO.
> >      >>>>>
> >      >>>>> What DB backend are you using for Sofia? Is it possible that
> >     that's
> >      >>>>> hanging for a moment? For example if you're running a backup
> >     on the DB
> >      >>>>> that blocks all writes to the DB while Sofia is trying to
> >     update the DB
> >      >>>>> that perhaps would cause this.
> >      >>>>>
> >      >>>>> Try running a SIP OPTIONS ping your your sofia profile from
> the
> >      >>>>> localhost during that time, which should exclude it being any
> >     issue on
> >      >>>>> the ethernet.
> >      >>>>>
> >      >>>>> -Steve
> >      >>>>>
> >      >>>>>
> >      >>>>>
> >      >>>>>
> >      >>>>> On 22 November 2012 19:31, Tim St. Pierre
> >     <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net
> >
> >      >>>>> <mailto:fs-list at communicatefreely.net
> >     <mailto:fs-list at communicatefreely.net>>> wrote:
> >      >>>>>
> >      >>>>>      Hello,
> >      >>>>>
> >      >>>>>      I'm having a bit of an odd problem.
> >      >>>>>
> >      >>>>>      Intermittently, often every 2-3 days or so, Freeswitch
> stops
> >      >>>>>      replying to SIP for about 5
> >      >>>>>      minutes.  I can't verify if it's EXACTLY 5 minutes, but
> >     it seems to
> >      >>>>>      be pretty close.
> >      >>>>>
> >      >>>>>      During this time, no new registrations or invites can
> >     happen, but
> >      >>>>>      existing calls stay
> >      >>>>>      connected for at least a minute or two.  In the logs,
> >     you can see
> >      >>>>>      calls slowly hanging up
> >      >>>>>      with "NORMAL_CLEARING".  In 5 minutes, everything starts
> >     up again
> >      >>>>>      with no word about it at
> >      >>>>>      all in the logs.
> >      >>>>>
> >      >>>>>      When calls resume, I notice that the number of sessions
> >     returned by
> >      >>>>>      the status command is
> >      >>>>>      one higher than the actual number sessions returned by
> show
> >      >>>>>      channels, or by looking in the
> >      >>>>>      database.  Every time this happens, the discrepancy
> >     increases by one.
> >      >>>>>
> >      >>>>>      The interruption happens on all SIP profiles, but calls
> >     originated
> >      >>>>>      from the socket API
> >      >>>>>      still work, insofar as they return with PROGRESS_TIMEOUT
> >     since the
> >      >>>>>      profiles are still
> >      >>>>>      running, but stuck.
> >      >>>>>
> >      >>>>>      We are using ODBC/MySQL for the core database, and the
> >     database
> >      >>>>>      server only runs this
> >      >>>>>      database and some basic PHP/xml-curl stuff.
> >      >>>>>
> >      >>>>>      We have 416 endpoints registered, and usually sit at
> >     about 30
> >      >>>>>      sessions during the day.
> >      >>>>>
> >      >>>>>      This never happens at night, only during busier times,
> >     but not
> >      >>>>>      necessarily busy hour.
> >      >>>>>
> >      >>>>>      I'm running on FreeBSD 8.2-RELEASE AMD 64(2 XEON cores,
> >     4G ram)
> >      >>>>>
> >      >>>>>      Freeswitch is 1.0.head (git-7531fed 2011-08-17 11-27-20
> >     -0500)
> >      >>>>>
> >      >>>>>      Yes, I know it's old and I'm trying to upgrade, but I'm
> >     still having
> >      >>>>>      some problems getting
> >      >>>>>      all my phones to work properly with 1.2 stable.  This is
> a
> >      >>>>>      production system, so I can't
> >      >>>>>      just blindly put out the newest release.  Mostly, I need
> >     to buy
> >      >>>>>      myself some time so that I
> >      >>>>>      can get the kinks worked out of the latest version and
> >     then upgrade
> >      >>>>>      the production box.
> >      >>>>>
> >      >>>>>      I'm grateful for any insights as to what could be
> >     happening, even if
> >      >>>>>      a solution is just a
> >      >>>>>      temporary workaround.
> >      >>>>>
> >      >>>>>      Thanks!
> >      >>>>>
> >      >>>>>      -Tim
> >      >>>>>
> >      >>>>>
> >      >>>>>
> >
> _________________________________________________________________________
> >      >>>>>      Professional FreeSWITCH Consulting Services:
> >      >>>>>      consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org> <mailto:consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>>
> >      >>>>>      http://www.freeswitchsolutions.com
> >      >>>>>
> >      >>>>>      FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> >      >>>>>      
> >      >>>>>
> >      >>>>>      Official FreeSWITCH Sites
> >      >>>>>      http://www.freeswitch.org
> >      >>>>>      http://wiki.freeswitch.org
> >      >>>>>      http://www.cluecon.com
> >      >>>>>
> >      >>>>>      FreeSWITCH-users mailing list
> >      >>>>>      FreeSWITCH-users at lists.freeswitch.org
> >     <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
> >      >>>>>
> >      >>>>>
> >      >>>>>
> >      >>>>>
> >
> ------------------------------------------------------------------------
> >      >>>>>
> >      >>>>>
> >
> _________________________________________________________________________
> >      >>>>> Professional FreeSWITCH Consulting Services:
> >      >>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >      >>>>> http://www.freeswitchsolutions.com
> >      >>>>>
> >      >>>>> 
> >      >>>>> 
> >      >>>>>
> >      >>>>> Official FreeSWITCH Sites
> >      >>>>> http://www.freeswitch.org
> >      >>>>> http://wiki.freeswitch.org
> >      >>>>> http://www.cluecon.com
> >      >>>>>
> >      >>>>> FreeSWITCH-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
> >      >>>>
> >
> _________________________________________________________________________
> >      >>>> Professional FreeSWITCH Consulting Services:
> >      >>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >      >>>> http://www.freeswitchsolutions.com
> >      >>>>
> >      >>>> 
> >      >>>> 
> >      >>>>
> >      >>>> Official FreeSWITCH Sites
> >      >>>> http://www.freeswitch.org
> >      >>>> http://wiki.freeswitch.org
> >      >>>> http://www.cluecon.com
> >      >>>>
> >      >>>> FreeSWITCH-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
> >      >>>
> >
> _________________________________________________________________________
> >      >>> Professional FreeSWITCH Consulting Services:
> >      >>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >      >>> http://www.freeswitchsolutions.com
> >      >>>
> >      >>> 
> >      >>> 
> >      >>>
> >      >>> Official FreeSWITCH Sites
> >      >>> http://www.freeswitch.org
> >      >>> http://wiki.freeswitch.org
> >      >>> http://www.cluecon.com
> >      >>>
> >      >>> FreeSWITCH-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
> >      >>
> >      >>
> >
> _________________________________________________________________________
> >      >> Professional FreeSWITCH Consulting Services:
> >      >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >      >> http://www.freeswitchsolutions.com
> >      >>
> >      >> 
> >      >> 
> >      >>
> >      >> Official FreeSWITCH Sites
> >      >> http://www.freeswitch.org
> >      >> http://wiki.freeswitch.org
> >      >> http://www.cluecon.com
> >      >>
> >      >> FreeSWITCH-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
> >      >
> >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     FreeSWITCH-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
> >
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From covici at ccs.covici.com  Fri Nov 30 00:42:45 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 29 Nov 2012 16:42:45 -0500
Subject: [Freeswitch-users] Compilation Issue
In-Reply-To: <CA+zebiYzT75EewBfBfWWvXDNpPcdc+sfToconYKv+PBvoBDnng@mail.gmail.com>
References: <50B7AA27.2000003@phoenixinternet.net>
	<CA+zebiYzT75EewBfBfWWvXDNpPcdc+sfToconYKv+PBvoBDnng@mail.gmail.com>
Message-ID: <13302.1354225365@ccs.covici.com>

Maybe only compile one at a time?

Komar, Jason <jkomar at jbox.ca> wrote:

> Had a similar thing on Gentoo. Seems to be a race condition. Stefan
> Knoblich (Gentoo FreeSWITCH overlay maintainer) did a workaround in the
> ebuild to get past it. I'm not sure what it entailed. Maybe if he sees this
> message, he could chime in on it.
> 
> Jason
> 
> 
> On Thu, Nov 29, 2012 at 11:32 AM, Gilbert T. Gutierrez, Jr. <
> mailing-lists at phoenixinternet.net> wrote:
> 
> > I am having an issue compiling 1.2 Stable this morning. My notes are all
> > below. When I compile a second time, it appears to compile ok. I am not
> > sure if I trust that compilation because some of the modules may not be
> > complete. What do you guys feel the problem may be?
> >
> > Gilbert
> >
> > Centos6 64bit
> >
> > The procedure I am following:
> > cd /usr/src
> > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
> > cd freeswitch
> > ./bootstrap.sh -j
> > ./configure --without-libcurl -C
> > make -j `cat /proc/cpuinfo |grep processor |wc -l`
> >
> >
> > The make first throws the following error...
> > "In file included from ./src/include/private/switch_core_pvt.h:44,
> >                   from src/switch.c:48:
> > /usr/src/freeswitch/libs/apr/include/apr_pools.h:37:17: error: apr.h: No
> > such file or directory"
> >
> > Everything following is an error terminating in "make: *** [all] Error 2"
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveayre at gmail.com  Fri Nov 30 00:46:53 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 29 Nov 2012 21:46:53 +0000
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in
 XML CDR's?
In-Reply-To: <1354221331.77886.YahooMailNeo@web160803.mail.bf1.yahoo.com>
References: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>
	<CAKzWOxUcpgaS29KWmSHg8FmMAraOVAa75i71sXdpbBC0S7EC_w@mail.gmail.com>
	<1354221331.77886.YahooMailNeo@web160803.mail.bf1.yahoo.com>
Message-ID: <CAFiqYukvcw50KpgEbHjBkFU=zMA-1TUFRHvA7CYzA1n7jzJPPw@mail.gmail.com>

jb will indeed be jitterbuffer, but not sure what it'll mean. Discarded by
jb, or left in jb at end of call maybe? Perhaps grepping the source for the
tag name'll shed some light.

I believe that skip *may* be packets ignored because FS isn't ready to
receive them yet. Eg receiving RTP on a call where nothing's consuming it,
or on a bridged call that hasn't been answered yet.

But this is guesswork...


On 29 November 2012 20:35, Mick Stevens <mickstevens at yahoo.com> wrote:

> Hi Michael &  Co,
>
> Many thanks for the prompt & positive response! Yes, more than happy to
> collate feedback from the global team & update the wiki accordingly (I'd
> like to be able to document what the field values between the > < indicate
> as well as just explain the field names if possible...) +anything else I
> can do to to contribute to the project...?
>
> Yes, now the wiki is back up I've managed to work out that cng_packet =
> comfort noise generation! Also from the wiki, possibly that the jb in rtp_audio_in_jb_packet_count
> & rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody
> else already knows this!).
>
> To provide the background to my original enquiry, I'm trying to identify
> if the rtp_audio_in_skip_packet_count in the following XML CDR extract is
> indicative of packet loss, or perhaps more accurately noticeable audio loss
> (as the packets haven't been lost, just ignored/"skipped"?: If anybody
> knows please speak up! [image: :) happy]
>
>     <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
>     <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
>     <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
>     <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
>     <rtp_audio_in_skip_packet_count>1609</rtp_audio_in_skip_packet_count>
>     <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
>     <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
>     <rtp_audio_in_cng_packet_count>15</rtp_audio_in_cng_packet_count>
>     <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
>     <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>
> Thank you in anticipation of enlightenment!
>
> #[image: :x lovestruck]FreeSWITCH
>
> Rgds, Mick
> Tel/SMS. +44(0)7967 594432
> Fax. +44(0)7053 452429
> Email/IM. mickstevens at yahoo.com
> Skype: mick_stevens
> www.facebook.com/mickstevens
> www.twitter.com/mickstevens
>
>   ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, 29 November 2012, 19:14
> *Subject:* Re: [Freeswitch-users] Explanation of rtp_audio_in / out
> fields in XML CDR's?
>
> Mick,
>
> This data is definitely not on the wiki - or anywhere else that I can see.
> I think we can crowdsource this to get the info collected and then add it
> to the mod_xml_curl wiki page<http://wiki.freeswitch.org/wiki/Mod_xml_cdr>.
> For the record, here's a quick dump of the fields that I took from an XML
> CDR:
>
>     <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
>     <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
>     <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
>     <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
>     <rtp_audio_in_skip_packet_count>18</rtp_audio_in_skip_packet_count>
>     <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
>     <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
>     <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
>     <rtp_audio_in_flush_packet_count>11</rtp_audio_in_flush_packet_count>
>     <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>     <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
>     <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
>     <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
>     <rtp_audio_out_media_packet_count>67</rtp_audio_out_media_packet_count>
>     <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
>     <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
>     <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>
>
> I think raw_bytes, media_bytes, packet_count, and media_packet_count are
> self-explanatory. I think cng_packet_count is probably self-explanatory
> too. My question on dtmf_packet_count would be whether it's only for
> RFC2833 packets (I suspect yes, but would like confirmation).
>
> If anyone knows these please reply to this email and Mick and I will get
> them documented on the wiki (right Mick? ;)
>
> rtp_audio_in_skip_packet_count
> rtp_audio_out_skip_packet_count
> rtp_audio_in_jb_packet_count
> rtp_audio_in_flush_packet_count
> rtp_audio_in_largest_jb_size
>
> Thanks all!
>
> -MC
>
> On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com>wrote:
>
> Hi Folks,
>
> I'm trying to use FS XML CDR's to diagnose historic audio problems. I
> think I can work out some of the rtp_audio_in / out fields (raw bytes &
> media bytes being nearly equal looks like a good sign) but am wondering
> about the skip, cng & flush fields for example?
>
> I have tried Googling this & can find evidence of other people having
> asked this question but not of the answer. I have also checked my FS 106 &
> Cookbook book's without success. The wiki appears to be down at the moment
> so my apologies if the answer lies there.
>
> I know how to do this in real time using wireshark etc but am interested
> in being able to do some analysis on historic problems reported by
> customers that aren't willing/able to replicate the problem in order for a
> protocol trace to be captured.
>
> Any help much appreciated!
>
> Rgds, Mick
> Tel/SMS. +44(0)7967 594432
> Fax. +44(0)7053 452429
> Email/IM. mickstevens at yahoo.com
> Skype: mick_stevens
> www.facebook.com/mickstevens
> www.twitter.com/mickstevens
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From richgration at gmail.com  Fri Nov 30 02:58:18 2012
From: richgration at gmail.com (Richard Gration)
Date: Thu, 29 Nov 2012 23:58:18 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAFiqYumcs0r9X06vSkOuujdeB13iYLanqBVjsr2ucLWWR2AaTA@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
	<CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
	<CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>
	<CAFiqYukX75FDSpAT9zdvnOBb_boJxQAZKJSk2QhuJKroY_LJng@mail.gmail.com>
	<CAAAWEqDAh=0bXG=+3gsneO+R6TmNW4BE2b2G=9P3rZqajY9wxQ@mail.gmail.com>
	<CAFiqYumcs0r9X06vSkOuujdeB13iYLanqBVjsr2ucLWWR2AaTA@mail.gmail.com>
Message-ID: <CAAAWEqBHRPHTfcG961KYTzwzz3eaaGiDu14v+SX4yL9rXJf28Q@mail.gmail.com>

On 29 November 2012 19:43, Steven Ayre <steveayre at gmail.com> wrote:
> I would recommend that you adjust your DB check code to run in a loop. CDR
> submission occurs after CHANNEL_HANGUP - so you have a race condition and
> may check the DB before the CDR has been committed. You'd need to handle a)
> trying again after a short delay if the CDR isn't there yet and b) a timeout
> in case xml_cdr fails and the CDR isn't available (or the script'll keep
> running until it is).

I agree the script needs work, particularly in error handling :-) I
also need to set an alarm before I go into read event loop I think.
I've had a few zombies to deal with so far ...

I tried using myevents but I didn't read anything. I might try and
suss that out at some point.

R

-- 
Once our basic material needs are met - in my utopia, anyway - life
becomes a perpetual celebration in which everyone has a talent to
contribute. But we cannot levitate ourselves into that blessed
condition by wishing it. We need to brace ourselves for a struggle
against terrifying obstacles, both of our own making and imposed by
the natural world. And the first step is to recover from the delusion
that is positive thinking.
       -- Barbara Ehrenreich


From richgration at gmail.com  Fri Nov 30 03:03:45 2012
From: richgration at gmail.com (Richard Gration)
Date: Fri, 30 Nov 2012 00:03:45 +0000
Subject: [Freeswitch-users] Problem accessing channel variables
In-Reply-To: <CAAAWEqBHRPHTfcG961KYTzwzz3eaaGiDu14v+SX4yL9rXJf28Q@mail.gmail.com>
References: <CAAAWEqC4+k-Sq-YC6qO7n7t+gJbq-48_9xazj108BM368SvDMQ@mail.gmail.com>
	<CAFiqYukfVmsGvrmqRU3m_0Rtv0C-1pLytfNpdpvHZ2g4fLeLcA@mail.gmail.com>
	<CAAAWEqA5GqzfMma8c9v9s1b=fvrh+4-4ns8WDoYyypquxT9B2w@mail.gmail.com>
	<CAFiqYukX75FDSpAT9zdvnOBb_boJxQAZKJSk2QhuJKroY_LJng@mail.gmail.com>
	<CAAAWEqDAh=0bXG=+3gsneO+R6TmNW4BE2b2G=9P3rZqajY9wxQ@mail.gmail.com>
	<CAFiqYumcs0r9X06vSkOuujdeB13iYLanqBVjsr2ucLWWR2AaTA@mail.gmail.com>
	<CAAAWEqBHRPHTfcG961KYTzwzz3eaaGiDu14v+SX4yL9rXJf28Q@mail.gmail.com>
Message-ID: <CAAAWEqBwPxQRe-FAfm_4=bJU8AgHaw947Nh_ssaSew=ec5oBtQ@mail.gmail.com>

Forgot to say, no I don't get the channel var in the hangup event. I
thought I would too.
-- 
Once our basic material needs are met - in my utopia, anyway - life
becomes a perpetual celebration in which everyone has a talent to
contribute. But we cannot levitate ourselves into that blessed
condition by wishing it. We need to brace ourselves for a struggle
against terrifying obstacles, both of our own making and imposed by
the natural world. And the first step is to recover from the delusion
that is positive thinking.
       -- Barbara Ehrenreich


From abaci64 at gmail.com  Fri Nov 30 03:05:24 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 29 Nov 2012 19:05:24 -0500
Subject: [Freeswitch-users] Trouble with voicemail say digits
In-Reply-To: <CAKzWOxX+=SDMt24qZ4EF48KLDao8Lv9OuC6-wuXZPuL3QdC-dQ@mail.gmail.com>
References: <CA+zebiY3-N37JnDNVcgonxT-Qw-+tY64uQw7zCNZSY9joy9Rxw@mail.gmail.com>
	<20121129044205.141c5071@mail.tritonwest.net>
	<CAKzWOxX+=SDMt24qZ4EF48KLDao8Lv9OuC6-wuXZPuL3QdC-dQ@mail.gmail.com>
Message-ID: <50B7F844.9090106@gmail.com>

There is a ticket for it.
http://jira.freeswitch.org/browse/FS-4878

On 11/29/2012 2:22 PM, Michael Collins wrote:
> This was a regression in mod_dptools that made it into 1.2.5.1. It was 
> fixed on Monday IIRC but with all the DDoS drama it flew under the 
> radar. This commit fixes it, so if you need a quick repair you can 
> remove these lines and recompile mod_dptools:
>
> http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=0b148a85b94be33fe70b692240e0d449a59f0ef2
>
> You should be able to do that without even stopping FS. I'll talk to 
> Ken and see if we can't get something more elegant in place soon.
>
> -MC
>
> On Wed, Nov 28, 2012 at 8:42 PM, Dave R. Kompel <drk at drkngs.net 
> <mailto:drk at drkngs.net>> wrote:
>
>     I thought it was me... Did something change whre the /8000/ got
>     removed from the path? Is there a point where you can't leave your
>     old configs in place when you update?
>     --Dave
>
>         ------------------------------------------------------------------------
>         *From:* Komar, Jason [mailto:jkomar at jbox.ca
>         <mailto:jkomar at jbox.ca>]
>         *To:* FreeSWITCH Users Help
>         [mailto:freeswitch-users at lists.freeswitch.org
>         <mailto:freeswitch-users at lists.freeswitch.org>]
>         *Sent:* Wed, 28 Nov 2012 19:47:05 -0800
>         *Subject:* [Freeswitch-users] Trouble with voicemail say digits
>
>
>         Just updated to FS 1.2.5.1 on Gentoo from 1.2.3. I have one
>         extension that uses the default voicemail greeting. Since I
>         updated, when it gets to the part where it says the extension
>         number, it cannot find the wav files for the digits. It is
>         looking for them in the digits folder and not in the frequency
>         subfolders (i.e. 16000  32000  48000  8000). As soon as it
>         hits this error, the voicemail app says goodbye and hangs up.
>
>         2012-11-28 20:35:17.068585 [DEBUG] switch_ivr_play_say.c:244
>         Handle say:[2003] (en:en)
>         2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>         [/opt/freeswitch/sounds/en/us/callie/digits/2.wav] does not exist.
>         2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>         [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
>         2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>         [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
>         2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>         [/opt/freeswitch/sounds/en/us/callie/digits/3.wav] does not exist.
>
>         Is there a config option to specify the frequency subfolder,
>         or did something not build correctly?
>
>         Thanks,
>         Jason Komar
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Fri Nov 30 03:55:07 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 16:55:07 -0800
Subject: [Freeswitch-users] sqlite3 and regex
In-Reply-To: <1354181295.13048.82.camel@marces.madrid.commsmundi.com>
References: <CCB9BB87.5EE38%krice@freeswitch.org>
	<1354119909.13048.56.camel@marces.madrid.commsmundi.com>
	<1413611559183349171@unknownmsgid>
	<1354181295.13048.82.camel@marces.madrid.commsmundi.com>
Message-ID: <CAKzWOxUTKGu6H2EQ6-4=hmmaCceS7g5AwVyxhBrp22c4c+EBwA@mail.gmail.com>

That's how it's done, people! FYI, Anthony added this today in these two
commits:
http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=081e261
http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=bce107b

You can also see the flow of the conversation between the person submitting
the patch and the devs who had suggestions/information to make the process
go more smoothly:

http://jira.freeswitch.org/browse/FS-4883

*THAT* is why we need Jira for bugs/features/patches and not merely the
mailing list.

Kudos to all who got this feature added!
-MC

On Thu, Nov 29, 2012 at 1:28 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
> Hi,
>
> Last night came up with this patch:
>
> http://jira.freeswitch.org/browse/FS-4883
>
> Is working in my fs, this way i can use REGEXP in my lua scripts :)
>
>
>
>
>
> On Thu, 2012-11-29 at 01:19 -0500, S. Scott wrote:
>
> That's EXACTLY what I'm hunting for!  My freeswitch build doesn't have a
> pcre.so anywhere on the disk.  The .../freeswitch/libs/pcre/ exists with
> many files but no .so.
>
>
> What does one need to do to get the sqlite3 and the pcre.so rolled (from
> the git ingredients)?  make() and stuff not my strong suit.
>
> Thanks,
> ?????
> iThing:  Big thumbs & little keys. Please excuse typo, spelling and
> grammar errors ? Last night I played a blank CD at full blast. The Mime
> next door went nuts.
>
>
>
> On 2012-11-28, at 23:53, Antonio <asilva at wirelessmundi.com> wrote:
>
>
>
>  Hi,
>
> Coudn't we load just an external extension to sqlite?
>
> for example, to have regular expression support you could load an external
> library like "sqlite3-pcre" (available for unbuntu), you can load from
> the console interface or directly in sql.
>
> console:
> sqlite> .load '/usr/lib/sqlite3/pcre.so
>
> sql in console:
> sqlite> select load_extension(''/usr/lib/sqlite3/pcre.so');
>
>
> In fs is not possible, when i try to do it from my lua script i have the
> following error:
>
> 2012-11-28 17:21:34.196304 [ERR] switch_core_sqldb.c:572 NATIVE SQL ERR
> [no such function: load_extension]
> select load_extension('/usr/lib/sqlite3/pcre.so');
>
>
> Since you have already a switch type of db ( core, odbc or pgsql) could be
> nice when using core, be able to use a few more functions available in
> sqlite?
>
>
> Thanks,
> Ant?nio
>
>
>
>
>
>
> On Fri, 2012-11-02 at 18:03 -0500, Ken Rice wrote:
>
> No and this wont happen anytime soon... The SQL interfaces for FreeSwitch
> are kept generic as we support more then just sqlite from common code and
> if we did it for sqlite we would have to make sure its implemented equally
> well for postgresql and mysql and mssql and any other database someone
> might want to use via ODBC
>
> K
>
>
> On 11/2/12 3:34 PM, "Scott" <8f27e956 at gmail.com> wrote:
>
>  "LIKE" notwithstanding, sqlite3 does not have a built-in true regex
> function; it does allow for a a c-language hook to one.  Given fs extensive
> use of the regex engine and of sqlite3, we're wondering if the hook is
> already written and rolled.  If so, can the rest of us hook it to our
> sqlite3 uses (e.g. from dial plan lua sqlite3).
>
> With thanks,
>
>
> ------------------------------
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org*
> *http://www.ClueCon.com*
> *http://www.OSTAG.org*
> irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>   --
>
> Un cordial saludo / Best regards,
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.com
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>   --
>
> Un cordial saludo / Best regards,
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Nov 30 03:57:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 16:57:04 -0800
Subject: [Freeswitch-users] mod_voicemail compilation error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2338E30@Mail-Kilo.squay.com>
References: <1FFF97C269757C458224B7C895F35F151CAFE9@cantor.std.visionutv.se>
	<592A9CF93E12394E8472A6CC66E66BF2337941@Mail-Kilo.squay.com>
	<592A9CF93E12394E8472A6CC66E66BF2338E30@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxVGNrb-Eo94nmaotNU=pxuesUnPv=4EzoxzmAmr5pp-kQ@mail.gmail.com>

Simple debug print statements should not be this difficult. Do you have
anyone who is even moderately skilled in C programming who could take a
look?

