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

Hello FreeSWITCHers!

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

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

Looking forward to next Wednesday!

-MC
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From Nabble_01394 at slickdeals.endjunk.com  Sat Oct  1 02:44:08 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Fri, 30 Sep 2011 15:44:08 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH Conference Call Oct 5: All about
	git
In-Reply-To: <CAKzWOxVHZfYM97CTko=KdgEf-=xWHH0hsX4wEjEaN1JCMg03Jw@mail.gmail.com>
References: <CAKzWOxVHZfYM97CTko=KdgEf-=xWHH0hsX4wEjEaN1JCMg03Jw@mail.gmail.com>
Message-ID: <1317422648667-6849446.post@n2.nabble.com>


mercutioviz wrote:
> Looking forward to next Wednesday!
Me 2. Love GIT so much that I just can't *g*et *it* over with! 

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-Conference-Call-Oct-5-All-about-git-tp6849238p6849446.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From alec.taylor6 at gmail.com  Sat Oct  1 08:17:01 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sat, 1 Oct 2011 14:17:01 +1000
Subject: [Freeswitch-users] FreeSWITCH Conference Call Oct 5: All about
	git
In-Reply-To: <CAKzWOxVHZfYM97CTko=KdgEf-=xWHH0hsX4wEjEaN1JCMg03Jw@mail.gmail.com>
References: <CAKzWOxVHZfYM97CTko=KdgEf-=xWHH0hsX4wEjEaN1JCMg03Jw@mail.gmail.com>
Message-ID: <CAO+9iGdfssnXZZtxCMNh6MSs+2uJXAG=Dbr6FZ707s_UqvJqPw@mail.gmail.com>

Should make it!

Which timezone is it being presented in?

On 10/1/11, Michael Collins <msc at freeswitch.org> wrote:
> Hello FreeSWITCHers!
>
> Travis Cross will be presenting on the subject of git. Many of us have a
> love-hate (or a hate-hate) relationship with git so Travis is going to help
> us get over the hump. The agenda page has a place where you can post
> questions:
>
> http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05#Featured_Presentation
>
> Looking forward to next Wednesday!
>
> -MC
>


From msc at freeswitch.org  Sat Oct  1 09:35:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Sep 2011 22:35:56 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Oct 5: All about
	git
In-Reply-To: <CAO+9iGdfssnXZZtxCMNh6MSs+2uJXAG=Dbr6FZ707s_UqvJqPw@mail.gmail.com>
References: <CAKzWOxVHZfYM97CTko=KdgEf-=xWHH0hsX4wEjEaN1JCMg03Jw@mail.gmail.com>
	<CAO+9iGdfssnXZZtxCMNh6MSs+2uJXAG=Dbr6FZ707s_UqvJqPw@mail.gmail.com>
Message-ID: <CAKzWOxW5ycaYaunvCfcpaeZ+L8hHD-kCFSe0ttLZO9eCxtuXRQ@mail.gmail.com>

The agenda page has the specific information on time and call-in numbers.
However, it is usually 1PM eastern, 10AM pacific.

-MC

On Fri, Sep 30, 2011 at 9:17 PM, Alec Taylor <alec.taylor6 at gmail.com> wrote:

> Should make it!
>
> Which timezone is it being presented in?
>
> On 10/1/11, Michael Collins <msc at freeswitch.org> wrote:
> > Hello FreeSWITCHers!
> >
> > Travis Cross will be presenting on the subject of git. Many of us have a
> > love-hate (or a hate-hate) relationship with git so Travis is going to
> help
> > us get over the hump. The agenda page has a place where you can post
> > questions:
> >
> >
> http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05#Featured_Presentation
> >
> > Looking forward to next Wednesday!
> >
> > -MC
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alec.taylor6 at gmail.com  Sat Oct  1 10:40:58 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sat, 1 Oct 2011 16:40:58 +1000
Subject: [Freeswitch-users] FreeSWITCH Conference Call Oct 5: All about
	git
In-Reply-To: <CAKzWOxW5ycaYaunvCfcpaeZ+L8hHD-kCFSe0ttLZO9eCxtuXRQ@mail.gmail.com>
References: <CAKzWOxVHZfYM97CTko=KdgEf-=xWHH0hsX4wEjEaN1JCMg03Jw@mail.gmail.com>
	<CAO+9iGdfssnXZZtxCMNh6MSs+2uJXAG=Dbr6FZ707s_UqvJqPw@mail.gmail.com>
	<CAKzWOxW5ycaYaunvCfcpaeZ+L8hHD-kCFSe0ttLZO9eCxtuXRQ@mail.gmail.com>
Message-ID: <CAO+9iGeRgZCOSSJrScfngFWmiBgqWz-9AhGfurJ5VNWmJj2jnQ@mail.gmail.com>

Cheers

On Sat, Oct 1, 2011 at 3:35 PM, Michael Collins <msc at freeswitch.org> wrote:
> The agenda page has the specific information on time and call-in numbers.
> However, it is usually 1PM eastern, 10AM pacific.
> -MC
>
> On Fri, Sep 30, 2011 at 9:17 PM, Alec Taylor <alec.taylor6 at gmail.com> wrote:
>>
>> Should make it!
>>
>> Which timezone is it being presented in?
>>
>> On 10/1/11, Michael Collins <msc at freeswitch.org> wrote:
>> > Hello FreeSWITCHers!
>> >
>> > Travis Cross will be presenting on the subject of git. Many of us have a
>> > love-hate (or a hate-hate) relationship with git so Travis is going to
>> > help
>> > us get over the hump. The agenda page has a place where you can post
>> > questions:
>> >
>> >
>> > http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05#Featured_Presentation
>> >
>> > Looking forward to next Wednesday!
>> >
>> > -MC
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tatianava at tut.by  Sat Oct  1 12:24:42 2011
From: tatianava at tut.by (vitaleg)
Date: Sat, 1 Oct 2011 01:24:42 -0700 (PDT)
Subject: [Freeswitch-users] Voicemail notifications not working
Message-ID: <1317457482603-6850183.post@n2.nabble.com>

Hello. Please help - I want to send notifications of new vm to my email. In
directory for the particular user set:

vm-notify-mailto, vm-notify-email-all-messages  params

and in switch.conf.xml:




But stil no notifications are sending. When missed vm-notify-mailto, there
was a notification that something wrong and notifications won`t work/ Then I
fixed it, warning disappeared. But still no notifications, no errors and so
on ( Please help.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Voicemail-notifications-not-working-tp6850183p6850183.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tatianava at tut.by  Sat Oct  1 14:49:20 2011
From: tatianava at tut.by (vitaleg)
Date: Sat, 1 Oct 2011 03:49:20 -0700 (PDT)
Subject: [Freeswitch-users] Voicemail notifications not working
In-Reply-To: <1317457482603-6850183.post@n2.nabble.com>
References: <1317457482603-6850183.post@n2.nabble.com>
Message-ID: <1317466160490-6850392.post@n2.nabble.com>

Solved - for some reason fromuser was set to user at domain, while I set it to
voicemail at mydomain.com. So gmail did not allowed to receive such emails. Now
changed domain to mydomain.com and it works. Still uncear about why my
settings for fromuser were ignored. By the way, tpl file was edited for
correct sender too.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Voicemail-notifications-not-working-tp6850183p6850392.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rupa at rupa.com  Sun Oct  2 07:20:14 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 1 Oct 2011 22:20:14 -0500
Subject: [Freeswitch-users] FreeSWITCH never hangup at nobal_amount
In-Reply-To: <1316090079670-6796618.post@n2.nabble.com>
References: <1316090079670-6796618.post@n2.nabble.com>
Message-ID: <CAJw9H-Nn=q=9ap6E98DXCJSTfJF7NH6SHtYXzT3asu2-ZL6LAQ@mail.gmail.com>

Did you get get resolved?

On Thu, Sep 15, 2011 at 7:34 AM, mg123 <nish.chn at gmail.com> wrote:
> freeswitch is not supporting billing during the call and never hangup at
> nobal_amount.
>
> nibblebil_conf.xml configurations
>
>
>
>
> During the call fs_cli display below lines for each heartbeat
>
> ? ? 2011-09-15 17:30:42.507425 [DEBUG] mod_nibblebill.c:573 Received
> request via SESSION_HEARTBEAT!
> ? ? 2011-09-15 17:30:42.507425 [DEBUG] mod_nibblebill.c:434 Attempting to
> bill at $1.00 per minute to account 1005
> ? ? 2011-09-15 17:31:42.544169 [DEBUG] mod_nibblebill.c:573 Received
> request via SESSION_HEARTBEAT!
> ? ? 2011-09-15 17:31:42.544169 [DEBUG] mod_nibblebill.c:434 Attempting to
> bill at $1.00 per minute to account 1005
>
> Please advice.
>
> Thanks.
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-never-hangup-at-nobal-amount-tp6796618p6796618.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From leon at scarlet-internet.nl  Sun Oct  2 13:08:54 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Sun, 2 Oct 2011 11:08:54 +0200
Subject: [Freeswitch-users] User management
In-Reply-To: <0EDCF8FF-EAD8-4F3B-B974-0A19FD953379@lyonl.com>
References: <ACCA0268-D0F0-42A0-A755-43745D9E1570@lyonl.com>
	<4E85440B.4020203@livecall.com>
	<0EDCF8FF-EAD8-4F3B-B974-0A19FD953379@lyonl.com>
Message-ID: <F5AF911D-ED18-44CF-98CA-8796C7F94798@scarlet-internet.nl>

Hello Chad,

You do get IP information through mod_xml_curl. You should get the following variables to your web application:

* action
* sip_profile
* sip_user_agent
* sip_auth_username
* sip_auth_realm
* sip_auth_nonce
* sip_auth_uri
* sip_contact_user
* sip_contact_host
* sip_to_user
* sip_to_host
* sip_to_port
* sip_from_user
* sip_from_host
* sip_from_port
* sip_request_user
* sip_request_host
* sip_request_port
* sip_auth_qop
* sip_auth_cnonce
* sip_auth_response
* sip_auth_method

Besides that you also get all sip headers that start with X-

But of course you can use any module that implements the xml handler directory binding such as:

* mod_erlang_event
* mod_perl
* mod_yaml
* mod_lua
* mod_managed
* mod_xml_curl
* mod_xml_ldap

I don't know how well each of these work, but I can tell that mod_lua (in combination with fetching data from a db) works great in our production environment - see an example here: http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh#User_Directory_XML

regards,

Leon


On Sep 30, 2011, at 7:31 AM, Chad Vogel wrote:

> I'm not looking to use curl, I would like to use some kind of scripting so i can apply custom rules to track and reject authentication based on rate and origin of authentication from source ip address
> 
> On Sep 29, 2011, at 11:22 PM, Jack wrote:
> 
>> Hi Chad,
>> mod_curl works great for that....
>> Jack
>> 
>> On 9/29/2011 5:47 PM, Chad Vogel wrote:
>>> Hello,
>>> 
>>> I am wondering if anyone has created any scripts to manage user authentication using a database?
>>> 
>>> Chad
>>> 
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Oct  2 20:57:43 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 2 Oct 2011 09:57:43 -0700
Subject: [Freeswitch-users] User management
In-Reply-To: <F5AF911D-ED18-44CF-98CA-8796C7F94798@scarlet-internet.nl>
References: <ACCA0268-D0F0-42A0-A755-43745D9E1570@lyonl.com>
	<4E85440B.4020203@livecall.com>
	<0EDCF8FF-EAD8-4F3B-B974-0A19FD953379@lyonl.com>
	<F5AF911D-ED18-44CF-98CA-8796C7F94798@scarlet-internet.nl>
Message-ID: <CAMWiMy_eQ=MzdEL5e2oUyQ3Gq+1Q_uebr5YwC74tRNUmpFLxLA@mail.gmail.com>

fusionpbx does a great job of user management if you're looking to
deploy this on a SME type of envrionment where SSO isn't a that much
of an issue

On Sun, Oct 2, 2011 at 2:08 AM, Leon de Rooij <leon at scarlet-internet.nl> wrote:
> Hello Chad,
> You do get IP information through mod_xml_curl. You should get the following
> variables to your web application:
> * action
> * sip_profile
> * sip_user_agent
> * sip_auth_username
> * sip_auth_realm
> * sip_auth_nonce
> * sip_auth_uri
> * sip_contact_user
> * sip_contact_host
> * sip_to_user
> * sip_to_host
> * sip_to_port
> * sip_from_user
> * sip_from_host
> * sip_from_port
> * sip_request_user
> * sip_request_host
> * sip_request_port
> * sip_auth_qop
> * sip_auth_cnonce
> * sip_auth_response
> * sip_auth_method
> Besides that you also get all sip headers that start with X-
> But of course you can use any module that implements the xml handler
> directory binding such as:
> * mod_erlang_event
> * mod_perl
> * mod_yaml
> * mod_lua
> * mod_managed
> * mod_xml_curl
> * mod_xml_ldap
> I don't know how well each of these work, but I can tell that mod_lua (in
> combination with fetching data from a db) works great in our production
> environment - see an example
> here:?http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh#User_Directory_XML
> regards,
> Leon
>
> On Sep 30, 2011, at 7:31 AM, Chad Vogel wrote:
>
> I'm not looking to use curl, I would like to use some kind of scripting so i
> can apply custom rules to track and reject authentication based on rate and
> origin of authentication from source ip address
>
> On Sep 29, 2011, at 11:22 PM, Jack wrote:
>
> Hi Chad,
>
> mod_curl works great for that....
>
> Jack
>
> On 9/29/2011 5:47 PM, Chad Vogel wrote:
>
> Hello,
>
> I am wondering if anyone has created any scripts to manage user
> authentication using a database?
>
> Chad
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 henrikaagaardsorensen at gmail.com  Sun Oct  2 21:48:06 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Sun, 2 Oct 2011 19:48:06 +0200
Subject: [Freeswitch-users] FreeSwitch fails with Kamailio Dispatcher in
	front.
Message-ID: <CAGH8SeaQtgu7eNh7aHa8RZzoBj14L_Fn9oR7sgFO+ESGUPK6EA@mail.gmail.com>

I've setup a Freeswitch which works when connecting devices directly to it.

But with a Kamailio Dispatcher setup in front, it fails. It reports the
error:

2011-09-30 08:47:04.309900 [ERR] sofia_reg.c:2136 Cannot locate any
authentication credentials to complete an authentication request for realm
'"10.0.1.7"'
I've setup tshark on FreeSwitch now to analyze the input from both
scenarios.

When connected directly to FreeSwitch, which works, this is my INVITE:
http://pastebin.com/p4CZJJz3

This is how the INVITE looks when it fails via Kamailio:
http://pastebin.com/9AFEsuVs

Can anyone help my out?
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From lloyd.aloysius at gmail.com  Mon Oct  3 05:01:23 2011
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sun, 2 Oct 2011 21:01:23 -0400
Subject: [Freeswitch-users] FreeSWITCH Voicemail - silence detection
Message-ID: <CAGvuWYa1VxFFaQWt7e0BQsu5JuaZbugaSzAve+i64S2Zv4Efbg@mail.gmail.com>

Hello,

I have the voicemail issue. When someone  soft spoken leaves a message they
get interrupted by the silence detection.

?to play this message press 1, to send this message press 2, to delete this
message and try again ??

I found two options in voicemail.conf. None of them make any difference.

<param name="record-silence-threshold" value="200"/>
<param name="record-silence-hits" value="2"/>

Is there anyway to increase the threshold or time out.

FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)


Thanks in advance.
Lloyd
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From frisch.alan at gmail.com  Mon Oct  3 08:02:17 2011
From: frisch.alan at gmail.com (Alan Frisch)
Date: Mon, 3 Oct 2011 00:02:17 -0400
Subject: [Freeswitch-users] MOD_LCR: IntraState and IntraLata Confusion.
Message-ID: <CAFoaFX6aaHJKNsfM48OhgRBp5n8jLLiZ2wzNQrCP3ZAChj_gdA@mail.gmail.com>

Been cutting my teeth on Mod_LCR today with inter/intra state costs and have
run into a little issue that I am sure one of the refined experts can help
with.

Now if I have a rate list that is like:

NPANXX,INTERSTATE,INTRASTATE
212203,0.00738,0.00716
...
203302,0.00604,0.00716

I assume I should load the INTRASTATE into the intrastate_rate MySQL field
(of course).... and would I then load the INTERSTATE into the rate field?
What do I do about the intralata_rate field?

If I leave intralata_rate 0 or NULL, then LCR fails.  I only want to rate on
inter/intra-state and not on LATA.  If tried setting the intralata_rate to
be the same as the interstate_rate which works fine in most examples, but
with the above data (one number being in NY and the other in CT) the LCR
selects an intralata rate instead of an interstate rate, due to the common
lata (though the states are different).

How can I get Mod_LCR to rate only by state?  I am just playing with the LCR
command in CLI and not in the dialplan yet.

Thanks.

AF.
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From valery.kalinin at gmail.com  Mon Oct  3 09:41:18 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Mon, 3 Oct 2011 11:41:18 +0600
Subject: [Freeswitch-users] My new web utility - Windows 7 gadget
Message-ID: <CAF=swh---deRqRKCyJGxhPgAd9bdTvXhzqA1qwRFFExmCr9SNA@mail.gmail.com>

I offer you my new web application for FreeSWITCH.
This simple utility runs as a gadget for Windows 7 or Vista.
When a call comes in, it displays a window with the name of the caller.
If desired, the program can be customized to integrate with your CRM system
or Helpdesk.
Download at:
https://sites.google.com/site/freeswitched/home/downloads
It consists of two parts: server and client.
The server part - a simple script fscp.php hosted on your web server that
has access to the server FreeSWITCH.
Allowing access to FreeSWITCH at the beginning of the file.
These are the three variables:
    $FreeSWITCHserver = '127 .0.0.1 ';
    $FreeSWITCHport = 8021;
    $FreeSWITCHpassword = 'ClueCon';

The client part is made as a gadget for Windows 7/Vista.
At its first launch should proceed with setup, specifying:
Server address:
    Full address of the script fscp.php, for example:
    http://192.168.0.1/fscp.php
Account:
    Account in FreeSWITCH, which calls for will be tracked, for example:
1000.

When a call window will be displayed by means of VBScript (modal) or
Javascript (the usual).

Tests and features are welcome!
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From jcgpoza at gmail.com  Mon Oct  3 11:33:52 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Mon, 3 Oct 2011 09:33:52 +0200
Subject: [Freeswitch-users] Customer SIP headers
In-Reply-To: <335D0A35-929D-4CAB-9E68-99FDE829C489@lyonl.com>
References: <335D0A35-929D-4CAB-9E68-99FDE829C489@lyonl.com>
Message-ID: <CAMDVdehRPV-6=z-+0TZ_0SBfV1PjPSjDpHa9APAJzx5M-DZ9Eg@mail.gmail.com>

Hello Chad,

I guess you are not supposed to do that because you must only use the
standard headers, otherwise you could have compatibility issues.

Use

<action application="set" data="sip_h_X-myheader"/>


To create your own standard headers. If you still want to create your own
headers I'm afraid you are gonna have to patch the mod_sofia.

Best Regards.



2011/9/30 Chad Vogel <cvogel at lyonl.com>

>  *hello,*
> *
> *
> *I'm looking to do custom sip headers. I wan't to send a 302 message and
> set the content header to something like this: Contact: <tel:+17208886435>
> ;expires=3600 Also on the From header I would like to add a isup-oli param
> to set ISUP cause codes.  **how can i do this? *
> *
> *
> *Chad*
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Oct  3 17:34:54 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Oct 2011 08:34:54 -0500
Subject: [Freeswitch-users] Customer SIP headers
In-Reply-To: <CAMDVdehRPV-6=z-+0TZ_0SBfV1PjPSjDpHa9APAJzx5M-DZ9Eg@mail.gmail.com>
References: <335D0A35-929D-4CAB-9E68-99FDE829C489@lyonl.com>
	<CAMDVdehRPV-6=z-+0TZ_0SBfV1PjPSjDpHa9APAJzx5M-DZ9Eg@mail.gmail.com>
Message-ID: <DD930196-125C-45C4-BA1F-D76144D63FF0@freeswitch.org>

look in do_invite in sofia_glue.c you have plenty of things you can modify with just variables but be warned you'll break a lot of things if you change stuff beyond what works.

/b

On Oct 3, 2011, at 2:33 AM, jcgpoza gonzalez wrote:

> Hello Chad,
> 
> I guess you are not supposed to do that because you must only use the
> standard headers, otherwise you could have compatibility issues.
> 
> Use
> 
> <action application="set" data="sip_h_X-myheader"/>
> 
> 
> To create your own standard headers. If you still want to create your own
> headers I'm afraid you are gonna have to patch the mod_sofia.
> 
> Best Regards.

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From jmoran at secureachsystems.com  Mon Oct  3 18:40:18 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Mon, 3 Oct 2011 10:40:18 -0400
Subject: [Freeswitch-users] Interaction between event handler and streamfile
	(avmd specifically)
Message-ID: <361E98F99D3CC3439EED59BC1924ED6954715A@SERVER2003.SecuReachSystems.local>

In short, I'm looking to stream a message twice unless a voicemail
beeps. If the voicemail beeps, then start over immediately and only play
the message once.

 

Pseudo-code:

 

session.execute("avmd", "start");

session.streamFile(playbackFileName); //1st play

session.streamFile(playbackFileName); //2nd play

session.execute("hangup","ORIGINATOR_CANCEL");

 

function onDTMF(s, type, obj, arg)

{  

  if(type == "event" && session.getVariable("avmd_detect") == "TRUE")

 {

  session.execute("avmd", "stop");

  session.streamFile(playbackFileName);

  return false;

 }

 return true; 
}

 

My question is what the effect and timing is of returning false or true
from the onDTMF function? Does the original streamFile continue playing?
Pause? end immediately? Do I have to set a variable, return false to
stop the original from playing, then do my streamfile logic OUTSIDE of
the onDTMF for proper flow? What happens if I never return true or
false, what does it do then?

 

Thanks,
Jason

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From igor.kuvaldin at studia52.ru  Sun Oct  2 12:09:11 2011
From: igor.kuvaldin at studia52.ru (=?utf-8?B?0JjQs9C+0YDRjCDQmtGD0LLQsNC70LTQuNC9?=)
Date: Sun, 02 Oct 2011 12:09:11 +0400 (MSD)
Subject: [Freeswitch-users] mod_callcenter, moh,
	and session progress (early media)
In-Reply-To: <f56cb06c-b5ef-4d9c-b97d-ad7f4a3e4984@main.studia52>
Message-ID: <0dd7ae2a-0fa3-45ed-9fe6-924533167e02@main.studia52>

Hello everyone.

How can I play moh in mod_callcenter in session progress?
Is there a bridge for the variables for such a possibility?
Or an analogue <action application="pre_answer"/>?


From phil.kim at valiant.com  Mon Oct  3 18:33:13 2011
From: phil.kim at valiant.com (Phil Kim)
Date: Mon, 3 Oct 2011 10:33:13 -0400
Subject: [Freeswitch-users] DTMF Problem with T-Mobile carrier
Message-ID: <C69FF59885AC464896A4B68500B6E97407C3221B@VAL-SV-EXCH-01.valiant.com>

Hello everybody,

 

I really need some help here!

 

I'm trying to use FS for making both inbound and outbound calls through
Broadvox.

Everything works perfectly except one particular situation with DTMF
problem. 

In case of inbound calling (I'm calling FS), everything works fine. Call
connects and I can send DTMF to FS and FS recognized those 100% (Testing
with softphone from my desktop, SIP phone in the office, and all
wireless carriers)

However, I am having trouble with outbound calling. I have set it up to
where FS makes an outbound call to certain number and once the call is
answered, we start playing some custom IVR. 

Here, it works fine with softphone from my desktop, SIP phones in the
office, AT&T, and Verizon. But for some reason FS will not recognize the
DTMF coming from T-Mobile cell phones.

Anyone has any idea? Below are my current setting from external.xml
which was recommended by Broadvox.

 

<param name="codec-prefs" value="PCMU,G729"/>

            <param name="dtmf-type" value="rfc2833"/>

            <param name="auto-rtp-bugs" value="clear"/>

            <param name="pass-rfc2833" value="true"/>      

            <param name="inbound-late-negotiation" value="true"/>

            

            <param name="sip-trace" value="no"/>

            <param name="sip-capture" value="no"/>        

            <param name="rfc2833-pt" value="101"/>

            <param name="sip-port" value="$${external_sip_port}"/>

            <param name="dialplan" value="XML"/>

            <param name="context" value="public"/>

            <param name="dtmf-duration" value="1000"/>

            <param name="inbound-codec-prefs"
value="$${global_codec_prefs}"/>

            <param name="outbound-codec-prefs"
value="$${outbound_codec_prefs}"/>

            <param name="hold-music" value="$${hold_music}"/>

            <param name="rtp-timer-name" value="soft"/>

 

 

 

Best regards,

 

Phil Kim

 

The information contained in this e-mail message, and any attachment
thereto, is the property of Valiant and is confidential and may not be
disclosed without our express permission. If you are not the intended
recipient or an employee or agent responsible for delivering this message to
the intended recipient, you are hereby notified that you have received this
message in error and that any review, dissemination, distribution or copying
of this message, or any attachment thereto, in whole or in part, is strictly
prohibited. If you have received this message in error, please immediately
notify us by telephone, fax or e-mail and delete the message and all of its
attachments. 

Thank you. 

Every effort is made to keep our network free from viruses. You should,
however, review this e-mail message, as well as any attachment thereto,
for viruses. We take no responsibility and have no liability for any
computer virus which may be transferred via this e-mail message.
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From kris at kriskinc.com  Mon Oct  3 20:07:00 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 3 Oct 2011 12:07:00 -0400
Subject: [Freeswitch-users] DTMF Problem with T-Mobile carrier
In-Reply-To: <C69FF59885AC464896A4B68500B6E97407C3221B@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E97407C3221B@VAL-SV-EXCH-01.valiant.com>
Message-ID: <CAKDfjgeXXnm-sV5yhB6GTg3sU+F6=dtMfmTWb3+iaDgh7rGOrA@mail.gmail.com>

Phil,

  Broadvox is known to do quite a few things that can affect DTMF
handling in non-straightforward ways.  This isn't a criticism of
Broadvox, it's just very common for carriers mixing upstreams
(especially when using Sonus).

  How far have you debugged this with Broadvox?  At minimum you'll
probably need a full SIP+RTP capture of the incoming call (especially
RFC2833 events) for anyone to help.

On Mon, Oct 3, 2011 at 10:33 AM, Phil Kim <phil.kim at valiant.com> wrote:
> Hello everybody,
>
>
>
> I really need some help here!
>
>
>
> I?m trying to use FS for making both inbound and outbound calls through
> Broadvox.
>
> Everything works perfectly except one particular situation with DTMF
> problem.
>
> In case of inbound calling (I?m calling FS), everything works fine. Call
> connects and I can send DTMF to FS and FS recognized those 100% (Testing
> with softphone from my desktop, SIP phone in the office, and all wireless
> carriers)
>
> However, I am having trouble with outbound calling. I have set it up to
> where FS makes an outbound call to certain number and once the call is
> answered, we start playing some custom IVR.
>
> Here, it works fine with softphone from my desktop, SIP phones in the
> office, AT&T, and Verizon. But for some reason FS will not recognize the
> DTMF coming from T-Mobile cell phones.
>
> Anyone has any idea? Below are my current setting from external.xml which
> was recommended by Broadvox.
>
>
>
> <param name="codec-prefs" value="PCMU,G729"/>
>
> ??????????? <param name="dtmf-type" value="rfc2833"/>
>
> ??????????? <param name="auto-rtp-bugs" value="clear"/>
>
> ??????????? <param name="pass-rfc2833" value="true"/>
>
> ??????????? <param name="inbound-late-negotiation" value="true"/>
>
>
>
> ??? ??????? <param name="sip-trace" value="no"/>
>
> ??? ??????? <param name="sip-capture" value="no"/>
>
> ??? ??????? <param name="rfc2833-pt" value="101"/>
>
> ??? ??????? <param name="sip-port" value="$${external_sip_port}"/>
>
> ??? ??????? <param name="dialplan" value="XML"/>
>
> ??? ??????? <param name="context" value="public"/>
>
> ??? ??????? <param name="dtmf-duration" value="1000"/>
>
> ?? ???????? <param name="inbound-codec-prefs"
> value="$${global_codec_prefs}"/>
>
> ??? ??????? <param name="outbound-codec-prefs"
> value="$${outbound_codec_prefs}"/>
>
> ?? ???????? <param name="hold-music" value="$${hold_music}"/>
>
> ??? ??????? <param name="rtp-timer-name" value="soft"/>
>
>
>
>
>
>
>
> Best regards,
>
>
>
> Phil Kim
>
>
>
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or copying
> of this message, or any attachment thereto, in whole or in part, is strictly
> prohibited. If you have received this message in error, please immediately
> notify us by telephone, fax or e-mail and delete the message and all of its
> attachments.
>
> Thank you.
>
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 phil.kim at valiant.com  Mon Oct  3 20:31:22 2011
From: phil.kim at valiant.com (Phil Kim)
Date: Mon, 3 Oct 2011 12:31:22 -0400
Subject: [Freeswitch-users] DTMF Problem with T-Mobile carrier
In-Reply-To: <CAKDfjgeXXnm-sV5yhB6GTg3sU+F6=dtMfmTWb3+iaDgh7rGOrA@mail.gmail.com>
References: <C69FF59885AC464896A4B68500B6E97407C3221B@VAL-SV-EXCH-01.valiant.com>
	<CAKDfjgeXXnm-sV5yhB6GTg3sU+F6=dtMfmTWb3+iaDgh7rGOrA@mail.gmail.com>
Message-ID: <C69FF59885AC464896A4B68500B6E97407C322EF@VAL-SV-EXCH-01.valiant.com>

Hi,

Thank you for your reply Kristian. 

I actually spent couple of hours with a tech support person from Broadvox. 

He recommended that I use 'rfc2833' for 'dtmf-type' since that was most reliable one. Then he was capturing traffics going in and out of my trunk and verified that signal were sent from Broadvox trunk in the same way for both AT&T and T-Mobile. I believed him since when I click from my T-Mobile cell phone, he could tell me which numbers I click when my FS was either not getting any or getting only some digits. 

Thing I really don't understand is that it really works perfectly fine when I call the FS from T-Mobile, but it does not when FS calls T-Mobile.
Does anyone know if there is any configuration I can change to control this?

For Kristian : I think I can get the SIP + RTP capture using WireShark. Can you help me if I send you the capture file?

Best regards,

Phil Kim

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: Monday, October 03, 2011 12:07 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF Problem with T-Mobile carrier

Phil,

  Broadvox is known to do quite a few things that can affect DTMF
handling in non-straightforward ways.  This isn't a criticism of
Broadvox, it's just very common for carriers mixing upstreams
(especially when using Sonus).

  How far have you debugged this with Broadvox?  At minimum you'll
probably need a full SIP+RTP capture of the incoming call (especially
RFC2833 events) for anyone to help.

On Mon, Oct 3, 2011 at 10:33 AM, Phil Kim <phil.kim at valiant.com> wrote:
> Hello everybody,
>
>
>
> I really need some help here!
>
>
>
> I'm trying to use FS for making both inbound and outbound calls through
> Broadvox.
>
> Everything works perfectly except one particular situation with DTMF
> problem.
>
> In case of inbound calling (I'm calling FS), everything works fine. Call
> connects and I can send DTMF to FS and FS recognized those 100% (Testing
> with softphone from my desktop, SIP phone in the office, and all wireless
> carriers)
>
> However, I am having trouble with outbound calling. I have set it up to
> where FS makes an outbound call to certain number and once the call is
> answered, we start playing some custom IVR.
>
> Here, it works fine with softphone from my desktop, SIP phones in the
> office, AT&T, and Verizon. But for some reason FS will not recognize the
> DTMF coming from T-Mobile cell phones.
>
> Anyone has any idea? Below are my current setting from external.xml which
> was recommended by Broadvox.
>
>
>
> <param name="codec-prefs" value="PCMU,G729"/>
>
> ??????????? <param name="dtmf-type" value="rfc2833"/>
>
> ??????????? <param name="auto-rtp-bugs" value="clear"/>
>
> ??????????? <param name="pass-rfc2833" value="true"/>
>
> ??????????? <param name="inbound-late-negotiation" value="true"/>
>
>
>
> ??? ??????? <param name="sip-trace" value="no"/>
>
> ??? ??????? <param name="sip-capture" value="no"/>
>
> ??? ??????? <param name="rfc2833-pt" value="101"/>
>
> ??? ??????? <param name="sip-port" value="$${external_sip_port}"/>
>
> ??? ??????? <param name="dialplan" value="XML"/>
>
> ??? ??????? <param name="context" value="public"/>
>
> ??? ??????? <param name="dtmf-duration" value="1000"/>
>
> ?? ???????? <param name="inbound-codec-prefs"
> value="$${global_codec_prefs}"/>
>
> ??? ??????? <param name="outbound-codec-prefs"
> value="$${outbound_codec_prefs}"/>
>
> ?? ???????? <param name="hold-music" value="$${hold_music}"/>
>
> ??? ??????? <param name="rtp-timer-name" value="soft"/>
>
>
>
>
>
>
>
> Best regards,
>
>
>
> Phil Kim
>
>
>
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or copying
> of this message, or any attachment thereto, in whole or in part, is strictly
> prohibited. If you have received this message in error, please immediately
> notify us by telephone, fax or e-mail and delete the message and all of its
> attachments.
>
> Thank you.
>
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
The information contained in this e-mail message, and any attachment
thereto, is the property of Valiant and is confidential and may not be
disclosed without our express permission. If you are not the intended
recipient or an employee or agent responsible for delivering this message to
the intended recipient, you are hereby notified that you have received this
message in error and that any review, dissemination, distribution or copying
of this message, or any attachment thereto, in whole or in part, is strictly
prohibited. If you have received this message in error, please immediately
notify us by telephone, fax or e-mail and delete the message and all of its
attachments. 

Thank you. 

Every effort is made to keep our network free from viruses. You should,
however, review this e-mail message, as well as any attachment thereto,
for viruses. We take no responsibility and have no liability for any
computer virus which may be transferred via this e-mail message.



From buscom123+fs at gmail.com  Mon Oct  3 23:14:59 2011
From: buscom123+fs at gmail.com (R H)
Date: Mon, 3 Oct 2011 13:14:59 -0600
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a single
	call
Message-ID: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>

Hey Everyone,

I am preparing to set up HA Failover to my production freeswitch
environment. In preparation for this I have been doing some testing in a
development environment and I encountered the following problem:

I currently have two sip-profiles set up that I am using full time, when I
enable odbc in the switch.conf.xml everything works correctly. When I enable
odbc in ONE of the sip-profiles everything works correctly. But when I
enable odbc in the second profile I begin to see two b-legs for ever call
that I initiate. Both b-legs are directed at the same client.

Does anyone have any idea why enabling odbc on two sofia profiles would
cause every inbound call to generate two outbound calls? I also decided to
test this problem by adding two more profiles for a total of 4. With 4
profiles I get 4 b-legs. What am I doing wrong?

Ryan
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From anthony.minessale at gmail.com  Mon Oct  3 23:23:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 Oct 2011 14:23:17 -0500
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
Message-ID: <CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>

could you elaborate more on what the B-legs are?
Are you doing bridges to an outbound entity, if so how?

do you have any traces doing:

sofia global siptrace on
and
console loglevel debug





On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
> Hey Everyone,
> I am preparing to set up HA Failover to my production freeswitch
> environment. In preparation for this I have been doing some testing in a
> development environment and I encountered the following problem:
> I currently have two sip-profiles set up that I am using full time, when I
> enable odbc in the switch.conf.xml everything works correctly. When I enable
> odbc in ONE of the sip-profiles everything works correctly. But when I
> enable odbc in the second profile I begin to see two b-legs for ever call
> that I initiate. Both b-legs are directed at the same client.
> Does anyone have any idea why enabling odbc on two sofia profiles would
> cause every inbound call to generate two outbound calls? I also decided to
> test this problem by adding two more profiles for a total of 4. With 4
> profiles I get 4 b-legs. What am I doing wrong?
> Ryan
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Mon Oct  3 23:26:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Oct 2011 12:26:33 -0700
Subject: [Freeswitch-users] FreeSWITCH Voicemail - silence detection
In-Reply-To: <CAGvuWYa1VxFFaQWt7e0BQsu5JuaZbugaSzAve+i64S2Zv4Efbg@mail.gmail.com>
References: <CAGvuWYa1VxFFaQWt7e0BQsu5JuaZbugaSzAve+i64S2Zv4Efbg@mail.gmail.com>
Message-ID: <CAKzWOxVZLX1JrzRwLdFtyNH35piVqhSasBd+OQz3AnEAwVE7YQ@mail.gmail.com>

Yes, you can control this. The threshold value is described here:
http://wiki.freeswitch.org/wiki/Mod_voicemail#record-silence-threshold

The silence hits is described here:
http://wiki.freeswitch.org/wiki/Mod_voicemail#record-silence-hits

Both of those link to the docs for the wait_for_silence dialplan tool, which
discusses these values in more detail:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence#Description

In a nutshell, if you lower the threshold value then that means the silence
needs to be "quieter" to be considered "silence." So, for a soft-spoken
voice you would want to lower the threshold. I would experiment with that.
Note that if you set the threshold too low then no silence will ever be
detected. As for the "hits" value, that just tweaks how much silence FS
needs to detect before it assumes there is silence on the line. The higher
the number of hits, the longer the silent period needs to be for FS to
assume it is "silence."

-MC

On Sun, Oct 2, 2011 at 6:01 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:

> Hello,
>
> I have the voicemail issue. When someone  soft spoken leaves a message they
> get interrupted by the silence detection.
>
> ?to play this message press 1, to send this message press 2, to delete this
> message and try again ??
>
> I found two options in voicemail.conf. None of them make any difference.
>
> <param name="record-silence-threshold" value="200"/>
> <param name="record-silence-hits" value="2"/>
>
> Is there anyway to increase the threshold or time out.
>
> FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)
>
>
> Thanks in advance.
> Lloyd
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mi.ke at null.net  Tue Oct  4 00:22:01 2011
From: mi.ke at null.net (Mi Ke)
Date: Mon, 03 Oct 2011 16:22:01 -0400
Subject: [Freeswitch-users] unusual behaviour of mod_com_g729
Message-ID: <20111003202201.309110@gmx.com>

Hi All,

 While having 2 static active calls (all 4 legs are g729 and no new inbound calls were hitting fs) I've noticed the following output in console:

 2011-10-03 20:03:53.346308 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a3e8 0x7f85c8008ee0
 2011-10-03 20:03:53.346308 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a428 0x7f85c8008ee0
 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a408 0x1d3cbe0
 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a408 (nil)
 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a408 0x1d3cbe0
 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a448 0x1d3cbe0
 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a428 (nil)
 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a428 0x7f85c8008ee0
 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a428 0x7f85c8008ee0
 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a468 0x7f85c8008ee0
 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a448 0x1d3cbe0
 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a448 (nil)
 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a448 0x1d3cbe0
 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a488 0x1d3cbe0
 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a468 (nil)
 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a468 0x7f85c8008ee0
 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a468 0x7f85c8008ee0
 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a4a8 0x7f85c8008ee0
 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a488 0x1d3cbe0
 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a488 (nil)
 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a488 0x1d3cbe0
 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a4c8 0x1d3cbe0
 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a4a8 (nil)
 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a4a8 0x7f85c8008ee0
 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a4a8 0x7f85c8008ee0
 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a4e8 0x7f85c8008ee0
 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a4c8 0x1d3cbe0
 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a4c8 (nil)
 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a4c8 0x1d3cbe0
 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a508 0x1d3cbe0
 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a4e8 (nil)
 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a4e8 0x7f85c8008ee0
 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a4e8 0x7f85c8008ee0
 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a528 0x7f85c8008ee0
 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a508 0x1d3cbe0
 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a508 (nil)
 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a508 0x1d3cbe0
 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a548 0x1d3cbe0
 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a528 (nil)
 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a528 0x7f85c8008ee0
 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a528 0x7f85c8008ee0
 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a568 0x7f85c8008ee0
 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a548 0x1d3cbe0
 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a548 (nil)
 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a548 0x1d3cbe0
 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a588 0x1d3cbe0
 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a568 (nil)
 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a568 0x7f85c8008ee0
 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a568 0x7f85c8008ee0
 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a5a8 0x7f85c8008ee0
 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a588 0x1d3cbe0
 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a588 (nil)
 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a588 0x1d3cbe0
 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a5c8 0x1d3cbe0
 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a5a8 (nil)
 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a5a8 0x7f85c8008ee0
 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a5a8 0x7f85c8008ee0
 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a5e8 0x7f85c8008ee0
 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a5c8 0x1d3cbe0
 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a5c8 (nil)
 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a5c8 0x1d3cbe0
 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a608 0x1d3cbe0
 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a5e8 (nil)
 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a5e8 0x7f85c8008ee0
 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0x7f85c864a5e8 0x7f85c8008ee0
 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:148 DECODER CREATE - 0x7f85c864a628 0x7f85c8008ee0
 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0x7f85c864a608 0x1d3cbe0
 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0x7f85c864a608 (nil)
 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0x7f85c864a608 0x1d3cbe0
 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x7f85c864a648 0x1d3cbe0
 ...

 Same continued until one of calls ended (with no errors in log). 

 Is this normal ?

 freeswitch at internal> version
 FreeSWITCH Version 1.0.head (git-7f04f7a 2011-09-28 00-35-06 -0500)

 Cheers / Mike
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From anthony.minessale at gmail.com  Tue Oct  4 02:01:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 Oct 2011 17:01:38 -0500
Subject: [Freeswitch-users] Interaction between event handler and
 streamfile (avmd specifically)
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6954715A@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED6954715A@SERVER2003.SecuReachSystems.local>
Message-ID: <CAKbxfG_3AhYmFiXmQtXbGSmf1Bc5oK0AA2oOQWRd2pK0=et3zQ@mail.gmail.com>

whatever you return from the callback besides null undefined blank
string etc will be returned from streamfile so you want to just return
something like "replay" vs "stop" vs nothing if you want it to keep
playing.  then you can decide from there.


On Mon, Oct 3, 2011 at 9:40 AM, Jason Moran <jmoran at secureachsystems.com> wrote:
> In short, I?m looking to stream a message twice unless a voicemail beeps. If
> the voicemail beeps, then start over immediately and only play the message
> once.
>
>
>
> Pseudo-code:
>
>
>
> session.execute("avmd", "start");
>
> session.streamFile(playbackFileName); //1st play
>
> session.streamFile(playbackFileName); //2nd play
>
> session.execute("hangup","ORIGINATOR_CANCEL");
>
>
>
> function onDTMF(s, type, obj, arg)
>
> {
>
> ? if(type == "event" && session.getVariable("avmd_detect") == "TRUE")
>
> ?{
>
> ? session.execute("avmd", "stop");
>
> ? session.streamFile(playbackFileName);
>
> ? return false;
>
> ?}
>
> ?return true;
> }
>
>
>
> My question is what the effect and timing is of returning false or true from
> the onDTMF function? Does the original streamFile continue playing? Pause?
> end immediately? Do I have to set a variable, return false to stop the
> original from playing, then do my streamfile logic OUTSIDE of the onDTMF for
> proper flow? What happens if I never return true or false, what does it do
> then?
>
>
>
> Thanks,
> Jason
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From pjintheusa at gmail.com  Tue Oct  4 02:21:54 2011
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 3 Oct 2011 18:21:54 -0400
Subject: [Freeswitch-users] no switch_event_get_header in FreeSwitch.dll
Message-ID: <CAGK4cYw7oYYu3ZOXuyMj+hNASsRrG7_jESMOsNYD+dJwR6Q_xw@mail.gmail.com>

Hi there,

On the lastest GIT head, compiling for 64 bit windows - when I try and start
FreeSWITCH I get an exception:

The procedure entry point switch_event_get_header could not be found in the
dyn lib FreeSwitch.dll

FreeSWITCH spears to start OK. And the build seems OK.

Any ideas in what I could be doing wrong?


Phil
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From buscom123+fs at gmail.com  Tue Oct  4 02:29:09 2011
From: buscom123+fs at gmail.com (R H)
Date: Mon, 3 Oct 2011 16:29:09 -0600
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
Message-ID: <CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>

Sure,

I have two soft phones registering as extensions 4001 and 4002 to the same
profile. When 4002 dials 4001 the softphone that is registered as 4001
received two calls simultaneously. This only happens when I enable odbc on
two profiles. If I disable odbc or only enable it for one profile everything
works fine. If I enable ODBC for 3 profiles I would get three invites to
4001, etc...

Attached is some logging output. Notice the parts *highlighted in red.*
*
*
Thank you for your help.

Ryan




On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> could you elaborate more on what the B-legs are?
> Are you doing bridges to an outbound entity, if so how?
>
> do you have any traces doing:
>
> sofia global siptrace on
> and
> console loglevel debug
>
>
>
>
>
> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
> > Hey Everyone,
> > I am preparing to set up HA Failover to my production freeswitch
> > environment. In preparation for this I have been doing some testing in a
> > development environment and I encountered the following problem:
> > I currently have two sip-profiles set up that I am using full time, when
> I
> > enable odbc in the switch.conf.xml everything works correctly. When I
> enable
> > odbc in ONE of the sip-profiles everything works correctly. But when I
> > enable odbc in the second profile I begin to see two b-legs for ever call
> > that I initiate. Both b-legs are directed at the same client.
> > Does anyone have any idea why enabling odbc on two sofia profiles would
> > cause every inbound call to generate two outbound calls? I also decided
> to
> > test this problem by adding two more profiles for a total of 4. With 4
> > profiles I get 4 b-legs. What am I doing wrong?
> > Ryan
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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 brian at freeswitch.org  Tue Oct  4 02:54:20 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Oct 2011 17:54:20 -0500
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
Message-ID: <247F216E-AED0-4BE4-BB56-8250A4DCA11E@freeswitch.org>

select * from sip_registrations;

/b

On Oct 3, 2011, at 5:29 PM, R H wrote:

> Sure,
> 
> I have two soft phones registering as extensions 4001 and 4002 to the same
> profile. When 4002 dials 4001 the softphone that is registered as 4001
> received two calls simultaneously. This only happens when I enable odbc on
> two profiles. If I disable odbc or only enable it for one profile everything
> works fine. If I enable ODBC for 3 profiles I would get three invites to
> 4001, etc...
> 
> Attached is some logging output. Notice the parts *highlighted in red.*
> *
> *
> Thank you for your help.
> 
> Ryan

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From lloyd.aloysius at gmail.com  Tue Oct  4 03:32:42 2011
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Mon, 3 Oct 2011 19:32:42 -0400
Subject: [Freeswitch-users] FreeSWITCH Voicemail - silence detection
In-Reply-To: <CAKzWOxVZLX1JrzRwLdFtyNH35piVqhSasBd+OQz3AnEAwVE7YQ@mail.gmail.com>
References: <CAGvuWYa1VxFFaQWt7e0BQsu5JuaZbugaSzAve+i64S2Zv4Efbg@mail.gmail.com>
	<CAKzWOxVZLX1JrzRwLdFtyNH35piVqhSasBd+OQz3AnEAwVE7YQ@mail.gmail.com>
Message-ID: <CAGvuWYYxzZyLWne+Q+S9zFFxZ_SPM3bi0ngerCmHr2kDHJXAsQ@mail.gmail.com>

Micheal,

Thank you for the references. I try to increase the threshold. So quiet
means lower . I just leave the silence-threshold as 200. Then increase the
hits to 10. That solve my issue.

<param name="record-silence-threshold" value="200"/>
<param name="record-silence-hits" value="10"/>

Thank you for the help

LLoyd


*
*



On Mon, Oct 3, 2011 at 3:26 PM, Michael Collins <msc at freeswitch.org> wrote:

> Yes, you can control this. The threshold value is described here:
> http://wiki.freeswitch.org/wiki/Mod_voicemail#record-silence-threshold
>
> The silence hits is described here:
> http://wiki.freeswitch.org/wiki/Mod_voicemail#record-silence-hits
>
> Both of those link to the docs for the wait_for_silence dialplan tool,
> which discusses these values in more detail:
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence#Description
>
> In a nutshell, if you lower the threshold value then that means the silence
> needs to be "quieter" to be considered "silence." So, for a soft-spoken
> voice you would want to lower the threshold. I would experiment with that.
> Note that if you set the threshold too low then no silence will ever be
> detected. As for the "hits" value, that just tweaks how much silence FS
> needs to detect before it assumes there is silence on the line. The higher
> the number of hits, the longer the silent period needs to be for FS to
> assume it is "silence."
>
> -MC
>
> On Sun, Oct 2, 2011 at 6:01 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:
>
>> Hello,
>>
>> I have the voicemail issue. When someone  soft spoken leaves a message
>> they get interrupted by the silence detection.
>>
>> ?to play this message press 1, to send this message press 2, to delete
>> this message and try again ??
>>
>> I found two options in voicemail.conf. None of them make any difference.
>>
>> <param name="record-silence-threshold" value="200"/>
>> <param name="record-silence-hits" value="2"/>
>>
>> Is there anyway to increase the threshold or time out.
>>
>> FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)
>>
>>
>> Thanks in advance.
>> Lloyd
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Oct  4 03:59:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 Oct 2011 18:59:28 -0500
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
Message-ID: <CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>

are you really using "mydomain.com" on all the profiles at the same time?

On Mon, Oct 3, 2011 at 5:29 PM, R H <buscom123+fs at gmail.com> wrote:
> Sure,
> I have two soft phones registering as extensions 4001 and 4002 to the same
> profile. When 4002 dials 4001 the softphone that is registered as 4001
> received two calls?simultaneously. This only happens when I enable odbc on
> two profiles. If I disable odbc or only enable it for one profile everything
> works fine. If I enable ODBC for 3 profiles I would get three invites to
> 4001, etc...
> Attached is some logging output. Notice the parts highlighted in red.
> Thank you for your help.
> Ryan
>
>
>
> On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> could you elaborate more on what the B-legs are?
>> Are you doing bridges to an outbound entity, if so how?
>>
>> do you have any traces doing:
>>
>> sofia global siptrace on
>> and
>> console loglevel debug
>>
>>
>>
>>
>>
>> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
>> > Hey Everyone,
>> > I am preparing to set up HA Failover to my production freeswitch
>> > environment. In preparation for this I have been doing some testing in a
>> > development environment and I encountered the following problem:
>> > I currently have two sip-profiles set up that I am using full time, when
>> > I
>> > enable odbc in the switch.conf.xml everything works correctly. When I
>> > enable
>> > odbc in ONE of the sip-profiles everything works correctly. But when I
>> > enable odbc in the second profile I begin to see two b-legs for ever
>> > call
>> > that I initiate. Both b-legs are directed at the same client.
>> > Does anyone have any idea why enabling odbc on two sofia profiles would
>> > cause every inbound call to generate two outbound calls? I also decided
>> > to
>> > test this problem by adding two more profiles for a total of 4. With 4
>> > profiles I get 4 b-legs. What am I doing wrong?
>> > Ryan
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Oct  4 04:00:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 Oct 2011 19:00:50 -0500
Subject: [Freeswitch-users] unusual behaviour of mod_com_g729
In-Reply-To: <20111003202201.309110@gmx.com>
References: <20111003202201.309110@gmx.com>
Message-ID: <CAKbxfG8ZYwm5efb+FjuW+qs_3tpLgPc9EmyA0YRMaL9M78UZ-A@mail.gmail.com>

taken out of context like that it would be hard to tell you should enter
console loglevel debug and include more of the log


On Mon, Oct 3, 2011 at 3:22 PM, Mi Ke <mi.ke at null.net> wrote:
> Hi All,
>
> While having 2 static active calls (all 4 legs are g729 and no new inbound
> calls were hitting fs) I've noticed the following output in console:
>
> 2011-10-03 20:03:53.346308 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a3e8 0x7f85c8008ee0
> 2011-10-03 20:03:53.346308 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a428 0x7f85c8008ee0
> 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a408 0x1d3cbe0
> 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a408 (nil)
> 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a408 0x1d3cbe0
> 2011-10-03 20:03:53.646311 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a448 0x1d3cbe0
> 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a428 (nil)
> 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a428 0x7f85c8008ee0
> 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a428 0x7f85c8008ee0
> 2011-10-03 20:03:53.906304 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a468 0x7f85c8008ee0
> 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a448 0x1d3cbe0
> 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a448 (nil)
> 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a448 0x1d3cbe0
> 2011-10-03 20:03:55.986327 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a488 0x1d3cbe0
> 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a468 (nil)
> 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a468 0x7f85c8008ee0
> 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a468 0x7f85c8008ee0
> 2011-10-03 20:03:57.086308 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a4a8 0x7f85c8008ee0
> 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a488 0x1d3cbe0
> 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a488 (nil)
> 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a488 0x1d3cbe0
> 2011-10-03 20:03:59.106306 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a4c8 0x1d3cbe0
> 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a4a8 (nil)
> 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a4a8 0x7f85c8008ee0
> 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a4a8 0x7f85c8008ee0
> 2011-10-03 20:04:00.146308 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a4e8 0x7f85c8008ee0
> 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a4c8 0x1d3cbe0
> 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a4c8 (nil)
> 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a4c8 0x1d3cbe0
> 2011-10-03 20:04:00.466304 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a508 0x1d3cbe0
> 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a4e8 (nil)
> 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a4e8 0x7f85c8008ee0
> 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a4e8 0x7f85c8008ee0
> 2011-10-03 20:04:01.206309 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a528 0x7f85c8008ee0
> 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a508 0x1d3cbe0
> 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a508 (nil)
> 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a508 0x1d3cbe0
> 2011-10-03 20:04:03.366307 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a548 0x1d3cbe0
> 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a528 (nil)
> 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a528 0x7f85c8008ee0
> 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a528 0x7f85c8008ee0
> 2011-10-03 20:04:04.046310 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a568 0x7f85c8008ee0
> 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a548 0x1d3cbe0
> 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a548 (nil)
> 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a548 0x1d3cbe0
> 2011-10-03 20:04:05.046309 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a588 0x1d3cbe0
> 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a568 (nil)
> 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a568 0x7f85c8008ee0
> 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a568 0x7f85c8008ee0
> 2011-10-03 20:04:05.406309 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a5a8 0x7f85c8008ee0
> 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a588 0x1d3cbe0
> 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a588 (nil)
> 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a588 0x1d3cbe0
> 2011-10-03 20:04:05.426308 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a5c8 0x1d3cbe0
> 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a5a8 (nil)
> 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a5a8 0x7f85c8008ee0
> 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a5a8 0x7f85c8008ee0
> 2011-10-03 20:04:05.746321 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a5e8 0x7f85c8008ee0
> 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a5c8 0x1d3cbe0
> 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a5c8 (nil)
> 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a5c8 0x1d3cbe0
> 2011-10-03 20:04:06.766307 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a608 0x1d3cbe0
> 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a5e8 (nil)
> 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a5e8 0x7f85c8008ee0
> 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:89 DECODER DESTROY -
> 0x7f85c864a5e8 0x7f85c8008ee0
> 2011-10-03 20:04:09.326303 [INFO] mod_com_g729.c:148 DECODER CREATE -
> 0x7f85c864a628 0x7f85c8008ee0
> 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x7f85c864a608 0x1d3cbe0
> 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x7f85c864a608 (nil)
> 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x7f85c864a608 0x1d3cbe0
> 2011-10-03 20:04:09.426307 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x7f85c864a648 0x1d3cbe0
> ...
>
> Same continued until one of calls ended (with no errors in log).
>
> Is this normal ?
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-7f04f7a 2011-09-28 00-35-06 -0500)
>
> Cheers / Mike
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From beffa at ieee.org  Tue Oct  4 11:25:08 2011
From: beffa at ieee.org (Federico Beffa)
Date: Tue, 4 Oct 2011 09:25:08 +0200
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <1317308654166-6844212.post@n2.nabble.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
	<1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
	<CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>
	<1317308654166-6844212.post@n2.nabble.com>
Message-ID: <CAKrPhPMA3cngUy9hkNyucbzOv=GwAjN6qWUQNFJ=K4+koEcAEA@mail.gmail.com>

Just in case someone will hit the same problem...

After some tests, I've found that, changing "bind_server_ip" from
"auto" to the local IP address of the FS server (192.168.X.X), appears
to solve my timeout problem.  (The account did not go offline for more
than 12 hours.)

To summarize my current configuration: I have a fix public IP assigned
to my router (NAT/Firewall). NAT is configured to forward ports 5080
and 5081 to the host on which FS is running and UPnP is disabled.

I'm running a FS version checked out from git the 20th of August.

Regards,
Fede

On Thu, Sep 29, 2011 at 5:04 PM, mazilo
<Nabble_01394 at slickdeals.endjunk.com> wrote:
> The only thing I can say is to update your FS with the latest git commit.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-offline-mod-dingaling-tp6839355p6844212.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jeff at jefflenk.com  Tue Oct  4 17:57:43 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 4 Oct 2011 06:57:43 -0700 (PDT)
Subject: [Freeswitch-users] no switch_event_get_header in FreeSwitch.dll
In-Reply-To: <CAGK4cYw7oYYu3ZOXuyMj+hNASsRrG7_jESMOsNYD+dJwR6Q_xw@mail.gmail.com>
References: <CAGK4cYw7oYYu3ZOXuyMj+hNASsRrG7_jESMOsNYD+dJwR6Q_xw@mail.gmail.com>
Message-ID: <1317736663240-6858734.post@n2.nabble.com>

VS2008 or VS2010? This is most likely caused by a SWIG problem.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-switch-event-get-header-in-FreeSwitch-dll-tp6856879p6858734.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Stefan.Weigel at allianz-warranty.com  Tue Oct  4 17:58:32 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Tue, 4 Oct 2011 15:58:32 +0200
Subject: [Freeswitch-users] Desktop phone - answer a ringing call via API
	call
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36E5F@AZWSMS03.azwarranty.int>

Hi all,



Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP 560) via api command ?

I tried with SendMsg <uuid> and 'answer' application, but it's not want I need/want.





Thanks and best regards,



Stefan

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From jeff at jefflenk.com  Tue Oct  4 17:59:19 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 4 Oct 2011 06:59:19 -0700 (PDT)
Subject: [Freeswitch-users] My new web utility - Windows 7 gadget
In-Reply-To: <CAF=swh---deRqRKCyJGxhPgAd9bdTvXhzqA1qwRFFExmCr9SNA@mail.gmail.com>
References: <CAF=swh---deRqRKCyJGxhPgAd9bdTvXhzqA1qwRFFExmCr9SNA@mail.gmail.com>
Message-ID: <1317736759665-6858743.post@n2.nabble.com>

Cool thanks for posting your sample!

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/My-new-web-utility-Windows-7-gadget-tp6853946p6858743.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Tue Oct  4 18:19:17 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 4 Oct 2011 16:19:17 +0200
Subject: [Freeswitch-users] My new web utility - Windows 7 gadget
In-Reply-To: <1317736759665-6858743.post@n2.nabble.com>
References: <CAF=swh---deRqRKCyJGxhPgAd9bdTvXhzqA1qwRFFExmCr9SNA@mail.gmail.com>
	<1317736759665-6858743.post@n2.nabble.com>
Message-ID: <CALXCt0rJaQ4oxJgTMyZKf165La5MjzAwBUH0DJ2nTbdQX3oSkQ@mail.gmail.com>

kewl!

-giovanni

On Tue, Oct 4, 2011 at 3:59 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
> Cool thanks for posting your sample!
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/My-new-web-utility-Windows-7-gadget-tp6853946p6858743.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From wayne at hamilton.net  Tue Oct  4 18:27:03 2011
From: wayne at hamilton.net (Wayne)
Date: Tue, 4 Oct 2011 09:27:03 -0500
Subject: [Freeswitch-users] Sending DTMF on B-leg WORKING
In-Reply-To: <B5489AF9DCC040E2AAA414038FFBB657@ccs.local>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local><CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com><9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local><CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com><F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local><CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com><C57A15E034154A9297EE8DF5EF752971@ccs.local><CAKzWOxXJ7J+Pi2iart1nHC8-OmwHkZ0vjH3iXsKvAKGdNOt7=Q@mail.gmail.com>
	<B5489AF9DCC040E2AAA414038FFBB657@ccs.local>
Message-ID: <99817380EE8745A5B22E84058810B705@ccs.local>

Just to answer this posts for others that are searching for the answer this
is what I did to make it work. 
For some reason after resetting my Sangoma card  with a power down and
restart of the server send_dtmf started working.
Here is  the XML that worked for me. Seems that the sleep and flush also got
rid of my random tones.
 
<extension name="test-tone" >
   <condition field="destination_number" expression="1(.*)" >
        <action application="set"
data="effective_caller_id_number=9025556747"/>
        <action application="sleep" data="500"/>
        <action application="flush_dtmf"/>
        <action application="export"
data="nolocal:execute_on_answer=send_dtmf W$1#@500" />
       <action application="bridge" data="freetdm/wp1/a/18005551234"/>
   </condition>
</extension>
 
Thanks to all that helped.
Wayne
 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wayne
Sent: Thursday, September 15, 2011 11:55 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


Okay,
 
I have tried send_dtmf and queue_dtmf and I can not get any DTMF to pass.
Would anyone be so kind and send me an extension that has it working on the
B-leg. I am getting nowhere.
 
Thanks Wayne 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, September 13, 2011 4:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


Okay, fair enough. Not sure what queue_dtmf is not working. You could always
try "execute_on_answer" and execute an extension that sends the dtmfs.  

-MC


On Tue, Sep 13, 2011 at 2:16 PM, Wayne <wayne at hamilton.net> wrote:



Sorry Michael,
 
When I started out I was sending the calls to a SIP provider. I couldn't
here the DTMF tones. For trouble shooting I moved it to a local PRI. I was
thinking that maybe this was an issue with the sip trunk and sending inband
or out of band but that doesn't seem to be the case.
 
Wayne

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, September 13, 2011 2:41 PM 

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg




On Tue, Sep 13, 2011 at 11:34 AM, Wayne <wayne at hamilton.net> wrote:



That's true I don't hear any tones when I answer the phone. I don't see any
messages in my log from queue_dtmf. Any help would be great.
 
Wayne


Okay, in your original post you said you are making an OB call on a SIP
trunk, yet in your example there is no SIP trunk. You've got an internal
user and a FreeTDM FXO port. Can you please clarify what you actually have?
Who's the caller and who's the callee?
-MC

 


 
 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, September 08, 2011 6:25 PM 

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


Are you saying that no DTMFs are being sent out when the bridge is initially
completed?  
-MC


On Wed, Sep 7, 2011 at 3:17 PM, Wayne <wayne at hamilton.net> wrote:



I have tried that 
<extension name="test-tone" >
   <condition field="destination_number" expression="(9025556747
<tel:%289025556747> )" >
       <action application="set" data="call_direction=outbound" />
       <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}" />
       <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}" />
       <action application="set" data="continue_on_fail=true" />
       <action application="queue_dtmf" data="0123456789" />
       <action application="bridge" data="freetdm/wp1/a/$1" />
   </condition>
</extension>
 
The debug tells me that this
tail -f freeswitch.log | grep -i dtmf
 
Dialplan:  <mailto:sofia/internal/1333 at 192.168.48.87>
sofia/internal/1333 at 192.168.48.87 Action queue_dtmf(0123456789)
EXECUTE  <mailto:sofia/internal/1333 at 192.168.48.87>
sofia/internal/1333 at 192.168.48.87 queue_dtmf(0123456789)
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3355 Set 2833 dtmf send
payload to 101
2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3360 Set 2833 dtmf receive
payload to 101
2011-09-07 17:13:45.116216 [DEBUG] ftdm_io.c:3714 [s1c1][1:1] Generating
DTMF [0123456789]
2011-09-07 17:13:47.876238 [DEBUG] ftmod_wanpipe.c:701 [s1c1][1:1] Enabled
DTMF events
2011-09-07 17:13:47.956228 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing
wanpipe DTMF: 9
2011-09-07 17:13:47.956228 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing DTMF 9
(debug = 0)
2011-09-07 17:13:47.956228 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
channel FreeTDM/1:1/9025556747 <tel:1%2F9025556747> 
2011-09-07 17:13:48.036235 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing
wanpipe DTMF: 9
2011-09-07 17:13:48.036235 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing DTMF 9
(debug = 0)
2011-09-07 17:13:48.036235 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
channel FreeTDM/1:1/9025556747 <tel:1%2F9025556747> 
 
So where would I look next.
 
Wayne


  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Wednesday, September 07, 2011 3:55 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


I think you looking for queue_dtmf.

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

-Avi Marcus



On Wed, Sep 7, 2011 at 11:47 PM, Wayne <wayne at hamilton.net> wrote:
> Hello All,
>
> I need to call out on a SIP trunk and when the call is answered send DTMF
> tones.
> I need to send the DTMF only on the outbound leg.
>
> Does anyone have a dialplan that will do that? Is it possible. I have only
> found one thread on it and did get much out of it.
>
> Thanks
> Wayne
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




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






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






FreeSWITCH-users mailing 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 henrikaagaardsorensen at gmail.com  Tue Oct  4 19:07:12 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Tue, 4 Oct 2011 17:07:12 +0200
Subject: [Freeswitch-users] Kamailio and FreeSwitch. Too many hops problem.
Message-ID: <CAGH8SeY5NRpNk+TDN3HPmf7J3u6VHtQerortqygcu4p7vCHEQA@mail.gmail.com>

I'm facing a problem with my current setup.

Currently I have a Kamailio server as dispatcher. It doesn't do anything to
the SIP packets except forwarding them.

FreeSwitch is receiving SIP packages from Kamailio, as Kamailio as setup as
a trunk to FreeSwitch.

I have some devices with attached numbers, they are external numbers.

When I call in to the voicemail, to hear hold music etc. everything works.
But when I'm trying to call other devices I get the error "Too Many Hops".

Can anyone help me figure out the problem?
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From buscom123+fs at gmail.com  Tue Oct  4 20:24:38 2011
From: buscom123+fs at gmail.com (R H)
Date: Tue, 4 Oct 2011 10:24:38 -0600
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
	<CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>
Message-ID: <CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>

In Response to Brian West, below are my sip registrations. These may have
slightly different values from the logs I sent yesterday simply because I
shut down last night and re-registered this morning. Same softphones, same
extensions, just new registrations:

-----------------------------------------------------------------------------
*select * from sip_registrations;*

MTAwYWEwYzhlNzc2MDAxYTUzNDhlY2EwZmEyZDFlZDE.,4001,mydomain.com,mydomain.com,"""user""
<sip:4001 at 192.168.99.164:61238;rinstance=e2d50bbe31452d30>",Registered(UDP),unknown,1317746818,X-Lite
4 release 4.1 stamp
63215,4001,192.168.122.236,internal,psiFreeswitch,192.168.99.164,61238,4001,
mydomain.com,4001,mydomain.com,192.168.122.236,psiFreeswitch

dQJifXuoqNBUlWCuZ8rCI6cu.Nn5F81W,4002,mydomain.com,mydomain.com,"""Test
User123"" <sip:cqgulpey at 192.168.99.164:57003>",Registered(UDP),unknown,1317743825,Blink
Lite 1.2.0
(MacOSX),4002,192.168.122.236,internal,psiFreeswitch,192.168.99.164,57003,4002,
mydomain.com,4002,mydomain.com,192.168.122.236,psiFreeswitch

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

In Response to Anthony Minessale,

"Mydomain.com" is simply for testing and will not be our go-live domain. The
particular switch I am testing on is located in a closed internal network
designed specifically for testing systems like this and nothing more. It is
set up using the instructions on the FreeSwitch wiki for Multi-Tenant
installations because we have the potential to go multi-tenant in the future
and we want to make sure multi-tenant setups work for the system we are
building.

As far as my profiles in development i have the following setup:

1) Internal - To emulate production this is limited use, no real phones will
ever connect from the local LAN in production. We have been testing
registrations on this profile anyway but not there is no plan to have phones
register here in production:

<domain name="all" alias="false" parse="false"/>



2) External - Most phones will register here, possibly to multiple domains
(tenants). These devices will be on other networks with possible natting
issues. This profile requires authentication to make calls.

3) External_Debug - When clients complain of connectivity issues we will be
asking them to move to this profile so that we can more easily monitor their
sip traffic and debug their issues. This profile will also require
authentication and allow registration to the many domains.

4) Unnamed Future External Profile - We will also most likely have another
profile for our upstream providers once we are ready to turn them on.






On Mon, Oct 3, 2011 at 5:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> are you really using "mydomain.com" on all the profiles at the same time?
>
> On Mon, Oct 3, 2011 at 5:29 PM, R H <buscom123+fs at gmail.com> wrote:
> > Sure,
> > I have two soft phones registering as extensions 4001 and 4002 to the
> same
> > profile. When 4002 dials 4001 the softphone that is registered as 4001
> > received two calls simultaneously. This only happens when I enable odbc
> on
> > two profiles. If I disable odbc or only enable it for one profile
> everything
> > works fine. If I enable ODBC for 3 profiles I would get three invites to
> > 4001, etc...
> > Attached is some logging output. Notice the parts highlighted in red.
> > Thank you for your help.
> > Ryan
> >
> >
> >
> > On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> could you elaborate more on what the B-legs are?
> >> Are you doing bridges to an outbound entity, if so how?
> >>
> >> do you have any traces doing:
> >>
> >> sofia global siptrace on
> >> and
> >> console loglevel debug
> >>
> >>
> >>
> >>
> >>
> >> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
> >> > Hey Everyone,
> >> > I am preparing to set up HA Failover to my production freeswitch
> >> > environment. In preparation for this I have been doing some testing in
> a
> >> > development environment and I encountered the following problem:
> >> > I currently have two sip-profiles set up that I am using full time,
> when
> >> > I
> >> > enable odbc in the switch.conf.xml everything works correctly. When I
> >> > enable
> >> > odbc in ONE of the sip-profiles everything works correctly. But when I
> >> > enable odbc in the second profile I begin to see two b-legs for ever
> >> > call
> >> > that I initiate. Both b-legs are directed at the same client.
> >> > Does anyone have any idea why enabling odbc on two sofia profiles
> would
> >> > cause every inbound call to generate two outbound calls? I also
> decided
> >> > to
> >> > test this problem by adding two more profiles for a total of 4. With 4
> >> > profiles I get 4 b-legs. What am I doing wrong?
> >> > Ryan
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From buscom123+fs at gmail.com  Tue Oct  4 20:26:12 2011
From: buscom123+fs at gmail.com (R H)
Date: Tue, 4 Oct 2011 10:26:12 -0600
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
	<CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>
	<CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>
Message-ID: <CAAtQP55kS=fA15=h3b7_pRLkxoS4u-XUneQcBFQH=udtioe1uA@mail.gmail.com>

Sometimes I hate Gmail hotkeys, I meant to finish the description of my
profiles but it sent before I was done:

External

<domain name="all" alias="false" parse="true"/>

External_Debug

<domain name="all" alias="false" parse="true"/>

Thank you guys again for your help. I'm not terribly familiar with C yet so
I am struggling to debug this on my own.

Ryan


On Tue, Oct 4, 2011 at 10:24 AM, R H <buscom123+fs at gmail.com> wrote:

> In Response to Brian West, below are my sip registrations. These may have
> slightly different values from the logs I sent yesterday simply because I
> shut down last night and re-registered this morning. Same softphones, same
> extensions, just new registrations:
>
>
> -----------------------------------------------------------------------------
> *select * from sip_registrations;*
>
> MTAwYWEwYzhlNzc2MDAxYTUzNDhlY2EwZmEyZDFlZDE.,4001,mydomain.com,
> mydomain.com,"""user"" <sip:4001 at 192.168.99.164:61238;rinstance=e2d50bbe31452d30>",Registered(UDP),unknown,1317746818,X-Lite
> 4 release 4.1 stamp
> 63215,4001,192.168.122.236,internal,psiFreeswitch,192.168.99.164,61238,4001,
> mydomain.com,4001,mydomain.com,192.168.122.236,psiFreeswitch
>
> dQJifXuoqNBUlWCuZ8rCI6cu.Nn5F81W,4002,mydomain.com,mydomain.com,"""Test
> User123"" <sip:cqgulpey at 192.168.99.164:57003>",Registered(UDP),unknown,1317743825,Blink
> Lite 1.2.0
> (MacOSX),4002,192.168.122.236,internal,psiFreeswitch,192.168.99.164,57003,4002,
> mydomain.com,4002,mydomain.com,192.168.122.236,psiFreeswitch
>
>
> -----------------------------------------------------------------------------
>
> In Response to Anthony Minessale,
>
> "Mydomain.com" is simply for testing and will not be our go-live domain.
> The particular switch I am testing on is located in a closed internal
> network designed specifically for testing systems like this and nothing
> more. It is set up using the instructions on the FreeSwitch wiki for
> Multi-Tenant installations because we have the potential to go multi-tenant
> in the future and we want to make sure multi-tenant setups work for the
> system we are building.
>
> As far as my profiles in development i have the following setup:
>
> 1) Internal - To emulate production this is limited use, no real phones
> will ever connect from the local LAN in production. We have been testing
> registrations on this profile anyway but not there is no plan to have phones
> register here in production:
>
> <domain name="all" alias="false" parse="false"/>
>
>
>
> 2) External - Most phones will register here, possibly to multiple domains
> (tenants). These devices will be on other networks with possible natting
> issues. This profile requires authentication to make calls.
>
> 3) External_Debug - When clients complain of connectivity issues we will be
> asking them to move to this profile so that we can more easily monitor their
> sip traffic and debug their issues. This profile will also require
> authentication and allow registration to the many domains.
>
> 4) Unnamed Future External Profile - We will also most likely have another
> profile for our upstream providers once we are ready to turn them on.
>
>
>
>
>
>
> On Mon, Oct 3, 2011 at 5:59 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> are you really using "mydomain.com" on all the profiles at the same time?
>>
>> On Mon, Oct 3, 2011 at 5:29 PM, R H <buscom123+fs at gmail.com> wrote:
>> > Sure,
>> > I have two soft phones registering as extensions 4001 and 4002 to the
>> same
>> > profile. When 4002 dials 4001 the softphone that is registered as 4001
>> > received two calls simultaneously. This only happens when I enable odbc
>> on
>> > two profiles. If I disable odbc or only enable it for one profile
>> everything
>> > works fine. If I enable ODBC for 3 profiles I would get three invites to
>> > 4001, etc...
>> > Attached is some logging output. Notice the parts highlighted in red.
>> > Thank you for your help.
>> > Ryan
>> >
>> >
>> >
>> > On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> could you elaborate more on what the B-legs are?
>> >> Are you doing bridges to an outbound entity, if so how?
>> >>
>> >> do you have any traces doing:
>> >>
>> >> sofia global siptrace on
>> >> and
>> >> console loglevel debug
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
>> >> > Hey Everyone,
>> >> > I am preparing to set up HA Failover to my production freeswitch
>> >> > environment. In preparation for this I have been doing some testing
>> in a
>> >> > development environment and I encountered the following problem:
>> >> > I currently have two sip-profiles set up that I am using full time,
>> when
>> >> > I
>> >> > enable odbc in the switch.conf.xml everything works correctly. When I
>> >> > enable
>> >> > odbc in ONE of the sip-profiles everything works correctly. But when
>> I
>> >> > enable odbc in the second profile I begin to see two b-legs for ever
>> >> > call
>> >> > that I initiate. Both b-legs are directed at the same client.
>> >> > Does anyone have any idea why enabling odbc on two sofia profiles
>> would
>> >> > cause every inbound call to generate two outbound calls? I also
>> decided
>> >> > to
>> >> > test this problem by adding two more profiles for a total of 4. With
>> 4
>> >> > profiles I get 4 b-legs. What am I doing wrong?
>> >> > Ryan
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From anthony.minessale at gmail.com  Tue Oct  4 20:27:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 Oct 2011 11:27:23 -0500
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
	<CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>
	<CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>
Message-ID: <CAKbxfG9Msun7_rdMEoSHk-rtpZ_bVJmdUoHWxFBDHDC5U2EaFg@mail.gmail.com>

ok but the db is shared so if they are all using mydomain.com it will
cross over so you should use distinct fake domains at least between
your profiles.