-MC

On Thu, Nov 29, 2012 at 3:15 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Even if i rebuild initially it was working, but when I change the c code
> and compile am getting the same error and voicemail stops working. I have
> taken the source code which is currently running in our production and did
> a build. ****
>
> The version we are using is FreeSWITCH Version 1.0.head (git-a0a77f8
> 2011-12-15 12-23-53 -0500). Since this is the one that is running in
> production I can?t install the latest version too.****
>
> ** **
>
> Please help me out in resolving this issue in the version that am using.
> Thanks****
>
> ** **
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana
> Venugopan
> *Sent:* 23 November 2012 12:25
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_voicemail compilation error****
>
> ** **
>
> Hi,****
>
> Thanks. But it worked when the freeswitch was build. After making changes
> and do a compile this is not supporting and giving this error. ****
>
> The same libfreeswitch.so might have supported
> switch_channel_expand_variables_check() during initial build as well right?
> ****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter Olsson
> *Sent:* 23 November 2012 10:46
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_voicemail compilation error****
>
>  ****
>
> Rebuild the entire project instead. It seems that the current
> libfreeswitch.so doesn?t support switch_channel_expand_variables_check().*
> ***
>
>  ****
>
> /Peter****
>
>  ****
>
>  ****
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Archana Venugopan
> *Skickat:* den 23 november 2012 11:03
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] mod_voicemail compilation error****
>
>  ****
>
> Hi,****
>
>  ****
>
> I tried putting debug statements in mod_voicemail.c file and tried to
> re-compile it but I got this msg and now its not going to voicemail.
> Earlier it was working properly but after this voicemail was not working.
> Please help.****
>
>  ****
>
> *[root at squay-laptop-1 freeswitch]# make mod_voicemail-install*****
>
>  /bin/sh /usr/local/src/freeswitch/quiet_libtool   --mode=install
> /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'****
>
> quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0
> /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0****
>
> quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
> libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 &&
> ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })****
>
> quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
> libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln
> -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })****
>
> quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
> /usr/local/freeswitch/lib/libfreeswitch.la****
>
> quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
> /usr/local/freeswitch/lib/libfreeswitch.a****
>
> quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
> ****
>
> quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a**
> **
>
> quiet_libtool: finish:
> PATH="/usr/lib/qt-3.3/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/sbin"
> ldconfig -n /usr/local/freeswitch/lib****
>
> ----------------------------------------------------------------------****
>
> Libraries have been installed in:****
>
>    /usr/local/freeswitch/lib****
>
>  ****
>
> If you ever happen to want to link against installed libraries****
>
> in a given directory, LIBDIR, you must either use libtool, and****
>
> specify the full pathname of the library, or use the `-LLIBDIR'****
>
> flag during linking and do at least one of the following:****
>
>    - add LIBDIR to the `LD_LIBRARY_PATH' environment variable****
>
>      during execution****
>
>    - add LIBDIR to the `LD_RUN_PATH' environment variable****
>
>      during linking****
>
>    - use the `-Wl,-rpath -Wl,LIBDIR' linker flag****
>
>    - have your system administrator add LIBDIR to `/etc/ld.so.conf'****
>
>  ****
>
> See any operating system documentation about shared libraries for****
>
> more information, such as the ld(1) and ld.so(8) manual pages.****
>
> ----------------------------------------------------------------------****
>
>  ****
>
> making install mod_voicemail****
>
> Compiling
> /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c...
> ****
>
> quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic
> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c
> /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c
> -fPIC -DPIC -o .libs/mod_voicemail.o****
>
> quiet_libtool: compile:  gcc -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic
> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c
> /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c
> -o mod_voicemail.o >/dev/null 2>&1****
>
> Creating mod_voicemail.la...****
>
> installing mod_voicemail.la****
>
> quiet_libtool: install: warning: relinking `mod_voicemail.la'****
>
>  ****
>
>  ****
>
> *freeswitch at internal> reload mod_voicemail*****
>
> +OK module unloaded****
>
> +OK Reloading XML****
>
> -ERR loading module [module load file routine returned an error]****
>
>  ****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:901 Deleting
> Application 'voicemail'****
>
> freeswitch at internal> 2012-11-23 09:58:31.448059 [DEBUG]
> switch_loadable_module.c:903 Write lock interface 'voicemail' to wait for
> existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'voicemail'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'voicemail' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'voicemail_inject'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'voicemail_inject' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_inject'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_inject' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_boxcount'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_boxcount' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_prefs'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_prefs' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_delete'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_delete' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_read'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_read' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_list'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_list' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_auth_login'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_auth_login' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_count'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_count' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_list'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_list' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_get'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_get' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_delete'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_delete' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_undelete'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_undelete' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_purge'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_purge' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_save'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_save' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_msg_forward'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_msg_forward' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_pref_greeting_set'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_pref_greeting_set' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_pref_recname_set'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_pref_recname_set' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_loadable_module.c:954 Deleting
> API Function 'vm_fsdb_pref_password_set'****
>
> 2012-11-23 09:58:31.448059 [DEBUG] switch_loadable_module.c:956 Write lock
> interface 'vm_fsdb_pref_password_set' to wait for existing references.****
>
> 2012-11-23 09:58:31.448059 [CONSOLE] switch_loadable_module.c:1765
> Stopping: mod_voicemail****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:442 Subclass
> reservation deleted for
> /usr/local/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:vm::maintenance
> ****
>
> 2012-11-23 09:58:31.448059 [NOTICE] switch_event.c:1889 Event Binding
> deleted for mod_voicemail:MESSAGE_QUERY****
>
> 2012-11-23 09:58:31.518046 [CONSOLE] mod_voicemail.c:3840 Event Thread
> Ended****
>
> 2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5759 Waiting for write
> lock (Profile default)****
>
> 2012-11-23 09:58:31.518046 [DEBUG] mod_voicemail.c:5762 Destroying Profile
> default****
>
> 2012-11-23 09:58:31.518046 [CONSOLE] switch_loadable_module.c:1785
> mod_voicemail unloaded.****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding
> tone_descriptor: 1****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 1, tone: CED_TONE(0)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 1, tone: SIT(1)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 1, tone: REORDER_TONE(2)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)***
> *
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 1, tone: BUSY_TONE(3)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding
> tone_descriptor: 44****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 44, tone: CED_TONE(0)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 44, tone: SIT(1)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 44, tone: REORDER_TONE(2)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 44, tone: BUSY_TONE(3)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:449 Adding
> tone_descriptor: 49****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 49, tone: CED_TONE(0)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 49, tone: SIT(1)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 49, tone: REORDER_TONE(2)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:470 Adding
> tone_descriptor: 49, tone: BUSY_TONE(3)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_spandsp.c:505 Adding
> tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)****
>
> 2012-11-23 09:58:32.768049 [INFO] mod_enum.c:775 ENUM Reloaded****
>
> 2012-11-23 09:58:32.768049 [CRIT] switch_loadable_module.c:1281 Error
> Loading module /usr/local/freeswitch/mod/mod_voicemail.so****
>
> **/usr/local/freeswitch/mod/mod_voicemail.so: undefined symbol:
> switch_channel_expand_variables_check******
>
> 2012-11-23 09:58:32.778219 [INFO] switch_time.c:1035 Timezone reloaded 530
> definitions****
>
>  ****
>
>  ****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> !DSPAM:50af4e1232761560677559! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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-- 
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Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Nov 30 04:03:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 17:03:35 -0800
Subject: [Freeswitch-users] Trouble with voicemail say digits
In-Reply-To: <50B7F844.9090106@gmail.com>
References: <CA+zebiY3-N37JnDNVcgonxT-Qw-+tY64uQw7zCNZSY9joy9Rxw@mail.gmail.com>
	<20121129044205.141c5071@mail.tritonwest.net>
	<CAKzWOxX+=SDMt24qZ4EF48KLDao8Lv9OuC6-wuXZPuL3QdC-dQ@mail.gmail.com>
	<50B7F844.9090106@gmail.com>
Message-ID: <CAKzWOxWVkMasQw5arQpieJXHn4K818Fg2VNntJ2ua-UahAR40g@mail.gmail.com>

Yep. That needs to be closed out. One of the devs will do that shortly.
-MC

On Thu, Nov 29, 2012 at 4:05 PM, Abaci <abaci64 at gmail.com> wrote:

>  There is a ticket for it.
> http://jira.freeswitch.org/browse/FS-4878
>
> On 11/29/2012 2:22 PM, Michael Collins wrote:
>
> This was a regression in mod_dptools that made it into 1.2.5.1. It was
> fixed on Monday IIRC but with all the DDoS drama it flew under the radar.
> This commit fixes it, so if you need a quick repair you can remove these
> lines and recompile mod_dptools:
>
>
> http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=0b148a85b94be33fe70b692240e0d449a59f0ef2
>
> You should be able to do that without even stopping FS. I'll talk to Ken
> and see if we can't get something more elegant in place soon.
>
> -MC
>
> On Wed, Nov 28, 2012 at 8:42 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>
>>  I thought it was me... Did something change whre the /8000/ got removed
>> from the path? Is there a point where you can't leave your old configs in
>> place when you update?
>>
>> --Dave
>>
>>  ------------------------------
>> *From:* Komar, Jason [mailto:jkomar at jbox.ca]
>> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org
>> ]
>> *Sent:* Wed, 28 Nov 2012 19:47:05 -0800
>> *Subject:* [Freeswitch-users] Trouble with voicemail say digits
>>
>>
>> Just updated to FS 1.2.5.1 on Gentoo from 1.2.3. I have one extension
>> that uses the default voicemail greeting. Since I updated, when it gets to
>> the part where it says the extension number, it cannot find the wav files
>> for the digits. It is looking for them in the digits folder and not in the
>> frequency subfolders (i.e. 16000  32000  48000  8000). As soon as it hits
>> this error, the voicemail app says goodbye and hangs up.
>>
>>  2012-11-28 20:35:17.068585 [DEBUG] switch_ivr_play_say.c:244 Handle
>> say:[2003] (en:en)
>> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>> [/opt/freeswitch/sounds/en/us/callie/digits/2.wav] does not exist.
>> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>> [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
>> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>> [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist.
>> 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File
>> [/opt/freeswitch/sounds/en/us/callie/digits/3.wav] does not exist.
>>
>>  Is there a config option to specify the frequency subfolder, or did
>> something not build correctly?
>>
>>  Thanks,
>> Jason Komar
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Nov 30 04:08:07 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 17:08:07 -0800
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <01f101cdce1a$79309130$6b91b390$@co.in>
References: <00a001cdc317$f58c2900$e0a47b00$@co.in>
	<00eb01cdc32f$e284b6c0$a78e2440$@co.in>
	<CAFVUaV_Sc407-Qs-c9f8jj3cXLsbxmJnLEuFLZz=zB-JS1V7mA@mail.gmail.com>
	<012b01cdcc88$fea5a6e0$fbf0f4a0$@co.in>
	<00c101cdcdeb$b38bbaa0$1aa32fe0$@co.in>
	<CAFiqYuk9cYxDMyxetKHdxixukk3u1TCpr3GhYBm4Ccxvt1ZkRQ@mail.gmail.com>
	<01f101cdce1a$79309130$6b91b390$@co.in>
Message-ID: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>

Answers inline...