On Tue, Oct 4, 2011 at 11:24 AM, R H <buscom123+fs at gmail.com> wrote:
> In Response to Brian West, below are my sip registrations. These may have
> slightly different values from the logs I sent yesterday simply because I
> shut down last night and re-registered this morning. Same softphones, same
> extensions, just new registrations:
> -----------------------------------------------------------------------------
> select * from sip_registrations;
> MTAwYWEwYzhlNzc2MDAxYTUzNDhlY2EwZmEyZDFlZDE.,4001,mydomain.com,mydomain.com,"""user""
> <sip:4001 at 192.168.99.164:61238;rinstance=e2d50bbe31452d30>",Registered(UDP),unknown,1317746818,X-Lite
> 4 release 4.1 stamp
> 63215,4001,192.168.122.236,internal,psiFreeswitch,192.168.99.164,61238,4001,mydomain.com,4001,mydomain.com,192.168.122.236,psiFreeswitch
> dQJifXuoqNBUlWCuZ8rCI6cu.Nn5F81W,4002,mydomain.com,mydomain.com,"""Test
> User123""
> <sip:cqgulpey at 192.168.99.164:57003>",Registered(UDP),unknown,1317743825,Blink
> Lite 1.2.0
> (MacOSX),4002,192.168.122.236,internal,psiFreeswitch,192.168.99.164,57003,4002,mydomain.com,4002,mydomain.com,192.168.122.236,psiFreeswitch
> -----------------------------------------------------------------------------
> In Response to Anthony Minessale,
> "Mydomain.com" is simply for testing and will not be our go-live domain. The
> particular switch I am testing on is located in a closed internal network
> designed specifically for testing systems like this and nothing more. It is
> set up using the instructions on the FreeSwitch wiki for Multi-Tenant
> installations because we have the potential to go multi-tenant in the future
> and we want to make sure multi-tenant setups work for the system we are
> building.
> As far as my profiles in development i have the following setup:
> 1) Internal - To emulate production this is limited use, no real phones will
> ever connect from the local LAN in production. We have been testing
> registrations on this profile anyway but not there is no plan to have phones
> register here in production:
> <domain name="all" alias="false" parse="false"/>
>
>
> 2) External - Most phones will register here, possibly to multiple domains
> (tenants). These devices will be on other networks with possible natting
> issues. This profile requires authentication to make calls.
> 3) External_Debug - When clients complain of connectivity issues we will be
> asking them to move to this profile so that we can more easily monitor their
> sip traffic and debug their issues. This profile will also require
> authentication and allow registration to the many domains.
> 4) Unnamed Future External Profile - We will also most likely have another
> profile for our upstream providers once we are ready to turn them on.
>
>
>
>
>
> On Mon, Oct 3, 2011 at 5:59 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> are you really using "mydomain.com" on all the profiles at the same time?
>>
>> On Mon, Oct 3, 2011 at 5:29 PM, R H <buscom123+fs at gmail.com> wrote:
>> > Sure,
>> > I have two soft phones registering as extensions 4001 and 4002 to the
>> > same
>> > profile. When 4002 dials 4001 the softphone that is registered as 4001
>> > received two calls?simultaneously. This only happens when I enable odbc
>> > on
>> > two profiles. If I disable odbc or only enable it for one profile
>> > everything
>> > works fine. If I enable ODBC for 3 profiles I would get three invites to
>> > 4001, etc...
>> > Attached is some logging output. Notice the parts highlighted in red.
>> > Thank you for your help.
>> > Ryan
>> >
>> >
>> >
>> > On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> could you elaborate more on what the B-legs are?
>> >> Are you doing bridges to an outbound entity, if so how?
>> >>
>> >> do you have any traces doing:
>> >>
>> >> sofia global siptrace on
>> >> and
>> >> console loglevel debug
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
>> >> > Hey Everyone,
>> >> > I am preparing to set up HA Failover to my production freeswitch
>> >> > environment. In preparation for this I have been doing some testing
>> >> > in a
>> >> > development environment and I encountered the following problem:
>> >> > I currently have two sip-profiles set up that I am using full time,
>> >> > when
>> >> > I
>> >> > enable odbc in the switch.conf.xml everything works correctly. When I
>> >> > enable
>> >> > odbc in ONE of the sip-profiles everything works correctly. But when
>> >> > I
>> >> > enable odbc in the second profile I begin to see two b-legs for ever
>> >> > call
>> >> > that I initiate. Both b-legs are directed at the same client.
>> >> > Does anyone have any idea why enabling odbc on two sofia profiles
>> >> > would
>> >> > cause every inbound call to generate two outbound calls? I also
>> >> > decided
>> >> > to
>> >> > test this problem by adding two more profiles for a total of 4. With
>> >> > 4
>> >> > profiles I get 4 b-legs. What am I doing wrong?
>> >> > Ryan
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From michaelt at callall.co.za  Tue Oct  4 14:51:39 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Tue, 4 Oct 2011 12:51:39 +0200
Subject: [Freeswitch-users] hangup error missing remote port
In-Reply-To: <CAK=jKZJ3cvdya8i-2A-ygRx12VK3d8ZSbLkxLDxPsA3uUxj2gg@mail.gmail.com>
References: <CAK=jKZJ3cvdya8i-2A-ygRx12VK3d8ZSbLkxLDxPsA3uUxj2gg@mail.gmail.com>
Message-ID: <CAFmdODgsp9z5FOtsXiHoE-g5vxpjhrKH_fEZBs=fwfDDSbrbkQ@mail.gmail.com>

Hi Allan,

 I am having the same problem with back to back FS boxes, I am also getting
port '0' sent in the m= SDP, in my case it is sent in the 183 SDP.

 Did you get to the bottom of the issue?

Thanks,

Michael

On Sun, Sep 4, 2011 at 6:44 AM, Allan Piske <zyryznet at gmail.com> wrote:

> Hi,
>
> I'm having some trouble discovering the source and solution to one problem.
>
> I setup a FS box for receiving faxes, and a already have another FS box as
> SBC.
>
>
> SPA3102   ---> opensips ---> FS_SBC --> opensips --> FS_FAX
> this works ok, call gets answered, i hear the fax tone, the SPA3102 changes
> to T38 and everyone is happy.
>
>
> Nortel CS2000   ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc)
> ---> FS_FAX
> but his doen't.
>
> This little devil CS2000 sends every invite with m=image header along with
> m=audio. With aparently FS doesn't like.
>
>
> like that:
>
> freeswitch at internal> recv 1215 bytes from udp/[10.143.82.250]:5060 at
> 08:26:08.226858:
>    ------------------------------------------------------------------------
>    INVITE sip:4730365302 at 10.143.82.253:5060;transport=UDP;user=phone
> SIP/2.0
>    Record-Route:
> <sip:10.143.82.250;lr=on;ftag=-45026-10d9cdb-2668a317-10d9cdb>
>    f: <sip:4784064435 at 10.150.65.16:5060
> ;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb
>    t: <sip:4730365302 at 10.143.82.250:5060;user=phone>
>    i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481
>    CSeq: 1 INVITE
>    User-agent: CS2000_NGSS/9.0
>    P-Asserted-Identity: <sip:4784064435 at 10.150.65.16;user=phone>
>    Allow:
> ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE
>    Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0
>    v: SIP/2.0/UDP
> PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d
>    Max-Forwards: 69
>    m: <sip:10.150.65.16:5060;transport=UDP>
>    k: 100rel
>    c: application/sdp
>    l: 420
>
>    v=0
>    o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202
>    s=-
>    p=+1 6135555555
>    c=IN IP4 10.152.204.202
>    t=0 0
>    m=audio 55920 RTP/AVP 18 8 101
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:20
>    a=fmtp:18 annexb=no
>    m=image 64112 udptl t38
>    a=T38FaxVersion:0
>    a=T38FaxMaxBuffer:1100
>    a=T38FaxMaxDatagram:612
>    a=T38MaxBitRate:14400
>    a=T38FaxRateManagement:transferredTCF
>    a=T38FaxUdpEC:t38UDPRedundancy
>
> But then we have a problem ...
>
>
>
>
> My beloved FS_SBC changes the SDP before sending it to the next hop .. as
> it should becase they are on diferent subnets and set to proxy_media
>
> 2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302
> Patched SDP
> ---
> v=0
> o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202
> s=-
> p=+1 6135555555
> c=IN IP4 10.152.204.202
> t=0 0
> m=audio 55920 RTP/AVP 18 8 101
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> m=image 64112 udptl t38
> a=T38FaxVersion:0
> a=T38FaxMaxBuffer:1100
> a=T38FaxMaxDatagram:612
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxUdpEC:t38UDPRedundancy
>
> +++
> v=0
> o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D
> s=FreeSWITCH
> p=+1 6135555555
> c=IN IP4 A.B.C.D
> t=0 0
> m=audio 19262 RTP/AVP 18 8 101
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> m=image 19262 udptl t38
> a=T38FaxVersion:0
> a=T38FaxMaxBuffer:1100
> a=T38FaxMaxDatagram:612
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxUdpEC:t38UDPRedundancy
>
>
> so we have now  2  m=   fields with the same port !!!!!
>
> then when it gets answered on the FS_FAX box .. we got disconnection on the
> FS_SBC box
>
>
> 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel
> sofia/tpa/4730365302 entering state [completing][200]
> 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP:
> v=0
> o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19
> s=FreeSWITCH
> c=IN IP4 189.45.192.19
> t=0 0
> m=audio 31436 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> m=image 0 udptl 19
>
> 2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel
> sofia/tpa/4730365302 entering state [ready][200]
> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830
> (sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE
> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal
> sofia/voxip/4784064435 at 10.150.65.16:5060 [BREAK]
> 2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel
> [sofia/tpa/4730365302] has been answered
> 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec
> sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits
> 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP
> [sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20
> 2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS ERROR:
> [Missing remote port]
> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563
> (sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP
> 2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup
> sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal
> sofia/tpa/4730365302 [KILL]
> 2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/tpa/4730365302 [BREAK]
>  2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate
> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed.
>  Cause: DESTINATION_OUT_OF_ORDER
>
>
> port = 0 !!!
>
> is that maybe a fault in opensips in the middle because the both m= fields
> with the same port ? , or one of the FS boxes ?
> something is broken here, sadly :(
>
> Appreciate any help.
>
> Allan
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From masood24 at gmail.com  Tue Oct  4 17:03:07 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Tue, 4 Oct 2011 16:03:07 +0300
Subject: [Freeswitch-users] Installing GIT for CentOS the link is down or
 give me directo link of latest version.
Message-ID: <CAMd3iJK2ZA12REJniWU7rOoVvSdmeyDeNLbom7umb_ae3+s19A@mail.gmail.com>

Dear All,

I am new to FreeSwitch and I have installed the old version 1.0.6 on
CentOS-5.5 32 Bit
>From wiki I read that to install the latest version of freeswitch should use
GIT.
Now to install GIT I found following link :
http://kernel.org/pub/software/scm/git/git-1.7.6.tar.bz2
but the link is down since last two weeks, I am using* CentOS 5.5 32 Bit
machine.*
Can any one tell me where can i find the GIT OR most importantly if some one
knows the direct link to download FreeSwitch Latest Version.
Like for older version this is the direct link :
http://files.freeswitch.org/freeswitch-1.0.6.tar.gz

Will appreciate your Help.

Thanks

-- 
*Regards,
Masood Ahmed
*
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From lloyd.aloysius at gmail.com  Tue Oct  4 20:51:34 2011
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Tue, 4 Oct 2011 12:51:34 -0400
Subject: [Freeswitch-users] Installing GIT for CentOS the link is down
 or give me directo link of latest version.
In-Reply-To: <CAMd3iJK2ZA12REJniWU7rOoVvSdmeyDeNLbom7umb_ae3+s19A@mail.gmail.com>
References: <CAMd3iJK2ZA12REJniWU7rOoVvSdmeyDeNLbom7umb_ae3+s19A@mail.gmail.com>
Message-ID: <CAGvuWYaRym2xk-7vTOzLnfqgiMeLsP73Ywpfmy0=UkBQv5fZuQ@mail.gmail.com>

1. GIT Latest Version

wget
http://www.codemonkey.org.uk/projects/git-snapshots/git/git-latest.tar.gz


2.FreeSWITCH Latest Version

git clone  git://git.freeswitch.org/freeswitch.git


Thanks
Lloyd

*
*



On Tue, Oct 4, 2011 at 9:03 AM, Masood Ahmed <masood24 at gmail.com> wrote:

> Dear All,
>
> I am new to FreeSwitch and I have installed the old version 1.0.6 on
> CentOS-5.5 32 Bit
> From wiki I read that to install the latest version of freeswitch should
> use GIT.
> Now to install GIT I found following link :
> http://kernel.org/pub/software/scm/git/git-1.7.6.tar.bz2
> but the link is down since last two weeks, I am using* CentOS 5.5 32 Bit
> machine.*
> Can any one tell me where can i find the GIT OR most importantly if some
> one knows the direct link to download FreeSwitch Latest Version.
> Like for older version this is the direct link :
> http://files.freeswitch.org/freeswitch-1.0.6.tar.gz
>
> Will appreciate your Help.
>
> Thanks
>
> --
> *Regards,
> Masood Ahmed
> *
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Tue Oct  4 21:12:18 2011
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 4 Oct 2011 13:12:18 -0400
Subject: [Freeswitch-users] no switch_event_get_header in FreeSwitch.dll
In-Reply-To: <1317736663240-6858734.post@n2.nabble.com>
References: <CAGK4cYw7oYYu3ZOXuyMj+hNASsRrG7_jESMOsNYD+dJwR6Q_xw@mail.gmail.com>
	<1317736663240-6858734.post@n2.nabble.com>
Message-ID: <CAGK4cYxLDCR1a3-=Akyi0iKBra8_=HOAy7B-p+mE6HLDizqpyg@mail.gmail.com>

VS2008 - OK - thanks for the pointer - let me take a look.

On Tue, Oct 4, 2011 at 9:57 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> VS2008 or VS2010? This is most likely caused by a SWIG problem.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/no-switch-event-get-header-in-FreeSwitch-dll-tp6856879p6858734.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Tue Oct  4 21:34:18 2011
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 4 Oct 2011 13:34:18 -0400
Subject: [Freeswitch-users] no switch_event_get_header in FreeSwitch.dll
In-Reply-To: <CAGK4cYxLDCR1a3-=Akyi0iKBra8_=HOAy7B-p+mE6HLDizqpyg@mail.gmail.com>
References: <CAGK4cYw7oYYu3ZOXuyMj+hNASsRrG7_jESMOsNYD+dJwR6Q_xw@mail.gmail.com>
	<1317736663240-6858734.post@n2.nabble.com>
	<CAGK4cYxLDCR1a3-=Akyi0iKBra8_=HOAy7B-p+mE6HLDizqpyg@mail.gmail.com>
Message-ID: <CAGK4cYy-9L0rz3DSOWoC-uXcYjMej6FJE8HrTwpnMhjWPbX2nA@mail.gmail.com>

Sorry - my bad. I still had mod_file string in there. Deleted dll and all is
good.

On Tue, Oct 4, 2011 at 1:12 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> VS2008 - OK - thanks for the pointer - let me take a look.
>
>
> On Tue, Oct 4, 2011 at 9:57 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>> VS2008 or VS2010? This is most likely caused by a SWIG problem.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/no-switch-event-get-header-in-FreeSwitch-dll-tp6856879p6858734.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From infos at madovsky.org  Tue Oct  4 21:37:14 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Oct 2011 13:37:14 -0400
Subject: [Freeswitch-users] persistant registration
Message-ID: <A8CF8B4A12F54E83AB8D701364B4DDE8@e1705>

Hi all,

sometimes sip registration stays persistant, even
if the phone is not connected anymore. I can see this line in sofia status profile

Status:         Registered(UDP)(unknown) EXP(2011-10-04 09:38:40) EXPSECS(-14152)

as you can see the expsecs is far under 0. 
any idea ?

Thanks
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From infos at madovsky.org  Tue Oct  4 21:41:39 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Oct 2011 13:41:39 -0400
Subject: [Freeswitch-users] persistant registration
Message-ID: <840C738DB8A44D018B6A0CAA981D28F4@e1705>

Ok I just deleted some sqlite files in db directory and everything is fine.
probably a corrupted db file.

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, October 04, 2011 1:37 PM
  Subject: persistant registration


  Hi all,

  sometimes sip registration stays persistant, even
  if the phone is not connected anymore. I can see this line in sofia status profile

  Status:         Registered(UDP)(unknown) EXP(2011-10-04 09:38:40) EXPSECS(-14152)

  as you can see the expsecs is far under 0. 
  any idea ?

  Thanks
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From victor.chukalovskiy at utoronto.ca  Tue Oct  4 21:55:16 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Tue, 04 Oct 2011 13:55:16 -0400
Subject: [Freeswitch-users] Adding new MOH stream - isn't reloadxml enough
	to make it effective?
Message-ID: <4E8B4884.90703@utoronto.ca>

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From randy.andrade at gmail.com  Tue Oct  4 23:02:23 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Tue, 4 Oct 2011 15:02:23 -0400
Subject: [Freeswitch-users] BLF on a Valet Park Orbit?
Message-ID: <CAJZcYrd6PLxjjhW=TRNt-aZb2yfCMis_bdtEPZo2k-HqEETUTg@mail.gmail.com>

Hey there,

I'm just wondering if anyone knows of a way to use the fifo / parking
functionality in Freeswitch to trigger BLF updates on a phone. IE, on a
Cisco SPA508, I have BLF / SD keys setup with the default Freeswitch
6001-6099 valet park spots (no I don't have keys for all 99 spots, I'm just
doing the first 5), with the normal syntax that I would use to monitor any
other extension in Freeswitch. Although the SD's work great for sending a
call to, and picking a call up from, each parking spot, it would be
extremely handy if each spot could send BLF updates so as to indicate when
someone was actively parked in that spot. So if I send someone to spot 6003,
the SD key for 6003 would light up with the BLF function that a normal
extension can be monitored with.

I'm just curious whether anyone has seen this working or created similar
functionality a different way.

Thanks!

Randy
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From infos at madovsky.org  Wed Oct  5 02:26:49 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 Oct 2011 18:26:49 -0400
Subject: [Freeswitch-users] RTCP packets not written
Message-ID: <A2F386BC5DAB4840A36C9CE52AD6E0F3@e1705>

Hi,

I noticed 

2011-10-04 14:24:28.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:24:33.512388 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:24:38.512378 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:24:43.512390 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:24:48.512362 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:24:53.512375 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:24:58.512364 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:03.512369 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:08.512386 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:13.512390 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:18.512364 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:23.512376 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:28.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:33.512365 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:38.512368 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:43.512372 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:48.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written
2011-10-04 14:25:53.512367 [DEBUG] switch_rtp.c:3910 RTCP packet not written


is it come from a permission file somewhere ?

Thanks
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From brad at tech21.com  Wed Oct  5 02:52:23 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 4 Oct 2011 15:52:23 -0700
Subject: [Freeswitch-users] My new web utility - Windows 7 gadget
In-Reply-To: <CALXCt0rJaQ4oxJgTMyZKf165La5MjzAwBUH0DJ2nTbdQX3oSkQ@mail.gmail.com>
References: <CAF=swh---deRqRKCyJGxhPgAd9bdTvXhzqA1qwRFFExmCr9SNA@mail.gmail.com>
	<1317736759665-6858743.post@n2.nabble.com>
	<CALXCt0rJaQ4oxJgTMyZKf165La5MjzAwBUH0DJ2nTbdQX3oSkQ@mail.gmail.com>
Message-ID: <CAGRr05-E-0CxGXxpbkDWKpkY75MhLFaMLP3nHzx2jVizkpD-7g@mail.gmail.com>

Git contrib!?

On Tue, Oct 4, 2011 at 7:19 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> kewl!
>
> -giovanni
>
> On Tue, Oct 4, 2011 at 3:59 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
> > Cool thanks for posting your sample!
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/My-new-web-utility-Windows-7-gadget-tp6853946p6858743.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brad at tech21.com  Wed Oct  5 03:11:54 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 4 Oct 2011 16:11:54 -0700
Subject: [Freeswitch-users] My new web utility - Windows 7 gadget
In-Reply-To: <CAGRr05-E-0CxGXxpbkDWKpkY75MhLFaMLP3nHzx2jVizkpD-7g@mail.gmail.com>
References: <CAF=swh---deRqRKCyJGxhPgAd9bdTvXhzqA1qwRFFExmCr9SNA@mail.gmail.com>
	<1317736759665-6858743.post@n2.nabble.com>
	<CALXCt0rJaQ4oxJgTMyZKf165La5MjzAwBUH0DJ2nTbdQX3oSkQ@mail.gmail.com>
	<CAGRr05-E-0CxGXxpbkDWKpkY75MhLFaMLP3nHzx2jVizkpD-7g@mail.gmail.com>
Message-ID: <CAGRr058-=DXDdz3v0Ay5kniswgy1vZMkJ425so6dZ=m-RG=3GA@mail.gmail.com>

Would be nice if the notifications had English translations. You can pretty
much assume a lot from the popup content.
Also, would be cool if it was a 'slide up' style window instead of an OK box
with info on it . I could see that mess with fullscreen applications.

On Tue, Oct 4, 2011 at 3:52 PM, Brad Mina <brad at tech21.com> wrote:

> Git contrib!?
>
>
> On Tue, Oct 4, 2011 at 7:19 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> kewl!
>>
>> -giovanni
>>
>> On Tue, Oct 4, 2011 at 3:59 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>> > Cool thanks for posting your sample!
>> >
>> > --
>> > View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/My-new-web-utility-Windows-7-gadget-tp6853946p6858743.html
>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From masood24 at gmail.com  Wed Oct  5 10:22:17 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Wed, 5 Oct 2011 09:22:17 +0300
Subject: [Freeswitch-users] Installing GIT for CentOS the link is down
	or give me
Message-ID: <CAMd3iJJ8n7h2rf6u9kHc3QSKnUXWx3eRZXK3=tj60xO-Bm966w@mail.gmail.com>

Thank you Lloyd Aloysius.


On Tue, Oct 4, 2011 at 10:02 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. Re: Installing GIT for CentOS the link is down or give me
>      directo link of latest version. (Lloyd Aloysius)
>   2. Re: no switch_event_get_header in FreeSwitch.dll (Phillip Jones)
>   3. Re: no switch_event_get_header in FreeSwitch.dll (Phillip Jones)
>   4. persistant registration (Madovsky)
>   5. Re: persistant registration (Madovsky)
>   6. Adding new MOH stream - isn't reloadxml enough    to make it
>      effective? (Victor Chukalovskiy)
>   7. BLF on a Valet Park Orbit? (Randy Andrade)
>
>
> ---------- Forwarded message ----------
> From: Lloyd Aloysius <lloyd.aloysius at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 4 Oct 2011 12:51:34 -0400
> Subject: Re: [Freeswitch-users] Installing GIT for CentOS the link is down
> or give me directo link of latest version.
>
> 1. GIT Latest Version
>
> wget
> http://www.codemonkey.org.uk/projects/git-snapshots/git/git-latest.tar.gz
>
>
> 2.FreeSWITCH Latest Version
>
> git clone  git://git.freeswitch.org/freeswitch.git
>
>
> Thanks
> Lloyd
>
> *
> *
>
>
>
> On Tue, Oct 4, 2011 at 9:03 AM, Masood Ahmed <masood24 at gmail.com> wrote:
>
>> Dear All,
>>
>> I am new to FreeSwitch and I have installed the old version 1.0.6 on
>> CentOS-5.5 32 Bit
>> From wiki I read that to install the latest version of freeswitch should
>> use GIT.
>> Now to install GIT I found following link :
>> http://kernel.org/pub/software/scm/git/git-1.7.6.tar.bz2
>> but the link is down since last two weeks, I am using* CentOS 5.5 32 Bit
>> machine.*
>> Can any one tell me where can i find the GIT OR most importantly if some
>> one knows the direct link to download FreeSwitch Latest Version.
>> Like for older version this is the direct link :
>> http://files.freeswitch.org/freeswitch-1.0.6.tar.gz
>>
>> Will appreciate your Help.
>>
>> Thanks
>>
>> --
>> *Regards,
>> Masood Ahmed
>> *
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: Phillip Jones <pjintheusa at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 4 Oct 2011 13:12:18 -0400
> Subject: Re: [Freeswitch-users] no switch_event_get_header in
> FreeSwitch.dll
> VS2008 - OK - thanks for the pointer - let me take a look.
>
> On Tue, Oct 4, 2011 at 9:57 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>> VS2008 or VS2010? This is most likely caused by a SWIG problem.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/no-switch-event-get-header-in-FreeSwitch-dll-tp6856879p6858734.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> ---------- Forwarded message ----------
> From: Phillip Jones <pjintheusa at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 4 Oct 2011 13:34:18 -0400
> Subject: Re: [Freeswitch-users] no switch_event_get_header in
> FreeSwitch.dll
> Sorry - my bad. I still had mod_file string in there. Deleted dll and all
> is good.
>
> On Tue, Oct 4, 2011 at 1:12 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> VS2008 - OK - thanks for the pointer - let me take a look.
>>
>>
>> On Tue, Oct 4, 2011 at 9:57 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>
>>> VS2008 or VS2010? This is most likely caused by a SWIG problem.
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/no-switch-event-get-header-in-FreeSwitch-dll-tp6856879p6858734.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> ---------- Forwarded message ----------
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 4 Oct 2011 13:37:14 -0400
> Subject: [Freeswitch-users] persistant registration
> **
> Hi all,
>
> sometimes sip registration stays persistant, even
> if the phone is not connected anymore. I can see this line in sofia status
> profile
>
> Status:         Registered(UDP)(unknown) EXP(2011-10-04 09:38:40)
> EXPSECS(-14152)
> as you can see the expsecs is far under 0.
> any idea ?
>
> Thanks
>
>
> ---------- Forwarded message ----------
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 4 Oct 2011 13:41:39 -0400
> Subject: Re: [Freeswitch-users] persistant registration
> **
> Ok I just deleted some sqlite files in db directory and everything is fine.
> probably a corrupted db file.
>
> thanks
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, October 04, 2011 1:37 PM
> *Subject:* persistant registration
>
> Hi all,
>
> sometimes sip registration stays persistant, even
> if the phone is not connected anymore. I can see this line in sofia status
> profile
>
> Status:         Registered(UDP)(unknown) EXP(2011-10-04 09:38:40)
> EXPSECS(-14152)
> as you can see the expsecs is far under 0.
> any idea ?
>
> Thanks
>
>
>
> ---------- Forwarded message ----------
> From: Victor Chukalovskiy <victor.chukalovskiy at utoronto.ca>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 04 Oct 2011 13:55:16 -0400
> Subject: [Freeswitch-users] Adding new MOH stream - isn't reloadxml enough
> to make it effective?
> **
> Hello,
>
> Just added a new stream in local_stream.conf.xml file.
> Upon "*reloadxml" *the new stream still doesn't show-up in output of "*
> show_local_stream"
> *Does it take more than "*reloadxml"* to make new stream available? Don't
> want to restart a production system, so please let me know the right
> command.
>
> Also, what exactly is <param name="chime-max" value="500"/> ?
> In-line comment says "*limit to how many seconds the file will play" *but
> wiki says  "*max number of "break-in" attempts". *These two seem to
> contradict each other, which one is the correct interpretation?
>
>
> Thank you,
> Victor
>
>
> ---------- Forwarded message ----------
> From: Randy Andrade <randy.andrade at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 4 Oct 2011 15:02:23 -0400
> Subject: [Freeswitch-users] BLF on a Valet Park Orbit?
> Hey there,
>
> I'm just wondering if anyone knows of a way to use the fifo / parking
> functionality in Freeswitch to trigger BLF updates on a phone. IE, on a
> Cisco SPA508, I have BLF / SD keys setup with the default Freeswitch
> 6001-6099 valet park spots (no I don't have keys for all 99 spots, I'm just
> doing the first 5), with the normal syntax that I would use to monitor any
> other extension in Freeswitch. Although the SD's work great for sending a
> call to, and picking a call up from, each parking spot, it would be
> extremely handy if each spot could send BLF updates so as to indicate when
> someone was actively parked in that spot. So if I send someone to spot 6003,
> the SD key for 6003 would light up with the BLF function that a normal
> extension can be monitored with.
>
> I'm just curious whether anyone has seen this working or created similar
> functionality a different way.
>
> Thanks!
>
> Randy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Masood Ahmed
*
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From msc at freeswitch.org  Wed Oct  5 20:19:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Oct 2011 09:19:44 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today: Git Tutorial
Message-ID: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>

Hello all!

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

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

Talk to you soon,
Michael
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From rrodolfos at gmail.com  Wed Oct  5 17:40:47 2011
From: rrodolfos at gmail.com (RrodolfoS .)
Date: Wed, 5 Oct 2011 09:10:47 -0430
Subject: [Freeswitch-users] Freeswitch fail to load the mod_spidermonkey on
 startup, undefined symbol: SSLv2_client_method
Message-ID: <CAMdaD0CUSpb19Au1rLTi=-TkwioQf3xVNjZuS-2kFW6vmDH5ig@mail.gmail.com>

List!,

Freeswitch fail to load the mod_spidermonkey on startup, throw the follow [CRIT]

2011-10-05 08:46:43.966241 [CRIT] switch_loadable_module.c:1275 Error
Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so
**/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined symbol:
SSLv2_client_method**

I try recompile the module with: make mod_spidermonkey-install

but continues fail to load spidermonkey module

how can i solve it?, i'm using debian wheezy amd64 linux/3.0.0-1-686-pae.

The freeswitch source was downloaded from git clone
git://git.freeswitch.org/freeswitch.git

thanks.

Rodolfo Sauce-Guinand


From jchavanton at gmail.com  Wed Oct  5 17:52:22 2011
From: jchavanton at gmail.com (Julien Chavanton)
Date: Wed, 5 Oct 2011 09:52:22 -0400
Subject: [Freeswitch-users] using sip-force-contact with a gateway
Message-ID: <CAKmcL2=QdynU-AtG2AM42HW2ko0E+-_6pHkKvKpn5YvYL3YbGw@mail.gmail.com>

Hi, I am dealing with an Asterisk server behind NAT that is not set
properly, with its external IP
I am asking them to fix it on there side but I am curious to know if there
is a way to deal with this implemented in FS

Freeswitch --> Internet --> NAT --> Asterisk

In the "200 OK" returned by Asterisk  as the private IP in the "Contact"
header
FS is returning ACK to the private IP.

INVITE --->
<--- 200 OK
---> ACK
<--- 200 OK
---> ACK
<--- 200 OK
---> ACK
....

It seems   <param name="apply-nat-acl" value="rfc1918"/><param
name="aggressive-nat-detection" value="true"/>
does

It seems sip-force-contact could patch the problem caused by the remote
gateway
Is there a way to us sip-force-contact with gateway instead of an entry in
the directory ?
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From msc at freeswitch.org  Wed Oct  5 20:31:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Oct 2011 09:31:15 -0700
Subject: [Freeswitch-users] Adding new MOH stream - isn't reloadxml
 enough to make it effective?
In-Reply-To: <4E8B4884.90703@utoronto.ca>
References: <4E8B4884.90703@utoronto.ca>
Message-ID: <CAKzWOxXnprV4vm5EBHs2ggd3S8kfkk9JyPPB4zbcUw9VuQRrQw@mail.gmail.com>

See the other commands provided by mod_local_stream:
start_local_stream <local_stream_name> [<path>] [<rate>] [<shuffle>]
[<prebuf>] [<channels>] [<interval>] [<timer_name>]
stop_local_stream <local_stream_name>

-MC

On Tue, Oct 4, 2011 at 10:55 AM, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

> **
> Hello,
>
> Just added a new stream in local_stream.conf.xml file.
> Upon "*reloadxml" *the new stream still doesn't show-up in output of "*
> show_local_stream"
> *Does it take more than "*reloadxml"* to make new stream available? Don't
> want to restart a production system, so please let me know the right
> command.
>
> Also, what exactly is <param name="chime-max" value="500"/> ?
> In-line comment says "*limit to how many seconds the file will play" *but
> wiki says  "*max number of "break-in" attempts". *These two seem to
> contradict each other, which one is the correct interpretation?
>
>
> Thank you,
> Victor
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct  5 20:42:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Oct 2011 09:42:03 -0700
Subject: [Freeswitch-users] Desktop phone - answer a ringing call via
	API call
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36E5F@AZWSMS03.azwarranty.int>
References: <AcyCnbOIxZuxEaeRQIyKEkToMpcvaQ==>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E5F@AZWSMS03.azwarranty.int>
Message-ID: <CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>

This would be a function of the phone. AFAICT, Polycom does not have a Webby
way of doing this. However, I think Brian West has a bead on using the
expansion port along with an Arduino board to do some manual/remote control
of the phone. If anyone has more information on remote controlling Polycoms
(or any other phones, for that matter) please let us know.

-MC

On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

>  Hi all,****
>
> ** **
>
> Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP
> 560) via api command ?****
>
> I tried with SendMsg <uuid> and ?answer? application, but it?s not want I
> need/want.****
>
> ** **
>
> ** **
>
> Thanks and best regards,****
>
> ** **
>
> Stefan****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Oct  5 21:00:52 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 5 Oct 2011 19:00:52 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Conference Call
	Today: Git Tutorial
In-Reply-To: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>
References: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>
Message-ID: <CALXCt0o9JF0OCSZ55D5J6Yt_g5_Mw8PczzPfU5wY9zz7OtGbMg@mail.gmail.com>

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

-giovanni


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

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From buscom123+fs at gmail.com  Wed Oct  5 21:54:39 2011
From: buscom123+fs at gmail.com (R H)
Date: Wed, 5 Oct 2011 11:54:39 -0600
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAKbxfG9Msun7_rdMEoSHk-rtpZ_bVJmdUoHWxFBDHDC5U2EaFg@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
	<CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>
	<CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>
	<CAKbxfG9Msun7_rdMEoSHk-rtpZ_bVJmdUoHWxFBDHDC5U2EaFg@mail.gmail.com>
Message-ID: <CAAtQP57w_FgGCFaSA3Sxooky0kTyoLdAkfjv227=AxxO=nBs9g@mail.gmail.com>

Hey Anthony, I appreciate your help on this. Your response was not clear
enough for me to understand how to accomplish what you are describing. Let's
assume I only have 2 profiles (Internal/External). Both profiles default to
the public context.

Now if I were to create two domains in the directory and name them "
internal.mydomain.com" and "external.mydomain.com" and force everyone who
registers to the internal profile to use the internal domain and everyone on
the external profile to use the external domain I would still get the same
problem.

If I have two clients register on the external profile using the external
domain and they call each other, with ODBC disabled everything works fine,
but with ODBC enabled I get two invites to the destination client.

The fact that I get different behavior with odbc enabled / disabled seems to
indicate a bug more than it indicates a mis-configuration.

Perhaps I am missing something? I don't quite understand the section in the
example xml that refers to domains. Is there something I need to do there?
The wiki does not mention it at all and the comments in the xml are not very
clear either:

<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="false" parse="true"/>
 </domains>

Thanks for your help,

Ryan

On Tue, Oct 4, 2011 at 10:27 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> ok but the db is shared so if they are all using mydomain.com it will
> cross over so you should use distinct fake domains at least between
> your profiles.
>
>
> On Tue, Oct 4, 2011 at 11:24 AM, R H <buscom123+fs at gmail.com> wrote:
> > In Response to Brian West, below are my sip registrations. These may have
> > slightly different values from the logs I sent yesterday simply because I
> > shut down last night and re-registered this morning. Same softphones,
> same
> > extensions, just new registrations:
> >
> -----------------------------------------------------------------------------
> > select * from sip_registrations;
> > MTAwYWEwYzhlNzc2MDAxYTUzNDhlY2EwZmEyZDFlZDE.,4001,mydomain.com,
> mydomain.com,"""user""
> > <sip:4001 at 192.168.99.164:61238
> ;rinstance=e2d50bbe31452d30>",Registered(UDP),unknown,1317746818,X-Lite
> > 4 release 4.1 stamp
> >
> 63215,4001,192.168.122.236,internal,psiFreeswitch,192.168.99.164,61238,4001,
> mydomain.com,4001,mydomain.com,192.168.122.236,psiFreeswitch
> > dQJifXuoqNBUlWCuZ8rCI6cu.Nn5F81W,4002,mydomain.com,mydomain.com,"""Test
> > User123""
> > <sip:cqgulpey at 192.168.99.164:57003
> >",Registered(UDP),unknown,1317743825,Blink
> > Lite 1.2.0
> >
> (MacOSX),4002,192.168.122.236,internal,psiFreeswitch,192.168.99.164,57003,4002,
> mydomain.com,4002,mydomain.com,192.168.122.236,psiFreeswitch
> >
> -----------------------------------------------------------------------------
> > In Response to Anthony Minessale,
> > "Mydomain.com" is simply for testing and will not be our go-live domain.
> The
> > particular switch I am testing on is located in a closed internal network
> > designed specifically for testing systems like this and nothing more. It
> is
> > set up using the instructions on the FreeSwitch wiki for Multi-Tenant
> > installations because we have the potential to go multi-tenant in the
> future
> > and we want to make sure multi-tenant setups work for the system we are
> > building.
> > As far as my profiles in development i have the following setup:
> > 1) Internal - To emulate production this is limited use, no real phones
> will
> > ever connect from the local LAN in production. We have been testing
> > registrations on this profile anyway but not there is no plan to have
> phones
> > register here in production:
> > <domain name="all" alias="false" parse="false"/>
> >
> >
> > 2) External - Most phones will register here, possibly to multiple
> domains
> > (tenants). These devices will be on other networks with possible natting
> > issues. This profile requires authentication to make calls.
> > 3) External_Debug - When clients complain of connectivity issues we will
> be
> > asking them to move to this profile so that we can more easily monitor
> their
> > sip traffic and debug their issues. This profile will also require
> > authentication and allow registration to the many domains.
> > 4) Unnamed Future External Profile - We will also most likely have
> another
> > profile for our upstream providers once we are ready to turn them on.
> >
> >
> >
> >
> >
> > On Mon, Oct 3, 2011 at 5:59 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> are you really using "mydomain.com" on all the profiles at the same
> time?
> >>
> >> On Mon, Oct 3, 2011 at 5:29 PM, R H <buscom123+fs at gmail.com> wrote:
> >> > Sure,
> >> > I have two soft phones registering as extensions 4001 and 4002 to the
> >> > same
> >> > profile. When 4002 dials 4001 the softphone that is registered as 4001
> >> > received two calls simultaneously. This only happens when I enable
> odbc
> >> > on
> >> > two profiles. If I disable odbc or only enable it for one profile
> >> > everything
> >> > works fine. If I enable ODBC for 3 profiles I would get three invites
> to
> >> > 4001, etc...
> >> > Attached is some logging output. Notice the parts highlighted in red.
> >> > Thank you for your help.
> >> > Ryan
> >> >
> >> >
> >> >
> >> > On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> could you elaborate more on what the B-legs are?
> >> >> Are you doing bridges to an outbound entity, if so how?
> >> >>
> >> >> do you have any traces doing:
> >> >>
> >> >> sofia global siptrace on
> >> >> and
> >> >> console loglevel debug
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
> >> >> > Hey Everyone,
> >> >> > I am preparing to set up HA Failover to my production freeswitch
> >> >> > environment. In preparation for this I have been doing some testing
> >> >> > in a
> >> >> > development environment and I encountered the following problem:
> >> >> > I currently have two sip-profiles set up that I am using full time,
> >> >> > when
> >> >> > I
> >> >> > enable odbc in the switch.conf.xml everything works correctly. When
> I
> >> >> > enable
> >> >> > odbc in ONE of the sip-profiles everything works correctly. But
> when
> >> >> > I
> >> >> > enable odbc in the second profile I begin to see two b-legs for
> ever
> >> >> > call
> >> >> > that I initiate. Both b-legs are directed at the same client.
> >> >> > Does anyone have any idea why enabling odbc on two sofia profiles
> >> >> > would
> >> >> > cause every inbound call to generate two outbound calls? I also
> >> >> > decided
> >> >> > to
> >> >> > test this problem by adding two more profiles for a total of 4.
> With
> >> >> > 4
> >> >> > profiles I get 4 b-legs. What am I doing wrong?
> >> >> > Ryan
> >> >> >
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Anthony Minessale II
> >> >>
> >> >> FreeSWITCH http://www.freeswitch.org/
> >> >> ClueCon http://www.cluecon.com/
> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>
> >> >> AIM: anthm
> >> >> MSN:anthony_minessale at hotmail.com
> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> >> IRC: irc.freenode.net #freeswitch
> >> >>
> >> >> FreeSWITCH Developer Conference
> >> >> sip:888 at conference.freeswitch.org
> >> >> googletalk:conf+888 at conference.freeswitch.org
> >> >> pstn:+19193869900
> >> >>
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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 msc at freeswitch.org  Wed Oct  5 22:28:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Oct 2011 11:28:36 -0700
Subject: [Freeswitch-users] Sending DTMF on B-leg WORKING
In-Reply-To: <99817380EE8745A5B22E84058810B705@ccs.local>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local>
	<CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>
	<9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>
	<CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com>
	<F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local>
	<CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com>
	<C57A15E034154A9297EE8DF5EF752971@ccs.local>
	<CAKzWOxXJ7J+Pi2iart1nHC8-OmwHkZ0vjH3iXsKvAKGdNOt7=Q@mail.gmail.com>
	<B5489AF9DCC040E2AAA414038FFBB657@ccs.local>
	<99817380EE8745A5B22E84058810B705@ccs.local>
Message-ID: <CAKzWOxV0czhxJmmv2zF=uf9dTFG9o-ZpgPHr=TuH5q4iK_evxQ@mail.gmail.com>

FYI, I added this to the queue_dtmf misc dp tools wiki page.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf#Alternate_Example_Using_execute_on_answer

-MC

On Tue, Oct 4, 2011 at 7:27 AM, Wayne <wayne at hamilton.net> wrote:

> **
> Just to answer this posts for others that are searching for the answer this
> is what I did to make it work.
> For some reason after resetting my Sangoma card  with a power down and
> restart of the server send_dtmf started working.
> Here is  the XML that worked for me. Seems that the sleep and flush also
> got rid of my random tones.
>
> <extension name="test-tone" >
>    <condition field="destination_number" expression="1(.*)" >
>         <action application="set" data="effective_caller_id_number=
> 9025556747"/>
>         <action application="sleep" data="500"/>
>         <action application="flush_dtmf"/>
>         <action application="export"
> data="nolocal:execute_on_answer=send_dtmf W$1#@500" />
>        <action application="bridge" data="freetdm/wp1/a/18005551234"/>
>    </condition>
> </extension>
>
> Thanks to all that helped.
> Wayne
>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Wayne
> *Sent:* Thursday, September 15, 2011 11:55 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>  Okay,
>
> I have tried send_dtmf and queue_dtmf and I can not get any DTMF to pass.
> Would anyone be so kind and send me an extension that has it working on the
> B-leg. I am getting nowhere.
>
> Thanks Wayne
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, September 13, 2011 4:29 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
> Okay, fair enough. Not sure what queue_dtmf is not working. You could
> always try "execute_on_answer" and execute an extension that sends the
> dtmfs.
>
> -MC
>
> On Tue, Sep 13, 2011 at 2:16 PM, Wayne <wayne at hamilton.net> wrote:
>
>> **
>> Sorry Michael,
>>
>> When I started out I was sending the calls to a SIP provider. I couldn't
>> here the DTMF tones. For trouble shooting I moved it to a local PRI. I was
>> thinking that maybe this was an issue with the sip trunk and sending inband
>> or out of band but that doesn't seem to be the case.
>>
>> Wayne
>>
>>  ------------------------------
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Tuesday, September 13, 2011 2:41 PM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>>
>>
>>
>> On Tue, Sep 13, 2011 at 11:34 AM, Wayne <wayne at hamilton.net> wrote:
>>
>>> **
>>> That's true I don't hear any tones when I answer the phone. I don't see
>>> any messages in my log from queue_dtmf. Any help would be great.
>>>
>>> Wayne
>>>
>> Okay, in your original post you said you are making an OB call on a SIP
>> trunk, yet in your example there is no SIP trunk. You've got an internal
>> user and a FreeTDM FXO port. Can you please clarify what you actually have?
>> Who's the caller and who's the callee?
>> -MC
>>
>>
>>
>>>
>>>
>>>
>>>  ------------------------------
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>>> Collins
>>> *Sent:* Thursday, September 08, 2011 6:25 PM
>>>
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>>>
>>>   Are you saying that no DTMFs are being sent out when the bridge is
>>> initially completed?
>>> -MC
>>>
>>> On Wed, Sep 7, 2011 at 3:17 PM, Wayne <wayne at hamilton.net> wrote:
>>>
>>>> **
>>>> I have tried that
>>>> <extension name="test-tone" >
>>>>    <condition field="destination_number" expression="(9025556747)" >
>>>>        <action application="set" data="call_direction=outbound" />
>>>>        <action application="set"
>>>> data="effective_caller_id_name=${outbound_caller_id_name}" />
>>>>        <action application="set"
>>>> data="effective_caller_id_number=${outbound_caller_id_number}" />
>>>>        <action application="set" data="continue_on_fail=true" />
>>>>        <action application="queue_dtmf" data="0123456789" />
>>>>        <action application="bridge" data="freetdm/wp1/a/$1" />
>>>>    </condition>
>>>> </extension>
>>>>
>>>> The debug tells me that this
>>>> tail -f freeswitch.log | grep -i dtmf
>>>>
>>>> Dialplan: sofia/internal/1333 at 192.168.48.87 Action
>>>> queue_dtmf(0123456789)
>>>> EXECUTE sofia/internal/1333 at 192.168.48.87 queue_dtmf(0123456789)
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
>>>> sofia/internal/1333 at 192.168.48.87 Queue dtmf
>>>> 2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3355 Set 2833 dtmf send
>>>> payload to 101
>>>> 2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3360 Set 2833 dtmf
>>>> receive payload to 101
>>>> 2011-09-07 17:13:45.116216 [DEBUG] ftdm_io.c:3714 [s1c1][1:1] Generating
>>>> DTMF [0123456789]
>>>> 2011-09-07 17:13:47.876238 [DEBUG] ftmod_wanpipe.c:701 [s1c1][1:1]
>>>> Enabled DTMF events
>>>> 2011-09-07 17:13:47.956228 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1]
>>>> Queuing wanpipe DTMF: 9
>>>> 2011-09-07 17:13:47.956228 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing
>>>> DTMF 9 (debug = 0)
>>>> 2011-09-07 17:13:47.956228 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
>>>> channel FreeTDM/1:1/9025556747
>>>> 2011-09-07 17:13:48.036235 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1]
>>>> Queuing wanpipe DTMF: 9
>>>> 2011-09-07 17:13:48.036235 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing
>>>> DTMF 9 (debug = 0)
>>>> 2011-09-07 17:13:48.036235 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
>>>> channel FreeTDM/1:1/9025556747
>>>>
>>>> So where would I look next.
>>>>
>>>> Wayne
>>>>
>>>>
>>>>  ------------------------------
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi
>>>> Marcus
>>>> *Sent:* Wednesday, September 07, 2011 3:55 PM
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>>>>
>>>>   I think you looking for queue_dtmf.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
>>>>
>>>> -Avi Marcus
>>>>
>>>>
>>>>
>>>> On Wed, Sep 7, 2011 at 11:47 PM, Wayne <wayne at hamilton.net> wrote:
>>>> > Hello All,
>>>> >
>>>> > I need to call out on a SIP trunk and when the call is answered send
>>>> DTMF
>>>> > tones.
>>>> > I need to send the DTMF only on the outbound leg.
>>>> >
>>>> > Does anyone have a dialplan that will do that? Is it possible. I have
>>>> only
>>>> > found one thread on it and did get much out of it.
>>>> >
>>>> > Thanks
>>>> > Wayne
>>>> >
>>>> >
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cliff at develix.com  Wed Oct  5 23:11:28 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 05 Oct 2011 12:11:28 -0700
Subject: [Freeswitch-users] mod_cdr_pg_csv not loading automatically
Message-ID: <1317841888.2482.9.camel@portable-evil>


I have this in modules.conf.xml:

    <!-- Event Handlers -->
    <load module="mod_cdr_pg_csv"/>
    <!-- <load module="mod_cdr_csv"/> -->
    <!-- <load module="mod_cdr_sqlite"/> -->
    <!-- <load module="mod_event_multicast"/> -->
    <load module="mod_event_socket"/>
    <!-- <load module="mod_event_zmq"/> -->
    <!-- <load module="mod_zeroconf"/> -->
    <!-- <load module="mod_erlang_event"/> -->
    <!-- <load module="mod_snmp"/> -->

but mod_cdr_pg doesn't autoload.  However, if I use fs_cli and "load
mod_cdr_pg_csv", it loads and works fine.

This is a fresh install from git yesterday.  Unfortunately I can't
restart FS until tonight (it's a production system) to see the startup
messages, but I wonder if anyone has seen this type of behavior (my
older system didn't have this issue).

Scientific Linux 6.1 x86_64.

Regards,
Cliff




From jmoran at secureachsystems.com  Wed Oct  5 23:17:33 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Wed, 5 Oct 2011 15:17:33 -0400
Subject: [Freeswitch-users] mod_avmd - Does this work for anyone on current
	builds?
Message-ID: <361E98F99D3CC3439EED59BC1924ED695471E9@SERVER2003.SecuReachSystems.local>

I tried out mod_avmd 6+ months back and had some success.  However, I
cannot seem to get it to trip any longer on newer builds of FS. The
avmd_detect variable is blank and no event fires on multiple carriers
and destination phones. Does anybody else have success on current/recent
builds of FS?

 

Thanks,
Jason

 

Example code:

function onInput(s, type, obj, arg)

{

   try {

        if(type == "event" && session.getVariable("avmd_detect") ==
"TRUE")

        {

              console_log("info", "Voicemail Detected\n"); 

              session.execute("avmd", "stop");

              session.execute("sleep", "200");

 
session.streamFile('/usr/local/freeswitch/recordings/playbackfile.wav');

              if(session.ready()) //Only hangup if onDTMF code has
finished.

            {

                session.execute("hangup","ORIGINATOR_CANCEL");
//session.hangup doesn't give the code we want

            }

            return false; // don't allow original playback file to
continue

        }

   }  catch (e) {

      console_log("err", e + "\n");

   }

   return true;

}

 

session = new Session('{
origination_caller_id_number=[deleted]}sofia/gateway/[deleted]/'+argv[0]
);

 

if (session.ready()) {

    session.execute("sleep","500");

    session.execute("avmd", "start");

    console_log("info", "avmd started\n"); 

 

 
session.streamFile('/usr/local/freeswitch/recordings/playbackfile.wav',
onInput);

    console_log("info","avmd_detect value
is:"+session.getVariable("avmd_detect")+"\n");

    if ( session.ready( ) ) {

        session.hangup( );

    }

}

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From cliff at develix.com  Wed Oct  5 23:21:09 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 05 Oct 2011 12:21:09 -0700
Subject: [Freeswitch-users] mod_native_file and speex
Message-ID: <1317842469.2482.16.camel@portable-evil>

I'm using Speex to bridge calls between an out-of-state Asterisk box and
a local FreeSWITCH IVR.  This works great, but I'm having to do needless
transcoding on the FS system to play the IVR .wav files.  I tried using
speexenc to generate speex versions of the files but the audio is
garbled (sounds like a screech).  No errors or warnings are generated on
the console when playing the file.  I'm only using 8000Hz, mono files
and generate the speex file with -n (narrowband) argument.

I believe speexenc generates the files in an ogg container, so perhaps
this is the problem, but I'm not sure how to get around it if it is.

In any case, if anyone knows of a recipe for creating speex files that
mod_native_file can use, I'd appreciate a pointer.

Regards,
Cliff



From cliff at develix.com  Thu Oct  6 00:13:04 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 05 Oct 2011 13:13:04 -0700
Subject: [Freeswitch-users] mod_cdr_pg_csv not loading automatically
In-Reply-To: <1317841888.2482.9.camel@portable-evil>
References: <1317841888.2482.9.camel@portable-evil>
Message-ID: <1317845584.2482.20.camel@portable-evil>

On Wed, 2011-10-05 at 12:11 -0700, Cliff Wells wrote:
> I have this in modules.conf.xml:
> 
>     <!-- Event Handlers -->
>     <load module="mod_cdr_pg_csv"/>
>     <!-- <load module="mod_cdr_csv"/> -->
>     <!-- <load module="mod_cdr_sqlite"/> -->
>     <!-- <load module="mod_event_multicast"/> -->
>     <load module="mod_event_socket"/>
>     <!-- <load module="mod_event_zmq"/> -->
>     <!-- <load module="mod_zeroconf"/> -->
>     <!-- <load module="mod_erlang_event"/> -->
>     <!-- <load module="mod_snmp"/> -->
> 
> but mod_cdr_pg doesn't autoload.  However, if I use fs_cli and "load
> mod_cdr_pg_csv", it loads and works fine.
> 
> This is a fresh install from git yesterday.  Unfortunately I can't
> restart FS until tonight (it's a production system) to see the startup
> messages, but I wonder if anyone has seen this type of behavior (my
> older system didn't have this issue).


Actually, grepping logs revealed this:

[NOTICE] switch_event.c:1889 Event Binding deleted for mod_cdr_pg_csv:TRAP
[CRIT] switch_loadable_module.c:1275 Error Loading module /usr/local/freeswitch/mod/mod_cdr_pg_csv.so

Not sure what this means, or why I can load it after the fact.

Cliff



From cliff at develix.com  Thu Oct  6 00:20:21 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 05 Oct 2011 13:20:21 -0700
Subject: [Freeswitch-users] mod_cdr_pg_csv not loading automatically
In-Reply-To: <1317845584.2482.20.camel@portable-evil>
References: <1317841888.2482.9.camel@portable-evil>
	<1317845584.2482.20.camel@portable-evil>
Message-ID: <1317846021.2482.23.camel@portable-evil>

On Wed, 2011-10-05 at 13:13 -0700, Cliff Wells wrote:
> On Wed, 2011-10-05 at 12:11 -0700, Cliff Wells wrote:
> > I have this in modules.conf.xml:
> > 
> >     <!-- Event Handlers -->
> >     <load module="mod_cdr_pg_csv"/>
> >     <!-- <load module="mod_cdr_csv"/> -->
> >     <!-- <load module="mod_cdr_sqlite"/> -->
> >     <!-- <load module="mod_event_multicast"/> -->
> >     <load module="mod_event_socket"/>
> >     <!-- <load module="mod_event_zmq"/> -->
> >     <!-- <load module="mod_zeroconf"/> -->
> >     <!-- <load module="mod_erlang_event"/> -->
> >     <!-- <load module="mod_snmp"/> -->
> > 
> > but mod_cdr_pg doesn't autoload.  However, if I use fs_cli and "load
> > mod_cdr_pg_csv", it loads and works fine.
> > 
> > This is a fresh install from git yesterday.  Unfortunately I can't
> > restart FS until tonight (it's a production system) to see the startup
> > messages, but I wonder if anyone has seen this type of behavior (my
> > older system didn't have this issue).
> 
> 
> Actually, grepping logs revealed this:
> 
> [NOTICE] switch_event.c:1889 Event Binding deleted for mod_cdr_pg_csv:TRAP
> [CRIT] switch_loadable_module.c:1275 Error Loading module /usr/local/freeswitch/mod/mod_cdr_pg_csv.so

Sorry, missed copy/paste (it's been a long night):

[CRIT] switch_loadable_module.c:1275 Error Loading module /usr/local/freeswitch/mod/mod_cdr_pg_csv.so
**/usr/lib64/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator**

Really bizarre (to me) that the module can be loaded once FS is fully up
and running.

Cliff



From rzhang at gosilverplus.com  Thu Oct  6 01:10:48 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Wed, 05 Oct 2011 14:10:48 -0700
Subject: [Freeswitch-users] please remove me from the mailing list
In-Reply-To: <CAKzWOxV0czhxJmmv2zF=uf9dTFG9o-ZpgPHr=TuH5q4iK_evxQ@mail.gmail.com>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local>	<CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>	<9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>	<CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com>	<F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local>	<CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com>	<C57A15E034154A9297EE8DF5EF752971@ccs.local>	<CAKzWOxXJ7J+Pi2iart1nHC8-OmwHkZ0vjH3iXsKvAKGdNOt7=Q@mail.gmail.com>	<B5489AF9DCC040E2AAA414038FFBB657@ccs.local>	<99817380EE8745A5B22E84058810B705@ccs.local>
	<CAKzWOxV0czhxJmmv2zF=uf9dTFG9o-ZpgPHr=TuH5q4iK_evxQ@mail.gmail.com>
Message-ID: <4E8CC7D8.4040406@gosilverplus.com>

On 10/5/2011 11:28 AM, Michael Collins wrote:
> FYI, I added this to the queue_dtmf misc dp tools wiki page.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf#Alternate_Example_Using_execute_on_answer
>
> -MC
>
> On Tue, Oct 4, 2011 at 7:27 AM, Wayne <wayne at hamilton.net 
> <mailto:wayne at hamilton.net>> wrote:
>
>     Just to answer this posts for others that are searching for the
>     answer this is what I did to make it work.
>     For some reason after resetting my Sangoma card  with a power down
>     and restart of the server send_dtmf started working.
>     Here is  the XML that worked for me. Seems that the sleep and
>     flush also got rid of my random tones.
>     <extension name="test-tone" >
>     <condition field="destination_number" expression="1(.*)" >
>     <action application="set"
>     data="effective_caller_id_number=9025556747 <tel:9025556747>"/>
>     <action application="sleep" data="500"/>
>     <action application="flush_dtmf"/>
>     <action application="export"
>     data="nolocal:execute_on_answer=send_dtmf W$1#@500
>     <mailto:W$1#@500>" />
>     <action application="bridge" data="freetdm/wp1/a/18005551234
>     <tel:18005551234>"/>
>     </condition>
>     </extension>
>     Thanks to all that helped.
>     Wayne
>
>     *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 *Wayne
>     *Sent:* Thursday, September 15, 2011 11:55 AM
>     *To:* 'FreeSWITCH Users Help'
>     *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>     Okay,
>     I have tried send_dtmf and queue_dtmf and I can not get any DTMF
>     to pass. Would anyone be so kind and send me an extension that has
>     it working on the B-leg. I am getting nowhere.
>     Thanks Wayne
>
>     *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 *Michael Collins
>     *Sent:* Tuesday, September 13, 2011 4:29 PM
>     *To:* FreeSWITCH Users Help
>     *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>     Okay, fair enough. Not sure what queue_dtmf is not working. You
>     could always try "execute_on_answer" and execute an extension that
>     sends the dtmfs.
>
>     -MC
>
>     On Tue, Sep 13, 2011 at 2:16 PM, Wayne <wayne at hamilton.net
>     <mailto:wayne at hamilton.net>> wrote:
>
>         Sorry Michael,
>         When I started out I was sending the calls to a SIP provider.
>         I couldn't here the DTMF tones. For trouble shooting I moved
>         it to a local PRI. I was thinking that maybe this was an issue
>         with the sip trunk and sending inband or out of band but that
>         doesn't seem to be the case.
>         Wayne
>
>         *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 *Michael Collins
>         *Sent:* Tuesday, September 13, 2011 2:41 PM
>
>         *To:* FreeSWITCH Users Help
>         *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>
>
>         On Tue, Sep 13, 2011 at 11:34 AM, Wayne <wayne at hamilton.net
>         <mailto:wayne at hamilton.net>> wrote:
>
>             That's true I don't hear any tones when I answer the
>             phone. I don't see any messages in my log from queue_dtmf.
>             Any help would be great.
>             Wayne
>
>         Okay, in your original post you said you are making an OB call
>         on a SIP trunk, yet in your example there is no SIP trunk.
>         You've got an internal user and a FreeTDM FXO port. Can you
>         please clarify what you actually have? Who's the caller and
>         who's the callee?
>         -MC
>
>
>
>             *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 *Michael Collins
>             *Sent:* Thursday, September 08, 2011 6:25 PM
>
>             *To:* FreeSWITCH Users Help
>             *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>             Are you saying that no DTMFs are being sent out when the
>             bridge is initially completed?
>             -MC
>
>             On Wed, Sep 7, 2011 at 3:17 PM, Wayne <wayne at hamilton.net
>             <mailto:wayne at hamilton.net>> wrote:
>
>                 I have tried that
>                 <extension name="test-tone" >
>                 <condition field="destination_number"
>                 expression="(9025556747 <tel:%289025556747>)" >
>                 <action application="set"
>                 data="call_direction=outbound" />
>                 <action application="set"
>                 data="effective_caller_id_name=${outbound_caller_id_name}"
>                 />
>                 <action application="set"
>                 data="effective_caller_id_number=${outbound_caller_id_number}"
>                 />
>                 <action application="set" data="continue_on_fail=true" />
>                 <action application="queue_dtmf" data="0123456789" />
>                 <action application="bridge" data="freetdm/wp1/a/$1" />
>                 </condition>
>                 </extension>
>                 The debug tells me that this
>                 tail -f freeswitch.log | grep -i dtmf
>                 Dialplan: sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Action
>                 queue_dtmf(0123456789)
>                 EXECUTE sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87>
>                 queue_dtmf(0123456789)
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:44.676204 [DEBUG]
>                 switch_channel.c:465 sofia/internal/1333 at 192.168.48.87
>                 <mailto:sofia/internal/1333 at 192.168.48.87> Queue dtmf
>                 2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3355
>                 Set 2833 dtmf send payload to 101
>                 2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3360
>                 Set 2833 dtmf receive payload to 101
>                 2011-09-07 17:13:45.116216 [DEBUG] ftdm_io.c:3714
>                 [s1c1][1:1] Generating DTMF [0123456789]
>                 2011-09-07 17:13:47.876238 [DEBUG] ftmod_wanpipe.c:701
>                 [s1c1][1:1] Enabled DTMF events
>                 2011-09-07 17:13:47.956228 [DEBUG]
>                 ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing wanpipe DTMF: 9
>                 2011-09-07 17:13:47.956228 [DEBUG] ftdm_io.c:3524
>                 [s1c1][1:1] Queuing DTMF 9 (debug = 0)
>                 2011-09-07 17:13:47.956228 [DEBUG] mod_freetdm.c:733
>                 Queuing DTMF [9] in channel FreeTDM/1:1/9025556747
>                 <tel:1%2F9025556747>
>                 2011-09-07 17:13:48.036235 [DEBUG]
>                 ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing wanpipe DTMF: 9
>                 2011-09-07 17:13:48.036235 [DEBUG] ftdm_io.c:3524
>                 [s1c1][1:1] Queuing DTMF 9 (debug = 0)
>                 2011-09-07 17:13:48.036235 [DEBUG] mod_freetdm.c:733
>                 Queuing DTMF [9] in channel FreeTDM/1:1/9025556747
>                 <tel:1%2F9025556747>
>                 So where would I look next.
>                 Wayne
>
>
>                 *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 *Avi Marcus
>                 *Sent:* Wednesday, September 07, 2011 3:55 PM
>                 *To:* FreeSWITCH Users Help
>                 *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>                 I think you looking for queue_dtmf.
>
>                 http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
>
>                 -Avi Marcus
>
>
>
>                 On Wed, Sep 7, 2011 at 11:47 PM, Wayne
>                 <wayne at hamilton.net <mailto:wayne at hamilton.net>> wrote:
>                 > Hello All,
>                 >
>                 > I need to call out on a SIP trunk and when the call
>                 is answered send DTMF
>                 > tones.
>                 > I need to send the DTMF only on the outbound leg.
>                 >
>                 > Does anyone have a dialplan that will do that? Is it
>                 possible. I have only
>                 > found one thread on it and did get much out of it.
>                 >
>                 > Thanks
>                 > Wayne
>                 >
>                 >
>                 >
>                 > FreeSWITCH-users mailing list
>                 > FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 >
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 >
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 > http://www.freeswitch.org
>                 >
>
>
>
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>
>
>
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org
>
>
>
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Oct  6 01:15:12 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 05 Oct 2011 16:15:12 -0500
Subject: [Freeswitch-users] please remove me from the mailing list
In-Reply-To: <4E8CC7D8.4040406@gosilverplus.com>
Message-ID: <CAB23310.1ED98%krice@freeswitch.org>

Ok, try reading at the bottom of every email that goes out on the lists...
Quoted below so it can be easily found


On 10/5/11 4:10 PM, "ran zhang" <rzhang at gosilverplus.com> wrote:

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

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From msc at freeswitch.org  Thu Oct  6 01:17:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Oct 2011 14:17:43 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Conference Call
 Today: Git Tutorial
In-Reply-To: <CALXCt0o9JF0OCSZ55D5J6Yt_g5_Mw8PczzPfU5wY9zz7OtGbMg@mail.gmail.com>
References: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>
	<CALXCt0o9JF0OCSZ55D5J6Yt_g5_Mw8PczzPfU5wY9zz7OtGbMg@mail.gmail.com>
Message-ID: <CAKzWOxXhDbvtsxSM7vpu8OQmGDh_0UiuBaFqX9qb2rDTAUYeTA@mail.gmail.com>

FYI, the recordings are up on the wiki!

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

-MC

On Wed, Oct 5, 2011 at 10:00 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> I'll try to join from cell, but please please please, do a recording anyway
> ;).
>
> -giovanni
>
>
> On 10/5/11, Michael Collins <msc at freeswitch.org> wrote:
> > Hello all!
> >
> > Today we have our very own Travis Cross who will be helping everyone get
> up
> > to speed with git. We highly recommend everyone who uses the FreeSWITCH
> > source to join us and get your git questions answered. The agenda page is
> > here:
> >
> > http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05
> >
> > Talk to you soon,
> > Michael
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darknesslabs at gmail.com  Thu Oct  6 01:55:12 2011
From: darknesslabs at gmail.com (Karol)
Date: Wed, 5 Oct 2011 17:55:12 -0400
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Conference Call
 Today: Git Tutorial
In-Reply-To: <CAKzWOxXhDbvtsxSM7vpu8OQmGDh_0UiuBaFqX9qb2rDTAUYeTA@mail.gmail.com>
References: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>
	<CALXCt0o9JF0OCSZ55D5J6Yt_g5_Mw8PczzPfU5wY9zz7OtGbMg@mail.gmail.com>
	<CAKzWOxXhDbvtsxSM7vpu8OQmGDh_0UiuBaFqX9qb2rDTAUYeTA@mail.gmail.com>
Message-ID: <CAFZs+v8BREg1xof6c+Dsqk9RbLK-B9ji79STa0e8p9QF-oH5mA@mail.gmail.com>

Files aren't found.

On Oct 5, 2011 5:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
> FYI, the recordings are up on the wiki!
>
> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls
>
> -MC
>
>
> On Wed, Oct 5, 2011 at 10:00 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:
>>
>> I'll try to join from cell, but please please please, do a recording
anyway ;).
>>
>> -giovanni
>>
>>
>> On 10/5/11, Michael Collins <msc at freeswitch.org> wrote:
>> > Hello all!
>> >
>> > Today we have our very own Travis Cross who will be helping everyone
get up
>> > to speed with git. We highly recommend everyone who uses the FreeSWITCH
>> > source to join us and get your git questions answered. The agenda page
is
>> > here:
>> >
>> > http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05
>> >
>> > Talk to you soon,
>> > Michael
>> >
>>
>> --
>> Sent from my mobile device
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct  6 03:19:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Oct 2011 16:19:01 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Conference Call
 Today: Git Tutorial
In-Reply-To: <CAFZs+v8BREg1xof6c+Dsqk9RbLK-B9ji79STa0e8p9QF-oH5mA@mail.gmail.com>
References: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>
	<CALXCt0o9JF0OCSZ55D5J6Yt_g5_Mw8PczzPfU5wY9zz7OtGbMg@mail.gmail.com>
	<CAKzWOxXhDbvtsxSM7vpu8OQmGDh_0UiuBaFqX9qb2rDTAUYeTA@mail.gmail.com>
	<CAFZs+v8BREg1xof6c+Dsqk9RbLK-B9ji79STa0e8p9QF-oH5mA@mail.gmail.com>
Message-ID: <CAKzWOxV-4ASRG3UNB1iJXK9DrU-_PiP_3m_JRGi=uEh_E_Ondg@mail.gmail.com>

Sorry, I typo'd the filenames when I created the links. All better now.
-MC

On Wed, Oct 5, 2011 at 2:55 PM, Karol <darknesslabs at gmail.com> wrote:

> Files aren't found.
>
> On Oct 5, 2011 5:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
> >
> > FYI, the recordings are up on the wiki!
> >
> > http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls
> >
> > -MC
> >
> >
> > On Wed, Oct 5, 2011 at 10:00 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> >>
> >> I'll try to join from cell, but please please please, do a recording
> anyway ;).
> >>
> >> -giovanni
> >>
> >>
> >> On 10/5/11, Michael Collins <msc at freeswitch.org> wrote:
> >> > Hello all!
> >> >
> >> > Today we have our very own Travis Cross who will be helping everyone
> get up
> >> > to speed with git. We highly recommend everyone who uses the
> FreeSWITCH
> >> > source to join us and get your git questions answered. The agenda page
> is
> >> > here:
> >> >
> >> > http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05
> >> >
> >> > Talk to you soon,
> >> > Michael
> >> >
> >>
> >> --
> >> Sent from my mobile device
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Thu Oct  6 08:42:52 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 6 Oct 2011 00:42:52 -0400
Subject: [Freeswitch-users] mod_native_file and speex
In-Reply-To: <1317842469.2482.16.camel@portable-evil>
References: <1317842469.2482.16.camel@portable-evil>
Message-ID: <55E34A90-B971-4C7F-8DA4-89E1B15EDF6E@avgs.ca>

Hi,

I made a small tool to encode files for use with mod_native_file, it's being compiled automatically as part of the normal install process.

Try that:
fs_encode -l mod_speex file.wav file.SPEEX

PS: The binary sits right next to fs_cli and freeswitch

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




On 2011-10-05, at 3:21 PM, Cliff Wells wrote:

> I'm using Speex to bridge calls between an out-of-state Asterisk box and
> a local FreeSWITCH IVR.  This works great, but I'm having to do needless
> transcoding on the FS system to play the IVR .wav files.  I tried using
> speexenc to generate speex versions of the files but the audio is
> garbled (sounds like a screech).  No errors or warnings are generated on
> the console when playing the file.  I'm only using 8000Hz, mono files
> and generate the speex file with -n (narrowband) argument.
> 
> I believe speexenc generates the files in an ogg container, so perhaps
> this is the problem, but I'm not sure how to get around it if it is.
> 
> In any case, if anyone knows of a recipe for creating speex files that
> mod_native_file can use, I'd appreciate a pointer.
> 
> Regards,
> Cliff
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cliff at develix.com  Thu Oct  6 09:08:45 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 05 Oct 2011 22:08:45 -0700
Subject: [Freeswitch-users] mod_native_file and speex
In-Reply-To: <55E34A90-B971-4C7F-8DA4-89E1B15EDF6E@avgs.ca>
References: <1317842469.2482.16.camel@portable-evil>
	<55E34A90-B971-4C7F-8DA4-89E1B15EDF6E@avgs.ca>
Message-ID: <1317877725.2256.0.camel@portable-evil>

Hi Mathieu,

I will give that a try, thanks!

Cliff

On Thu, 2011-10-06 at 00:42 -0400, Mathieu Rene wrote:
> Hi,
> 
> I made a small tool to encode files for use with mod_native_file, it's being compiled automatically as part of the normal install process.
> 
> Try that:
> fs_encode -l mod_speex file.wav file.SPEEX
> 
> PS: The binary sits right next to fs_cli and freeswitch
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2011-10-05, at 3:21 PM, Cliff Wells wrote:
> 
> > I'm using Speex to bridge calls between an out-of-state Asterisk box and
> > a local FreeSWITCH IVR.  This works great, but I'm having to do needless
> > transcoding on the FS system to play the IVR .wav files.  I tried using
> > speexenc to generate speex versions of the files but the audio is
> > garbled (sounds like a screech).  No errors or warnings are generated on
> > the console when playing the file.  I'm only using 8000Hz, mono files
> > and generate the speex file with -n (narrowband) argument.
> > 
> > I believe speexenc generates the files in an ogg container, so perhaps
> > this is the problem, but I'm not sure how to get around it if it is.
> > 
> > In any case, if anyone knows of a recipe for creating speex files that
> > mod_native_file can use, I'd appreciate a pointer.
> > 
> > Regards,
> > Cliff
> > 
> > 
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 beffa at ieee.org  Thu Oct  6 11:18:23 2011
From: beffa at ieee.org (Federico Beffa)
Date: Thu, 6 Oct 2011 09:18:23 +0200
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <CAKrPhPMA3cngUy9hkNyucbzOv=GwAjN6qWUQNFJ=K4+koEcAEA@mail.gmail.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
	<1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
	<CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>
	<1317308654166-6844212.post@n2.nabble.com>
	<CAKrPhPMA3cngUy9hkNyucbzOv=GwAjN6qWUQNFJ=K4+koEcAEA@mail.gmail.com>
Message-ID: <CAKrPhPM4MV5XQ8oBop-PvwAZbMQM9S422wH-UMDK3n1whvVf+w@mail.gmail.com>

Unfortunately I was singing victory too early. My problem still persists...

The git version I'm running is from the 20th of August. In the mean
time have there been changes which could affect mod_dingaling? Looking
in the git commit comments I have the impression that the answer is
no.

Regards,
Fede

On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa at ieee.org> wrote:
> Just in case someone will hit the same problem...
>
> After some tests, I've found that, changing "bind_server_ip" from
> "auto" to the local IP address of the FS server (192.168.X.X), appears
> to solve my timeout problem. ?(The account did not go offline for more
> than 12 hours.)
>
> To summarize my current configuration: I have a fix public IP assigned
> to my router (NAT/Firewall). NAT is configured to forward ports 5080
> and 5081 to the host on which FS is running and UPnP is disabled.
>
> I'm running a FS version checked out from git the 20th of August.
>
> Regards,
> Fede
>
> On Thu, Sep 29, 2011 at 5:04 PM, mazilo
> <Nabble_01394 at slickdeals.endjunk.com> wrote:
>> The only thing I can say is to update your FS with the latest git commit.
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
>> --
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-offline-mod-dingaling-tp6839355p6844212.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From cliff at develix.com  Thu Oct  6 12:09:25 2011
From: cliff at develix.com (Cliff Wells)
Date: Thu, 06 Oct 2011 01:09:25 -0700
Subject: [Freeswitch-users] mod_native_file and speex
In-Reply-To: <55E34A90-B971-4C7F-8DA4-89E1B15EDF6E@avgs.ca>
References: <1317842469.2482.16.camel@portable-evil>
	<55E34A90-B971-4C7F-8DA4-89E1B15EDF6E@avgs.ca>
Message-ID: <1317888565.2256.4.camel@portable-evil>

On Thu, 2011-10-06 at 00:42 -0400, Mathieu Rene wrote:
> Hi,
> 
> I made a small tool to encode files for use with mod_native_file, it's being compiled automatically as part of the normal install process.
> 
> Try that:
> fs_encode -l mod_speex file.wav file.SPEEX

Similar result: file appears to play successfully, but only silence is
heard.  As you can see, it gets done playing rather quickly (message is
actually about 20s in duration).

2011-10-05 23:59:37.357566 [INFO] mod_native_file.c:94 Opening File [/usr/local/freeswitch/scripts/jovan-message.SPEEX] 8000hz
2011-10-05 23:59:37.357566 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms
2011-10-05 23:59:37.357566 [DEBUG] switch_ivr_play_say.c:1672 done playing file


Cliff





From Nabble_01394 at slickdeals.endjunk.com  Thu Oct  6 17:43:21 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Thu, 6 Oct 2011 06:43:21 -0700 (PDT)
Subject: [Freeswitch-users] 3pcc
Message-ID: <1317908601432-6865965.post@n2.nabble.com>

On Line #66 of the /conf/sip_profiles/external.xml/ file, the 3pcc global
option is commented out by default. I don't know if this 3pcc is a
/fortunate/ or /unfortunate/ option to have, but it will make 
http://whistlephone.com WhistlePhone  to work properly. Without the 3pcc
option, there will be no incoming calls from  http://whistlephone.com
WhistlePhone  (FS CLI will dump some messages as shown below) unless FS
registers to all  http://whistlephone.com WhistlePhone  proxy servers as
shown in 
http://wiki.freeswitch.org/wiki/Provider_Configuration:_WhistlePhone here .
The question I have is if it is safe to enable the 3pcc option. If it is,
then I don't mind to go ahead to update the wiki page.