On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

>   Hi,****
>
> ** **
>
> I am using valet_park. I?ve configure a IVR menu of an extension, based on
> user?s input call is forwarded to other extensions.****
>
> ** **
>
> Extension on which end-users will call ?****
>
> ** **
>
> <extension name="ivr_nitin">
>       <condition field="destination_number" expression="^5002$">
>         <action application="answer"/>
>         <action application="sleep" data="500"/>
>         <action application="ivr" data="nitin_ivr"/>
>       </condition>
> </extension>****
>
> ** **
>
> The IVR configuration XML is ?****
>
> ** **
>
> <include>
>         <menu name="nitin_ivr"
>                 greet-long="say:Welcome to Newgen General Insurance
> Company. Press 1 for Changing Address, 2 for Changing Nominee or 3 for
> Close Policy."
>                 greet-short="say:Welcome to Newgen. Press 1 for Changing
> Address, 2 for Changing Nominee or 3 for Close Policy."
>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>                 exit-sound="voicemail/vm-goodbye.wav"
>                 confirm-macro=""
>                 confirm-key=""
>                 tts-engine="flite"
>                 tts-voice="slt"
>                 confirm-attempts="3"
>                 timeout="3000"
>                 inter-digit-timeout="2000"
>                 max-failures="3"
>                 max-timeouts="3"
>                 digit-len="4">
>
>                 <entry action="menu-exec-app" digits="1" param="transfer
> 450 XML default"/>    <!-- Change Address -->
>                 <entry action="menu-exec-app" digits="2" param="transfer
> 451 XML default"/>    <!-- Change Nominee -->
>                 <entry action="menu-exec-app" digits="3" param="transfer
> 452 XML default"/>    <!-- Close Policy -->
>
>         </menu>
> </include>
>
> ****
>
> Once user presses ?1?, call is forwarded to 450, for this extension
> dialplan entry is ?****
>
> ** **
>
> <extension>
>           <condition field="destination_number" expression="^(450)$">
>           <action application="play_and_get_digits" data="10 16 3 3000 #
> say:'Press your account number, followed by hash key' say:'Wrong Input' res
> \d+" />
>           <action application="phrase" data="spell,${res}"/>
>           <action application="db" data="insert/testapp/newcall1/${res}"
> />
>               <action application="set" data="caller_id_name=Change
> Address Request" />
>               <action application="set" data="call_timeout=60" />
>               <action application="set" data="originate_timeout=60" />
>               <action application="valet_park" data="my_lot auto in 8501
> 8599" />           ****
>
>           </condition>
> </extension>****
>
> ** **
>
> Here, the call is parked at any available extension between 8501 to 8599.*
> ***
>
> ** **
>
> Then I?ve set up an extension to pick up calls ?****
>
> ** **
>
> <extension name="park-out">
>    <condition field="destination_number" expression="^(85\d\d)$">
>      <action application="answer"/>
>      <action application="valet_park" data="my_lot $1"/>
>    </condition>
>  </extension>****
>
> ** **
>
> I have a few questions ?****
>
> ** **
>
> **1.       **Valet_park parks the call on any available extension between
> 8501 to 8599 (<action application="valet_park" data="my_lot auto in 8501
> 8599" />). Is there any way to let me know on which extension the call have
> been parked?
>
Using 'auto in' the system will announce the parking location. If you are
sending a call in from an IVR then the caller will hear their park
location. The only way to know where the call went would be to watch the
event socket for relevant valet events.

> ****
>
> **2.       **How can I get the number from which call was made in
> extension 450. The idea is to use the caller number as key and entered
> value as value while making entry in database (<action application="db"
> data="insert/testapp/newcall1/${res}" />)?****
>
Do you mean the caller id number? That's literally in channel variable
${caller_id_number}

> **3.       **If two users call on extension 5002 (where IVR menu is
> played), what will happen? Will the second user have to wait for first to
> finish or whether both will be connected parallel?
>
Both can be in the IVR at the same time and they won't affect each other at
all.

-MC

> ****
>
> ** **
>
> Please help me out.****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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-- 
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Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From ntomer at newgen.co.in  Fri Nov 30 05:09:44 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 30 Nov 2012 07:39:44 +0530
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>
Message-ID: <1354241384.50b81568a4579@mx.newgen.co.in>

Dear Michael,

Thanks for your help.

About thsis - "Using 'auto in' the system will announce the parking
location. If you are sending a call in from an IVR then the caller
will hear their park location. The only way to know where the call
went would be to watch the event socket for relevant valet events. "

Yes, right now the extension where call is parked, is announced to
the caller. I don't want that to happen. I don't want it announced to
caller, rather I want it retruned to me, so that I can store it in
database. So that my agents can pick the call after seeing the
extension where it is parked.

Please tell me more details about how to watch the event socket for
valet events.

Regards

Nitin

On Friday, 30-11-2012 on 6:38 Michael Collins wrote:

Answers inline...

On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer  wrote:

Hi,

?

I am using valet_park. I?ve configure a IVR menu of an extension,
based on user?s input call is forwarded to other extensions.

?

Extension on which end-users will call ?

?

????? 
??????? 
??????? 
??? ????
????? 

?

The IVR configuration XML is ?

?

??????? 
?
??????????????? ??? 
??????????????? ??? 
??????????????? ??? 

??????? 

Once user presses ?1?, call is forwarded to 450, for this
extension dialplan entry is ?

?

????????? 
??? ????? 
??? ????? 
??? ????? ??? 
????????????? 
????????????? 
????????????? 
????????????? ?????????? 

????????? 

?

Here, the call is parked at any available extension between 8501 to
8599.

?

Then I?ve set up an extension to pick up calls ?

?

?? 
???? 
???? 
?? 
?

?

I have a few questions ?

?

1.?????? Valet_park parks the call on any available extension
between 8501 to 8599 (). Is there any way to let me know on which
extension the call have been parked?

Using 'auto in' the system will announce the parking location. If you
are sending a call in from an IVR then the caller will hear their park
location. The only way to know where the call went would be to watch
the event socket for relevant valet events. 

2.?????? How can I get the number from which call was made in
extension 450. The idea is to use the caller number as key and entered
value as value while making entry in database ()?

Do you mean the caller id number? That's literally in channel
variable ${caller_id_number} 

3.?????? If two users call on extension 5002 (where IVR menu is
played), what will happen? Will the second user have to wait for first
to finish or whether both will be connected parallel?

Both can be in the IVR at the same time and they won't affect each
other at all.

-MC 

?

Please help me out.

?

Regards

?

Nitin

Disclaimer :- This E-mail And Any Attachment May Contain
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_________________________________________________________________________
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consulting at freeswitch.org
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Official FreeSWITCH Sites
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Twitter: @mercutioviz
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http://www.ClueCon.com
http://www.OSTAG.org



Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.


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From msc at freeswitch.org  Fri Nov 30 05:59:40 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Nov 2012 18:59:40 -0800
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <1354241384.50b81568a4579@mx.newgen.co.in>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>
	<1354241384.50b81568a4579@mx.newgen.co.in>
Message-ID: <CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>

As far as I know you can't suppress the announcement of the location to the
caller.

For the event socket you have a lot of homework to do. I recommend:
FS Book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>,
chapter 9
FS Cookbook <http://link.packtpub.com/nuIOlX>, chapter 4
Wiki event socket (see link on left)

For a really quick dive into what events look like:
launch fs_cli and type:
/log 0
/events plain all

You'll see EVERY event that the system throws. Try this to narrow it down
just to valet events:

/filter Event-Class valet_parking::info

I typed most of this off the top of my head, so standard
disclaimer<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>applies.
Hope this helps you get started!
-MC


On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

> Dear Michael,
>
> Thanks for your help.
>
> About thsis - "Using 'auto in' the system will announce the parking
> location. If you are sending a call in from an IVR then the caller will
> hear their park location. The only way to know where the call went would be
> to watch the event socket for relevant valet events. "
>
> Yes, right now the extension where call is parked, is announced to the
> caller. I don't want that to happen. I don't want it announced to caller,
> rather I want it retruned to me, so that I can store it in database. So
> that my agents can pick the call after seeing the extension where it is
> parked.
>
> Please tell me more details about how to watch the event socket for valet
> events.
>
> Regards
>
> Nitin
>
> On Friday, 30-11-2012 on 6:38 Michael Collins wrote:
>
> Answers inline...
>
> On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
>
>>   Hi,****
>>
>> ** **
>>
>> I am using valet_park. I?ve configure a IVR menu of an extension, based
>> on user?s input call is forwarded to other extensions.****
>>
>> ** **
>>
>> Extension on which end-users will call ?****
>>
>> ** **
>>
>> <extension name="ivr_nitin">
>>       <condition field="destination_number" expression="^5002$">
>>         <action application="answer"/>
>>         <action application="sleep" data="500"/>
>>         <action application="ivr" data="nitin_ivr"/>
>>       </condition>
>> </extension>****
>>
>> ** **
>>
>> The IVR configuration XML is ?****
>>
>> ** **
>>
>> <include>
>>         <menu name="nitin_ivr"
>>                 greet-long="say:Welcome to Newgen General Insurance
>> Company. Press 1 for Changing Address, 2 for Changing Nominee or 3 for
>> Close Policy."
>>                 greet-short="say:Welcome to Newgen. Press 1 for Changing
>> Address, 2 for Changing Nominee or 3 for Close Policy."
>>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>                 exit-sound="voicemail/vm-goodbye.wav"
>>                 confirm-macro=""
>>                 confirm-key=""
>>                 tts-engine="flite"
>>                 tts-voice="slt"
>>                 confirm-attempts="3"
>>                 timeout="3000"
>>                 inter-digit-timeout="2000"
>>                 max-failures="3"
>>                 max-timeouts="3"
>>                 digit-len="4">
>>
>>                 <entry action="menu-exec-app" digits="1" param="transfer
>> 450 XML default"/>    <!-- Change Address -->
>>                 <entry action="menu-exec-app" digits="2" param="transfer
>> 451 XML default"/>    <!-- Change Nominee -->
>>                 <entry action="menu-exec-app" digits="3" param="transfer
>> 452 XML default"/>    <!-- Close Policy -->
>>
>>         </menu>
>> </include>
>>
>> ****
>>
>> Once user presses ?1?, call is forwarded to 450, for this extension
>> dialplan entry is ?****
>>
>> ** **
>>
>> <extension>
>>           <condition field="destination_number" expression="^(450)$">
>>           <action application="play_and_get_digits" data="10 16 3 3000 #
>> say:'Press your account number, followed by hash key' say:'Wrong Input' res
>> \d+" />
>>           <action application="phrase" data="spell,${res}"/>
>>           <action application="db" data="insert/testapp/newcall1/${res}"
>> />
>>               <action application="set" data="caller_id_name=Change
>> Address Request" />
>>               <action application="set" data="call_timeout=60" />
>>               <action application="set" data="originate_timeout=60" />
>>               <action application="valet_park" data="my_lot auto in 8501
>> 8599" />           ****
>>
>>           </condition>
>> </extension>****
>>
>> ** **
>>
>> Here, the call is parked at any available extension between 8501 to 8599.
>> ****
>>
>> ** **
>>
>> Then I?ve set up an extension to pick up calls ?****
>>
>> ** **
>>
>> <extension name="park-out">
>>    <condition field="destination_number" expression="^(85\d\d)$">
>>      <action application="answer"/>
>>      <action application="valet_park" data="my_lot $1"/>
>>    </condition>
>>  </extension>****
>>
>> ** **
>>
>> I have a few questions ?****
>>
>> ** **
>>
>> **1.       **Valet_park parks the call on any available extension
>> between 8501 to 8599 (<action application="valet_park" data="my_lot auto
>> in 8501 8599" />). Is there any way to let me know on which extension the
>> call have been parked?
>>
> Using 'auto in' the system will announce the parking location. If you are
> sending a call in from an IVR then the caller will hear their park
> location. The only way to know where the call went would be to watch the
> event socket for relevant valet events.
>
>> ****
>>
>> **2.       **How can I get the number from which call was made in
>> extension 450. The idea is to use the caller number as key and entered
>> value as value while making entry in database (<action application="db"
>> data="insert/testapp/newcall1/${res}" />)?****
>>
> Do you mean the caller id number? That's literally in channel variable
> ${caller_id_number}
>
>> **3.       **If two users call on extension 5002 (where IVR menu is
>> played), what will happen? Will the second user have to wait for first to
>> finish or whether both will be connected parallel?
>>
> Both can be in the IVR at the same time and they won't affect each other
> at all.
>
> -MC
>
>> ****
>>
>> ** **
>>
>> Please help me out.****
>>
>> ** **
>>
>> Regards****
>>
>> ** **
>>
>> Nitin****
>>
>> Disclaimer :- This e-mail and any attachment may contain confidential,
>> proprietary or legally privileged information. If you are not the original
>> intended recipient and have erroneously received this message, you are
>> prohibited from using, copying, altering or disclosing the content of this
>> message. Please delete it immediately and notify the sender. Newgen
>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
>> damage arising from the use of the information transmitted by this email
>> including damages from virus and further acknowledges that no binding
>> nature of the message shall be implied or assumed unless the sender does so
>> expressly with due authority of NSTL.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From sparklezou at 163.com  Fri Nov 30 06:19:48 2012
From: sparklezou at 163.com (sparklezou)
Date: Fri, 30 Nov 2012 11:19:48 +0800
Subject: [Freeswitch-users] About the display problem during calling
Message-ID: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com>

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!
 

2012-11-30



sparklezou
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From ntomer at newgen.co.in  Fri Nov 30 07:33:01 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 30 Nov 2012 10:03:01 +0530
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>	<1354241384.50b81568a4579@mx.newgen.co.in>
	<CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
Message-ID: <00ff01cdceb3$ca4a7ec0$5edf7c40$@co.in>

Dear Michael,

 

Thanks for your response.