-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/3pcc-tp6865965p6865965.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Thu Oct  6 18:11:36 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 6 Oct 2011 07:11:36 -0700
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <CAKrPhPM4MV5XQ8oBop-PvwAZbMQM9S422wH-UMDK3n1whvVf+w@mail.gmail.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
	<1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
	<CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>
	<1317308654166-6844212.post@n2.nabble.com>
	<CAKrPhPMA3cngUy9hkNyucbzOv=GwAjN6qWUQNFJ=K4+koEcAEA@mail.gmail.com>
	<CAKrPhPM4MV5XQ8oBop-PvwAZbMQM9S422wH-UMDK3n1whvVf+w@mail.gmail.com>
Message-ID: <CAMWiMy9Oew+MsMFHp8+XsBTxhya3ba9yiQ7sGMWS4RRgK2KpCg@mail.gmail.com>

Changes in the FreeSWITCH core could affect the way how mod_dingaling
functions. You might want to give the latest git a shot and see what
happens. Personally, updating to the latest git isn't the solution to
all problems ;)

On Thu, Oct 6, 2011 at 12:18 AM, Federico Beffa <beffa at ieee.org> wrote:
> Unfortunately I was singing victory too early. My problem still persists...
>
> The git version I'm running is from the 20th of August. In the mean
> time have there been changes which could affect mod_dingaling? Looking
> in the git commit comments I have the impression that the answer is
> no.
>
> Regards,
> Fede
>
> On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa at ieee.org> wrote:
>> Just in case someone will hit the same problem...
>>
>> After some tests, I've found that, changing "bind_server_ip" from
>> "auto" to the local IP address of the FS server (192.168.X.X), appears
>> to solve my timeout problem. ?(The account did not go offline for more
>> than 12 hours.)
>>
>> To summarize my current configuration: I have a fix public IP assigned
>> to my router (NAT/Firewall). NAT is configured to forward ports 5080
>> and 5081 to the host on which FS is running and UPnP is disabled.
>>
>> I'm running a FS version checked out from git the 20th of August.
>>
>> Regards,
>> Fede
>>
>> On Thu, Sep 29, 2011 at 5:04 PM, mazilo
>> <Nabble_01394 at slickdeals.endjunk.com> wrote:
>>> The only thing I can say is to update your FS with the latest git commit.
>>>
>>> -----
>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
>>> --
>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-offline-mod-dingaling-tp6839355p6844212.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 johnrose at google.hm  Thu Oct  6 20:27:17 2011
From: johnrose at google.hm (John Rose)
Date: Thu, 6 Oct 2011 10:27:17 -0600
Subject: [Freeswitch-users] Playing with mod_sms
Message-ID: <008e01cc8444$d10c0450$73240cf0$@google.hm>

 

It's awesome that we have this module now.  Kudos to all those that worked
on it.

 

I'm going to start playing with this a bit. Currently I'm using some of the
SIP MESSAGE functionally with a tweaked Freeswitch 1.0.6+ sofia_presence.c
to receive SIP MESSAGE ESL events from unsolicited addresses and have to
figure out somehow on confirming that 200 OK's were received on MESSAGE send
and not failures, a 404 or whatever. Hopefully this new module will help
with that.

 

Any pointers would be appreciated.

 

John

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From leon at scarlet-internet.nl  Thu Oct  6 18:54:30 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 6 Oct 2011 16:54:30 +0200
Subject: [Freeswitch-users] sofia auth challenge loglevel warning to debug10
	& fail2ban
Message-ID: <EF52D25A-E690-463C-AFB7-3FA247C9981B@scarlet-internet.nl>

Hi all,

I noticed that the loglevel for sofia auth challenge was changed from warning to debug10 about two weeks ago - which I really like btw, it shouldn't be a warning

http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/endpoints/mod_sofia/sofia_reg.c?r1=1b9b3456ad01e93bcefe8f5f67a19307c3bad808&r2=7333d46d5b90bc5d94a2bf83169997f2c2322112

But doesn't this change break fail2ban for the ppl who use that ? The regex should at least be changed on the wiki:

http://wiki.freeswitch.org/wiki/Fail2ban#Edit_Config_Files

But I think most people don't log debug level to files, so fail2ban won't be triggered even if the regex is changed, right ?

Don't know if it's a problem, maybe I'm overlooking something, just wanted to let everyone know..

regards,

Leon
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From popov-k at speechpro.com  Thu Oct  6 13:44:24 2011
From: popov-k at speechpro.com (Kirill Popov)
Date: Thu, 06 Oct 2011 13:44:24 +0400
Subject: [Freeswitch-users] How I can get recog result
Message-ID: <4E8D7878.3060109@speechpro.com>

Hello!

I am just starting to learn FreeSWITCH few days ago.
Before I have some experience in Asterisk and now I don't understand 
somethings.

I try to use module mod_unimrcp in FreeSWITCH.
I made simple dialplan and starting recognition
<action application="detect_speech" data="unimrcp /usr/local/
freeswitch/grammar/digits"/>
It works.

I see NLSML in console FS, but I don't understand how I can use it.
In Asterisk NLSML-answer set in variable ${RECOG_RESULT}. How in FS?
There isn't such variable.

In Asterisk I can get this variable from perl-script, parse it, and
get utterance, confidence, etc. How I can get these parameters in FS?


-- 
Best regards,
Kirill


From Briac.GAUTIERDECHARNACE at supinfo.com  Thu Oct  6 13:23:31 2011
From: Briac.GAUTIERDECHARNACE at supinfo.com (=?iso-8859-1?Q?GAUTIER_DE_CHARNAC=C9_Briac?=)
Date: Thu, 6 Oct 2011 09:23:31 +0000
Subject: [Freeswitch-users] Freeswitch mod_lua and email
In-Reply-To: <9CE5761907AF3242B8E5F41A2F99BBB20884DF@AMSPRD0104MB137.eurprd01.prod.exchangelabs.com>
References: <9CE5761907AF3242B8E5F41A2F99BBB20884CA@AMSPRD0104MB137.eurprd01.prod.exchangelabs.com>,
	<9CE5761907AF3242B8E5F41A2F99BBB20884DF@AMSPRD0104MB137.eurprd01.prod.exchangelabs.com>
Message-ID: <9CE5761907AF3242B8E5F41A2F99BBB2088509@AMSPRD0104MB137.eurprd01.prod.exchangelabs.com>

Hello,
I am trying to send an email from a lua script with this line :

freeswitch.email(to, from, headers, body, file, convert_cmd, convert_ext)

But when I execute the script I have an error :

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

011-10-06 11:01:18.320247 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/voicemail.lua:43: attempt to call field 'email' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/voicemail.lua:43: in main chunk

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

Do you have an idea how solve this problem ?

Thanks

Briac GAUTIER DE CHARNACE
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From x.liu at hw.ac.uk  Thu Oct  6 19:30:18 2011
From: x.liu at hw.ac.uk (xl127)
Date: Thu, 06 Oct 2011 16:30:18 +0100
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not work as
	expected
Message-ID: <4E8DC98A.8040408@hw.ac.uk>

Hello,

I am trying to bridge multiple calls to multiple Java ESL outbound 
socket applications.

Basically I want to check if the first socket app is engaged then the 
call should be transferred
to the second socket app and so on.

For initial test, I tried two concurrent calls, the problem is both 
calls are always answered
by the first outbound socket.

I have following setting in the dialplan:

<extension name="sipgateDialedIn">
<condition field="destination_number" expression="^5800$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="loopback/5801"/>
<action application="bridge" data="loopback/5802"/>

<action application="set" data="tts_engine=flite"/>
<action application="set" data="tts_voice=kal"/>
<action application="speak" data="Sorry, I am busy at the moment. Please 
try again later."/>
<action application="sleep" data="10"/>
<action application="hangup"/>
</condition>
</extension>

<extension name="myOutboundESL5801">
<condition field="destination_number" expression="^5801$">
<action application="socket" data="127.0.0.1:9981 async full"/>
</condition>
</extension>

<extension name="myOutboundESL5802">
<condition field="destination_number" expression="^5802$">
<action application="socket" data="127.0.0.1:9982 async full"/>
</condition>
</extension>

I expect the first call to 5800 would be asnwered by extension 5801, the 
second concurrent call to 5800 will be
answered by extension 5802, and the third call gets a busy message.

I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if 
the socket is engaged?

I've tried and googled a while but without success.

Any advices please?

Many thanks!
Xing



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012




From anthony.minessale at gmail.com  Thu Oct  6 19:43:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Oct 2011 10:43:39 -0500
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <4E8DC98A.8040408@hw.ac.uk>
References: <4E8DC98A.8040408@hw.ac.uk>
Message-ID: <CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>

Really the socket app has not limit to how many simultaneous
connections it can handle.
You could try mod_limit, or make the remote end of you application
fail when there are too many connections.


On Thu, Oct 6, 2011 at 10:30 AM, xl127 <x.liu at hw.ac.uk> wrote:
> Hello,
>
> I am trying to bridge multiple calls to multiple Java ESL outbound
> socket applications.
>
> Basically I want to check if the first socket app is engaged then the
> call should be transferred
> to the second socket app and so on.
>
> For initial test, I tried two concurrent calls, the problem is both
> calls are always answered
> by the first outbound socket.
>
> I have following setting in the dialplan:
>
> <extension name="sipgateDialedIn">
> <condition field="destination_number" expression="^5800$">
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="bridge" data="loopback/5801"/>
> <action application="bridge" data="loopback/5802"/>
>
> <action application="set" data="tts_engine=flite"/>
> <action application="set" data="tts_voice=kal"/>
> <action application="speak" data="Sorry, I am busy at the moment. Please
> try again later."/>
> <action application="sleep" data="10"/>
> <action application="hangup"/>
> </condition>
> </extension>
>
> <extension name="myOutboundESL5801">
> <condition field="destination_number" expression="^5801$">
> <action application="socket" data="127.0.0.1:9981 async full"/>
> </condition>
> </extension>
>
> <extension name="myOutboundESL5802">
> <condition field="destination_number" expression="^5802$">
> <action application="socket" data="127.0.0.1:9982 async full"/>
> </condition>
> </extension>
>
> I expect the first call to 5800 would be asnwered by extension 5801, the
> second concurrent call to 5800 will be
> answered by extension 5802, and the third call gets a busy message.
>
> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
> the socket is engaged?
>
> I've tried and googled a while but without success.
>
> Any advices please?
>
> Many thanks!
> Xing
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
> Heriot-Watt University is the Sunday Times
> Scottish University of the Year 2011-2012
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Oct  6 19:45:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Oct 2011 10:45:11 -0500
Subject: [Freeswitch-users] sofia auth challenge loglevel warning to
 debug10 & fail2ban
In-Reply-To: <EF52D25A-E690-463C-AFB7-3FA247C9981B@scarlet-internet.nl>
References: <EF52D25A-E690-463C-AFB7-3FA247C9981B@scarlet-internet.nl>
Message-ID: <CAKbxfG89PhWQDXs9N8qbdZeR7TR71MOdTfTrb-XgRE7HxCS6tA@mail.gmail.com>

Yes its currently an issue we are trying to decide about.  See our homepage.


On Thu, Oct 6, 2011 at 9:54 AM, Leon de Rooij <leon at scarlet-internet.nl> wrote:
> Hi all,
> I noticed that the loglevel for sofia auth challenge was changed from
> warning to debug10 about two weeks ago - which I really like btw, it
> shouldn't be a warning
> http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/endpoints/mod_sofia/sofia_reg.c?r1=1b9b3456ad01e93bcefe8f5f67a19307c3bad808&r2=7333d46d5b90bc5d94a2bf83169997f2c2322112
> But doesn't this change break fail2ban for the ppl who use that ? The regex
> should at least be changed on the wiki:
> http://wiki.freeswitch.org/wiki/Fail2ban#Edit_Config_Files
> But I think most people don't log debug level to files, so fail2ban won't be
> triggered even if the regex is changed, right ?
> Don't know if it's a problem, maybe I'm overlooking something, just wanted
> to let everyone know..
> regards,
> Leon
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From infos at madovsky.org  Thu Oct  6 19:49:41 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 Oct 2011 11:49:41 -0400
Subject: [Freeswitch-users] RTCP packets not written
Message-ID: <2E3C85AE646E4480BF01EB16FB5C82CC@e1705>

if I understand the RFC,
RTCP is only for RTP application statistics so
it's not really necessary ?

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, October 04, 2011 6:26 PM
  Subject: RTCP packets not written


  Hi,

  I noticed 

  2011-10-04 14:24:28.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:24:33.512388 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:24:38.512378 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:24:43.512390 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:24:48.512362 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:24:53.512375 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:24:58.512364 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:03.512369 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:08.512386 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:13.512390 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:18.512364 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:23.512376 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:28.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:33.512365 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:38.512368 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:43.512372 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:48.512366 [DEBUG] switch_rtp.c:3910 RTCP packet not written
  2011-10-04 14:25:53.512367 [DEBUG] switch_rtp.c:3910 RTCP packet not written


  is it come from a permission file somewhere ?

  Thanks
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From victor.chukalovskiy at utoronto.ca  Thu Oct  6 19:51:05 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 06 Oct 2011 11:51:05 -0400
Subject: [Freeswitch-users] Adding new MOH stream - isn't reloadxml
 enough to make it effective?
In-Reply-To: <CAKzWOxXnprV4vm5EBHs2ggd3S8kfkk9JyPPB4zbcUw9VuQRrQw@mail.gmail.com>
References: <4E8B4884.90703@utoronto.ca>
	<CAKzWOxXnprV4vm5EBHs2ggd3S8kfkk9JyPPB4zbcUw9VuQRrQw@mail.gmail.com>
Message-ID: <4E8DCE69.3050600@utoronto.ca>

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From yehavi.bourvine at gmail.com  Thu Oct  6 20:28:16 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 6 Oct 2011 18:28:16 +0200
Subject: [Freeswitch-users] callcenter: getting the number of users waiting
Message-ID: <CAAtWKeEe21M=gomkWw+sF5ZfbXTF+WbTidJ=rtnH5G0-dEc7eA@mail.gmail.com>

Hello,

  I am implementing a callcenter using the mod_callcenter. I would like the
user to hear some text/music while waiting and from time to time to announce
his/her place in the queue.
Is it possible by some simple mean, or do I have to use an external
application with uuid_broadcast to do that? Is there a table that holds the
list of currently waiting incoming calls?

                                Thanks, __Yehavi:
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From x.liu at hw.ac.uk  Thu Oct  6 20:28:41 2011
From: x.liu at hw.ac.uk (xl127)
Date: Thu, 06 Oct 2011 17:28:41 +0100
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
Message-ID: <4E8DD739.5090104@hw.ac.uk>

Thanks for the suggestions!

I thought about and tried to make my outbound socket application
fail e.g. by send a hangup message but it didn't work.

What I did is that I set a public variable channelUUID with initial 
value null in MyOutboundHandler,
I check if it is null in the handleConnectionResponse method. If it not 
null, a hangup command is sent to FS.
But it looks like each call channel creates its own Java thread, so the 
channelUUID is always null before the connection.
So my approach doesn't work.

Any idea about checking if the remote application is answering a call?

Thanks!
Xing


On 06/10/11 16:43, Anthony Minessale wrote:
> Really the socket app has not limit to how many simultaneous
> connections it can handle.
> You could try mod_limit, or make the remote end of you application
> fail when there are too many connections.
>
>
> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk>  wrote:
>> Hello,
>>
>> I am trying to bridge multiple calls to multiple Java ESL outbound
>> socket applications.
>>
>> Basically I want to check if the first socket app is engaged then the
>> call should be transferred
>> to the second socket app and so on.
>>
>> For initial test, I tried two concurrent calls, the problem is both
>> calls are always answered
>> by the first outbound socket.
>>
>> I have following setting in the dialplan:
>>
>> <extension name="sipgateDialedIn">
>> <condition field="destination_number" expression="^5800$">
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="bridge" data="loopback/5801"/>
>> <action application="bridge" data="loopback/5802"/>
>>
>> <action application="set" data="tts_engine=flite"/>
>> <action application="set" data="tts_voice=kal"/>
>> <action application="speak" data="Sorry, I am busy at the moment. Please
>> try again later."/>
>> <action application="sleep" data="10"/>
>> <action application="hangup"/>
>> </condition>
>> </extension>
>>
>> <extension name="myOutboundESL5801">
>> <condition field="destination_number" expression="^5801$">
>> <action application="socket" data="127.0.0.1:9981 async full"/>
>> </condition>
>> </extension>
>>
>> <extension name="myOutboundESL5802">
>> <condition field="destination_number" expression="^5802$">
>> <action application="socket" data="127.0.0.1:9982 async full"/>
>> </condition>
>> </extension>
>>
>> I expect the first call to 5800 would be asnwered by extension 5801, the
>> second concurrent call to 5800 will be
>> answered by extension 5802, and the third call gets a busy message.
>>
>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>> the socket is engaged?
>>
>> I've tried and googled a while but without success.
>>
>> Any advices please?
>>
>> Many thanks!
>> Xing
>>
>>
>>
>> --
>> Heriot-Watt University is a Scottish charity
>> registered under charity number SC000278.
>>
>> Heriot-Watt University is the Sunday Times
>> Scottish University of the Year 2011-2012
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012




From msc at freeswitch.org  Thu Oct  6 21:08:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Oct 2011 10:08:48 -0700
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <4E8DD739.5090104@hw.ac.uk>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
Message-ID: <CAKzWOxUTR3EOr+fUaC3RXXcz7G5Rq49S4ZgiDbTb1WfeDLqxgg@mail.gmail.com>

Perhaps you could rely on FreeSWITCH's global_setvar and global_getvar APIs.
You could perhaps create a global var whose name is the uuid and then put
some meaningful value in it. You would just need to make absolutely certain
that you clear out that variable manually, otherwise you'll have a lot of
junk variables.

-MC

On Thu, Oct 6, 2011 at 9:28 AM, xl127 <x.liu at hw.ac.uk> wrote:

> Thanks for the suggestions!
>
> I thought about and tried to make my outbound socket application
> fail e.g. by send a hangup message but it didn't work.
>
> What I did is that I set a public variable channelUUID with initial
> value null in MyOutboundHandler,
> I check if it is null in the handleConnectionResponse method. If it not
> null, a hangup command is sent to FS.
> But it looks like each call channel creates its own Java thread, so the
> channelUUID is always null before the connection.
> So my approach doesn't work.
>
> Any idea about checking if the remote application is answering a call?
>
> Thanks!
> Xing
>
>
> On 06/10/11 16:43, Anthony Minessale wrote:
> > Really the socket app has not limit to how many simultaneous
> > connections it can handle.
> > You could try mod_limit, or make the remote end of you application
> > fail when there are too many connections.
> >
> >
> > On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk>  wrote:
> >> Hello,
> >>
> >> I am trying to bridge multiple calls to multiple Java ESL outbound
> >> socket applications.
> >>
> >> Basically I want to check if the first socket app is engaged then the
> >> call should be transferred
> >> to the second socket app and so on.
> >>
> >> For initial test, I tried two concurrent calls, the problem is both
> >> calls are always answered
> >> by the first outbound socket.
> >>
> >> I have following setting in the dialplan:
> >>
> >> <extension name="sipgateDialedIn">
> >> <condition field="destination_number" expression="^5800$">
> >> <action application="set" data="continue_on_fail=true"/>
> >> <action application="set" data="hangup_after_bridge=true"/>
> >> <action application="bridge" data="loopback/5801"/>
> >> <action application="bridge" data="loopback/5802"/>
> >>
> >> <action application="set" data="tts_engine=flite"/>
> >> <action application="set" data="tts_voice=kal"/>
> >> <action application="speak" data="Sorry, I am busy at the moment. Please
> >> try again later."/>
> >> <action application="sleep" data="10"/>
> >> <action application="hangup"/>
> >> </condition>
> >> </extension>
> >>
> >> <extension name="myOutboundESL5801">
> >> <condition field="destination_number" expression="^5801$">
> >> <action application="socket" data="127.0.0.1:9981 async full"/>
> >> </condition>
> >> </extension>
> >>
> >> <extension name="myOutboundESL5802">
> >> <condition field="destination_number" expression="^5802$">
> >> <action application="socket" data="127.0.0.1:9982 async full"/>
> >> </condition>
> >> </extension>
> >>
> >> I expect the first call to 5800 would be asnwered by extension 5801, the
> >> second concurrent call to 5800 will be
> >> answered by extension 5802, and the third call gets a busy message.
> >>
> >> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
> >> the socket is engaged?
> >>
> >> I've tried and googled a while but without success.
> >>
> >> Any advices please?
> >>
> >> Many thanks!
> >> Xing
> >>
> >>
> >>
> >> --
> >> Heriot-Watt University is a Scottish charity
> >> registered under charity number SC000278.
> >>
> >> Heriot-Watt University is the Sunday Times
> >> Scottish University of the Year 2011-2012
> >>
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
> Heriot-Watt University is the Sunday Times
> Scottish University of the Year 2011-2012
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From buscom123+fs at gmail.com  Thu Oct  6 21:25:01 2011
From: buscom123+fs at gmail.com (R H)
Date: Thu, 6 Oct 2011 11:25:01 -0600
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAAtQP57w_FgGCFaSA3Sxooky0kTyoLdAkfjv227=AxxO=nBs9g@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
	<CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>
	<CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>
	<CAKbxfG9Msun7_rdMEoSHk-rtpZ_bVJmdUoHWxFBDHDC5U2EaFg@mail.gmail.com>
	<CAAtQP57w_FgGCFaSA3Sxooky0kTyoLdAkfjv227=AxxO=nBs9g@mail.gmail.com>
Message-ID: <CAAtQP54nnfKtixAS-+r-mJv4hW==f=M8MWUy7GML0bmsn3VR0w@mail.gmail.com>

For the benefit of anyone who may run into this in the future. I resolved
the issue by providing each sip_profile it's own database schema. This does
not seem like the best solution but it works.

Thanks for the feedback!

Ryan


On Wed, Oct 5, 2011 at 11:54 AM, R H <buscom123+fs at gmail.com> wrote:

> Hey Anthony, I appreciate your help on this. Your response was not clear
> enough for me to understand how to accomplish what you are describing. Let's
> assume I only have 2 profiles (Internal/External). Both profiles default to
> the public context.
>
> Now if I were to create two domains in the directory and name them "
> internal.mydomain.com" and "external.mydomain.com" and force everyone who
> registers to the internal profile to use the internal domain and everyone on
> the external profile to use the external domain I would still get the same
> problem.
>
> If I have two clients register on the external profile using the external
> domain and they call each other, with ODBC disabled everything works fine,
> but with ODBC enabled I get two invites to the destination client.
>
> The fact that I get different behavior with odbc enabled / disabled seems
> to indicate a bug more than it indicates a mis-configuration.
>
> Perhaps I am missing something? I don't quite understand the section in the
> example xml that refers to domains. Is there something I need to do there?
> The wiki does not mention it at all and the comments in the xml are not very
> clear either:
>
> <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="false" parse="true"/>
>  </domains>
>
> Thanks for your help,
>
> Ryan
>
> On Tue, Oct 4, 2011 at 10:27 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> ok but the db is shared so if they are all using mydomain.com it will
>> cross over so you should use distinct fake domains at least between
>> your profiles.
>>
>>
>> On Tue, Oct 4, 2011 at 11:24 AM, R H <buscom123+fs at gmail.com> wrote:
>> > In Response to Brian West, below are my sip registrations. These may
>> have
>> > slightly different values from the logs I sent yesterday simply because
>> I
>> > shut down last night and re-registered this morning. Same softphones,
>> same
>> > extensions, just new registrations:
>> >
>> -----------------------------------------------------------------------------
>> > select * from sip_registrations;
>> > MTAwYWEwYzhlNzc2MDAxYTUzNDhlY2EwZmEyZDFlZDE.,4001,mydomain.com,
>> mydomain.com,"""user""
>> > <sip:4001 at 192.168.99.164:61238
>> ;rinstance=e2d50bbe31452d30>",Registered(UDP),unknown,1317746818,X-Lite
>> > 4 release 4.1 stamp
>> >
>> 63215,4001,192.168.122.236,internal,psiFreeswitch,192.168.99.164,61238,4001,
>> mydomain.com,4001,mydomain.com,192.168.122.236,psiFreeswitch
>> > dQJifXuoqNBUlWCuZ8rCI6cu.Nn5F81W,4002,mydomain.com,mydomain.com,"""Test
>> > User123""
>> > <sip:cqgulpey at 192.168.99.164:57003
>> >",Registered(UDP),unknown,1317743825,Blink
>> > Lite 1.2.0
>> >
>> (MacOSX),4002,192.168.122.236,internal,psiFreeswitch,192.168.99.164,57003,4002,
>> mydomain.com,4002,mydomain.com,192.168.122.236,psiFreeswitch
>> >
>> -----------------------------------------------------------------------------
>> > In Response to Anthony Minessale,
>> > "Mydomain.com" is simply for testing and will not be our go-live domain.
>> The
>> > particular switch I am testing on is located in a closed internal
>> network
>> > designed specifically for testing systems like this and nothing more. It
>> is
>> > set up using the instructions on the FreeSwitch wiki for Multi-Tenant
>> > installations because we have the potential to go multi-tenant in the
>> future
>> > and we want to make sure multi-tenant setups work for the system we are
>> > building.
>> > As far as my profiles in development i have the following setup:
>> > 1) Internal - To emulate production this is limited use, no real phones
>> will
>> > ever connect from the local LAN in production. We have been testing
>> > registrations on this profile anyway but not there is no plan to have
>> phones
>> > register here in production:
>> > <domain name="all" alias="false" parse="false"/>
>> >
>> >
>> > 2) External - Most phones will register here, possibly to multiple
>> domains
>> > (tenants). These devices will be on other networks with possible natting
>> > issues. This profile requires authentication to make calls.
>> > 3) External_Debug - When clients complain of connectivity issues we will
>> be
>> > asking them to move to this profile so that we can more easily monitor
>> their
>> > sip traffic and debug their issues. This profile will also require
>> > authentication and allow registration to the many domains.
>> > 4) Unnamed Future External Profile - We will also most likely have
>> another
>> > profile for our upstream providers once we are ready to turn them on.
>> >
>> >
>> >
>> >
>> >
>> > On Mon, Oct 3, 2011 at 5:59 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> are you really using "mydomain.com" on all the profiles at the same
>> time?
>> >>
>> >> On Mon, Oct 3, 2011 at 5:29 PM, R H <buscom123+fs at gmail.com> wrote:
>> >> > Sure,
>> >> > I have two soft phones registering as extensions 4001 and 4002 to the
>> >> > same
>> >> > profile. When 4002 dials 4001 the softphone that is registered as
>> 4001
>> >> > received two calls simultaneously. This only happens when I enable
>> odbc
>> >> > on
>> >> > two profiles. If I disable odbc or only enable it for one profile
>> >> > everything
>> >> > works fine. If I enable ODBC for 3 profiles I would get three invites
>> to
>> >> > 4001, etc...
>> >> > Attached is some logging output. Notice the parts highlighted in red.
>> >> > Thank you for your help.
>> >> > Ryan
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> could you elaborate more on what the B-legs are?
>> >> >> Are you doing bridges to an outbound entity, if so how?
>> >> >>
>> >> >> do you have any traces doing:
>> >> >>
>> >> >> sofia global siptrace on
>> >> >> and
>> >> >> console loglevel debug
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
>> >> >> > Hey Everyone,
>> >> >> > I am preparing to set up HA Failover to my production freeswitch
>> >> >> > environment. In preparation for this I have been doing some
>> testing
>> >> >> > in a
>> >> >> > development environment and I encountered the following problem:
>> >> >> > I currently have two sip-profiles set up that I am using full
>> time,
>> >> >> > when
>> >> >> > I
>> >> >> > enable odbc in the switch.conf.xml everything works correctly.
>> When I
>> >> >> > enable
>> >> >> > odbc in ONE of the sip-profiles everything works correctly. But
>> when
>> >> >> > I
>> >> >> > enable odbc in the second profile I begin to see two b-legs for
>> ever
>> >> >> > call
>> >> >> > that I initiate. Both b-legs are directed at the same client.
>> >> >> > Does anyone have any idea why enabling odbc on two sofia profiles
>> >> >> > would
>> >> >> > cause every inbound call to generate two outbound calls? I also
>> >> >> > decided
>> >> >> > to
>> >> >> > test this problem by adding two more profiles for a total of 4.
>> With
>> >> >> > 4
>> >> >> > profiles I get 4 b-legs. What am I doing wrong?
>> >> >> > Ryan
>> >> >> >
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> FreeSWITCH http://www.freeswitch.org/
>> >> >> ClueCon http://www.cluecon.com/
>> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>
>> >> >> AIM: anthm
>> >> >> MSN:anthony_minessale at hotmail.com
>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> IRC: irc.freenode.net #freeswitch
>> >> >>
>> >> >> FreeSWITCH Developer Conference
>> >> >> sip:888 at conference.freeswitch.org
>> >> >> googletalk:conf+888 at conference.freeswitch.org
>> >> >> pstn:+19193869900
>> >> >>
>> >> >>
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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 anthony.minessale at gmail.com  Thu Oct  6 21:39:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Oct 2011 12:39:15 -0500
Subject: [Freeswitch-users] Sofia ODBC causes multiple b-legs on a
	single call
In-Reply-To: <CAAtQP57w_FgGCFaSA3Sxooky0kTyoLdAkfjv227=AxxO=nBs9g@mail.gmail.com>
References: <CAAtQP56E_AtM+Q0iN_O5VTRyHnabu9w-O0ZJ+BuC=MGZuP_APQ@mail.gmail.com>
	<CAKbxfG_=ZSkeXo3LK0vnednW3V35zqQW-u-Dc02EH+bepcpTig@mail.gmail.com>
	<CAAtQP56p0jkqfp+ur8LN5Z+qS9btJXcGZmtGLmAd0S7jGeSr1Q@mail.gmail.com>
	<CAKbxfG__9FmiidH0u4eCnAo+u=zN89MCYM6xWyK0RmG_CU1rfA@mail.gmail.com>
	<CAAtQP57wOv-RVpA-VYx4vuQvjepvph=34MM4fMzG5G78nC_ZEQ@mail.gmail.com>
	<CAKbxfG9Msun7_rdMEoSHk-rtpZ_bVJmdUoHWxFBDHDC5U2EaFg@mail.gmail.com>
	<CAAtQP57w_FgGCFaSA3Sxooky0kTyoLdAkfjv227=AxxO=nBs9g@mail.gmail.com>
Message-ID: <CAKbxfG8de35dxu7MOvJCdeBSTDssdKvVfoLgrSpeQABFqfR7Fg@mail.gmail.com>

did you set both 'force-register-domain' and
'force-register-db-domain' set to the respective domains?
force-register-domain only forces which domain it uses for auth
force-register-db-domain actually controls what it writes to the db.

or
set neither param on both profiles and put the correct domains in the
actual sip packets not using the same domain in any way between the
profiles even for testing.  Domains are meant to be unique.


My explanation before was simply do not use the same domain name for
both profiles because the method you are using to call the endpoints
will scan every profile for the domain name you supply and hence it
will find more than one.

If you do the above and still have a problem you need to collect the
data and post it to jira http://jira.freeswitch.org




On Wed, Oct 5, 2011 at 12:54 PM, R H <buscom123+fs at gmail.com> wrote:
> Hey Anthony, I appreciate your help on this. Your response was not clear
> enough for me to understand how to accomplish what you are describing. Let's
> assume I only have 2 profiles (Internal/External). Both profiles default to
> the public context.
>
> Now if I were to create two domains in the directory and name them
> "internal.mydomain.com" and "external.mydomain.com" and force everyone who
> registers to the internal profile to use the internal domain and everyone on
> the external profile to use the external domain I would still get the same
> problem.
>
> If I have two clients register on the external profile using the external
> domain and they call each other, with ODBC disabled everything works fine,
> but with ODBC enabled I get two invites to the destination client.
>
> The fact that I get different behavior with odbc enabled / disabled seems to
> indicate a bug more than it indicates a mis-configuration.
>
> Perhaps I am missing something? I don't quite understand the section in the
> example xml that refers to domains. Is there something I need to do there?
> The wiki does not mention it at all and the comments in the xml are not very
> clear either:
>
> <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="false" parse="true"/>
> ?</domains>
>
> Thanks for your help,
>
> Ryan
>
> On Tue, Oct 4, 2011 at 10:27 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> ok but the db is shared so if they are all using mydomain.com it will
>> cross over so you should use distinct fake domains at least between
>> your profiles.
>>
>>
>> On Tue, Oct 4, 2011 at 11:24 AM, R H <buscom123+fs at gmail.com> wrote:
>> > In Response to Brian West, below are my sip registrations. These may
>> > have
>> > slightly different values from the logs I sent yesterday simply because
>> > I
>> > shut down last night and re-registered this morning. Same softphones,
>> > same
>> > extensions, just new registrations:
>> >
>> > -----------------------------------------------------------------------------
>> > select * from sip_registrations;
>> >
>> > MTAwYWEwYzhlNzc2MDAxYTUzNDhlY2EwZmEyZDFlZDE.,4001,mydomain.com,mydomain.com,"""user""
>> >
>> > <sip:4001 at 192.168.99.164:61238;rinstance=e2d50bbe31452d30>",Registered(UDP),unknown,1317746818,X-Lite
>> > 4 release 4.1 stamp
>> >
>> > 63215,4001,192.168.122.236,internal,psiFreeswitch,192.168.99.164,61238,4001,mydomain.com,4001,mydomain.com,192.168.122.236,psiFreeswitch
>> > dQJifXuoqNBUlWCuZ8rCI6cu.Nn5F81W,4002,mydomain.com,mydomain.com,"""Test
>> > User123""
>> >
>> > <sip:cqgulpey at 192.168.99.164:57003>",Registered(UDP),unknown,1317743825,Blink
>> > Lite 1.2.0
>> >
>> > (MacOSX),4002,192.168.122.236,internal,psiFreeswitch,192.168.99.164,57003,4002,mydomain.com,4002,mydomain.com,192.168.122.236,psiFreeswitch
>> >
>> > -----------------------------------------------------------------------------
>> > In Response to Anthony Minessale,
>> > "Mydomain.com" is simply for testing and will not be our go-live domain.
>> > The
>> > particular switch I am testing on is located in a closed internal
>> > network
>> > designed specifically for testing systems like this and nothing more. It
>> > is
>> > set up using the instructions on the FreeSwitch wiki for Multi-Tenant
>> > installations because we have the potential to go multi-tenant in the
>> > future
>> > and we want to make sure multi-tenant setups work for the system we are
>> > building.
>> > As far as my profiles in development i have the following setup:
>> > 1) Internal - To emulate production this is limited use, no real phones
>> > will
>> > ever connect from the local LAN in production. We have been testing
>> > registrations on this profile anyway but not there is no plan to have
>> > phones
>> > register here in production:
>> > <domain name="all" alias="false" parse="false"/>
>> >
>> >
>> > 2) External - Most phones will register here, possibly to multiple
>> > domains
>> > (tenants). These devices will be on other networks with possible natting
>> > issues. This profile requires authentication to make calls.
>> > 3) External_Debug - When clients complain of connectivity issues we will
>> > be
>> > asking them to move to this profile so that we can more easily monitor
>> > their
>> > sip traffic and debug their issues. This profile will also require
>> > authentication and allow registration to the many domains.
>> > 4) Unnamed Future External Profile - We will also most likely have
>> > another
>> > profile for our upstream providers once we are ready to turn them on.
>> >
>> >
>> >
>> >
>> >
>> > On Mon, Oct 3, 2011 at 5:59 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> are you really using "mydomain.com" on all the profiles at the same
>> >> time?
>> >>
>> >> On Mon, Oct 3, 2011 at 5:29 PM, R H <buscom123+fs at gmail.com> wrote:
>> >> > Sure,
>> >> > I have two soft phones registering as extensions 4001 and 4002 to the
>> >> > same
>> >> > profile. When 4002 dials 4001 the softphone that is registered as
>> >> > 4001
>> >> > received two calls?simultaneously. This only happens when I enable
>> >> > odbc
>> >> > on
>> >> > two profiles. If I disable odbc or only enable it for one profile
>> >> > everything
>> >> > works fine. If I enable ODBC for 3 profiles I would get three invites
>> >> > to
>> >> > 4001, etc...
>> >> > Attached is some logging output. Notice the parts highlighted in red.
>> >> > Thank you for your help.
>> >> > Ryan
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Oct 3, 2011 at 1:23 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> could you elaborate more on what the B-legs are?
>> >> >> Are you doing bridges to an outbound entity, if so how?
>> >> >>
>> >> >> do you have any traces doing:
>> >> >>
>> >> >> sofia global siptrace on
>> >> >> and
>> >> >> console loglevel debug
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Mon, Oct 3, 2011 at 2:14 PM, R H <buscom123+fs at gmail.com> wrote:
>> >> >> > Hey Everyone,
>> >> >> > I am preparing to set up HA Failover to my production freeswitch
>> >> >> > environment. In preparation for this I have been doing some
>> >> >> > testing
>> >> >> > in a
>> >> >> > development environment and I encountered the following problem:
>> >> >> > I currently have two sip-profiles set up that I am using full
>> >> >> > time,
>> >> >> > when
>> >> >> > I
>> >> >> > enable odbc in the switch.conf.xml everything works correctly.
>> >> >> > When I
>> >> >> > enable
>> >> >> > odbc in ONE of the sip-profiles everything works correctly. But
>> >> >> > when
>> >> >> > I
>> >> >> > enable odbc in the second profile I begin to see two b-legs for
>> >> >> > ever
>> >> >> > call
>> >> >> > that I initiate. Both b-legs are directed at the same client.
>> >> >> > Does anyone have any idea why enabling odbc on two sofia profiles
>> >> >> > would
>> >> >> > cause every inbound call to generate two outbound calls? I also
>> >> >> > decided
>> >> >> > to
>> >> >> > test this problem by adding two more profiles for a total of 4.
>> >> >> > With
>> >> >> > 4
>> >> >> > profiles I get 4 b-legs. What am I doing wrong?
>> >> >> > Ryan
>> >> >> >
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> FreeSWITCH http://www.freeswitch.org/
>> >> >> ClueCon http://www.cluecon.com/
>> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >>
>> >> >> AIM: anthm
>> >> >> MSN:anthony_minessale at hotmail.com
>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> IRC: irc.freenode.net #freeswitch
>> >> >>
>> >> >> FreeSWITCH Developer Conference
>> >> >> sip:888 at conference.freeswitch.org
>> >> >> googletalk:conf+888 at conference.freeswitch.org
>> >> >> pstn:+19193869900
>> >> >>
>> >> >>
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Oct  6 23:20:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Oct 2011 14:20:26 -0500
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <4E8DD739.5090104@hw.ac.uk>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
Message-ID: <CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>

don't actually hang it up just don't connect the socket or connnect it
and disconnect it right away and the app will fail on in the
dial-plan.

again you could try mod_limit otherwise.


On Thu, Oct 6, 2011 at 11:28 AM, xl127 <x.liu at hw.ac.uk> wrote:
> Thanks for the suggestions!
>
> I thought about and tried to make my outbound socket application
> fail e.g. by send a hangup message but it didn't work.
>
> What I did is that I set a public variable channelUUID with initial
> value null in MyOutboundHandler,
> I check if it is null in the handleConnectionResponse method. If it not
> null, a hangup command is sent to FS.
> But it looks like each call channel creates its own Java thread, so the
> channelUUID is always null before the connection.
> So my approach doesn't work.
>
> Any idea about checking if the remote application is answering a call?
>
> Thanks!
> Xing
>
>
> On 06/10/11 16:43, Anthony Minessale wrote:
>> Really the socket app has not limit to how many simultaneous
>> connections it can handle.
>> You could try mod_limit, or make the remote end of you application
>> fail when there are too many connections.
>>
>>
>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk> ?wrote:
>>> Hello,
>>>
>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>> socket applications.
>>>
>>> Basically I want to check if the first socket app is engaged then the
>>> call should be transferred
>>> to the second socket app and so on.
>>>
>>> For initial test, I tried two concurrent calls, the problem is both
>>> calls are always answered
>>> by the first outbound socket.
>>>
>>> I have following setting in the dialplan:
>>>
>>> <extension name="sipgateDialedIn">
>>> <condition field="destination_number" expression="^5800$">
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="bridge" data="loopback/5801"/>
>>> <action application="bridge" data="loopback/5802"/>
>>>
>>> <action application="set" data="tts_engine=flite"/>
>>> <action application="set" data="tts_voice=kal"/>
>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>> try again later."/>
>>> <action application="sleep" data="10"/>
>>> <action application="hangup"/>
>>> </condition>
>>> </extension>
>>>
>>> <extension name="myOutboundESL5801">
>>> <condition field="destination_number" expression="^5801$">
>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>> </condition>
>>> </extension>
>>>
>>> <extension name="myOutboundESL5802">
>>> <condition field="destination_number" expression="^5802$">
>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>> </condition>
>>> </extension>
>>>
>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>> second concurrent call to 5800 will be
>>> answered by extension 5802, and the third call gets a busy message.
>>>
>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>> the socket is engaged?
>>>
>>> I've tried and googled a while but without success.
>>>
>>> Any advices please?
>>>
>>> Many thanks!
>>> Xing
>>>
>>>
>>>
>>> --
>>> Heriot-Watt University is a Scottish charity
>>> registered under charity number SC000278.
>>>
>>> Heriot-Watt University is the Sunday Times
>>> Scottish University of the Year 2011-2012
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
> Heriot-Watt University is the Sunday Times
> Scottish University of the Year 2011-2012
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From cvogel at lyonl.com  Thu Oct  6 23:59:33 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Thu, 6 Oct 2011 19:59:33 +0000
Subject: [Freeswitch-users] Call bridge problem
Message-ID: <23FE1289-CFC1-4C5E-AAAC-7DCB759BDD3E@lyonl.com>

Hello,

I'm having a problem bridging out calls, FS is sending 500 Internal Server Error to our gateway. Any thoughts why this is?


<include>
  <context name="public">
    <extension name="unloop">
      <condition field="${unroll_loops}" expression="^true$"/>
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}"/>
      </condition>
    </extension>
    <extension name="outside_call" continue="true">
      <condition>
        <action application="set" data="outside_call=true"/>
        <action application="set" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
      </condition>
    </extension>

    <extension name="call_debug" continue="true">
      <condition field="${call_debug}" expression="^true$" break="never">
        <action application="info"/>
      </condition>
    </extension>
    <extension name="4142211800">
      <condition field="destination_number" expression="^(\+?14142211800)$">
        <action application="set" data="effective_caller_id_name=LyonL"/>
        <action application="set" data="effective_caller_id_number=+14142211800"/>
        <action application="bridge" data="sofia/gateway/L3CN/+15618911806"/>
      </condition>
    </extension>
  </context>
</include>



<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <gateway name="L3CN">
      <param name="username" value="1-F2la9"/>
      <param name="password" value="password"/>
      <param name="realm" value="BroadWorks"/>
      <param name="proxy" value="4.55.35.60:5070"/>
      <param name="from-domain" value="69.65.114.181:5060"/>
      <param name="dtmf-type" value="rfc2833"/>
      <param name="extension-in-contact" value="true"/>
      <param name="caller-id-in-from" value="true"/>
      <param name="register" value="false"/>
    </gateway>
  </gateways>

  <aliases>
    <!--
    <alias name="outbound"/>
    <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
    <!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!--<param name="disable-srv503" value="true"/>-->
    <!-- This could be set to "passive" -->
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>

    <param name="user-agent-string" value="LyonL"/>

    <!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="greedy"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="NDLB-force-rport" value="server-only"/>
    <!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    -->
    <param name="rtp-ip" value="$${external_rtp_ip}"/>
    <param name="sip-ip" value="$${external_sip_ip}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!--<param name="enable-3pcc" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>


send 1283 bytes to udp/[4.55.35.60]:5070 at 19:50:28.998153:
   ------------------------------------------------------------------------
   INVITE sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
   Via: SIP/2.0/UDP 69.65.114.181;rport;branch=z9hG4bK62S7Fr2Um459j
   Max-Forwards: 8
   From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
   To: <sip:+15618911806 at 4.55.35.60:5070>
   Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
   CSeq: 18629251 INVITE
   Contact: <sip:1-F2la9 at 69.65.114.181:5060;transport=udp;gw=L3CN>
   Expires: 3600
   User-Agent: LyonL
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE
FER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Authorization: Digest username="1-F2la9", realm="BroadWorks", nonce="BroadWor
ksXgtg5vbouT4lpitdBW", cnonce="SKxoW2r3Ei+9o+X8ky5idg", algorithm=MD5, uri="sip:
+15618911806 at 4.55.35.60:5070", response="9f0325808936e22ae050921c176f2730", qop=
auth, nc=00000001
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 229
   X-FS-Support: update_display
   Remote-Party-ID: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;party=calling;
screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1317910968 1317910969 IN IP4 69.65.114.181
   s=FreeSWITCH
   c=IN IP4 69.65.114.181
   t=0 0
   m=audio 19660 RTP/AVP 0 8 18 101 13
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 297 bytes from udp/[4.55.35.60]:5070 at 19:50:29.002153:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 69.65.114.181;received=69.65.114.181;branch=z9hG4bK62S7Fr2Um
459j;rport=5060
   From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
   To: <sip:+15618911806 at 4.55.35.60:5070>
   Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
   CSeq: 18629251 INVITE

   ------------------------------------------------------------------------
recv 468 bytes from udp/[4.55.35.60]:5070 at 19:50:29.008153:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 69.65.114.181;received=69.65.114.181;branch=z9hG4bK62S7Fr2Um
459j;rport=5060
   From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
   To: <sip:+15618911806 at 4.55.35.60:5070>;tag=SDcpmkf99-1282242359-1317930631679

   Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
   CSeq: 18629251 INVITE
   WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXgtg5vbpbTeh3n5wBW",algo
rithm=MD5,realm="BroadWorks"
   Content-Length: 0

   ------------------------------------------------------------------------
send 367 bytes to udp/[4.55.35.60]:5070 at 19:50:29.008153:
   ------------------------------------------------------------------------
   ACK sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
   Via: SIP/2.0/UDP 69.65.114.181;rport;branch=z9hG4bK62S7Fr2Um459j
   Max-Forwards: 8
   From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
   To: <sip:+15618911806 at 4.55.35.60:5070>;tag=SDcpmkf99-1282242359-1317930631679

   Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
   CSeq: 18629251 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2011-10-06 19:50:28.979152 [NOTICE] sofia.c:5897 Hangup sofia/external/+15618911
806 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
2011-10-06 19:50:28.979152 [INFO] mod_dptools.c:2810 Originate Failed.  Cause: N
ORMAL_UNSPECIFIED
2011-10-06 19:50:28.979152 [NOTICE] mod_dptools.c:2929 Hangup sofia/external/+15
618911806 at 4.55.35.60<mailto:618911806 at 4.55.35.60> [CS_EXECUTE] [NORMAL_UNSPECIFIED]
send 380 bytes to udp/[4.55.35.60]:5070 at 19:50:29.010153:
   ------------------------------------------------------------------------
   SIP/2.0 500 Internal Server Error
   Via: SIP/2.0/UDP 4.55.35.60:5070;branch=z9hG4bKtil5pp004o3hvjs9n2c1.1
   From: <sip:+15618911806 at 4.55.35.60;user=phone>;tag=SDesfje01-1691283656-13179
30631601-
   To: "4142211800 4142211800" <sip:+14142211800 at 69.65.114.181>;tag=jQQ2QrmerFD9
D
   Call-ID: SDesfje01-bc0436bf1c3c496f2c13c86f66280e15-v3000i1
   CSeq: 761577689 INVITE
   Content-Length: 0

   ------------------------------------------------------------------------
2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1367 Session 19 (sofia
/external/+15618911806 at 4.55.35.60) Ended
2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1369 Close Channel sof
ia/external/+15618911806 at 4.55.35.60<mailto:ia/external/+15618911806 at 4.55.35.60> [CS_DESTROY]
2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1367 Session 20 (sofia
/external/+15618911806) Ended
2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1369 Close Channel sof
ia/external/+15618911806 [CS_DESTROY]
-------------- next part --------------
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From Hector.Geraldino at ip-soft.net  Fri Oct  7 01:34:40 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Thu, 6 Oct 2011 17:34:40 -0400
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
	<CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02206ED4A1@NY1-EXMB-01.ip-soft.net>

This might sound a little bit Off-Topic but, why are you willing to bind only one call per running instance? I cannot imagine any user scenario where this is desirable. Maybe as a proof of concept is Ok, but having an equivalence between calls and instances mmm... I don't know.

Can you please tell me what are you willing to accomplish? I'm really curious about it.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Thursday, October 06, 2011 3:20 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected

don't actually hang it up just don't connect the socket or connnect it
and disconnect it right away and the app will fail on in the
dial-plan.

again you could try mod_limit otherwise.


On Thu, Oct 6, 2011 at 11:28 AM, xl127 <x.liu at hw.ac.uk> wrote:
> Thanks for the suggestions!
>
> I thought about and tried to make my outbound socket application
> fail e.g. by send a hangup message but it didn't work.
>
> What I did is that I set a public variable channelUUID with initial
> value null in MyOutboundHandler,
> I check if it is null in the handleConnectionResponse method. If it not
> null, a hangup command is sent to FS.
> But it looks like each call channel creates its own Java thread, so the
> channelUUID is always null before the connection.
> So my approach doesn't work.
>
> Any idea about checking if the remote application is answering a call?
>
> Thanks!
> Xing
>
>
> On 06/10/11 16:43, Anthony Minessale wrote:
>> Really the socket app has not limit to how many simultaneous
>> connections it can handle.
>> You could try mod_limit, or make the remote end of you application
>> fail when there are too many connections.
>>
>>
>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk> ?wrote:
>>> Hello,
>>>
>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>> socket applications.
>>>
>>> Basically I want to check if the first socket app is engaged then the
>>> call should be transferred
>>> to the second socket app and so on.
>>>
>>> For initial test, I tried two concurrent calls, the problem is both
>>> calls are always answered
>>> by the first outbound socket.
>>>
>>> I have following setting in the dialplan:
>>>
>>> <extension name="sipgateDialedIn">
>>> <condition field="destination_number" expression="^5800$">
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="bridge" data="loopback/5801"/>
>>> <action application="bridge" data="loopback/5802"/>
>>>
>>> <action application="set" data="tts_engine=flite"/>
>>> <action application="set" data="tts_voice=kal"/>
>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>> try again later."/>
>>> <action application="sleep" data="10"/>
>>> <action application="hangup"/>
>>> </condition>
>>> </extension>
>>>
>>> <extension name="myOutboundESL5801">
>>> <condition field="destination_number" expression="^5801$">
>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>> </condition>
>>> </extension>
>>>
>>> <extension name="myOutboundESL5802">
>>> <condition field="destination_number" expression="^5802$">
>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>> </condition>
>>> </extension>
>>>
>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>> second concurrent call to 5800 will be
>>> answered by extension 5802, and the third call gets a busy message.
>>>
>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>> the socket is engaged?
>>>
>>> I've tried and googled a while but without success.
>>>
>>> Any advices please?
>>>
>>> Many thanks!
>>> Xing
>>>
>>>
>>>
>>> --
>>> Heriot-Watt University is a Scottish charity
>>> registered under charity number SC000278.
>>>
>>> Heriot-Watt University is the Sunday Times
>>> Scottish University of the Year 2011-2012
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
> Heriot-Watt University is the Sunday Times
> Scottish University of the Year 2011-2012
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


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


From cmrienzo at gmail.com  Fri Oct  7 02:49:06 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 6 Oct 2011 18:49:06 -0400
Subject: [Freeswitch-users] playing files through a remote URL
In-Reply-To: <93166BD9-5C05-4E41-A017-CBED60057429@gmail.com>
References: <AANLkTikYdXfOVYF6=vUsj8gB7kHUuwOddrYy_dFg6ByG@mail.gmail.com>
	<AANLkTik-d6Wb2dyStNSF-rVKCvaHzXq4pWKsYRKA9sf7@mail.gmail.com>
	<93166BD9-5C05-4E41-A017-CBED60057429@gmail.com>
Message-ID: <CAD-qRkgpU5SoY_=rgJMxRqH+2UEODH0eLZ7v4a0qDO1ZXBbqAg@mail.gmail.com>

Holger,

This is an old thread and I assume you solved your problem, but I'd like to
mention that mod_http_cache is now available and can fetch audio from a
webserver for playback.

http://wiki.freeswitch.org/wiki/Mod_http_cache

Regards,

Chris


On Mon, Jan 10, 2011 at 10:12 PM, Chris Rienzo <cmrienzo at gmail.com> wrote:

> Maybe you could write a script to download the file.  There are plenty of
> command line tools to do http get (curl, wget).
>
>
>
> On Jan 10, 2011, at 14:09, Holger Esser <hesser4900 at gmail.com> wrote:
>
> Hello,
>
> I am trying to write a simple ivr script that answers the call, plays a
> greeting and collects a few digits.
> Is it possible to fetch the audio through a remote URL (i.e. <http://messageserver/en.us/greeting.wav>
> http://messageServer/en.us/greeting.wav)?
>
> Or does it have to reside on the local file system or a mapped drive?
>
> Kind regards,
>
> Holger
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cliff at develix.com  Fri Oct  7 06:40:29 2011
From: cliff at develix.com (Cliff Wells)
Date: Thu, 06 Oct 2011 19:40:29 -0700
Subject: [Freeswitch-users] mod_native_file and speex
In-Reply-To: <1317888565.2256.4.camel@portable-evil>
References: <1317842469.2482.16.camel@portable-evil>
	<55E34A90-B971-4C7F-8DA4-89E1B15EDF6E@avgs.ca>
	<1317888565.2256.4.camel@portable-evil>
Message-ID: <1317955229.2256.13.camel@portable-evil>

On Thu, 2011-10-06 at 01:09 -0700, Cliff Wells wrote:
> On Thu, 2011-10-06 at 00:42 -0400, Mathieu Rene wrote:
> > Hi,
> > 
> > I made a small tool to encode files for use with mod_native_file, it's being compiled automatically as part of the normal install process.
> > 
> > Try that:
> > fs_encode -l mod_speex file.wav file.SPEEX
> 
> Similar result: file appears to play successfully, but only silence is
> heard.  As you can see, it gets done playing rather quickly (message is
> actually about 20s in duration).
> 
> 2011-10-05 23:59:37.357566 [INFO] mod_native_file.c:94 Opening File [/usr/local/freeswitch/scripts/jovan-message.SPEEX] 8000hz
> 2011-10-05 23:59:37.357566 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms

I just noticed the previous line states "Codec Activated L16 at 8000hz".
Why would the L16 codec be selected?  

Regards,
Cliff




From Stefan.Weigel at allianz-warranty.com  Fri Oct  7 12:42:32 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 7 Oct 2011 10:42:32 +0200
Subject: [Freeswitch-users] Desktop phone - answer a ringing call
	via	API call
In-Reply-To: <CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
References: <AcyCnbOIxZuxEaeRQIyKEkToMpcvaQ==>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E5F@AZWSMS03.azwarranty.int>
	<CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>

Hi Michael, list,



that's the point.

I would really appreciate a solution for that "problem".

We're using FreeSwitch in an callcenter environment and people used to handle the calls through some kind of CTI interface before we switched to FreeSWITCH. So it would be great to have this feature. But there's no possibility to control Polycom phones that way.



Hints on this topic are welcome!





Thanks and best regards



Stefan



Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.





Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins
Gesendet: Mittwoch, 5. Oktober 2011 18:42
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call



This would be a function of the phone. AFAICT, Polycom does not have a Webby way of doing this. However, I think Brian West has a bead on using the expansion port along with an Arduino board to do some manual/remote control of the phone. If anyone has more information on remote controlling Polycoms (or any other phones, for that matter) please let us know.



-MC

On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi all,



Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP 560) via api command ?

I tried with SendMsg <uuid> and 'answer' application, but it's not want I need/want.





Thanks and best regards,



Stefan



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 x.liu at hw.ac.uk  Fri Oct  7 14:55:00 2011
From: x.liu at hw.ac.uk (xl127)
Date: Fri, 07 Oct 2011 11:55:00 +0100
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02206ED4A1@NY1-EXMB-01.ip-soft.net>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
	<CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD02206ED4A1@NY1-EXMB-01.ip-soft.net>
Message-ID: <4E8EDA84.5000806@hw.ac.uk>

Hi Hector,

Your points make sense but I have some specific issues for my 
application so I thought of binding only one call per runnning instance.

1. My outbound socket application is one component/server of our Spoken 
Dialogue System (SDS).
We have four ports Nuance ASR/TTS licenses, each port is specific to one 
machine( yeah, we have four seperate license files for four machines).
It seems to be difficult to specify which channel uses which Nuance MRCP 
profile if simultenous calls are answered by only one instance.

2. It seems to me that the log info would be mixed for simultenous calls 
if running one instance.

3. My socket app will be communicating with other components/servers of 
the SDS via ICE middleware (http://www.zeroc.com/).
It will register to the central ICE server when starting the system via 
a Topic name which is a static name given by the developer,
it will keep publishing ASR results on that topic so the other 
components listening on the topic will get the results,
and it also subscribes and listens to other topics from other components 
for TTS.

In simultenous calls to one running instance scenario, the 
distinguishable thing between channels would be the channel UUID which I 
could use
as a part of the topic name, but it is dynamic and other servers could 
not know it when registering to the central server.

I could let the all channels publish results with a same static topic 
name and channel UUID, the interested parties could parse the results
they want based on the UUID, but I worried this would cause more burdens 
to the whole system.


4. Even if running one instance, I still need some ways to check and 
control the maximum number of calls as we have limited ASR/TTS resources.

Cheers,
Xing

On 06/10/11 22:34, Hector Geraldino wrote:
> This might sound a little bit Off-Topic but, why are you willing to bind only one call per running instance? I cannot imagine any user scenario where this is desirable. Maybe as a proof of concept is Ok, but having an equivalence between calls and instances mmm... I don't know.
>
> Can you please tell me what are you willing to accomplish? I'm really curious about it.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Thursday, October 06, 2011 3:20 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
>
> don't actually hang it up just don't connect the socket or connnect it
> and disconnect it right away and the app will fail on in the
> dial-plan.
>
> again you could try mod_limit otherwise.
>
>
> On Thu, Oct 6, 2011 at 11:28 AM, xl127<x.liu at hw.ac.uk>  wrote:
>> Thanks for the suggestions!
>>
>> I thought about and tried to make my outbound socket application
>> fail e.g. by send a hangup message but it didn't work.
>>
>> What I did is that I set a public variable channelUUID with initial
>> value null in MyOutboundHandler,
>> I check if it is null in the handleConnectionResponse method. If it not
>> null, a hangup command is sent to FS.
>> But it looks like each call channel creates its own Java thread, so the
>> channelUUID is always null before the connection.
>> So my approach doesn't work.
>>
>> Any idea about checking if the remote application is answering a call?
>>
>> Thanks!
>> Xing
>>
>>
>> On 06/10/11 16:43, Anthony Minessale wrote:
>>> Really the socket app has not limit to how many simultaneous
>>> connections it can handle.
>>> You could try mod_limit, or make the remote end of you application
>>> fail when there are too many connections.
>>>
>>>
>>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk>    wrote:
>>>> Hello,
>>>>
>>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>>> socket applications.
>>>>
>>>> Basically I want to check if the first socket app is engaged then the
>>>> call should be transferred
>>>> to the second socket app and so on.
>>>>
>>>> For initial test, I tried two concurrent calls, the problem is both
>>>> calls are always answered
>>>> by the first outbound socket.
>>>>
>>>> I have following setting in the dialplan:
>>>>
>>>> <extension name="sipgateDialedIn">
>>>> <condition field="destination_number" expression="^5800$">
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="bridge" data="loopback/5801"/>
>>>> <action application="bridge" data="loopback/5802"/>
>>>>
>>>> <action application="set" data="tts_engine=flite"/>
>>>> <action application="set" data="tts_voice=kal"/>
>>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>>> try again later."/>
>>>> <action application="sleep" data="10"/>
>>>> <action application="hangup"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5801">
>>>> <condition field="destination_number" expression="^5801$">
>>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5802">
>>>> <condition field="destination_number" expression="^5802$">
>>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>>> second concurrent call to 5800 will be
>>>> answered by extension 5802, and the third call gets a busy message.
>>>>
>>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>>> the socket is engaged?
>>>>
>>>> I've tried and googled a while but without success.
>>>>
>>>> Any advices please?
>>>>
>>>> Many thanks!
>>>> Xing
>>>>
>>>>
>>>>
>>>> --
>>>> Heriot-Watt University is a Scottish charity
>>>> registered under charity number SC000278.
>>>>
>>>> Heriot-Watt University is the Sunday Times
>>>> Scottish University of the Year 2011-2012
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>> --
>> Heriot-Watt University is a Scottish charity
>> registered under charity number SC000278.
>>
>> Heriot-Watt University is the Sunday Times
>> Scottish University of the Year 2011-2012
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012




From x.liu at hw.ac.uk  Fri Oct  7 15:03:44 2011
From: x.liu at hw.ac.uk (xl127)
Date: Fri, 07 Oct 2011 12:03:44 +0100
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
	<CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
Message-ID: <4E8EDC90.3030102@hw.ac.uk>

Thanks, Anthony!

I will check David Varnes's Java ESL API again to see how to check 
number of channel opend and how to disconnect the new channel.

Yeah I will look into mod_limit as well.

Cheers,
Xing



On 06/10/11 20:20, Anthony Minessale wrote:
> don't actually hang it up just don't connect the socket or connnect it
> and disconnect it right away and the app will fail on in the
> dial-plan.
>
> again you could try mod_limit otherwise.
>
>
> On Thu, Oct 6, 2011 at 11:28 AM, xl127<x.liu at hw.ac.uk>  wrote:
>> Thanks for the suggestions!
>>
>> I thought about and tried to make my outbound socket application
>> fail e.g. by send a hangup message but it didn't work.
>>
>> What I did is that I set a public variable channelUUID with initial
>> value null in MyOutboundHandler,
>> I check if it is null in the handleConnectionResponse method. If it not
>> null, a hangup command is sent to FS.
>> But it looks like each call channel creates its own Java thread, so the
>> channelUUID is always null before the connection.
>> So my approach doesn't work.
>>
>> Any idea about checking if the remote application is answering a call?
>>
>> Thanks!
>> Xing
>>
>>
>> On 06/10/11 16:43, Anthony Minessale wrote:
>>> Really the socket app has not limit to how many simultaneous
>>> connections it can handle.
>>> You could try mod_limit, or make the remote end of you application
>>> fail when there are too many connections.
>>>
>>>
>>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk>    wrote:
>>>> Hello,
>>>>
>>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>>> socket applications.
>>>>
>>>> Basically I want to check if the first socket app is engaged then the
>>>> call should be transferred
>>>> to the second socket app and so on.
>>>>
>>>> For initial test, I tried two concurrent calls, the problem is both
>>>> calls are always answered
>>>> by the first outbound socket.
>>>>
>>>> I have following setting in the dialplan:
>>>>
>>>> <extension name="sipgateDialedIn">
>>>> <condition field="destination_number" expression="^5800$">
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="bridge" data="loopback/5801"/>
>>>> <action application="bridge" data="loopback/5802"/>
>>>>
>>>> <action application="set" data="tts_engine=flite"/>
>>>> <action application="set" data="tts_voice=kal"/>
>>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>>> try again later."/>
>>>> <action application="sleep" data="10"/>
>>>> <action application="hangup"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5801">
>>>> <condition field="destination_number" expression="^5801$">
>>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5802">
>>>> <condition field="destination_number" expression="^5802$">
>>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>>> second concurrent call to 5800 will be
>>>> answered by extension 5802, and the third call gets a busy message.
>>>>
>>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>>> the socket is engaged?
>>>>
>>>> I've tried and googled a while but without success.
>>>>
>>>> Any advices please?
>>>>
>>>> Many thanks!
>>>> Xing
>>>>
>>>>
>>>>
>>>> --
>>>> Heriot-Watt University is a Scottish charity
>>>> registered under charity number SC000278.
>>>>
>>>> Heriot-Watt University is the Sunday Times
>>>> Scottish University of the Year 2011-2012
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>> --
>> Heriot-Watt University is a Scottish charity
>> registered under charity number SC000278.
>>
>> Heriot-Watt University is the Sunday Times
>> Scottish University of the Year 2011-2012
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012




From x.liu at hw.ac.uk  Fri Oct  7 15:13:05 2011
From: x.liu at hw.ac.uk (xl127)
Date: Fri, 07 Oct 2011 12:13:05 +0100
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <CAKzWOxUTR3EOr+fUaC3RXXcz7G5Rq49S4ZgiDbTb1WfeDLqxgg@mail.gmail.com>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
	<CAKzWOxUTR3EOr+fUaC3RXXcz7G5Rq49S4ZgiDbTb1WfeDLqxgg@mail.gmail.com>
Message-ID: <4E8EDEC1.5040708@hw.ac.uk>

Thanks Michael!

I am not quite clear how to check if the outbound socket is answering a 
call by using FS global_setvar and global_getvar APIs,
but will have a look at them.

Cheers,
Xing


On 06/10/11 18:08, Michael Collins wrote:
> Perhaps you could rely on FreeSWITCH's global_setvar and global_getvar 
> APIs. You could perhaps create a global var whose name is the uuid and 
> then put some meaningful value in it. You would just need to make 
> absolutely certain that you clear out that variable manually, 
> otherwise you'll have a lot of junk variables.
>
> -MC
>
> On Thu, Oct 6, 2011 at 9:28 AM, xl127 <x.liu at hw.ac.uk 
> <mailto:x.liu at hw.ac.uk>> wrote:
>
>     Thanks for the suggestions!
>
>     I thought about and tried to make my outbound socket application
>     fail e.g. by send a hangup message but it didn't work.
>
>     What I did is that I set a public variable channelUUID with initial
>     value null in MyOutboundHandler,
>     I check if it is null in the handleConnectionResponse method. If
>     it not
>     null, a hangup command is sent to FS.
>     But it looks like each call channel creates its own Java thread,
>     so the
>     channelUUID is always null before the connection.
>     So my approach doesn't work.
>
>     Any idea about checking if the remote application is answering a call?
>
>     Thanks!
>     Xing
>
>
>     On 06/10/11 16:43, Anthony Minessale wrote:
>     > Really the socket app has not limit to how many simultaneous
>     > connections it can handle.
>     > You could try mod_limit, or make the remote end of you application
>     > fail when there are too many connections.
>     >
>     >
>     > On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk
>     <mailto:x.liu at hw.ac.uk>>  wrote:
>     >> Hello,
>     >>
>     >> I am trying to bridge multiple calls to multiple Java ESL outbound
>     >> socket applications.
>     >>
>     >> Basically I want to check if the first socket app is engaged
>     then the
>     >> call should be transferred
>     >> to the second socket app and so on.
>     >>
>     >> For initial test, I tried two concurrent calls, the problem is both
>     >> calls are always answered
>     >> by the first outbound socket.
>     >>
>     >> I have following setting in the dialplan:
>     >>
>     >> <extension name="sipgateDialedIn">
>     >> <condition field="destination_number" expression="^5800$">
>     >> <action application="set" data="continue_on_fail=true"/>
>     >> <action application="set" data="hangup_after_bridge=true"/>
>     >> <action application="bridge" data="loopback/5801"/>
>     >> <action application="bridge" data="loopback/5802"/>
>     >>
>     >> <action application="set" data="tts_engine=flite"/>
>     >> <action application="set" data="tts_voice=kal"/>
>     >> <action application="speak" data="Sorry, I am busy at the
>     moment. Please
>     >> try again later."/>
>     >> <action application="sleep" data="10"/>
>     >> <action application="hangup"/>
>     >> </condition>
>     >> </extension>
>     >>
>     >> <extension name="myOutboundESL5801">
>     >> <condition field="destination_number" expression="^5801$">
>     >> <action application="socket" data="*MailScanner has detected a
>     possible fraud attempt from "127.0.0.1:9981" claiming to be*
>     127.0.0.1:9981 <http://127.0.0.1:9981> async full"/>
>     >> </condition>
>     >> </extension>
>     >>
>     >> <extension name="myOutboundESL5802">
>     >> <condition field="destination_number" expression="^5802$">
>     >> <action application="socket" data="*MailScanner has detected a
>     possible fraud attempt from "127.0.0.1:9982" claiming to be*
>     127.0.0.1:9982 <http://127.0.0.1:9982> async full"/>
>     >> </condition>
>     >> </extension>
>     >>
>     >> I expect the first call to 5800 would be asnwered by extension
>     5801, the
>     >> second concurrent call to 5800 will be
>     >> answered by extension 5802, and the third call gets a busy message.
>     >>
>     >> I use Java ESL client-0.9.2 libraries. How do I check in the
>     dialplan if
>     >> the socket is engaged?
>     >>
>     >> I've tried and googled a while but without success.
>     >>
>     >> Any advices please?
>     >>
>     >> Many thanks!
>     >> Xing
>     >>
>     >>
>     >>
>     >> --
>     >> Heriot-Watt University is a Scottish charity
>     >> registered under charity number SC000278.
>     >>
>     >> Heriot-Watt University is the Sunday Times
>     >> Scottish University of the Year 2011-2012
>     >>
>     >>
>     >>
>     >>
>     >> 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
>     >>
>     >
>     >
>
>
>
>     --
>     Heriot-Watt University is a Scottish charity
>     registered under charity number SC000278.
>
>     Heriot-Watt University is the Sunday Times
>     Scottish University of the Year 2011-2012
>
>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012


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From justlikeef at gmail.com  Fri Oct  7 16:50:29 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Fri, 7 Oct 2011 08:50:29 -0400
Subject: [Freeswitch-users]
 =?iso-8859-15?q?Desktop_phone_-_answer_a_ringi?=
 =?iso-8859-15?q?ng_call_via=09API_call?=
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
References: <AcyCnbOIxZuxEaeRQIyKEkToMpcvaQ==>
	<CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
Message-ID: <201110070850.30154.justlikeef@gmail.com>

It looks like Polycoms can be told to "auto answer" with the SIP header "Alert-Info: Ring Answer";

If the "CTI Interface" had the UUID of the call and the extension, then a quick API call or LUA script could add the sip header, then reinvite the phone causing it to answer.

Check out the Paging and Intercom page on the WIKI for more info on the auto answer setup.

On Friday 07 October 2011 04:42:32 Weigel, Stefan wrote:
> Hi Michael, list,
> 
> 
> 
> that's the point.
> 
> I would really appreciate a solution for that "problem".
> 
> We're using FreeSwitch in an callcenter environment and people used to handle the calls through some kind of CTI interface before we switched to FreeSWITCH. So it would be great to have this feature. But there's no possibility to control Polycom phones that way.
> 
> 
> 
> Hints on this topic are welcome!
> 
> 
> 
> 
> 
> Thanks and best regards
> 
> 
> 
> Stefan
> 
> 
> 
> Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.
> 
> 
> 
> 
> 
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins
> Gesendet: Mittwoch, 5. Oktober 2011 18:42
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call
> 
> 
> 
> This would be a function of the phone. AFAICT, Polycom does not have a Webby way of doing this. However, I think Brian West has a bead on using the expansion port along with an Arduino board to do some manual/remote control of the phone. If anyone has more information on remote controlling Polycoms (or any other phones, for that matter) please let us know.
> 
> 
> 
> -MC
> 
> On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:
> 
> Hi all,
> 
> 
> 
> Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP 560) via api command ?
> 
> I tried with SendMsg <uuid> and 'answer' application, but it's not want I need/want.
> 
> 
> 
> 
> 
> Thanks and best regards,
> 
> 
> 
> Stefan
> 
> 
> 
> 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 Stefan.Weigel at allianz-warranty.com  Fri Oct  7 17:30:50 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 7 Oct 2011 15:30:50 +0200
Subject: [Freeswitch-users] Desktop phone - answer a ringing call
	via	API call
In-Reply-To: <201110070850.30154.justlikeef@gmail.com>
References: <AcyCnbOIxZuxEaeRQIyKEkToMpcvaQ==>
	<CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
	<201110070850.30154.justlikeef@gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36E84@AZWSMS03.azwarranty.int>

Hi Rob, list,



very funny, it's exactly what I'm trying to do at the moment.



In my dialplan I put:



<action application="export" data="alert_info=aa<http://www.notused.com;info=alert-external;x-line-id=0>"/>



In sip.cfg I put:



 <alertInfo voIpProt.SIP.alertInfo.1.value="aa" voIpProt.SIP.alertInfo.1.class="3" />

[..]

<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" />



When tracing I can see:



INVITE sip:567 at 192.168.1.1:5060 SIP/2.0

[..]

Alert-Info: <aa>

[..]



The '<>' tags are appended automatically. So I tried to modify sip.cfg a little bit:



<alertInfo voIpProt.SIP.alertInfo.1.value="&lt;aa&gt;" voIpProt.SIP.alertInfo.1.class="3" />



But it's not working. In the documentation I found a remark:



If set to 1, auto-answer is enabled for all SIP

calls.

Note: This parameter is supported on the

Polycom VVX 1500 only.



So I don't know if this feature will work for SIP calls with my 560.









Thanks and best regards,



Stefan





Von: Rob Hutton [mailto:justlikeef at gmail.com]
Gesendet: Freitag, 7. Oktober 2011 14:50
An: freeswitch-users at lists.freeswitch.org
Cc: Weigel, Stefan
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call



It looks like Polycoms can be told to "auto answer" with the SIP header "Alert-Info: Ring Answer";



If the "CTI Interface" had the UUID of the call and the extension, then a quick API call or LUA script could add the sip header, then reinvite the phone causing it to answer.



Check out the Paging and Intercom page on the WIKI for more info on the auto answer setup.



On Friday 07 October 2011 04:42:32 Weigel, Stefan wrote:

> Hi Michael, list,

>

>

>

> that's the point.

>

> I would really appreciate a solution for that "problem".

>

> We're using FreeSwitch in an callcenter environment and people used to handle the calls through some kind of CTI interface before we switched to FreeSWITCH. So it would be great to have this feature. But there's no possibility to control Polycom phones that way.

>

>

>

> Hints on this topic are welcome!

>

>

>

>

>

> Thanks and best regards

>

>

>

> Stefan

>

>

>

> Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.

>

>

>

>

>

> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins

> Gesendet: Mittwoch, 5. Oktober 2011 18:42

> An: FreeSWITCH Users Help

> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call

>

>

>

> This would be a function of the phone. AFAICT, Polycom does not have a Webby way of doing this. However, I think Brian West has a bead on using the expansion port along with an Arduino board to do some manual/remote control of the phone. If anyone has more information on remote controlling Polycoms (or any other phones, for that matter) please let us know.

>

>

>

> -MC

>

> On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

>

> Hi all,

>

>

>

> Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP 560) via api command ?

>

> I tried with SendMsg <uuid> and 'answer' application, but it's not want I need/want.

>

>

>

>

>

> Thanks and best regards,

>

>

>

> Stefan

>

>

>

> 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 cmrienzo at gmail.com  Fri Oct  7 17:35:31 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 7 Oct 2011 09:35:31 -0400
Subject: [Freeswitch-users] How I can get recog result
In-Reply-To: <4E8D7878.3060109@speechpro.com>
References: <4E8D7878.3060109@speechpro.com>
Message-ID: <CAD-qRkjondjXVrDEbVv2CEEjUR7uLaCh=stzKYdQ_H6=s8cEPQ@mail.gmail.com>

This is almost identical to a recent question in the UniMRCP mailing
list...  anyway, here's the repeated answer for the benefit of search
engine...

There currently aren't any dialplan applications that will do what you
want.  The recognition result is fired as a "DETECTED_SPEECH" event.  See
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech .

You can get the event either using the event socket library (ESL) or by
running a javascript or lua script that uses an input callback.  See
http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts or
http://wiki.freeswitch.org/wiki/ESL for more details.

Chris


On Thu, Oct 6, 2011 at 5:44 AM, Kirill Popov <popov-k at speechpro.com> wrote:

> Hello!
>
> I am just starting to learn FreeSWITCH few days ago.
> Before I have some experience in Asterisk and now I don't understand
> somethings.
>
> I try to use module mod_unimrcp in FreeSWITCH.
> I made simple dialplan and starting recognition
> <action application="detect_speech" data="unimrcp /usr/local/
> freeswitch/grammar/digits"/>
> It works.
>
> I see NLSML in console FS, but I don't understand how I can use it.
> In Asterisk NLSML-answer set in variable ${RECOG_RESULT}. How in FS?
> There isn't such variable.
>
> In Asterisk I can get this variable from perl-script, parse it, and
> get utterance, confidence, etc. How I can get these parameters in FS?
>
>
> --
> Best regards,
> Kirill
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Stefan.Weigel at allianz-warranty.com  Fri Oct  7 17:45:48 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 7 Oct 2011 15:45:48 +0200
Subject: [Freeswitch-users] Desktop phone - answer a ringing call via
 API call
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36E84@AZWSMS03.azwarranty.int>
References: <AcyCnbOIxZuxEaeRQIyKEkToMpcvaQ==>
	<CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
	<201110070850.30154.justlikeef@gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E84@AZWSMS03.azwarranty.int>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36E85@AZWSMS03.azwarranty.int>

Hi Rob, list,



<action application="export" data="sip_auto_answer=true"/>



did the trick!



Somebody could append 'Polycom 560' to the models working with this on the WIKI-page http://wiki.freeswitch.org/wiki/Intercom



I will play around, maybe I can do it now...







Thanks for the hint!