 

If I can't suppress the announcement of location to caller, then I will have
to look at an alternative approach.

 

I make entries for every location on which a call is parked in my database,
so will it be possible for me to do -

 

1.       Suppose I have assigned extension 8501 to 8599 for parking.

2.       Run a for loop, from 8501 to 8599

3.       Check whether an entry exists for that extension in the database

4.       If an entry exists, it means a call is parked at that extension,
move to next extension

5.       If no entry exists, it means extension is free. Park the call there
and make entry in database

6.       Break the loop

 

Will it work? And if it will, please guide me on how to do that. I am new to
Freeswitch and not very well-versed  with it.

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, November 30, 2012 8:30 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] valet_park help needed

 

As far as I know you can't suppress the announcement of the location to the
caller.

For the event socket you have a lot of homework to do. I recommend:
FS
<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel
ephony-systems/book>  Book, chapter 9
FS Cookbook <http://link.packtpub.com/nuIOlX> , chapter 4
Wiki event socket (see link on left)

For a really quick dive into what events look like:
launch fs_cli and type:
/log 0
/events plain all

You'll see EVERY event that the system throws. Try this to narrow it down
just to valet events:

/filter Event-Class valet_parking::info

I typed most of this off the top of my head, so standard
<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>
disclaimer applies. Hope this helps you get started!
-MC



On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Dear Michael,

Thanks for your help.

About thsis - "Using 'auto in' the system will announce the parking
location. If you are sending a call in from an IVR then the caller will hear
their park location. The only way to know where the call went would be to
watch the event socket for relevant valet events. "

Yes, right now the extension where call is parked, is announced to the
caller. I don't want that to happen. I don't want it announced to caller,
rather I want it retruned to me, so that I can store it in database. So that
my agents can pick the call after seeing the extension where it is parked.

Please tell me more details about how to watch the event socket for valet
events.

Regards

Nitin

On Friday, 30-11-2012 on 6:38 Michael Collins wrote:

Answers inline...

On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Hi,

 

I am using valet_park. I've configure a IVR menu of an extension, based on
user's input call is forwarded to other extensions.

 

Extension on which end-users will call -

 

<extension name="ivr_nitin">
      <condition field="destination_number" expression="^5002$">
        <action application="answer"/>
        <action application="sleep" data="500"/>
        <action application="ivr" data="nitin_ivr"/>
      </condition>
</extension>

 

The IVR configuration XML is -

 

<include>
        <menu name="nitin_ivr"
                greet-long="say:Welcome to Newgen General Insurance Company.
Press 1 for Changing Address, 2 for Changing Nominee or 3 for Close Policy."
                greet-short="say:Welcome to Newgen. Press 1 for Changing
Address, 2 for Changing Nominee or 3 for Close Policy."
                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
                exit-sound="voicemail/vm-goodbye.wav"
                confirm-macro=""
                confirm-key=""
                tts-engine="flite"
                tts-voice="slt"
                confirm-attempts="3"
                timeout="3000"
                inter-digit-timeout="2000"
                max-failures="3"
                max-timeouts="3"
                digit-len="4">
 
                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Change Address -->
                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Change Nominee -->
                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Close Policy -->

        </menu>
</include>

Once user presses "1", call is forwarded to 450, for this extension dialplan
entry is -

 

<extension>
          <condition field="destination_number" expression="^(450)$">
          <action application="play_and_get_digits" data="10 16 3 3000 #
say:'Press your account number, followed by hash key' say:'Wrong Input' res
\d+" />
          <action application="phrase" data="spell,${res}"/>
          <action application="db" data="insert/testapp/newcall1/${res}" />

              <action application="set" data="caller_id_name=Change Address
Request" />
              <action application="set" data="call_timeout=60" />
              <action application="set" data="originate_timeout=60" />
              <action application="valet_park" data="my_lot auto in 8501
8599" />           

          </condition>
</extension>

 

Here, the call is parked at any available extension between 8501 to 8599.

 

Then I've set up an extension to pick up calls -

 

<extension name="park-out">
   <condition field="destination_number" expression="^(85\d\d)$">
     <action application="answer"/>
     <action application="valet_park" data="my_lot $1"/>
   </condition>
 </extension>

 

I have a few questions -

 

1.       Valet_park parks the call on any available extension between 8501
to 8599 (<action application="valet_park" data="my_lot auto in 8501 8599"
/>). Is there any way to let me know on which extension the call have been
parked?

Using 'auto in' the system will announce the parking location. If you are
sending a call in from an IVR then the caller will hear their park location.
The only way to know where the call went would be to watch the event socket
for relevant valet events. 

2.       How can I get the number from which call was made in extension 450.
The idea is to use the caller number as key and entered value as value while
making entry in database (<action application="db"
data="insert/testapp/newcall1/${res}" />)?

Do you mean the caller id number? That's literally in channel variable
${caller_id_number} 

3.       If two users call on extension 5002 (where IVR menu is played),
what will happen? Will the second user have to wait for first to finish or
whether both will be connected parallel?

Both can be in the IVR at the same time and they won't affect each other at
all.

-MC 

 

Please help me out.

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org




Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From william at xofap.com  Fri Nov 30 09:35:54 2012
From: william at xofap.com (William Alianto)
Date: Fri, 30 Nov 2012 13:35:54 +0700
Subject: [Freeswitch-users] Routing call to another interface
Message-ID: <CAGTd76ERO92fAFPLwCFgTzY5r3_s=OnJBYZHtX21u8Drs+wm6Q@mail.gmail.com>

Hi,

I got a server with this configuration :


Gateway
^
|
|
|
| eth 2 (10.x.x.x)
|
|
FS<-------->eth1 (180.x.x.x) <------->Internet<---------->User

Now I need to route the call to/from gateway. Would anybody help me to
configure a dialplan for this problem?

Regards
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From sparklezou at 163.com  Fri Nov 30 10:53:03 2012
From: sparklezou at 163.com (sparklezou)
Date: Fri, 30 Nov 2012 15:53:03 +0800
Subject: [Freeswitch-users] About the display problem during calling
In-Reply-To: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com>
References: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com>
Message-ID: <79e311e1.6599.13b504fd0fa.Coremail.sparklezou@163.com>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
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From steveayre at gmail.com  Fri Nov 30 12:29:57 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 30 Nov 2012 09:29:57 +0000
Subject: [Freeswitch-users] Compilation Issue
In-Reply-To: <50B7AA27.2000003@phoenixinternet.net>
References: <50B7AA27.2000003@phoenixinternet.net>
Message-ID: <CAFiqYum6P9_JTdAV5h8p8qkGHi0NZv=ujf--j00xM5DXN=EWaw@mail.gmail.com>

Does it succeed if you don't use a parallel build?

(I've got a completely separate project at the moment compiling that gets
an error with -j that disappears when it's not used).


On 29 November 2012 18:32, Gilbert T. Gutierrez, Jr. <
mailing-lists at phoenixinternet.net> wrote:

> I am having an issue compiling 1.2 Stable this morning. My notes are all
> below. When I compile a second time, it appears to compile ok. I am not
> sure if I trust that compilation because some of the modules may not be
> complete. What do you guys feel the problem may be?
>
> Gilbert
>
> Centos6 64bit
>
> The procedure I am following:
> cd /usr/src
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
> cd freeswitch
> ./bootstrap.sh -j
> ./configure --without-libcurl -C
> make -j `cat /proc/cpuinfo |grep processor |wc -l`
>
>
> The make first throws the following error...
> "In file included from ./src/include/private/switch_core_pvt.h:44,
>                   from src/switch.c:48:
> /usr/src/freeswitch/libs/apr/include/apr_pools.h:37:17: error: apr.h: No
> such file or directory"
>
> Everything following is an error terminating in "make: *** [all] Error 2"
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From NuwanW at unifybusiness.co.uk  Fri Nov 30 13:02:45 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Fri, 30 Nov 2012 10:02:45 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D24F7@BARXB0003.UnifyBusiness.local>

Hi Anthony,

I have attached the log file to the JIRA case (FS-4884) as you requested . Please let me know if you need any more information.

Thank you,
Nuwan.

This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From doreme202002 at yahoo.com  Fri Nov 30 13:12:14 2012
From: doreme202002 at yahoo.com (hala alramli)
Date: Fri, 30 Nov 2012 10:12:14 +0000 (GMT)
Subject: [Freeswitch-users] GEMopen and freeswitch for voice call and sms
Message-ID: <1354270334.42856.YahooMailNeo@web28802.mail.ir2.yahoo.com>

Hello all,

i want to ask if ?Huawei E355 and E369 dongle??compatible with?freeswich and GSMopen for SMS and voice calls.

thanks
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From ntomer at newgen.co.in  Fri Nov 30 14:35:28 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 30 Nov 2012 17:05:28 +0530
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <00ff01cdceb3$ca4a7ec0$5edf7c40$@co.in>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>	<1354241384.50b81568a4579@mx.newgen.co.in>	<CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
	<00ff01cdceb3$ca4a7ec0$5edf7c40$@co.in>
Message-ID: <01d601cdceee$cc35f940$64a1ebc0$@co.in>

Hi Michael,

 

I've decided to do it through Lua. I've written a Lua script which will
check whether an entry for an extension exists in the database. And if it
doesn't, it'll assume that extension as free and will park the call there.

 

I've made the following changes in the dialplan -

 

<extension>

<condition field="destination_number" expression="^(450)$">

<action application="play_and_get_digits" data="10 16 3 3000 # say:'Press
your account number, followed by hash key' say:'Wrong Input' res \d+" />

            <action application="phrase" data="spell,${res}"/>

            <action application="set" data="caller_id_name=Change Address
Request" />

            <action application="set" data="call_timeout=60" />

            <action application="set" data="originate_timeout=60" />

            <action inline="true" application="lua" data="test.lua "/>

            <action application="db"
data="insert/testapp/${parkednumber}/${res}" />

<action application="valet_park" data="my_lot ${parkednumber}" />

</condition>

  </extension>

 

Test.lua:

 

local dbh = freeswitch.Dbh("freeswitch-connector", "root", "system123#")

 

if dbh:connected() then

    if stream then

        stream:write("result - database connected\n\n")

    else

      print("database connected")

    end

else  

    if stream then

      stream:write("result - database error\n")

    else

      print("database error")

    end

    return

end

 

-- set variable - or print to console if no session is available 

local function sv(key, val)

   if session then

       session:setVariable("parkednumber", val)

       print("value set in session")

   elseif stream then

       stream:write(string.format("%25s : %s\n", key, val))

   else -- a script executed using luarun does not have a stream

       print(key .. " : " .. val)

   end

end

 

number = 8501

 

while(number < 8600)

local my_query = "select data from db_data where data_key = '" .. tries ..
"'"

 

      assert(dbh:query(my_query, function(row)

            for key, val in pairs(row) do     -- in this example only one

row with one column will be returned

                        sv(key, val)                   -- so here key =
'user'

            end

      end))

      tries = tries + 1

end

 

I want to execute the above query, and check whether any result came. If
results came, then I want to proceed to next number. And if no result came,
then I want to set it in the session variable and break from the while loop.
But I could only find the syntax for assert, no way to check whether the
query returned any result or not. Please help me out.

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Friday, November 30, 2012 10:03 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] valet_park help needed

 

Dear Michael,

 

Thanks for your response.

 

If I can't suppress the announcement of location to caller, then I will have
to look at an alternative approach.

 

I make entries for every location on which a call is parked in my database,
so will it be possible for me to do -

 

1.       Suppose I have assigned extension 8501 to 8599 for parking.

2.       Run a for loop, from 8501 to 8599

3.       Check whether an entry exists for that extension in the database

4.       If an entry exists, it means a call is parked at that extension,
move to next extension

5.       If no entry exists, it means extension is free. Park the call there
and make entry in database

6.       Break the loop

 

Will it work? And if it will, please guide me on how to do that. I am new to
Freeswitch and not very well-versed  with it.

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, November 30, 2012 8:30 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] valet_park help needed

 

As far as I know you can't suppress the announcement of the location to the
caller.

For the event socket you have a lot of homework to do. I recommend:
FS
<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel
ephony-systems/book>  Book, chapter 9
FS Cookbook <http://link.packtpub.com/nuIOlX> , chapter 4
Wiki event socket (see link on left)

For a really quick dive into what events look like:
launch fs_cli and type:
/log 0
/events plain all

You'll see EVERY event that the system throws. Try this to narrow it down
just to valet events:

/filter Event-Class valet_parking::info

I typed most of this off the top of my head, so standard
<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>
disclaimer applies. Hope this helps you get started!
-MC

On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Dear Michael,

Thanks for your help.

About thsis - "Using 'auto in' the system will announce the parking
location. If you are sending a call in from an IVR then the caller will hear
their park location. The only way to know where the call went would be to
watch the event socket for relevant valet events. "

Yes, right now the extension where call is parked, is announced to the
caller. I don't want that to happen. I don't want it announced to caller,
rather I want it retruned to me, so that I can store it in database. So that
my agents can pick the call after seeing the extension where it is parked.

Please tell me more details about how to watch the event socket for valet
events.

Regards

Nitin

On Friday, 30-11-2012 on 6:38 Michael Collins wrote:

Answers inline...

On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Hi,

 

I am using valet_park. I've configure a IVR menu of an extension, based on
user's input call is forwarded to other extensions.

 

Extension on which end-users will call -

 

<extension name="ivr_nitin">
      <condition field="destination_number" expression="^5002$">
        <action application="answer"/>
        <action application="sleep" data="500"/>
        <action application="ivr" data="nitin_ivr"/>
      </condition>
</extension>

 

The IVR configuration XML is -

 

<include>
        <menu name="nitin_ivr"
                greet-long="say:Welcome to Newgen General Insurance Company.
Press 1 for Changing Address, 2 for Changing Nominee or 3 for Close Policy."
                greet-short="say:Welcome to Newgen. Press 1 for Changing
Address, 2 for Changing Nominee or 3 for Close Policy."
                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
                exit-sound="voicemail/vm-goodbye.wav"
                confirm-macro=""
                confirm-key=""
                tts-engine="flite"
                tts-voice="slt"
                confirm-attempts="3"
                timeout="3000"
                inter-digit-timeout="2000"
                max-failures="3"
                max-timeouts="3"
                digit-len="4">
 
                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Change Address -->
                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Change Nominee -->
                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Close Policy -->

        </menu>
</include>

Once user presses "1", call is forwarded to 450, for this extension dialplan
entry is -

 

<extension>
          <condition field="destination_number" expression="^(450)$">
          <action application="play_and_get_digits" data="10 16 3 3000 #
say:'Press your account number, followed by hash key' say:'Wrong Input' res
\d+" />
          <action application="phrase" data="spell,${res}"/>
          <action application="db" data="insert/testapp/newcall1/${res}" />

              <action application="set" data="caller_id_name=Change Address
Request" />
              <action application="set" data="call_timeout=60" />
              <action application="set" data="originate_timeout=60" />
              <action application="valet_park" data="my_lot auto in 8501
8599" />           

          </condition>
</extension>

 

Here, the call is parked at any available extension between 8501 to 8599.

 

Then I've set up an extension to pick up calls -

 

<extension name="park-out">
   <condition field="destination_number" expression="^(85\d\d)$">
     <action application="answer"/>
     <action application="valet_park" data="my_lot $1"/>
   </condition>
 </extension>

 

I have a few questions -

 

1.       Valet_park parks the call on any available extension between 8501
to 8599 (<action application="valet_park" data="my_lot auto in 8501 8599"
/>). Is there any way to let me know on which extension the call have been
parked?

Using 'auto in' the system will announce the parking location. If you are
sending a call in from an IVR then the caller will hear their park location.
The only way to know where the call went would be to watch the event socket
for relevant valet events. 

2.       How can I get the number from which call was made in extension 450.
The idea is to use the caller number as key and entered value as value while
making entry in database (<action application="db"
data="insert/testapp/newcall1/${res}" />)?

Do you mean the caller id number? That's literally in channel variable
${caller_id_number} 

3.       If two users call on extension 5002 (where IVR menu is played),
what will happen? Will the second user have to wait for first to finish or
whether both will be connected parallel?

Both can be in the IVR at the same time and they won't affect each other at
all.

-MC 

 

Please help me out.

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


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




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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From krice at freeswitch.org  Fri Nov 30 17:04:36 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 30 Nov 2012 08:04:36 -0600
Subject: [Freeswitch-users] Brief show notes for today in the world of
 FreeSWITCH <PLEASE READ ME>
Message-ID: <CCDE1914.613DB%krice@freeswitch.org>

Hey Guys,

Coming up today FreeSWITCH 1.2.5.2 this is another minor release to catch a
few regressions we missed in 1.2.5.1. Yes we are aware of them and this will
be released sometime today, please watch the mailing list and other ways you
get your FS news. 


Also Don?t forget today is Friday, so the Friday FreeForAll will be in Full
Effect today.

Join us at sip:888 at conference.freeswitch.org and see what the FreeForAll is
all about

And speaking of 888, 9888 in your Default Example Config FreeSWITCH installs
is one way to get to the FreeSWITCH conference bridge. While we do
appreciate that you may want to call 888 on occation to test, please refrain
from calling 888 with 10 or 20 concurrent calls this is a public conference
bridge for the community and spamming it only makes is less available for
others to use.

K






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

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From Hector.Geraldino at ipsoft.com  Fri Nov 30 20:06:52 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 30 Nov 2012 17:06:52 +0000
Subject: [Freeswitch-users] FreeSWITCH suddenly stopped accepting calls
Message-ID: <ADF32A4A0F985A428C85B277B8E1C9D80F48B7EA@NY1-EXMB-02.ip-soft.net>

Hello everyone,

I have a FreeSWITCH instance that has been running for months without an issue, until last Monday. The setup is for a hotline where we receive incoming calls transferred from the Client Cisco UCM to our FreeSWITCH instance, and the calls are handled by a custom Java application via ESL.

As I said it was working flawlessly for weeks, but last Monday FreeSWITCH just stopped accepting new calls. I was able to see the SIP messages flowing back and forth from FS<->UCM, but looks like FreeSWITCH was unable to create new channels. Once I restarted FreeSWITCH everything was back to normal. Here are links to the SIP/console logs I was able to capture which covers both scenarios (non-working FS and working FS):

Non-working: http://pastebin.freeswitch.org/20275
Working: http://pastebin.freeswitch.org/20274

As you can see, in the non-working scenario no channels were created on FreeSWITCH, so no 200 ACK message was sent back to UCM after the INVITE, which caused the UCM to send a CANCEL message after a few seconds and a 487 Request terminated from FS -> UCM. Restarting the application fixed the issue, but I have a couple of questions regarding this:


1)      We have a lot of monitoring in place for all our applications, and I could probably alerted our Operations team if FreeSWITCH crashed or some error was logged in the FS log. However I was unable to found anything meaningful in the logs that indicates any sort of error. It just suddenly stopped creating new channels, and we get the alert/complain from the client - ouch!

2)      I was also not sure what other information I should collect when something like this happens. As the application didn't crash it didn't create any coredump files. I know that restarting it fixes the problem, but that is not what the client wants to hear.

Any recommendations on what to do/look at to proactively detect this kind of issues and/or how to collect the appropriate info to get the root cause will be greatly appreciated.

Thank you all for your time,
Hector
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From mailing-lists at phoenixinternet.net  Fri Nov 30 21:06:15 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Fri, 30 Nov 2012 11:06:15 -0700
Subject: [Freeswitch-users] Compilation Issue
In-Reply-To: <CAFiqYum6P9_JTdAV5h8p8qkGHi0NZv=ujf--j00xM5DXN=EWaw@mail.gmail.com>
References: <50B7AA27.2000003@phoenixinternet.net>
	<CAFiqYum6P9_JTdAV5h8p8qkGHi0NZv=ujf--j00xM5DXN=EWaw@mail.gmail.com>
Message-ID: <50B8F597.40301@phoenixinternet.net>

Steven,
I just ran make by itself (left off -j `cat /proc/cpuinfo |grep 
processor |wc -l`). The system made clean.

Gilbert

On 11/30/2012 2:29 AM, Steven Ayre wrote:
> Does it succeed if you don't use a parallel build?
>
> (I've got a completely separate project at the moment compiling that 
> gets an error with -j that disappears when it's not used).
>
>
> On 29 November 2012 18:32, Gilbert T. Gutierrez, Jr. 
> <mailing-lists at phoenixinternet.net 
> <mailto:mailing-lists at phoenixinternet.net>> wrote:
>
>     I am having an issue compiling 1.2 Stable this morning. My notes
>     are all
>     below. When I compile a second time, it appears to compile ok. I
>     am not
>     sure if I trust that compilation because some of the modules may
>     not be
>     complete. What do you guys feel the problem may be?
>
>     Gilbert
>
>     Centos6 64bit
>
>     The procedure I am following:
>     cd /usr/src
>     git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
>     <http://git.freeswitch.org/freeswitch.git>
>     cd freeswitch
>     ./bootstrap.sh -j
>     ./configure --without-libcurl -C
>     make -j `cat /proc/cpuinfo |grep processor |wc -l`
>
>
>     The make first throws the following error...
>     "In file included from ./src/include/private/switch_core_pvt.h:44,
>                       from src/switch.c:48:
>     /usr/src/freeswitch/libs/apr/include/apr_pools.h:37:17: error:
>     apr.h: No
>     such file or directory"
>
>     Everything following is an error terminating in "make: *** [all]
>     Error 2"
>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Fri Nov 30 21:20:41 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 30 Nov 2012 18:20:41 +0000
Subject: [Freeswitch-users] FreeSWITCH suddenly stopped accepting calls
In-Reply-To: <ADF32A4A0F985A428C85B277B8E1C9D80F48B7EA@NY1-EXMB-02.ip-soft.net>
References: <ADF32A4A0F985A428C85B277B8E1C9D80F48B7EA@NY1-EXMB-02.ip-soft.net>
Message-ID: <CAFiqYungZ9o2oAEYsBkTBYQ8itX=yDDEdFe6W7RZo2+q+-yR2A@mail.gmail.com>

>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-46f097c 2011-08-07 01-33-26
> -0400


That's very old, you should update. There'll be a lot of bugfixes since
then, and the age will make it harder to support.

1)      **We have a lot of monitoring in place for all our applications,
> and I could probably alerted our Operations team if FreeSWITCH crashed or
> some error was logged in the FS log. However I was unable to found anything
> meaningful in the logs that indicates any sort of error. It just suddenly
> stopped creating new channels, and we get the alert/complain from the
> client ? ouch!


You could try sending a SIP OPTIONS packet from a monitoring system and
check for a reply. Consecutive requests with no reply (timeout) could
indicate a problem. One isn't enough in case the packet gets dropped.

2)      **I was also not sure what other information I should collect when
> something like this happens. As the application didn?t crash it didn?t
> create any coredump files. I know that restarting it fixes the problem, but
> that is not what the client wants to hear.


Check out gcore - it can create a coredump of a running process. Very
useful for debugging.

If you're compiling from git, you'll need to keep the source you compiled
from (gdb needs the source to interpret the coredump).

-Steve




On 30 November 2012 17:06, Hector Geraldino <Hector.Geraldino at ipsoft.com>wrote:

>  Hello everyone,****
>
> ** **
>
> I have a FreeSWITCH instance that has been running for months without an
> issue, until last Monday. The setup is for a hotline where we receive
> incoming calls transferred from the Client Cisco UCM to our FreeSWITCH
> instance, and the calls are handled by a custom Java application via ESL.*
> ***
>
> ** **
>
> As I said it was working flawlessly for weeks, but last Monday FreeSWITCH
> just stopped accepting new calls. I was able to see the SIP messages
> flowing back and forth from FS<->UCM, but looks like FreeSWITCH was unable
> to create new channels. Once I restarted FreeSWITCH everything was back to
> normal. Here are links to the SIP/console logs I was able to capture which
> covers both scenarios (non-working FS and working FS):****
>
> ** **
>
> Non-working: http://pastebin.freeswitch.org/20275****
>
> Working: http://pastebin.freeswitch.org/20274****
>
> ** **
>
> As you can see, in the non-working scenario no channels were created on
> FreeSWITCH, so no 200 ACK message was sent back to UCM after the INVITE,
> which caused the UCM to send a CANCEL message after a few seconds and a 487
> Request terminated from FS -> UCM. Restarting the application fixed the
> issue, but I have a couple of questions regarding this:****
>
> ** **
>
> **1)      **We have a lot of monitoring in place for all our
> applications, and I could probably alerted our Operations team if
> FreeSWITCH crashed or some error was logged in the FS log. However I was
> unable to found anything meaningful in the logs that indicates any sort of
> error. It just suddenly stopped creating new channels, and we get the
> alert/complain from the client ? ouch!****
>
> **2)      **I was also not sure what other information I should collect
> when something like this happens. As the application didn?t crash it didn?t
> create any coredump files. I know that restarting it fixes the problem, but
> that is not what the client wants to hear.****
>
> ** **
>
> Any recommendations on what to do/look at to proactively detect this kind
> of issues and/or how to collect the appropriate info to get the root cause
> will be greatly appreciated. ****
>
> ** **
>
> Thank you all for your time,****
>
> Hector****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Nov 30 21:25:48 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 30 Nov 2012 18:25:48 +0000
Subject: [Freeswitch-users] FreeSWITCH suddenly stopped accepting calls
In-Reply-To: <CAFiqYungZ9o2oAEYsBkTBYQ8itX=yDDEdFe6W7RZo2+q+-yR2A@mail.gmail.com>
References: <ADF32A4A0F985A428C85B277B8E1C9D80F48B7EA@NY1-EXMB-02.ip-soft.net>
	<CAFiqYungZ9o2oAEYsBkTBYQ8itX=yDDEdFe6W7RZo2+q+-yR2A@mail.gmail.com>
Message-ID: <CAFiqYumtwWZufNFAxxaDinbgTGSksmojoUNykdBAmY+tKFFo-A@mail.gmail.com>

If you do get a coredump from gcore, see
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29

You will see in the backtrace what each thread is doing, which may
highlight what's causing the non-response.