Best regards,



Stefan



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Weigel, Stefan
Gesendet: Freitag, 7. Oktober 2011 15:31
An: 'FreeSWITCH Users Help'
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call



Hi Rob, list,



very funny, it's exactly what I'm trying to do at the moment.



In my dialplan I put:



<action application="export" data="alert_info=aa<http://www.notused.com;info=alert-external;x-line-id=0>"/>



In sip.cfg I put:



 <alertInfo voIpProt.SIP.alertInfo.1.value="aa" voIpProt.SIP.alertInfo.1.class="3" />

[..]

<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" />



When tracing I can see:



INVITE sip:567 at 192.168.1.1:5060 SIP/2.0

[..]

Alert-Info: <aa>

[..]



The '<>' tags are appended automatically. So I tried to modify sip.cfg a little bit:



<alertInfo voIpProt.SIP.alertInfo.1.value="&lt;aa&gt;" voIpProt.SIP.alertInfo.1.class="3" />



But it's not working. In the documentation I found a remark:



If set to 1, auto-answer is enabled for all SIP

calls.

Note: This parameter is supported on the

Polycom VVX 1500 only.



So I don't know if this feature will work for SIP calls with my 560.









Thanks and best regards,



Stefan





Von: Rob Hutton [mailto:justlikeef at gmail.com]
Gesendet: Freitag, 7. Oktober 2011 14:50
An: freeswitch-users at lists.freeswitch.org
Cc: Weigel, Stefan
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call



It looks like Polycoms can be told to "auto answer" with the SIP header "Alert-Info: Ring Answer";



If the "CTI Interface" had the UUID of the call and the extension, then a quick API call or LUA script could add the sip header, then reinvite the phone causing it to answer.



Check out the Paging and Intercom page on the WIKI for more info on the auto answer setup.



On Friday 07 October 2011 04:42:32 Weigel, Stefan wrote:

> Hi Michael, list,

>

>

>

> that's the point.

>

> I would really appreciate a solution for that "problem".

>

> We're using FreeSwitch in an callcenter environment and people used to handle the calls through some kind of CTI interface before we switched to FreeSWITCH. So it would be great to have this feature. But there's no possibility to control Polycom phones that way.

>

>

>

> Hints on this topic are welcome!

>

>

>

>

>

> Thanks and best regards

>

>

>

> Stefan

>

>

>

> Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.

>

>

>

>

>

> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins

> Gesendet: Mittwoch, 5. Oktober 2011 18:42

> An: FreeSWITCH Users Help

> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call

>

>

>

> This would be a function of the phone. AFAICT, Polycom does not have a Webby way of doing this. However, I think Brian West has a bead on using the expansion port along with an Arduino board to do some manual/remote control of the phone. If anyone has more information on remote controlling Polycoms (or any other phones, for that matter) please let us know.

>

>

>

> -MC

>

> On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

>

> Hi all,

>

>

>

> Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP 560) via api command ?

>

> I tried with SendMsg <uuid> and 'answer' application, but it's not want I need/want.

>

>

>

>

>

> Thanks and best regards,

>

>

>

> Stefan

>

>

>

> 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 boris at tagnet.ru  Fri Oct  7 18:40:16 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 07 Oct 2011 20:40:16 +0600
Subject: [Freeswitch-users] continue_on_fail
Message-ID: <4E8F0F50.8010201@tagnet.ru>

Hello!

     Have read docs but misundertood :( I use mod_lcr which return two 
or more gateways to the same destination. And when the user is busy with 
the first gateway the call continues with the second and so on. How may 
I to not continue gateways if user is busy?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From jeff at jefflenk.com  Fri Oct  7 18:56:21 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 7 Oct 2011 07:56:21 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch mod_lua and email
In-Reply-To: <9CE5761907AF3242B8E5F41A2F99BBB2088509@AMSPRD0104MB137.eurprd01.prod.exchangelabs.com>
References: <9CE5761907AF3242B8E5F41A2F99BBB2088509@AMSPRD0104MB137.eurprd01.prod.exchangelabs.com>
Message-ID: <1317999381500-6869637.post@n2.nabble.com>

Is that your own voicemail.lua file? Otherwise what version of FS are you
running?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-mod-lua-and-email-tp6866280p6869637.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chris at ghosttelecom.com  Fri Oct  7 19:03:24 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Fri, 7 Oct 2011 16:03:24 +0100
Subject: [Freeswitch-users] Break command not working
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF35A179@SVR01.ghosttelecom.local>

Hi,

 

I have the following scenario...

 

application='preanswer'

application='set' data='execute_on_answer=break'

application='sched_broadcast' data='+1 /path/file'

application='bridge' data='sofia...'

 

This works fine in that it pre answers the call and plays back a welcome
message prior to normal early media ringtone.

 

The problem I have is that if the call is answered quickly the playback
just continues until finished.

 

I would have thought that application='set'
data='execute_on_answer=break' should stop the playback on answer but
the break command doesn't have any effect. I have tried break all to no
effect.

 

Should this work?

 

Any ideas on how to make this work?

 

Many thanks 

 

Chris

 

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From brad at tech21.com  Fri Oct  7 19:35:54 2011
From: brad at tech21.com (Brad Mina)
Date: Fri, 7 Oct 2011 08:35:54 -0700
Subject: [Freeswitch-users] Desktop phone - answer a ringing call via
	API call
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36E85@AZWSMS03.azwarranty.int>
References: <CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
	<201110070850.30154.justlikeef@gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E84@AZWSMS03.azwarranty.int>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E85@AZWSMS03.azwarranty.int>
Message-ID: <CAGRr05_y9VKqDpw1y-gzP7SV6ffSL8wH9h_E9Vg7M3_aRfdOQw@mail.gmail.com>

Could you possibly document on the wiki exactly what you did and link to it
from this thread?

On Fri, Oct 7, 2011 at 6:45 AM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

>  Hi Rob, list,****
>
> ** **
>
> <action application="export" data="sip_auto_answer=true"/>****
>
> ** **
>
> did the trick!****
>
> ** **
>
> Somebody could append ?Polycom 560? to the models working with this on the
> WIKI-page http://wiki.freeswitch.org/wiki/Intercom****
>
> ** **
>
> I will play around, maybe I can do it now?****
>
> ** **
>
> ** **
>
> ** **
>
> Thanks for the hint!****
>
> ** **
>
> ** **
>
> Best regards,****
>
> ** **
>
> Stefan****
>
> ** **
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Weigel,
> Stefan
> *Gesendet:* Freitag, 7. Oktober 2011 15:31
>
> *An:* 'FreeSWITCH Users Help'
> *Betreff:* Re: [Freeswitch-users] Desktop phone - answer a ringing call
> via API call****
>
>  ** **
>
> Hi Rob, list,****
>
> ** **
>
> very funny, it?s exactly what I?m trying to do at the moment.****
>
> ** **
>
> In my dialplan I put:****
>
> ** **
>
> <action application="export" data="alert_info=aa <http://www.notused.com;info=alert-external;x-line-id=0>"/>****
>
> ** **
>
> In sip.cfg I put:****
>
> ** **
>
>  <alertInfo voIpProt.SIP.alertInfo.1.value="aa"
> voIpProt.SIP.alertInfo.1.class="3" />****
>
> [..]****
>
> <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" />****
>
> ** **
>
> When tracing I can see:****
>
> ** **
>
> INVITE sip:567 at 192.168.1.1:5060 SIP/2.0****
>
> [..]****
>
> Alert-Info: <aa>****
>
> [..]****
>
> ** **
>
> The ?<>? tags are appended automatically. So I tried to modify sip.cfg a
> little bit:****
>
> ** **
>
> <alertInfo voIpProt.SIP.alertInfo.1.value="&lt;aa&gt;"
> voIpProt.SIP.alertInfo.1.class="3" />****
>
> ** **
>
> But it?s not working. In the documentation I found a remark:****
>
> ** **
>
> If set to 1, auto-answer is enabled for all SIP****
>
> calls.****
>
> *Note**: This parameter is supported on the*
>
> *Polycom VVX 1500 only.*****
>
> ** **
>
> So I don?t know if this feature will work for SIP calls with my 560.****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> Thanks and best regards,****
>
> ** **
>
> Stefan****
>
> ** **
>
> ** **
>
> *Von:* Rob Hutton [mailto:justlikeef at gmail.com]
> *Gesendet:* Freitag, 7. Oktober 2011 14:50
> *An:* freeswitch-users at lists.freeswitch.org
> *Cc:* Weigel, Stefan
> *Betreff:* Re: [Freeswitch-users] Desktop phone - answer a ringing call
> via API call****
>
> ** **
>
> It looks like Polycoms can be told to "auto answer" with the SIP header "Alert-Info:
> Ring Answer";****
>
> ** **
>
> If the "CTI Interface" had the UUID of the call and the extension, then a
> quick API call or LUA script could add the sip header, then reinvite the
> phone causing it to answer.****
>
> ** **
>
> Check out the Paging and Intercom page on the WIKI for more info on the
> auto answer setup.****
>
> ** **
>
> On Friday 07 October 2011 04:42:32 Weigel, Stefan wrote:****
>
> > Hi Michael, list,****
>
> > ****
>
> > ****
>
> > ****
>
> > that's the point.****
>
> > ****
>
> > I would really appreciate a solution for that "problem".****
>
> > ****
>
> > We're using FreeSwitch in an callcenter environment and people used to
> handle the calls through some kind of CTI interface before we switched to
> FreeSWITCH. So it would be great to have this feature. But there's no
> possibility to control Polycom phones that way.****
>
> > ****
>
> > ****
>
> > ****
>
> > Hints on this topic are welcome!****
>
> > ****
>
> > ****
>
> > ****
>
> > ****
>
> > ****
>
> > Thanks and best regards****
>
> > ****
>
> > ****
>
> > ****
>
> > Stefan****
>
> > ****
>
> > ****
>
> > ****
>
> > Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.****
>
> > ****
>
> > ****
>
> > ****
>
> > ****
>
> > ****
>
> > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael
> Collins****
>
> > Gesendet: Mittwoch, 5. Oktober 2011 18:42****
>
> > An: FreeSWITCH Users Help****
>
> > Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via
> API call****
>
> > ****
>
> > ****
>
> > ****
>
> > This would be a function of the phone. AFAICT, Polycom does not have a
> Webby way of doing this. However, I think Brian West has a bead on using the
> expansion port along with an Arduino board to do some manual/remote control
> of the phone. If anyone has more information on remote controlling Polycoms
> (or any other phones, for that matter) please let us know.****
>
> > ****
>
> > ****
>
> > ****
>
> > -MC****
>
> > ****
>
> > On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan <
> Stefan.Weigel at allianz-warranty.com<mailto:
> Stefan.Weigel at allianz-warranty.com>> wrote:****
>
> > ****
>
> > Hi all,****
>
> > ****
>
> > ****
>
> > ****
>
> > Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP
> 560) via api command ?****
>
> > ****
>
> > I tried with SendMsg <uuid> and 'answer' application, but it's not want I
> need/want.****
>
> > ****
>
> > ****
>
> > ****
>
> > ****
>
> > ****
>
> > Thanks and best regards,****
>
> > ****
>
> > ****
>
> > ****
>
> > Stefan****
>
> > ****
>
> > ****
>
> > ****
>
> > FreeSWITCH-users mailing list****
>
> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>****
>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> ****
>
> > http://www.freeswitch.org****
>
> > ****
>
> > ****
>
> > ****
>
> > ****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Oct  7 20:47:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 11:47:19 -0500
Subject: [Freeswitch-users] Break command not working
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF35A179@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF35A179@SVR01.ghosttelecom.local>
Message-ID: <CAKbxfG90QQTtnsB0DsxMipPWpkJZXemTsKJ5fiDsPwqjgz9MuA@mail.gmail.com>

you can't execute the break because it's busy playing the file, you
can do the break on the other leg though,
This is more of what you want.

    <extension name="1235">
      <condition field="destination_number" expression="^1235$">
        <action application="set" data="execute_on_media=playback::/path/file"/>
        <action application="pre_answer"/>
        <action application="bridge"
data="{ignore_early_media=true,api_on_answer='uuid_break
${uuid}'}sofia/internal/foo at bar.com"/>
      </condition>
    </extension>


On Fri, Oct 7, 2011 at 10:03 AM, Chris Martineau <chris at ghosttelecom.com> wrote:
> Hi,
>
>
>
> I have the following scenario...
>
>
>
> application='preanswer'
>
> application='set' data='execute_on_answer=break'
>
> application='sched_broadcast' data='+1 /path/file'
>
> application='bridge' data=?sofia...?
>
>
>
> This works fine in that it pre answers the call and plays back a welcome
> message prior to normal early media ringtone.
>
>
>
> The problem I have is that if the call is answered quickly the playback just
> continues until finished.
>
>
>
> I would have thought that application='set' data='execute_on_answer=break'
> should stop the playback on answer but the break command doesn?t have any
> effect. I have tried break all to no effect.
>
>
>
> Should this work?
>
>
>
> Any ideas on how to make this work?
>
>
>
> Many thanks
>
>
>
> Chris
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Fri Oct  7 21:19:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 12:19:37 -0500
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <4E8EDC90.3030102@hw.ac.uk>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
	<CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
	<4E8EDC90.3030102@hw.ac.uk>
Message-ID: <CAKbxfG9WO20aEdbo25T3C7L23Ws5hv8txH_3QkuZmQmh87gc2w@mail.gmail.com>

really you should just fork in one process or use the supplied ivrd to
launch a script connected to the sockets by stdin/stdout....
but I tried to answer your question for what it was rather than steer
you off course but I do think your approach is somewhat messed up ;)


On Fri, Oct 7, 2011 at 6:03 AM, xl127 <x.liu at hw.ac.uk> wrote:
> Thanks, Anthony!
>
> I will check David Varnes's Java ESL API again to see how to check
> number of channel opend and how to disconnect the new channel.
>
> Yeah I will look into mod_limit as well.
>
> Cheers,
> Xing
>
>
>
> On 06/10/11 20:20, Anthony Minessale wrote:
>> don't actually hang it up just don't connect the socket or connnect it
>> and disconnect it right away and the app will fail on in the
>> dial-plan.
>>
>> again you could try mod_limit otherwise.
>>
>>
>> On Thu, Oct 6, 2011 at 11:28 AM, xl127<x.liu at hw.ac.uk> ?wrote:
>>> Thanks for the suggestions!
>>>
>>> I thought about and tried to make my outbound socket application
>>> fail e.g. by send a hangup message but it didn't work.
>>>
>>> What I did is that I set a public variable channelUUID with initial
>>> value null in MyOutboundHandler,
>>> I check if it is null in the handleConnectionResponse method. If it not
>>> null, a hangup command is sent to FS.
>>> But it looks like each call channel creates its own Java thread, so the
>>> channelUUID is always null before the connection.
>>> So my approach doesn't work.
>>>
>>> Any idea about checking if the remote application is answering a call?
>>>
>>> Thanks!
>>> Xing
>>>
>>>
>>> On 06/10/11 16:43, Anthony Minessale wrote:
>>>> Really the socket app has not limit to how many simultaneous
>>>> connections it can handle.
>>>> You could try mod_limit, or make the remote end of you application
>>>> fail when there are too many connections.
>>>>
>>>>
>>>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk> ? ?wrote:
>>>>> Hello,
>>>>>
>>>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>>>> socket applications.
>>>>>
>>>>> Basically I want to check if the first socket app is engaged then the
>>>>> call should be transferred
>>>>> to the second socket app and so on.
>>>>>
>>>>> For initial test, I tried two concurrent calls, the problem is both
>>>>> calls are always answered
>>>>> by the first outbound socket.
>>>>>
>>>>> I have following setting in the dialplan:
>>>>>
>>>>> <extension name="sipgateDialedIn">
>>>>> <condition field="destination_number" expression="^5800$">
>>>>> <action application="set" data="continue_on_fail=true"/>
>>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>>> <action application="bridge" data="loopback/5801"/>
>>>>> <action application="bridge" data="loopback/5802"/>
>>>>>
>>>>> <action application="set" data="tts_engine=flite"/>
>>>>> <action application="set" data="tts_voice=kal"/>
>>>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>>>> try again later."/>
>>>>> <action application="sleep" data="10"/>
>>>>> <action application="hangup"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> <extension name="myOutboundESL5801">
>>>>> <condition field="destination_number" expression="^5801$">
>>>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> <extension name="myOutboundESL5802">
>>>>> <condition field="destination_number" expression="^5802$">
>>>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>>>> second concurrent call to 5800 will be
>>>>> answered by extension 5802, and the third call gets a busy message.
>>>>>
>>>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>>>> the socket is engaged?
>>>>>
>>>>> I've tried and googled a while but without success.
>>>>>
>>>>> Any advices please?
>>>>>
>>>>> Many thanks!
>>>>> Xing
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Heriot-Watt University is a Scottish charity
>>>>> registered under charity number SC000278.
>>>>>
>>>>> Heriot-Watt University is the Sunday Times
>>>>> Scottish University of the Year 2011-2012
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>>
>>> --
>>> Heriot-Watt University is a Scottish charity
>>> registered under charity number SC000278.
>>>
>>> Heriot-Watt University is the Sunday Times
>>> Scottish University of the Year 2011-2012
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
> Heriot-Watt University is the Sunday Times
> Scottish University of the Year 2011-2012
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Fri Oct  7 21:59:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 12:59:27 -0500
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <CAKrPhPM4MV5XQ8oBop-PvwAZbMQM9S422wH-UMDK3n1whvVf+w@mail.gmail.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
	<1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
	<CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>
	<1317308654166-6844212.post@n2.nabble.com>
	<CAKrPhPMA3cngUy9hkNyucbzOv=GwAjN6qWUQNFJ=K4+koEcAEA@mail.gmail.com>
	<CAKrPhPM4MV5XQ8oBop-PvwAZbMQM9S422wH-UMDK3n1whvVf+w@mail.gmail.com>
Message-ID: <CAKbxfG8wM64mXMAN2WtjW=YcvjE9RN-WrzcVNUv8tNjrkLXYcA@mail.gmail.com>

There is no keep alive functionality in the core XMPP library we use
in the core of libdingaling (IKSEMEL)

One option would to have a NAT that doesn't disconnect active
connections just because they are idle (which seems bad to me).

Otherwise you could work on a patch, request a feature or come up with
some cron job maybe to use the chat api command to send a message to
your server from cron or something as a workaround.



On Thu, Oct 6, 2011 at 2:18 AM, Federico Beffa <beffa at ieee.org> wrote:
> Unfortunately I was singing victory too early. My problem still persists...
>
> The git version I'm running is from the 20th of August. In the mean
> time have there been changes which could affect mod_dingaling? Looking
> in the git commit comments I have the impression that the answer is
> no.
>
> Regards,
> Fede
>
> On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa at ieee.org> wrote:
>> Just in case someone will hit the same problem...
>>
>> After some tests, I've found that, changing "bind_server_ip" from
>> "auto" to the local IP address of the FS server (192.168.X.X), appears
>> to solve my timeout problem. ?(The account did not go offline for more
>> than 12 hours.)
>>
>> To summarize my current configuration: I have a fix public IP assigned
>> to my router (NAT/Firewall). NAT is configured to forward ports 5080
>> and 5081 to the host on which FS is running and UPnP is disabled.
>>
>> I'm running a FS version checked out from git the 20th of August.
>>
>> Regards,
>> Fede
>>
>> On Thu, Sep 29, 2011 at 5:04 PM, mazilo
>> <Nabble_01394 at slickdeals.endjunk.com> wrote:
>>> The only thing I can say is to update your FS with the latest git commit.
>>>
>>> -----
>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
>>> --
>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-offline-mod-dingaling-tp6839355p6844212.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From saurabh4ubuntu at gmail.com  Fri Oct  7 08:37:10 2011
From: saurabh4ubuntu at gmail.com (Ubuntu Rocks)
Date: Fri, 7 Oct 2011 10:07:10 +0530
Subject: [Freeswitch-users] rtp issues with mod_dingaling
Message-ID: <CAPr=v4MXo-Rg0pKh0wQ9yCK1Y3QStKfU+gY4L6gUaunqGgma9Q@mail.gmail.com>

Hi All,

I am trying to get mod_dingaling working with gtalk.
I  have configured extention 1004 in jingle_profiles/client.xml.
I make a call from another gtalk user,the call lands up fine at the
extention I have
configured.But there is no audio.I searched a lot and have tried everything
I found
on the net including using different STUN servers, using different values
for candidate-acl,disable-rtp-auto-adjust. etc.

                                        I enabled debugging in mod_dingaling
and found in the xmpp messages that along with a
local RTP ,and a stun RTP a relay RTP port is being advertized by the google
server,I ran wireshark on my freeswitch server and looks like
the audio is going to the stun RTP port not to the realy RTP port.
Can this be a problem?

Does gtalk want the audio to be sent directly to the concerned gtalk client
or does it have to go through the google server?
The FreeSwitch vesrion I am using is ' FreeSWITCH version: 1.0.7
(svn-exported)'.


Please find below my jingle_profiles/client.xml.

 <include>
  <!-- Client Profile (Original mode) -->
  <!-- to use this profile take the x- away from the open and close tags so
its <profile> and </profile> -->
  <profile type="client">
    <param name="name" value="jclient"/>
    <param name="login" value="XXXXXXXX at gmail.com/gtalk"/>
    <param name="password" value="XXXXX"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="auto-login" value="true"/>
    <!-- SASL "plain" or "md5" -->
    <param name="sasl" value="plain"/>
    <!-- if the server where the jabber is hosted is not the same as the one
in the jid -->
    <param name="server" value="talk.google.com"/>
    <!-- Enable TLS or not -->
    <param name="tls" value="true"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
   <param name="disable-rtp-auto-adjust" value="true"/>

    <!-- default extension (if one cannot be determined) -->
    <param name="exten" value="1004"/>
    <!-- VAD choose one -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <param name="vad" value="both"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <param name="candidate-acl" value="wan.auto"/>
    <param name="local-network-acl" value="localnet.auto"/>
  </profile>
</include>


I will appreciate any help or pointers on this.

--
Thanks
Saurabh
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From msc at freeswitch.org  Sat Oct  8 00:22:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Oct 2011 13:22:38 -0700
Subject: [Freeswitch-users] Call bridge problem
In-Reply-To: <23FE1289-CFC1-4C5E-AAAC-7DCB759BDD3E@lyonl.com>
References: <23FE1289-CFC1-4C5E-AAAC-7DCB759BDD3E@lyonl.com>
Message-ID: <CAKzWOxUemzAA3RKt+JLVD4zXgoJbjOHm1VwEKe6QRPL16JzBEg@mail.gmail.com>

Snag a console debug log along with the SIP trace and drop it into
pastebin.freeswitch.org. Be sure to use "FreeSWITCH Log" as the syntax
highlighting. Put the URL to that pb in this thread and the gang will take a
look.

-MC

On Thu, Oct 6, 2011 at 12:59 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>   Hello,
>
>  I'm having a problem bridging out calls, FS is sending 500 Internal
> Server Error to our gateway. Any thoughts why this is?
>
>
>  <include>
>    <context name="public">
>      <extension name="unloop">
>        <condition field="${unroll_loops}" expression="^true$"/>
>        <condition field="${sip_looped_call}" expression="^true$">
>          <action application="deflect" data="${destination_number}"/>
>        </condition>
>      </extension>
>      <extension name="outside_call" continue="true">
>        <condition>
>          <action application="set" data="outside_call=true"/>
>          <action application="set" data="RFC2822_DATE=${strftime(%a, %d %b
> %Y %T %z)}"/>
>        </condition>
>      </extension>
>
>      <extension name="call_debug" continue="true">
>        <condition field="${call_debug}" expression="^true$" break="never">
>          <action application="info"/>
>        </condition>
>      </extension>
>      <extension name="4142211800">
>        <condition field="destination_number" expression="^(\+?14142211800
> )$">
>          <action application="set" data="effective_caller_id_name=LyonL"/>
>          <action application="set" data="effective_caller_id_number=
> +14142211800"/>
>          <action application="bridge" data="sofia/gateway/L3CN/
> +15618911806"/>
>        </condition>
>      </extension>
>    </context>
>  </include>
>
>
>
>  <profile name="external">
>    <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>    <!-- This profile is only for outbound registrations to providers -->
>    <gateways>
>      <gateway name="L3CN">
>        <param name="username" value="1-F2la9"/>
>        <param name="password" value="password"/>
>        <param name="realm" value="BroadWorks"/>
>        <param name="proxy" value="4.55.35.60:5070"/>
>        <param name="from-domain" value="69.65.114.181:5060"/>
>        <param name="dtmf-type" value="rfc2833"/>
>        <param name="extension-in-contact" value="true"/>
>        <param name="caller-id-in-from" value="true"/>
>        <param name="register" value="false"/>
>      </gateway>
>    </gateways>
>
>    <aliases>
>      <!--
>      <alias name="outbound"/>
>      <alias name="nat"/>
>      -->
>    </aliases>
>
>    <domains>
>      <domain name="all" alias="false" parse="true"/>
>    </domains>
>
>    <settings>
>      <param name="debug" value="0"/>
>      <!-- If you want FreeSWITCH to shutdown if this profile fails to
> load, uncomment the next line. -->
>      <!-- <param name="shutdown-on-fail" value="true"/> -->
>      <param name="sip-trace" value="no"/>
>      <param name="sip-capture" value="no"/>
>      <param name="rfc2833-pt" value="101"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="public"/>
>      <param name="dtmf-duration" value="2000"/>
>      <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>      <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>      <param name="hold-music" value="$${hold_music}"/>
>      <param name="rtp-timer-name" value="soft"/>
>      <!--<param name="enable-100rel" value="true"/>-->
>      <!--<param name="disable-srv503" value="true"/>-->
>      <!-- This could be set to "passive" -->
>      <param name="local-network-acl" value="localnet.auto"/>
>      <param name="manage-presence" value="false"/>
>
>      <param name="user-agent-string" value="LyonL"/>
>
>      <!-- used to share presence info across sofia profiles
>  manage-presence needs to be set to passive on this profile
>  if you want it to behave as if it were the internal profile
>  for presence.
>      -->
>      <!-- Name of the db to use for this profile -->
>      <!--<param name="dbname" value="share_presence"/>-->
>      <!--<param name="presence-hosts" value="$${domain}"/>-->
>      <!--<param name="force-register-domain" value="$${domain}"/>-->
>      <!--all inbound reg will stored in the db using this domain -->
>      <!--<param name="force-register-db-domain" value="$${domain}"/>-->
>      <!-- ************************************************* -->
>
>      <!--<param name="aggressive-nat-detection" value="true"/>-->
>      <param name="inbound-codec-negotiation" value="greedy"/>
>      <param name="nonce-ttl" value="60"/>
>      <param name="auth-calls" value="false"/>
>      <param name="NDLB-force-rport" value="server-only"/>
>      <!--
>  DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
>      -->
>      <param name="rtp-ip" value="$${external_rtp_ip}"/>
>      <param name="sip-ip" value="$${external_sip_ip}"/>
>      <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>      <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>      <param name="sip-port" value="$${external_sip_port}"/>
>      <param name="rtp-timeout-sec" value="300"/>
>      <param name="rtp-hold-timeout-sec" value="1800"/>
>      <!--<param name="enable-3pcc" value="true"/>-->
>
>      <!-- TLS: disabled by default, set to "true" to enable -->
>      <param name="tls" value="$${external_ssl_enable}"/>
>      <!-- additional bind parameters for TLS -->
>      <param name="tls-bind-params" value="transport=tls"/>
>      <!-- Port to listen on for TLS requests. (5081 will be used if
> unspecified) -->
>      <param name="tls-sip-port" value="$${external_tls_port}"/>
>      <!-- Location of the agent.pem and cafile.pem ssl certificates
> (needed for TLS server) -->
>      <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
>      <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
> work with TLSv1 -->
>      <param name="tls-version" value="$${sip_tls_version}"/>
>
>    </settings>
>  </profile>
>
>
>  send 1283 bytes to udp/[4.55.35.60]:5070 at 19:50:28.998153:
>    ------------------------------------------------------------------------
>    INVITE sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
>    Via: SIP/2.0/UDP 69.65.114.181;rport;branch=z9hG4bK62S7Fr2Um459j
>    Max-Forwards: 8
>    From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
>    To: <sip:+15618911806 at 4.55.35.60:5070>
>    Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
>    CSeq: 18629251 INVITE
>    Contact: <sip:1-F2la9 at 69.65.114.181:5060;transport=udp;gw=L3CN>
>    Expires: 3600
>    User-Agent: LyonL
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, RE
> FER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Authorization: Digest username="1-F2la9", realm="BroadWorks",
> nonce="BroadWor
> ksXgtg5vbouT4lpitdBW", cnonce="SKxoW2r3Ei+9o+X8ky5idg", algorithm=MD5,
> uri="sip:
> +15618911806 at 4.55.35.60:5070",
> response="9f0325808936e22ae050921c176f2730", qop=
> auth, nc=00000001
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 229
>    X-FS-Support: update_display
>    Remote-Party-ID: "LyonL" <sip:+14142211800 at 69.65.114.181:5060
> >;party=calling;
> screen=yes;privacy=off
>
>     v=0
>    o=FreeSWITCH 1317910968 1317910969 IN IP4 69.65.114.181
>    s=FreeSWITCH
>    c=IN IP4 69.65.114.181
>    t=0 0
>    m=audio 19660 RTP/AVP 0 8 18 101 13
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 297 bytes from udp/[4.55.35.60]:5070 at 19:50:29.002153:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP
> 69.65.114.181;received=69.65.114.181;branch=z9hG4bK62S7Fr2Um
> 459j;rport=5060
>    From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
>    To: <sip:+15618911806 at 4.55.35.60:5070>
>    Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
>    CSeq: 18629251 INVITE
>
>
>  ------------------------------------------------------------------------
> recv 468 bytes from udp/[4.55.35.60]:5070 at 19:50:29.008153:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP
> 69.65.114.181;received=69.65.114.181;branch=z9hG4bK62S7Fr2Um
> 459j;rport=5060
>    From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
>    To: <sip:+15618911806 at 4.55.35.60:5070
> >;tag=SDcpmkf99-1282242359-1317930631679
>
>     Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
>    CSeq: 18629251 INVITE
>    WWW-Authenticate: DIGEST
> qop="auth",nonce="BroadWorksXgtg5vbpbTeh3n5wBW",algo
> rithm=MD5,realm="BroadWorks"
>    Content-Length: 0
>
>
>  ------------------------------------------------------------------------
> send 367 bytes to udp/[4.55.35.60]:5070 at 19:50:29.008153:
>    ------------------------------------------------------------------------
>    ACK sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
>    Via: SIP/2.0/UDP 69.65.114.181;rport;branch=z9hG4bK62S7Fr2Um459j
>    Max-Forwards: 8
>    From: "LyonL" <sip:+14142211800 at 69.65.114.181:5060>;tag=K0gUSK5HNr3US
>    To: <sip:+15618911806 at 4.55.35.60:5070
> >;tag=SDcpmkf99-1282242359-1317930631679
>
>     Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276
>    CSeq: 18629251 ACK
>    Content-Length: 0
>
>
>  ------------------------------------------------------------------------
> 2011-10-06 19:50:28.979152 [NOTICE] sofia.c:5897 Hangup
> sofia/external/+15618911
> 806 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
> 2011-10-06 19:50:28.979152 [INFO] mod_dptools.c:2810 Originate Failed.
>  Cause: N
> ORMAL_UNSPECIFIED
> 2011-10-06 19:50:28.979152 [NOTICE] mod_dptools.c:2929 Hangup
> sofia/external/+15
> 618911806 at 4.55.35.60 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> send 380 bytes to udp/[4.55.35.60]:5070 at 19:50:29.010153:
>    ------------------------------------------------------------------------
>    SIP/2.0 500 Internal Server Error
>    Via: SIP/2.0/UDP 4.55.35.60:5070;branch=z9hG4bKtil5pp004o3hvjs9n2c1.1
>    From: <sip:+15618911806 at 4.55.35.60;user=phone
> >;tag=SDesfje01-1691283656-13179
> 30631601-
>    To: "4142211800 4142211800" <sip:+14142211800 at 69.65.114.181
> >;tag=jQQ2QrmerFD9
> D
>    Call-ID: SDesfje01-bc0436bf1c3c496f2c13c86f66280e15-v3000i1
>    CSeq: 761577689 INVITE
>    Content-Length: 0
>
>
>  ------------------------------------------------------------------------
> 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1367 Session 19
> (sofia
> /external/+15618911806 at 4.55.35.60) Ended
> 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1369 Close
> Channel sof
> ia/external/+15618911806 at 4.55.35.60 [CS_DESTROY]
> 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1367 Session 20
> (sofia
> /external/+15618911806) Ended
> 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1369 Close
> Channel sof
> ia/external/+15618911806 [CS_DESTROY]
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fcintrono at gmail.com  Sat Oct  8 01:05:24 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Fri, 7 Oct 2011 16:05:24 -0500
Subject: [Freeswitch-users] Newbie question...
Message-ID: <CANtw_ZNnE9v7+EHhnjE+8V9_UFoxuwWNJ=bj6Jiiu4AuMWpVdg@mail.gmail.com>

Ok guys, I am a newbie in VOIP stuff and in freeswitch of course.

I got the freeswitch book and installed freeswitch 1.0.6 in my windows 7 pro
rig.

I also installed XLite 4 but when I try to register the extension is telling
me: "Acount failed to enable" SIP 408 error.

I am sure I stoped my firewall and  even with that I am getting same error.

I have not changed anything from the default configuration, the extension I
am trying to register is extension 1000, with password 1234.


?Could you help me please?

Thanks in advance.
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From Hector.Geraldino at ip-soft.net  Sat Oct  8 01:10:02 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Fri, 7 Oct 2011 17:10:02 -0400
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <4E8EDA84.5000806@hw.ac.uk>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
	<CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD02206ED4A1@NY1-EXMB-01.ip-soft.net>
	<4E8EDA84.5000806@hw.ac.uk>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02209F7DF3@NY1-EXMB-01.ip-soft.net>

Hi,

Most of the issues you've can be addressed with a 'single instanced' application, but I understand why you designed the application to work this way. If at any moment you think about enabling this application to handle several calls, you can overtake these issues in various ways, for example:

- Controlling the number of calls: you can register all answered calls (uuids) in a data structure (list, map), remove them when disconnected, and decide when to accept a new call by getting the collection size and comparing it to your parameter. If you ever get more Nuance licenses, increasing the number of managed calls would be as easy as change a parameter in a property file.

- You can have as many MRCP profiles as you want defined in the configuration xml files on mrcp_profiles/ directory. Coming up with the algorithm to assign channels to MRCP engines (Nuance server/port) won't be that hard. I can think of one: having all mrcp profiles in a map with a flag (in use/free), when the call comes I lock the first available profile so it's attached to this call. When the call is disconnected the profile is freed.

- log mixed, sure. But, is this really a problem? The same with the publishing of topics, you can replicate the same behavior of having multiple instances in a single one. I mean, if you make an abstraction and say: one instance of my java application equals one thread of my new java app you'll see how easy is to overcome this and other problems.

Sorry for such a long message, but as many others I feel real passion about application design topics. My intention is never to criticize what you've done, as each mind is a different world.  

Anyway, good luck with that!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xl127
Sent: Friday, October 07, 2011 6:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected

Hi Hector,

Your points make sense but I have some specific issues for my 
application so I thought of binding only one call per runnning instance.

1. My outbound socket application is one component/server of our Spoken 
Dialogue System (SDS).
We have four ports Nuance ASR/TTS licenses, each port is specific to one 
machine( yeah, we have four seperate license files for four machines).
It seems to be difficult to specify which channel uses which Nuance MRCP 
profile if simultenous calls are answered by only one instance.

2. It seems to me that the log info would be mixed for simultenous calls 
if running one instance.

3. My socket app will be communicating with other components/servers of 
the SDS via ICE middleware (http://www.zeroc.com/).
It will register to the central ICE server when starting the system via 
a Topic name which is a static name given by the developer,
it will keep publishing ASR results on that topic so the other 
components listening on the topic will get the results,
and it also subscribes and listens to other topics from other components 
for TTS.

In simultenous calls to one running instance scenario, the 
distinguishable thing between channels would be the channel UUID which I 
could use
as a part of the topic name, but it is dynamic and other servers could 
not know it when registering to the central server.

I could let the all channels publish results with a same static topic 
name and channel UUID, the interested parties could parse the results
they want based on the UUID, but I worried this would cause more burdens 
to the whole system.


4. Even if running one instance, I still need some ways to check and 
control the maximum number of calls as we have limited ASR/TTS resources.