I highly suggest as your very first step you upgrade to 1.2.5.2. This on
its own may stop your issue if it's down to a bug that was fixed, and if
you're still having the issue will put you on a version that can be
supported.


On 30 November 2012 18:20, Steven Ayre <steveayre at gmail.com> wrote:

> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-46f097c 2011-08-07 01-33-26
>> -0400
>
>
> That's very old, you should update. There'll be a lot of bugfixes since
> then, and the age will make it harder to support.
>
>  1)      **We have a lot of monitoring in place for all our applications,
>> and I could probably alerted our Operations team if FreeSWITCH crashed or
>> some error was logged in the FS log. However I was unable to found anything
>> meaningful in the logs that indicates any sort of error. It just suddenly
>> stopped creating new channels, and we get the alert/complain from the
>> client ? ouch!
>
>
> You could try sending a SIP OPTIONS packet from a monitoring system and
> check for a reply. Consecutive requests with no reply (timeout) could
> indicate a problem. One isn't enough in case the packet gets dropped.
>
> 2)      **I was also not sure what other information I should collect
>> when something like this happens. As the application didn?t crash it didn?t
>> create any coredump files. I know that restarting it fixes the problem, but
>> that is not what the client wants to hear.
>
>
> Check out gcore - it can create a coredump of a running process. Very
> useful for debugging.
>
> If you're compiling from git, you'll need to keep the source you compiled
> from (gdb needs the source to interpret the coredump).
>
> -Steve
>
>
>
>
> On 30 November 2012 17:06, Hector Geraldino <Hector.Geraldino at ipsoft.com>wrote:
>
>>  Hello everyone,****
>>
>> ** **
>>
>> I have a FreeSWITCH instance that has been running for months without an
>> issue, until last Monday. The setup is for a hotline where we receive
>> incoming calls transferred from the Client Cisco UCM to our FreeSWITCH
>> instance, and the calls are handled by a custom Java application via ESL.
>> ****
>>
>> ** **
>>
>> As I said it was working flawlessly for weeks, but last Monday FreeSWITCH
>> just stopped accepting new calls. I was able to see the SIP messages
>> flowing back and forth from FS<->UCM, but looks like FreeSWITCH was unable
>> to create new channels. Once I restarted FreeSWITCH everything was back to
>> normal. Here are links to the SIP/console logs I was able to capture which
>> covers both scenarios (non-working FS and working FS):****
>>
>> ** **
>>
>> Non-working: http://pastebin.freeswitch.org/20275****
>>
>> Working: http://pastebin.freeswitch.org/20274****
>>
>> ** **
>>
>> As you can see, in the non-working scenario no channels were created on
>> FreeSWITCH, so no 200 ACK message was sent back to UCM after the INVITE,
>> which caused the UCM to send a CANCEL message after a few seconds and a 487
>> Request terminated from FS -> UCM. Restarting the application fixed the
>> issue, but I have a couple of questions regarding this:****
>>
>> ** **
>>
>> **1)      **We have a lot of monitoring in place for all our
>> applications, and I could probably alerted our Operations team if
>> FreeSWITCH crashed or some error was logged in the FS log. However I was
>> unable to found anything meaningful in the logs that indicates any sort of
>> error. It just suddenly stopped creating new channels, and we get the
>> alert/complain from the client ? ouch!****
>>
>> **2)      **I was also not sure what other information I should collect
>> when something like this happens. As the application didn?t crash it didn?t
>> create any coredump files. I know that restarting it fixes the problem, but
>> that is not what the client wants to hear.****
>>
>> ** **
>>
>> Any recommendations on what to do/look at to proactively detect this kind
>> of issues and/or how to collect the appropriate info to get the root cause
>> will be greatly appreciated. ****
>>
>> ** **
>>
>> Thank you all for your time,****
>>
>> Hector****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From ksrigo at gmail.com  Fri Nov 30 14:54:21 2012
From: ksrigo at gmail.com (Srigo)
Date: Fri, 30 Nov 2012 03:54:21 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch is crashing when httapi playback
 application is executed
In-Reply-To: <1354210814465-7585041.post@n2.nabble.com>
References: <1354210814465-7585041.post@n2.nabble.com>
Message-ID: <1354276461718-7585075.post@n2.nabble.com>

Hi Michael,

I already opened a bug report here
http://jira.freeswitch.org/browse/FS-4886.
Thank you for all details.

Srigo



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-is-crashing-when-httapi-playback-application-is-executed-tp7585041p7585075.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Nov 30 21:52:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Nov 2012 10:52:28 -0800
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <01d601cdceee$cc35f940$64a1ebc0$@co.in>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>
	<1354241384.50b81568a4579@mx.newgen.co.in>
	<CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
	<00ff01cdceb3$ca4a7ec0$5edf7c40$@co.in>
	<01d601cdceee$cc35f940$64a1ebc0$@co.in>
Message-ID: <CAKzWOxXoRQGVSA5TAAf+Xb1+foO5wJ20ax-nn=NRa-EBTpp7KQ@mail.gmail.com>

I haven't had a chance to familiarize my self with the Lua/dbh way of
connecting to a database so I'll have to defer to Chad and the other
hardcore Lua gurus. This much I can tell you: there is a valet_info API
that you can run at fs_cli or from Lua that will tell you what parking
spots are currently in use.

-MC

On Fri, Nov 30, 2012 at 3:35 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

>  Hi Michael,****
>
> ** **
>
> I?ve decided to do it through Lua. I?ve written a Lua script which will
> check whether an entry for an extension exists in the database. And if it
> doesn?t, it?ll assume that extension as free and will park the call there.
> ****
>
> ** **
>
> I?ve made the following changes in the dialplan ?****
>
> ** **
>
> <extension>****
>
> <condition field="destination_number" expression="^(450)$">****
>
> <action application="play_and_get_digits" data="10 16 3 3000 # say:'Press
> your account number, followed by hash key' say:'Wrong Input' res \d+" />**
> **
>
>             <action application="phrase" data="spell,${res}"/>****
>
>             <action application="set" data="caller_id_name=Change Address
> Request" />****
>
>             <action application="set" data="call_timeout=60" />****
>
>             <action application="set" data="originate_timeout=60" />****
>
>             <action inline="true" application="lua" data="test.lua "/>****
>
>             <action application="db"
> data="insert/testapp/${parkednumber}/${res}" />****
>
> <action application="valet_park" data="my_lot ${parkednumber}" />****
>
> </condition>****
>
>   </extension>****
>
> ** **
>
> Test.lua:****
>
> ** **
>
> local dbh = freeswitch.Dbh("freeswitch-connector", "root", "system123#")**
> **
>
> ** **
>
> if dbh:connected() then****
>
>     if stream then****
>
>         stream:write("result - database connected\n\n")****
>
>     else****
>
>       print("database connected")****
>
>     end****
>
> else  ****
>
>     if stream then****
>
>       stream:write("result - database error\n")****
>
>     else****
>
>       print("database error")****
>
>     end****
>
>     return****
>
> end****
>
> ** **
>
> -- set variable - or print to console if no session is available ****
>
> local function sv(key, val)****
>
>    if session then****
>
>        session:setVariable("parkednumber", val)****
>
>        print("value set in session")****
>
>    elseif stream then****
>
>        stream:write(string.format("%25s : %s\n", key, val))****
>
>    else -- a script executed using luarun does not have a stream****
>
>        print(key .. " : " .. val)****
>
>    end****
>
> end****
>
> ** **
>
> number = 8501****
>
> ** **
>
> while(number < 8600)****
>
> local my_query = "select data from db_data where data_key = '" .. tries ..
> ???****
>
> ** **
>
>       assert(dbh:query(my_query, function(row)****
>
>             for key, val in pairs(row) do     -- in this example only one*
> ***
>
> row with one column will be returned****
>
>                         sv(key, val)                   -- so here key =
> 'user'****
>
>             end****
>
>       end))****
>
>       tries = tries + 1****
>
> end****
>
> ** **
>
> I want to execute the above query, and check whether any result came. If
> results came, then I want to proceed to next number. And if no result came,
> then I want to set it in the session variable and break from the while
> loop. But I could only find the syntax for assert, no way to check whether
> the query returned any result or not. Please help me out.****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nitin Tomer
> *Sent:* Friday, November 30, 2012 10:03 AM
>
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] valet_park help needed****
>
>  ** **
>
> Dear Michael,****
>
> ** **
>
> Thanks for your response.****
>
> ** **
>
> If I can?t suppress the announcement of location to caller, then I will
> have to look at an alternative approach.****
>
> ** **
>
> I make entries for every location on which a call is parked in my
> database, so will it be possible for me to do ?****
>
> ** **
>
> **1.       **Suppose I have assigned extension 8501 to 8599 for parking.**
> **
>
> **2.       **Run a for loop, from 8501 to 8599****
>
> **3.       **Check whether an entry exists for that extension in the
> database****
>
> **4.       **If an entry exists, it means a call is parked at that
> extension, move to next extension****
>
> **5.       **If no entry exists, it means extension is free. Park the
> call there and make entry in database****
>
> **6.       **Break the loop****
>
> ** **
>
> Will it work? And if it will, please guide me on how to do that. I am new
> to Freeswitch and not very well-versed  with it.****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, November 30, 2012 8:30 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] valet_park help needed****
>
> ** **
>
> As far as I know you can't suppress the announcement of the location to
> the caller.
>
> For the event socket you have a lot of homework to do. I recommend:
> FS Book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>,
> chapter 9
> FS Cookbook <http://link.packtpub.com/nuIOlX>, chapter 4
> Wiki event socket (see link on left)
>
> For a really quick dive into what events look like:
> launch fs_cli and type:
> /log 0
> /events plain all
>
> You'll see EVERY event that the system throws. Try this to narrow it down
> just to valet events:
>
> /filter Event-Class valet_parking::info
>
> I typed most of this off the top of my head, so standard disclaimer<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>applies. Hope this helps you get started!
> -MC****
>
> On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:*
> ***
>
> Dear Michael,
>
> Thanks for your help.
>
> About thsis - "Using 'auto in' the system will announce the parking
> location. If you are sending a call in from an IVR then the caller will
> hear their park location. The only way to know where the call went would be
> to watch the event socket for relevant valet events. "
>
> Yes, right now the extension where call is parked, is announced to the
> caller. I don't want that to happen. I don't want it announced to caller,
> rather I want it retruned to me, so that I can store it in database. So
> that my agents can pick the call after seeing the extension where it is
> parked.
>
> Please tell me more details about how to watch the event socket for valet
> events.
>
> Regards
>
> Nitin
>
> On Friday, 30-11-2012 on 6:38 Michael Collins wrote:****
>
> Answers inline...****
>
> On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:*
> ***
>
> Hi,****
>
>  ****
>
> I am using valet_park. I?ve configure a IVR menu of an extension, based on
> user?s input call is forwarded to other extensions.****
>
>  ****
>
> Extension on which end-users will call ?****
>
>  ****
>
> <extension name="ivr_nitin">
>       <condition field="destination_number" expression="^5002$">
>         <action application="answer"/>
>         <action application="sleep" data="500"/>
>         <action application="ivr" data="nitin_ivr"/>
>       </condition>
> </extension>****
>
>  ****
>
> The IVR configuration XML is ?****
>
>  ****
>
> <include>
>         <menu name="nitin_ivr"
>                 greet-long="say:Welcome to Newgen General Insurance
> Company. Press 1 for Changing Address, 2 for Changing Nominee or 3 for
> Close Policy."
>                 greet-short="say:Welcome to Newgen. Press 1 for Changing
> Address, 2 for Changing Nominee or 3 for Close Policy."
>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>                 exit-sound="voicemail/vm-goodbye.wav"
>                 confirm-macro=""
>                 confirm-key=""
>                 tts-engine="flite"
>                 tts-voice="slt"
>                 confirm-attempts="3"
>                 timeout="3000"
>                 inter-digit-timeout="2000"
>                 max-failures="3"
>                 max-timeouts="3"
>                 digit-len="4">
>
>                 <entry action="menu-exec-app" digits="1" param="transfer
> 450 XML default"/>    <!-- Change Address -->
>                 <entry action="menu-exec-app" digits="2" param="transfer
> 451 XML default"/>    <!-- Change Nominee -->
>                 <entry action="menu-exec-app" digits="3" param="transfer
> 452 XML default"/>    <!-- Close Policy -->
>
>         </menu>
> </include>****
>
> Once user presses ?1?, call is forwarded to 450, for this extension
> dialplan entry is ?****
>
>  ****
>
> <extension>
>           <condition field="destination_number" expression="^(450)$">
>           <action application="play_and_get_digits" data="10 16 3 3000 #
> say:'Press your account number, followed by hash key' say:'Wrong Input' res
> \d+" />
>           <action application="phrase" data="spell,${res}"/>
>           <action application="db" data="insert/testapp/newcall1/${res}"
> />
>               <action application="set" data="caller_id_name=Change
> Address Request" />
>               <action application="set" data="call_timeout=60" />
>               <action application="set" data="originate_timeout=60" />
>               <action application="valet_park" data="my_lot auto in 8501
> 8599" />           ****
>
>           </condition>
> </extension>****
>
>  ****
>
> Here, the call is parked at any available extension between 8501 to 8599.*
> ***
>
>  ****
>
> Then I?ve set up an extension to pick up calls ?****
>
>  ****
>
> <extension name="park-out">
>    <condition field="destination_number" expression="^(85\d\d)$">
>      <action application="answer"/>
>      <action application="valet_park" data="my_lot $1"/>
>    </condition>
>  </extension>****
>
>  ****
>
> I have a few questions ?****
>
>  ****
>
> 1.       Valet_park parks the call on any available extension between
> 8501 to 8599 (<action application="valet_park" data="my_lot auto in 8501
> 8599" />). Is there any way to let me know on which extension the call have
> been parked?****
>
> Using 'auto in' the system will announce the parking location. If you are
> sending a call in from an IVR then the caller will hear their park
> location. The only way to know where the call went would be to watch the
> event socket for relevant valet events. ****
>
>   2.       How can I get the number from which call was made in extension
> 450. The idea is to use the caller number as key and entered value as value
> while making entry in database (<action application="db"
> data="insert/testapp/newcall1/${res}" />)?****
>
>  Do you mean the caller id number? That's literally in channel variable
> ${caller_id_number} ****
>
>   3.       If two users call on extension 5002 (where IVR menu is
> played), what will happen? Will the second user have to wait for first to
> finish or whether both will be connected parallel?****
>
>  Both can be in the IVR at the same time and they won't affect each other
> at all.
>
> -MC ****
>
>    ****
>
> Please help me out.****
>
>  ****
>
> Regards****
>
>  ****
>
> Nitin****
>
> ** **
>  Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
> ** **
>  Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
> ** **
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
> ** **
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Nov 30 21:57:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Nov 2012 10:57:36 -0800
Subject: [Freeswitch-users] Routing call to another interface
In-Reply-To: <CAGTd76ERO92fAFPLwCFgTzY5r3_s=OnJBYZHtX21u8Drs+wm6Q@mail.gmail.com>
References: <CAGTd76ERO92fAFPLwCFgTzY5r3_s=OnJBYZHtX21u8Drs+wm6Q@mail.gmail.com>
Message-ID: <CAKzWOxVWpN20xDzQNH2Ctuku+W+UTht2Qy9eB809zH0_BbVKtQ@mail.gmail.com>

Do you already have a sip profile created for 10.x.x.x interface? If you do
then you can route calls out that interface. You can also create a gateway
for that profile. Let's say the profile for 10.x.x.x is called "outside1".
You can send calls out like this:

<action application="bridge" data="sofia/profile/outside1/
18005551212 at domain.com"/>

If you've created a gateway then it's just the standard gateway syntax.
More info here<http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings>
.

-MC

On Thu, Nov 29, 2012 at 10:35 PM, William Alianto <william at xofap.com> wrote:

> Hi,
>
> I got a server with this configuration :
>
>
> Gateway
> ^
> |
> |
> |
> | eth 2 (10.x.x.x)
> |
> |
> FS<-------->eth1 (180.x.x.x) <------->Internet<---------->User
>
> Now I need to route the call to/from gateway. Would anybody help me to
> configure a dialplan for this problem?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Fri Nov 30 22:03:38 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Nov 2012 13:03:38 -0600
Subject: [Freeswitch-users] FreeSWITCH suddenly stopped accepting calls
In-Reply-To: <ADF32A4A0F985A428C85B277B8E1C9D80F48B7EA@NY1-EXMB-02.ip-soft.net>
References: <ADF32A4A0F985A428C85B277B8E1C9D80F48B7EA@NY1-EXMB-02.ip-soft.net>
Message-ID: <CAKbxfG9_eLwjq0XywxsOU-WoxubfPRPv2YNDjnZShdPvWfJp0Q@mail.gmail.com>

Reproduce it on latest GIT HEAD.
Then if you catch it happening, gcore the process and post the bt on a jira
search FS wiki for debugging FS and look for obtaining a backtrace



On Fri, Nov 30, 2012 at 11:06 AM, Hector Geraldino <
Hector.Geraldino at ipsoft.com> wrote:

>  Hello everyone,****
>
> ** **
>
> I have a FreeSWITCH instance that has been running for months without an
> issue, until last Monday. The setup is for a hotline where we receive
> incoming calls transferred from the Client Cisco UCM to our FreeSWITCH
> instance, and the calls are handled by a custom Java application via ESL.*
> ***
>
> ** **
>
> As I said it was working flawlessly for weeks, but last Monday FreeSWITCH
> just stopped accepting new calls. I was able to see the SIP messages
> flowing back and forth from FS<->UCM, but looks like FreeSWITCH was unable
> to create new channels. Once I restarted FreeSWITCH everything was back to
> normal. Here are links to the SIP/console logs I was able to capture which
> covers both scenarios (non-working FS and working FS):****
>
> ** **
>
> Non-working: http://pastebin.freeswitch.org/20275****
>
> Working: http://pastebin.freeswitch.org/20274****
>
> ** **
>
> As you can see, in the non-working scenario no channels were created on
> FreeSWITCH, so no 200 ACK message was sent back to UCM after the INVITE,
> which caused the UCM to send a CANCEL message after a few seconds and a 487
> Request terminated from FS -> UCM. Restarting the application fixed the
> issue, but I have a couple of questions regarding this:****
>
> ** **
>
> **1)      **We have a lot of monitoring in place for all our
> applications, and I could probably alerted our Operations team if
> FreeSWITCH crashed or some error was logged in the FS log. However I was
> unable to found anything meaningful in the logs that indicates any sort of
> error. It just suddenly stopped creating new channels, and we get the
> alert/complain from the client ? ouch!****
>
> **2)      **I was also not sure what other information I should collect
> when something like this happens. As the application didn?t crash it didn?t
> create any coredump files. I know that restarting it fixes the problem, but
> that is not what the client wants to hear.****
>
> ** **
>
> Any recommendations on what to do/look at to proactively detect this kind
> of issues and/or how to collect the appropriate info to get the root cause
> will be greatly appreciated. ****
>
> ** **
>
> Thank you all for your time,****
>
> Hector****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From josefu at gmail.com  Fri Nov 30 22:28:39 2012
From: josefu at gmail.com (=?ISO-8859-1?Q?Jose_Fco=2E_Irles_Dur=E1?=)
Date: Fri, 30 Nov 2012 20:28:39 +0100
Subject: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
Message-ID: <CAJOPU6XN2Wh0ROUEK45-4W-8XmEsM7qiomXFtU+odi5Vr2Qk2g@mail.gmail.com>

Hello everybody, first of all, sorry for my english.

I have a test server with the mod_cdr_mongodb backend configured.

I'm trying to parse the cdrs with the mongodb driver for java, but i
have a trouble with a part of the json that freeswitch sends (mongo
saves the json document without errors). I haven't an API function to
extract some data. Firstly I thought that the problem was in the java
driver but I'm not sure.

The problem is in the "callflow" part, with the json standard[1], a
json document can't have two keys with the same name, but
mod_cdr_mongodb builds the json with a 'n' callflow objects.
Also it happends inside callflow object, with origination, originator
and originatee objects.

mod_cdr_mongodb saves the json with this format:
http://pastebin.com/VRz6s0eb

I modified the source of the backend adding json arrays around this
objects and now I can parse without problems.

The output with the modifications:
http://pastebin.com/v05FN9Ta

I'm in the correct way or I'm missing something?

[1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2

Regards
--
Jose Fco. Irles Dur?


From spencer at 5ninesolutions.com  Fri Nov 30 22:30:27 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 30 Nov 2012 11:30:27 -0800
Subject: [Freeswitch-users] Caller Name with FreeTDM libpri
Message-ID: <F31EBEBC-5C9B-43BE-8282-91BC976D8809@5ninesolutions.com>

Hello,
I have an ISDN PRI where the caller name is sent in a FACILITY message after the initial SETUP message.  Does anyone know how I can access this?  I'm using FreeTDM in DAHDI mode with libpri.

Thanks,
Spencer



From vbvbrj at gmail.com  Fri Nov 30 23:00:45 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Fri, 30 Nov 2012 22:00:45 +0200
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <CAKzWOxXoRQGVSA5TAAf+Xb1+foO5wJ20ax-nn=NRa-EBTpp7KQ@mail.gmail.com>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>
	<1354241384.50b81568a4579@mx.newgen.co.in>
	<CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
	<00ff01cdceb3$ca4a7ec0$5edf7c40$@co.in>
	<01d601cdceee$cc35f940$64a1ebc0$@co.in>
	<CAKzWOxXoRQGVSA5TAAf+Xb1+foO5wJ20ax-nn=NRa-EBTpp7KQ@mail.gmail.com>
Message-ID: <50B9106D.9080609@gmail.com>

> On Fri, Nov 30, 2012 at 3:35 AM, Nitin Tomer <ntomer at newgen.co.in
> <mailto:ntomer at newgen.co.in>> wrote:

>     I want to execute the above query, and check whether any result
>     came. If results came, then I want to proceed to next number. And if
>     no result came, then I want to set it in the session variable and
>     break from the while loop. But I could only find the syntax for
>     assert, no way to check whether the query returned any result or
>     not. Please help me out.____

This is how I do it:

querystring="SELECT `name` FROM `table` WHERE `id`="..id.."
dbh:query(querystring,function(row) result=row.name end)
if result~=nil then
	-- do something with the result get
end

-- 
Mimiko desu.


From jason.holden at start.ca  Fri Nov 30 23:01:49 2012
From: jason.holden at start.ca (Jason Holden)
Date: Fri, 30 Nov 2012 15:01:49 -0500
Subject: [Freeswitch-users] moh
Message-ID: <0ED15667341B47038B6A428015DEE70E@bob>

Is there a way that moh is always playing in the background so when I place
a user on hold they are not always starting at the beginning of the hold
file?

 

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From msc at freeswitch.org  Fri Nov 30 23:40:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Nov 2012 12:40:43 -0800
Subject: [Freeswitch-users] moh
In-Reply-To: <0ED15667341B47038B6A428015DEE70E@bob>
References: <0ED15667341B47038B6A428015DEE70E@bob>
Message-ID: <CAKzWOxWTkPQTcAvooWnA=yG2vDAtMHZMf5nbxgP4wm6Y4LdrLA@mail.gmail.com>

If you're using the example configs then it's already like that. The hold
music is a stream and it's constantly playing. How are you doing your hold
music?
-MC

On Fri, Nov 30, 2012 at 12:01 PM, Jason Holden <jason.holden at start.ca>wrote:

>  Is there a way that moh is always playing in the background so when I
> place a user on hold they are not always starting at the beginning of the
> hold file?****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From brian at freeswitch.org  Fri Nov 30 23:48:09 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Nov 2012 14:48:09 -0600
Subject: [Freeswitch-users] Caller Name with FreeTDM libpri
In-Reply-To: <F31EBEBC-5C9B-43BE-8282-91BC976D8809@5ninesolutions.com>
References: <F31EBEBC-5C9B-43BE-8282-91BC976D8809@5ninesolutions.com>
Message-ID: <5F8689C5-3C82-4762-BD2B-DFE9659C19B0@freeswitch.org>

facility
facility-timeout

Are the params you want.

--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 30, 2012, at 1:30 PM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:

> Hello,
> I have an ISDN PRI where the caller name is sent in a FACILITY message after the initial SETUP message.  Does anyone know how I can access this?  I'm using FreeTDM in DAHDI mode with libpri.
> 
> Thanks,
> Spencer
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Fri Nov 30 23:48:38 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Nov 2012 14:48:38 -0600
Subject: [Freeswitch-users] moh
In-Reply-To: <0ED15667341B47038B6A428015DEE70E@bob>
References: <0ED15667341B47038B6A428015DEE70E@bob>
Message-ID: <E759777F-1C38-4489-BDBE-996394C5A828@freeswitch.org>

Use local_stream like the default configs do.

--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900
ISN: 410*543





On Nov 30, 2012, at 2:01 PM, Jason Holden <jason.holden at start.ca> wrote:

> Is there a way that moh is always playing in the background so when I place a user on hold they are not always starting at the beginning of the hold file?
>