Cheers,
Xing

On 06/10/11 22:34, Hector Geraldino wrote:
> This might sound a little bit Off-Topic but, why are you willing to bind only one call per running instance? I cannot imagine any user scenario where this is desirable. Maybe as a proof of concept is Ok, but having an equivalence between calls and instances mmm... I don't know.
>
> Can you please tell me what are you willing to accomplish? I'm really curious about it.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Thursday, October 06, 2011 3:20 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
>
> don't actually hang it up just don't connect the socket or connnect it
> and disconnect it right away and the app will fail on in the
> dial-plan.
>
> again you could try mod_limit otherwise.
>
>
> On Thu, Oct 6, 2011 at 11:28 AM, xl127<x.liu at hw.ac.uk>  wrote:
>> Thanks for the suggestions!
>>
>> I thought about and tried to make my outbound socket application
>> fail e.g. by send a hangup message but it didn't work.
>>
>> What I did is that I set a public variable channelUUID with initial
>> value null in MyOutboundHandler,
>> I check if it is null in the handleConnectionResponse method. If it not
>> null, a hangup command is sent to FS.
>> But it looks like each call channel creates its own Java thread, so the
>> channelUUID is always null before the connection.
>> So my approach doesn't work.
>>
>> Any idea about checking if the remote application is answering a call?
>>
>> Thanks!
>> Xing
>>
>>
>> On 06/10/11 16:43, Anthony Minessale wrote:
>>> Really the socket app has not limit to how many simultaneous
>>> connections it can handle.
>>> You could try mod_limit, or make the remote end of you application
>>> fail when there are too many connections.
>>>
>>>
>>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk>    wrote:
>>>> Hello,
>>>>
>>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>>> socket applications.
>>>>
>>>> Basically I want to check if the first socket app is engaged then the
>>>> call should be transferred
>>>> to the second socket app and so on.
>>>>
>>>> For initial test, I tried two concurrent calls, the problem is both
>>>> calls are always answered
>>>> by the first outbound socket.
>>>>
>>>> I have following setting in the dialplan:
>>>>
>>>> <extension name="sipgateDialedIn">
>>>> <condition field="destination_number" expression="^5800$">
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="bridge" data="loopback/5801"/>
>>>> <action application="bridge" data="loopback/5802"/>
>>>>
>>>> <action application="set" data="tts_engine=flite"/>
>>>> <action application="set" data="tts_voice=kal"/>
>>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>>> try again later."/>
>>>> <action application="sleep" data="10"/>
>>>> <action application="hangup"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5801">
>>>> <condition field="destination_number" expression="^5801$">
>>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5802">
>>>> <condition field="destination_number" expression="^5802$">
>>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>>> second concurrent call to 5800 will be
>>>> answered by extension 5802, and the third call gets a busy message.
>>>>
>>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>>> the socket is engaged?
>>>>
>>>> I've tried and googled a while but without success.
>>>>
>>>> Any advices please?
>>>>
>>>> Many thanks!
>>>> Xing
>>>>
>>>>
>>>>
>>>> --
>>>> Heriot-Watt University is a Scottish charity
>>>> registered under charity number SC000278.
>>>>
>>>> Heriot-Watt University is the Sunday Times
>>>> Scottish University of the Year 2011-2012
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>> --
>> Heriot-Watt University is a Scottish charity
>> registered under charity number SC000278.
>>
>> Heriot-Watt University is the Sunday Times
>> Scottish University of the Year 2011-2012
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012




FreeSWITCH-users mailing list
FreeSWITCH-users at 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  Sat Oct  8 01:19:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Oct 2011 14:19:45 -0700
Subject: [Freeswitch-users] Newbie question...
In-Reply-To: <CANtw_ZNnE9v7+EHhnjE+8V9_UFoxuwWNJ=bj6Jiiu4AuMWpVdg@mail.gmail.com>
References: <CANtw_ZNnE9v7+EHhnjE+8V9_UFoxuwWNJ=bj6Jiiu4AuMWpVdg@mail.gmail.com>
Message-ID: <CAKzWOxXgOC4+0tZmZRdPJobQXXOef7voubUcuUnU-GwbKT0uiA@mail.gmail.com>

Welcome to FreeSWITCH and VoIP! I hope you don't get scared away too easily.
:)

First question: is your xlite on the same computer as your FreeSWITCH
install? Usually you don't want your softphone and FS to be on the same
computer, however if you are simply doing some testing and trying to learn
then it should be okay.

Here is where the fun begins. A SIP 408 means that the request timed out, so
the xlite is sending messages to the FS server and is not getting anything
back. Be sure that you use the correct IP address. The best thing to do
would be to start fs_cli.exe (which connects you to the FS command line) and
then type this command:

eval ${domain}

It should return an IP address - use that IP address in the xlite account
configuration.

Let us know how it goes!

-MC

On Fri, Oct 7, 2011 at 2:05 PM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Ok guys, I am a newbie in VOIP stuff and in freeswitch of course.
>
> I got the freeswitch book and installed freeswitch 1.0.6 in my windows 7
> pro rig.
>
> I also installed XLite 4 but when I try to register the extension is
> telling me: "Acount failed to enable" SIP 408 error.
>
> I am sure I stoped my firewall and  even with that I am getting same
> error.
>
> I have not changed anything from the default configuration, the extension I
> am trying to register is extension 1000, with password 1234.
>
>
> ?Could you help me please?
>
> Thanks in advance.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Sat Oct  8 01:55:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 16:55:50 -0500
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
 work as expected
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02209F7DF3@NY1-EXMB-01.ip-soft.net>
References: <4E8DC98A.8040408@hw.ac.uk>
	<CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com>
	<4E8DD739.5090104@hw.ac.uk>
	<CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD02206ED4A1@NY1-EXMB-01.ip-soft.net>
	<4E8EDA84.5000806@hw.ac.uk>
	<6A6B4C284AD15042B429EB9D904544AD02209F7DF3@NY1-EXMB-01.ip-soft.net>
Message-ID: <CAKbxfG-w-F+Ar-zFUr0x8AURGXjL_CHDbYvaVac5TuctP+KbeQ@mail.gmail.com>

My only real point is that pushing subsequent calls into the dial-plan
mechanism of FreeSWITCH and failing over to another "slot" is not
going to work very well in the FreeSWITCH architecture coming from
first hand experience of developing it myself.  Rather than question
the motive I just restate that you can use one call to the socket app
and limit the connections on the other side in a forking process or
use mod_limit ....


On Fri, Oct 7, 2011 at 4:10 PM, Hector Geraldino
<Hector.Geraldino at ip-soft.net> wrote:
> Hi,
>
> Most of the issues you've can be addressed with a 'single instanced' application, but I understand why you designed the application to work this way. If at any moment you think about enabling this application to handle several calls, you can overtake these issues in various ways, for example:
>
> - Controlling the number of calls: you can register all answered calls (uuids) in a data structure (list, map), remove them when disconnected, and decide when to accept a new call by getting the collection size and comparing it to your parameter. If you ever get more Nuance licenses, increasing the number of managed calls would be as easy as change a parameter in a property file.
>
> - You can have as many MRCP profiles as you want defined in the configuration xml files on mrcp_profiles/ directory. Coming up with the algorithm to assign channels to MRCP engines (Nuance server/port) won't be that hard. I can think of one: having all mrcp profiles in a map with a flag (in use/free), when the call comes I lock the first available profile so it's attached to this call. When the call is disconnected the profile is freed.
>
> - log mixed, sure. But, is this really a problem? The same with the publishing of topics, you can replicate the same behavior of having multiple instances in a single one. I mean, if you make an abstraction and say: one instance of my java application equals one thread of my new java app you'll see how easy is to overcome this and other problems.
>
> Sorry for such a long message, but as many others I feel real passion about application design topics. My intention is never to criticize what you've done, as each mind is a different world.
>
> Anyway, good luck with that!
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xl127
> Sent: Friday, October 07, 2011 6:55 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
>
> Hi Hector,
>
> Your points make sense but I have some specific issues for my
> application so I thought of binding only one call per runnning instance.
>
> 1. My outbound socket application is one component/server of our Spoken
> Dialogue System (SDS).
> We have four ports Nuance ASR/TTS licenses, each port is specific to one
> machine( yeah, we have four seperate license files for four machines).
> It seems to be difficult to specify which channel uses which Nuance MRCP
> profile if simultenous calls are answered by only one instance.
>
> 2. It seems to me that the log info would be mixed for simultenous calls
> if running one instance.
>
> 3. My socket app will be communicating with other components/servers of
> the SDS via ICE middleware (http://www.zeroc.com/).
> It will register to the central ICE server when starting the system via
> a Topic name which is a static name given by the developer,
> it will keep publishing ASR results on that topic so the other
> components listening on the topic will get the results,
> and it also subscribes and listens to other topics from other components
> for TTS.
>
> In simultenous calls to one running instance scenario, the
> distinguishable thing between channels would be the channel UUID which I
> could use
> as a part of the topic name, but it is dynamic and other servers could
> not know it when registering to the central server.
>
> I could let the all channels publish results with a same static topic
> name and channel UUID, the interested parties could parse the results
> they want based on the UUID, but I worried this would cause more burdens
> to the whole system.
>
>
> 4. Even if running one instance, I still need some ways to check and
> control the maximum number of calls as we have limited ASR/TTS resources.
>
> Cheers,
> Xing
>
> On 06/10/11 22:34, Hector Geraldino wrote:
>> This might sound a little bit Off-Topic but, why are you willing to bind only one call per running instance? I cannot imagine any user scenario where this is desirable. Maybe as a proof of concept is Ok, but having an equivalence between calls and instances mmm... I don't know.
>>
>> Can you please tell me what are you willing to accomplish? I'm really curious about it.
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
>> Sent: Thursday, October 06, 2011 3:20 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
>>
>> don't actually hang it up just don't connect the socket or connnect it
>> and disconnect it right away and the app will fail on in the
>> dial-plan.
>>
>> again you could try mod_limit otherwise.
>>
>>
>> On Thu, Oct 6, 2011 at 11:28 AM, xl127<x.liu at hw.ac.uk> ?wrote:
>>> Thanks for the suggestions!
>>>
>>> I thought about and tried to make my outbound socket application
>>> fail e.g. by send a hangup message but it didn't work.
>>>
>>> What I did is that I set a public variable channelUUID with initial
>>> value null in MyOutboundHandler,
>>> I check if it is null in the handleConnectionResponse method. If it not
>>> null, a hangup command is sent to FS.
>>> But it looks like each call channel creates its own Java thread, so the
>>> channelUUID is always null before the connection.
>>> So my approach doesn't work.
>>>
>>> Any idea about checking if the remote application is answering a call?
>>>
>>> Thanks!
>>> Xing
>>>
>>>
>>> On 06/10/11 16:43, Anthony Minessale wrote:
>>>> Really the socket app has not limit to how many simultaneous
>>>> connections it can handle.
>>>> You could try mod_limit, or make the remote end of you application
>>>> fail when there are too many connections.
>>>>
>>>>
>>>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk> ? ?wrote:
>>>>> Hello,
>>>>>
>>>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>>>> socket applications.
>>>>>
>>>>> Basically I want to check if the first socket app is engaged then the
>>>>> call should be transferred
>>>>> to the second socket app and so on.
>>>>>
>>>>> For initial test, I tried two concurrent calls, the problem is both
>>>>> calls are always answered
>>>>> by the first outbound socket.
>>>>>
>>>>> I have following setting in the dialplan:
>>>>>
>>>>> <extension name="sipgateDialedIn">
>>>>> <condition field="destination_number" expression="^5800$">
>>>>> <action application="set" data="continue_on_fail=true"/>
>>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>>> <action application="bridge" data="loopback/5801"/>
>>>>> <action application="bridge" data="loopback/5802"/>
>>>>>
>>>>> <action application="set" data="tts_engine=flite"/>
>>>>> <action application="set" data="tts_voice=kal"/>
>>>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>>>> try again later."/>
>>>>> <action application="sleep" data="10"/>
>>>>> <action application="hangup"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> <extension name="myOutboundESL5801">
>>>>> <condition field="destination_number" expression="^5801$">
>>>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> <extension name="myOutboundESL5802">
>>>>> <condition field="destination_number" expression="^5802$">
>>>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>>>> second concurrent call to 5800 will be
>>>>> answered by extension 5802, and the third call gets a busy message.
>>>>>
>>>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>>>> the socket is engaged?
>>>>>
>>>>> I've tried and googled a while but without success.
>>>>>
>>>>> Any advices please?
>>>>>
>>>>> Many thanks!
>>>>> Xing
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Heriot-Watt University is a Scottish charity
>>>>> registered under charity number SC000278.
>>>>>
>>>>> Heriot-Watt University is the Sunday Times
>>>>> Scottish University of the Year 2011-2012
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>>
>>> --
>>> Heriot-Watt University is a Scottish charity
>>> registered under charity number SC000278.
>>>
>>> Heriot-Watt University is the Sunday Times
>>> Scottish University of the Year 2011-2012
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
> Heriot-Watt University is the Sunday Times
> Scottish University of the Year 2011-2012
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Sat Oct  8 02:05:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 17:05:01 -0500
Subject: [Freeswitch-users] Desktop phone - answer a ringing call via
	API call
In-Reply-To: <CAGRr05_y9VKqDpw1y-gzP7SV6ffSL8wH9h_E9Vg7M3_aRfdOQw@mail.gmail.com>
References: <CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
	<201110070850.30154.justlikeef@gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E84@AZWSMS03.azwarranty.int>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E85@AZWSMS03.azwarranty.int>
	<CAGRr05_y9VKqDpw1y-gzP7SV6ffSL8wH9h_E9Vg7M3_aRfdOQw@mail.gmail.com>
Message-ID: <CAKbxfG_jf-vjfMUg_etYivmjc_B2rkHWHSWTRYVJ4wDzxpAv4w@mail.gmail.com>

if you want to let the phone ring normally then answer it once its
ringing you can use this FSAPI

uuid_phone_event <uuid of ringing call> talk


On Fri, Oct 7, 2011 at 10:35 AM, Brad Mina <brad at tech21.com> wrote:
> Could you possibly document on the wiki exactly what you did and link to it
> from this thread?
>
> On Fri, Oct 7, 2011 at 6:45 AM, Weigel, Stefan
> <Stefan.Weigel at allianz-warranty.com> wrote:
>>
>> Hi Rob, list,
>>
>>
>>
>> <action application="export" data="sip_auto_answer=true"/>
>>
>>
>>
>> did the trick!
>>
>>
>>
>> Somebody could append ?Polycom 560? to the models working with this on the
>> WIKI-page http://wiki.freeswitch.org/wiki/Intercom
>>
>>
>>
>> I will play around, maybe I can do it now?
>>
>>
>>
>>
>>
>>
>>
>> Thanks for the hint!
>>
>>
>>
>>
>>
>> Best regards,
>>
>>
>>
>> Stefan
>>
>>
>>
>> Von: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
>> Weigel, Stefan
>> Gesendet: Freitag, 7. Oktober 2011 15:31
>>
>> An: 'FreeSWITCH Users Help'
>> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via
>> API call
>>
>>
>>
>> Hi Rob, list,
>>
>>
>>
>> very funny, it?s exactly what I?m trying to do at the moment.
>>
>>
>>
>> In my dialplan I put:
>>
>>
>>
>> <action application="export" data="alert_info=aa"/>
>>
>>
>>
>> In sip.cfg I put:
>>
>>
>>
>> ?<alertInfo voIpProt.SIP.alertInfo.1.value="aa"
>> voIpProt.SIP.alertInfo.1.class="3" />
>>
>> [..]
>>
>> <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" />
>>
>>
>>
>> When tracing I can see:
>>
>>
>>
>> INVITE sip:567 at 192.168.1.1:5060 SIP/2.0
>>
>> [..]
>>
>> Alert-Info: <aa>
>>
>> [..]
>>
>>
>>
>> The ?<>? tags are appended automatically. So I tried to modify sip.cfg a
>> little bit:
>>
>>
>>
>> <alertInfo voIpProt.SIP.alertInfo.1.value="&lt;aa&gt;"
>> voIpProt.SIP.alertInfo.1.class="3" />
>>
>>
>>
>> But it?s not working. In the documentation I found a remark:
>>
>>
>>
>> If set to 1, auto-answer is enabled for all SIP
>>
>> calls.
>>
>> Note: This parameter is supported on the
>>
>> Polycom VVX 1500 only.
>>
>>
>>
>> So I don?t know if this feature will work for SIP calls with my 560.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Thanks and best regards,
>>
>>
>>
>> Stefan
>>
>>
>>
>>
>>
>> Von: Rob Hutton [mailto:justlikeef at gmail.com]
>> Gesendet: Freitag, 7. Oktober 2011 14:50
>> An: freeswitch-users at lists.freeswitch.org
>> Cc: Weigel, Stefan
>> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via
>> API call
>>
>>
>>
>> It looks like Polycoms can be told to "auto answer" with the SIP header
>> "Alert-Info: Ring Answer";
>>
>>
>>
>> If the "CTI Interface" had the UUID of the call and the extension, then a
>> quick API call or LUA script could add the sip header, then reinvite the
>> phone causing it to answer.
>>
>>
>>
>> Check out the Paging and Intercom page on the WIKI for more info on the
>> auto answer setup.
>>
>>
>>
>> On Friday 07 October 2011 04:42:32 Weigel, Stefan wrote:
>>
>> > Hi Michael, list,
>>
>> >
>>
>> >
>>
>> >
>>
>> > that's the point.
>>
>> >
>>
>> > I would really appreciate a solution for that "problem".
>>
>> >
>>
>> > We're using FreeSwitch in an callcenter environment and people used to
>> > handle the calls through some kind of CTI interface before we switched to
>> > FreeSWITCH. So it would be great to have this feature. But there's no
>> > possibility to control Polycom phones that way.
>>
>> >
>>
>> >
>>
>> >
>>
>> > Hints on this topic are welcome!
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> > Thanks and best regards
>>
>> >
>>
>> >
>>
>> >
>>
>> > Stefan
>>
>> >
>>
>> >
>>
>> >
>>
>> > Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> > Von: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
>> > Michael Collins
>>
>> > Gesendet: Mittwoch, 5. Oktober 2011 18:42
>>
>> > An: FreeSWITCH Users Help
>>
>> > Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call
>> > via API call
>>
>> >
>>
>> >
>>
>> >
>>
>> > This would be a function of the phone. AFAICT, Polycom does not have a
>> > Webby way of doing this. However, I think Brian West has a bead on using the
>> > expansion port along with an Arduino board to do some manual/remote control
>> > of the phone. If anyone has more information on remote controlling Polycoms
>> > (or any other phones, for that matter) please let us know.
>>
>> >
>>
>> >
>>
>> >
>>
>> > -MC
>>
>> >
>>
>> > On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan
>> > <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>>
>> > wrote:
>>
>> >
>>
>> > Hi all,
>>
>> >
>>
>> >
>>
>> >
>>
>> > Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP
>> > 560) via api command ?
>>
>> >
>>
>> > I tried with SendMsg <uuid> and 'answer' application, but it's not want
>> > I need/want.
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> > Thanks and best regards,
>>
>> >
>>
>> >
>>
>> >
>>
>> > Stefan
>>
>> >
>>
>> >
>>
>> >
>>
>> > FreeSWITCH-users mailing list
>>
>> >
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> > http://www.freeswitch.org
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From roger.castaldo at gmail.com  Sat Oct  8 06:59:30 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Fri, 7 Oct 2011 22:59:30 -0400
Subject: [Freeswitch-users] Newbie question...
In-Reply-To: <CAKzWOxXgOC4+0tZmZRdPJobQXXOef7voubUcuUnU-GwbKT0uiA@mail.gmail.com>
References: <CANtw_ZNnE9v7+EHhnjE+8V9_UFoxuwWNJ=bj6Jiiu4AuMWpVdg@mail.gmail.com>
	<CAKzWOxXgOC4+0tZmZRdPJobQXXOef7voubUcuUnU-GwbKT0uiA@mail.gmail.com>
Message-ID: <CAAYf2LuaRb80pWaddjns-WM0CUBvOwnXgAc-mt3JVrF5ERseQg@mail.gmail.com>

On a side note as well one thing I noticed when testing freeswitch on
windows 7 is I have always run it using the run as administrator to
ensure it can bind to things properly, otherwise it sometimes seems to
act all funky weird.

On Fri, Oct 7, 2011 at 5:19 PM, Michael Collins <msc at freeswitch.org> wrote:
> Welcome to FreeSWITCH and VoIP! I hope you don't get scared away too easily.
> :)
> First question: is your xlite on the same computer as your FreeSWITCH
> install? Usually you don't want your softphone and FS to be on the same
> computer, however if you are simply doing some testing and trying to learn
> then it should be okay.
> Here is where the fun begins. A SIP 408 means that the request timed out, so
> the xlite is sending messages to the FS server and is not getting anything
> back. Be sure that you use the correct IP address. The best thing to do
> would be to start fs_cli.exe (which connects you to the FS command line) and
> then type this command:
> eval ${domain}
> It should return an IP address - use that IP address in the xlite account
> configuration.
> Let us know how it goes!
> -MC
>
> On Fri, Oct 7, 2011 at 2:05 PM, Francisco Javier Cintr?n Olgu?n
> <fcintrono at gmail.com> wrote:
>>
>> Ok guys, I am a newbie in VOIP stuff and in freeswitch of course.
>> I got the freeswitch book and installed freeswitch 1.0.6 in my windows 7
>> pro rig.
>> I also installed XLite 4 but when I try to register the extension is
>> telling me: "Acount failed to enable" SIP 408 error.
>> I am sure I stoped my firewall and ?even with that I am getting same
>> error.
>> I have not changed anything from the default configuration, the extension
>> I am trying to register is extension 1000, with password 1234.
>>
>> ?Could you help me please?
>> Thanks in advance.
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Stefan.Weigel at allianz-warranty.com  Sat Oct  8 16:13:38 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Sat, 8 Oct 2011 14:13:38 +0200
Subject: [Freeswitch-users] Desktop phone - answer a ringing call
	via	API call
In-Reply-To: <CAGRr05_y9VKqDpw1y-gzP7SV6ffSL8wH9h_E9Vg7M3_aRfdOQw@mail.gmail.com>
References: <CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
	<201110070850.30154.justlikeef@gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E84@AZWSMS03.azwarranty.int>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E85@AZWSMS03.azwarranty.int>,
	<CAGRr05_y9VKqDpw1y-gzP7SV6ffSL8wH9h_E9Vg7M3_aRfdOQw@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3C88CF1@AZWSMS03.azwarranty.int>

Hi Brad, list,

as Anthony wrote a little bit later this day, with phones that understand TALK/HOLD notify messages, just use
api cal 'uuid_phone_event <uuid> talk|hold'.

What I did before:

with 'show channels I'm getting the uuid of the incoming leg bridged to the extension (which is already ringing).
Further on I set a variable with the dialed extension to this uuid:

freeswitch at internal<mailto:freeswitch at internal>> uuid_setvar 8ca38212-c500-482b-83ec-2d87ad515180 dialed_number 812
+OK

I wrote a new dialplan extension:

<include>
  <extension name="AUTOANSWER">
      <condition field="${dialed_number}" expression="^(\d\d\d)$" break="on-true">
          <action application="export" data="sip_auto_answer=true"/>
          <action application="transfer" data="$1 XML user_lines"/>
      </condition>
  </extension>
</include>

Then I transfer the call to this extension, it's exporting the variable and the phone answers.

freeswitch at internal<mailto:freeswitch at internal>> uuid_transfer 8ca38212-c500-482b-83ec-2d87ad515180 -both AUTOANSWER XML public
+OK
Problematic is call statistics (mod_cdr_csv in my case) because it's generating two entries for this single call.

Can I do this maybe with api call 'uuid_broadcast' ?


I don't know if it's worth to put on the WIKI, but if there's interest, I will do it.


Best regards,

Stefan

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] im Auftrag von Brad Mina [brad at tech21.com]
Gesendet: Freitag, 7. Oktober 2011 17:35
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call

Could you possibly document on the wiki exactly what you did and link to it from this thread?

On Fri, Oct 7, 2011 at 6:45 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:
Hi Rob, list,

<action application="export" data="sip_auto_answer=true"/>

did the trick!

Somebody could append ?Polycom 560? to the models working with this on the WIKI-page http://wiki.freeswitch.org/wiki/Intercom

I will play around, maybe I can do it now?



Thanks for the hint!


Best regards,

Stefan

Von: 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>] Im Auftrag von Weigel, Stefan
Gesendet: Freitag, 7. Oktober 2011 15:31

An: 'FreeSWITCH Users Help'
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call

Hi Rob, list,

very funny, it?s exactly what I?m trying to do at the moment.

In my dialplan I put:


<action application="export" data="alert_info=aa<UrlBlockedError.aspx>"/>

In sip.cfg I put:

 <alertInfo voIpProt.SIP.alertInfo.1.value="aa" voIpProt.SIP.alertInfo.1.class="3" />
[..]
<AUTO_ANSWER se.rt.3.name<http://se.rt.3.name>="Auto Answer" se.rt.3.type="answer" />

When tracing I can see:

INVITE sip:567 at 192.168.1.1:5060<http://sip:567 at 192.168.1.1:5060> SIP/2.0
[..]
Alert-Info: <aa>
[..]

The ?<>? tags are appended automatically. So I tried to modify sip.cfg a little bit:

<alertInfo voIpProt.SIP.alertInfo.1.value="&lt;aa&gt;" voIpProt.SIP.alertInfo.1.class="3" />

But it?s not working. In the documentation I found a remark:

If set to 1, auto-answer is enabled for all SIP
calls.
Note: This parameter is supported on the
Polycom VVX 1500 only.

So I don?t know if this feature will work for SIP calls with my 560.




Thanks and best regards,

Stefan


Von: Rob Hutton [mailto:justlikeef at gmail.com<mailto:justlikeef at gmail.com>]
Gesendet: Freitag, 7. Oktober 2011 14:50
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Cc: Weigel, Stefan
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call


It looks like Polycoms can be told to "auto answer" with the SIP header "Alert-Info: Ring Answer";



If the "CTI Interface" had the UUID of the call and the extension, then a quick API call or LUA script could add the sip header, then reinvite the phone causing it to answer.



Check out the Paging and Intercom page on the WIKI for more info on the auto answer setup.



On Friday 07 October 2011 04:42:32 Weigel, Stefan wrote:

> Hi Michael, list,

>

>

>

> that's the point.

>

> I would really appreciate a solution for that "problem".

>

> We're using FreeSwitch in an callcenter environment and people used to handle the calls through some kind of CTI interface before we switched to FreeSWITCH. So it would be great to have this feature. But there's no possibility to control Polycom phones that way.

>

>

>

> Hints on this topic are welcome!

>

>

>

>

>

> Thanks and best regards

>

>

>

> Stefan

>

>

>

> Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.

>

>

>

>

>

> Von: 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>] Im Auftrag von Michael Collins

> Gesendet: Mittwoch, 5. Oktober 2011 18:42

> An: FreeSWITCH Users Help

> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via API call

>

>

>

> This would be a function of the phone. AFAICT, Polycom does not have a Webby way of doing this. However, I think Brian West has a bead on using the expansion port along with an Arduino board to do some manual/remote control of the phone. If anyone has more information on remote controlling Polycoms (or any other phones, for that matter) please let us know.

>

>

>

> -MC

>

> On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com><mailto:Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>>> wrote:

>

> Hi all,

>

>

>

> Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP 560) via api command ?

>

> I tried with SendMsg <uuid> and 'answer' application, but it's not want I need/want.

>

>

>

>

>

> Thanks and best regards,

>

>

>

> Stefan

>

>

>

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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

> http://www.freeswitch.org

>

>

>

>




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


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From Stefan.Weigel at allianz-warranty.com  Sat Oct  8 16:13:45 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Sat, 8 Oct 2011 14:13:45 +0200
Subject: [Freeswitch-users] Desktop phone - answer a ringing call
	via	API call
In-Reply-To: <CAKbxfG_jf-vjfMUg_etYivmjc_B2rkHWHSWTRYVJ4wDzxpAv4w@mail.gmail.com>
References: <CAKzWOxUvBXFyguKB=rAtPNsurZQQGLWM2wOzUTQe6utiQ8JkaQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E82@AZWSMS03.azwarranty.int>
	<201110070850.30154.justlikeef@gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E84@AZWSMS03.azwarranty.int>
	<5003D7D3E06F514E8C682F18D223265C04D3B36E85@AZWSMS03.azwarranty.int>
	<CAGRr05_y9VKqDpw1y-gzP7SV6ffSL8wH9h_E9Vg7M3_aRfdOQw@mail.gmail.com>,
	<CAKbxfG_jf-vjfMUg_etYivmjc_B2rkHWHSWTRYVJ4wDzxpAv4w@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3C88CF2@AZWSMS03.azwarranty.int>

Hi Anthony,

wonderful! It's exactly what I need :)
As long as the hardware knows to handle the NOTIFY messages it's working like a charm.


Thanks a lot!


Best regards,

Stefan

________________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] im Auftrag von Anthony Minessale [anthony.minessale at gmail.com]
Gesendet: Samstag, 8. Oktober 2011 00:05
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via       API call

if you want to let the phone ring normally then answer it once its
ringing you can use this FSAPI

uuid_phone_event <uuid of ringing call> talk


On Fri, Oct 7, 2011 at 10:35 AM, Brad Mina <brad at tech21.com> wrote:
> Could you possibly document on the wiki exactly what you did and link to it
> from this thread?
>
> On Fri, Oct 7, 2011 at 6:45 AM, Weigel, Stefan
> <Stefan.Weigel at allianz-warranty.com> wrote:
>>
>> Hi Rob, list,
>>
>>
>>
>> <action application="export" data="sip_auto_answer=true"/>
>>
>>
>>
>> did the trick!
>>
>>
>>
>> Somebody could append ?Polycom 560? to the models working with this on the
>> WIKI-page http://wiki.freeswitch.org/wiki/Intercom
>>
>>
>>
>> I will play around, maybe I can do it now?
>>
>>
>>
>>
>>
>>
>>
>> Thanks for the hint!
>>
>>
>>
>>
>>
>> Best regards,
>>
>>
>>
>> Stefan
>>
>>
>>
>> Von: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
>> Weigel, Stefan
>> Gesendet: Freitag, 7. Oktober 2011 15:31
>>
>> An: 'FreeSWITCH Users Help'
>> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via
>> API call
>>
>>
>>
>> Hi Rob, list,
>>
>>
>>
>> very funny, it?s exactly what I?m trying to do at the moment.
>>
>>
>>
>> In my dialplan I put:
>>
>>
>>
>> <action application="export" data="alert_info=aa"/>
>>
>>
>>
>> In sip.cfg I put:
>>
>>
>>
>>  <alertInfo voIpProt.SIP.alertInfo.1.value="aa"
>> voIpProt.SIP.alertInfo.1.class="3" />
>>
>> [..]
>>
>> <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer" />
>>
>>
>>
>> When tracing I can see:
>>
>>
>>
>> INVITE sip:567 at 192.168.1.1:5060 SIP/2.0
>>
>> [..]
>>
>> Alert-Info: <aa>
>>
>> [..]
>>
>>
>>
>> The ?<>? tags are appended automatically. So I tried to modify sip.cfg a
>> little bit:
>>
>>
>>
>> <alertInfo voIpProt.SIP.alertInfo.1.value="&lt;aa&gt;"
>> voIpProt.SIP.alertInfo.1.class="3" />
>>
>>
>>
>> But it?s not working. In the documentation I found a remark:
>>
>>
>>
>> If set to 1, auto-answer is enabled for all SIP
>>
>> calls.
>>
>> Note: This parameter is supported on the
>>
>> Polycom VVX 1500 only.
>>
>>
>>
>> So I don?t know if this feature will work for SIP calls with my 560.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Thanks and best regards,
>>
>>
>>
>> Stefan
>>
>>
>>
>>
>>
>> Von: Rob Hutton [mailto:justlikeef at gmail.com]
>> Gesendet: Freitag, 7. Oktober 2011 14:50
>> An: freeswitch-users at lists.freeswitch.org
>> Cc: Weigel, Stefan
>> Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call via
>> API call
>>
>>
>>
>> It looks like Polycoms can be told to "auto answer" with the SIP header
>> "Alert-Info: Ring Answer";
>>
>>
>>
>> If the "CTI Interface" had the UUID of the call and the extension, then a
>> quick API call or LUA script could add the sip header, then reinvite the
>> phone causing it to answer.
>>
>>
>>
>> Check out the Paging and Intercom page on the WIKI for more info on the
>> auto answer setup.
>>
>>
>>
>> On Friday 07 October 2011 04:42:32 Weigel, Stefan wrote:
>>
>> > Hi Michael, list,
>>
>> >
>>
>> >
>>
>> >
>>
>> > that's the point.
>>
>> >
>>
>> > I would really appreciate a solution for that "problem".
>>
>> >
>>
>> > We're using FreeSwitch in an callcenter environment and people used to
>> > handle the calls through some kind of CTI interface before we switched to
>> > FreeSWITCH. So it would be great to have this feature. But there's no
>> > possibility to control Polycom phones that way.
>>
>> >
>>
>> >
>>
>> >
>>
>> > Hints on this topic are welcome!
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> > Thanks and best regards
>>
>> >
>>
>> >
>>
>> >
>>
>> > Stefan
>>
>> >
>>
>> >
>>
>> >
>>
>> > Btw.: FreeSWITCH rocks! It's so flexible, unbelievable.
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> > Von: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
>> > Michael Collins
>>
>> > Gesendet: Mittwoch, 5. Oktober 2011 18:42
>>
>> > An: FreeSWITCH Users Help
>>
>> > Betreff: Re: [Freeswitch-users] Desktop phone - answer a ringing call
>> > via API call
>>
>> >
>>
>> >
>>
>> >
>>
>> > This would be a function of the phone. AFAICT, Polycom does not have a
>> > Webby way of doing this. However, I think Brian West has a bead on using the
>> > expansion port along with an Arduino board to do some manual/remote control
>> > of the phone. If anyone has more information on remote controlling Polycoms
>> > (or any other phones, for that matter) please let us know.
>>
>> >
>>
>> >
>>
>> >
>>
>> > -MC
>>
>> >
>>
>> > On Tue, Oct 4, 2011 at 6:58 AM, Weigel, Stefan
>> > <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>>
>> > wrote:
>>
>> >
>>
>> > Hi all,
>>
>> >
>>
>> >
>>
>> >
>>
>> > Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP
>> > 560) via api command ?
>>
>> >
>>
>> > I tried with SendMsg <uuid> and 'answer' application, but it's not want
>> > I need/want.
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>> > Thanks and best regards,
>>
>> >
>>
>> >
>>
>> >
>>
>> > Stefan
>>
>> >
>>
>> >
>>
>> >
>>
>> > FreeSWITCH-users mailing list
>>
>> >
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> > http://www.freeswitch.org
>>
>> >
>>
>> >
>>
>> >
>>
>> >
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

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 fcintrono at gmail.com  Sat Oct  8 19:32:30 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Sat, 8 Oct 2011 10:32:30 -0500
Subject: [Freeswitch-users] Newbie question...
In-Reply-To: <CAKzWOxXgOC4+0tZmZRdPJobQXXOef7voubUcuUnU-GwbKT0uiA@mail.gmail.com>
References: <CANtw_ZNnE9v7+EHhnjE+8V9_UFoxuwWNJ=bj6Jiiu4AuMWpVdg@mail.gmail.com>
	<CAKzWOxXgOC4+0tZmZRdPJobQXXOef7voubUcuUnU-GwbKT0uiA@mail.gmail.com>
Message-ID: <CANtw_ZO+q74+bQwipQh9VYWhXXBZhN085QQUvD8GcLVhgBfhmA@mail.gmail.com>

Thank you for your help.

Well, Yes I installed Xlite in the same windows 7 rig in which I installed
freeswitch. And yes I am trying to learn what is VOIP, I  read that
freeswitch could run in windows so I bought freeswitch book and here I am
trying to understand it.

this is the answer of the eval command:

10.10.100.31

I put this IP in my xlite account, in fact I put 127.0.0.1 too, but it
always failing with SIP 408 code.

I changed freeswitch to run with administrator account, but I am getting
same SIP 408 error.

?What am I doing wrong?

Thanks in advance.
Have a nice day.

On Fri, Oct 7, 2011 at 4:19 PM, Michael Collins <msc at freeswitch.org> wrote:

> Welcome to FreeSWITCH and VoIP! I hope you don't get scared away too
> easily. :)
>
> First question: is your xlite on the same computer as your FreeSWITCH
> install? Usually you don't want your softphone and FS to be on the same
> computer, however if you are simply doing some testing and trying to learn
> then it should be okay.
>
> Here is where the fun begins. A SIP 408 means that the request timed out,
> so the xlite is sending messages to the FS server and is not getting
> anything back. Be sure that you use the correct IP address. The best thing
> to do would be to start fs_cli.exe (which connects you to the FS command
> line) and then type this command:
>
> eval ${domain}
>
> It should return an IP address - use that IP address in the xlite account
> configuration.
>
> Let us know how it goes!
>
> -MC
>
> On Fri, Oct 7, 2011 at 2:05 PM, Francisco Javier Cintr?n Olgu?n <
> fcintrono at gmail.com> wrote:
>
>> Ok guys, I am a newbie in VOIP stuff and in freeswitch of course.
>>
>> I got the freeswitch book and installed freeswitch 1.0.6 in my windows 7
>> pro rig.
>>
>> I also installed XLite 4 but when I try to register the extension is
>> telling me: "Acount failed to enable" SIP 408 error.
>>
>> I am sure I stoped my firewall and  even with that I am getting same
>> error.
>>
>> I have not changed anything from the default configuration, the extension
>> I am trying to register is extension 1000, with password 1234.
>>
>>
>> ?Could you help me please?
>>
>> Thanks in advance.
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hynek.cihlar at gmail.com  Sat Oct  8 12:14:18 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Sat, 8 Oct 2011 10:14:18 +0200
Subject: [Freeswitch-users] Half the call recording
Message-ID: <CAJh8XdUj=vMPZQrrBBZ6QcswStU2TUOjjV7id7LAirswmWezAQ@mail.gmail.com>

I cannot find information on recording half the call, only a page which
states, that split recording of the caller/callee is possible with
freeswitch. Reading the documentation about recording API commands I
couldn't find any info, whether freeswittch can record only one leg of a
two-legged called or even how to do it.

Any help is appreciated.

Hynek
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From admin at blindi.net  Sat Oct  8 20:43:22 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 8 Oct 2011 18:43:22 +0200 (CEST)
Subject: [Freeswitch-users] Why does group_confirm_file 3 times from?
Message-ID: <Pine.LNX.4.64.1110081830350.17472@tomback.antagus.de>

Hi guys,
    I'm sorry, because my ISP was temporarily unavailable, have been lost
    some mails to me.
I have installed the git version from fs.
But the Problem is the same.

I connect via sip.
My Extension is:

<extension name="callscreen">
<condition field="destination_number" expression="^callscreen">
  <action application="set" data="group_confirm_file=playback
  $${base_dir}/sounds/anrufe/achtung-dringend-call.wav"/>
   <action application="set" data="group_confirm_key=exec"/>
<action application="bridge"  data="loopback/0/operator1"/>
</condition>
</extension>

Fs plays before connecting this file 3 times from.
    No matter what digit I press.
Fs repeate the soundfile 3 times.
  I don.t find a parameter to  turn off this.
    Can someone help me please?
thanks.

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From beffa at ieee.org  Sat Oct  8 20:57:06 2011
From: beffa at ieee.org (Federico Beffa)
Date: Sat, 8 Oct 2011 18:57:06 +0200
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <CAKbxfG8wM64mXMAN2WtjW=YcvjE9RN-WrzcVNUv8tNjrkLXYcA@mail.gmail.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
	<1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
	<CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>
	<1317308654166-6844212.post@n2.nabble.com>
	<CAKrPhPMA3cngUy9hkNyucbzOv=GwAjN6qWUQNFJ=K4+koEcAEA@mail.gmail.com>
	<CAKrPhPM4MV5XQ8oBop-PvwAZbMQM9S422wH-UMDK3n1whvVf+w@mail.gmail.com>
	<CAKbxfG8wM64mXMAN2WtjW=YcvjE9RN-WrzcVNUv8tNjrkLXYcA@mail.gmail.com>
Message-ID: <CAKrPhPNO2Xn_-LcbWc+f=JsC6+76kNYcm54zpe-ZHEBskEHueA@mail.gmail.com>

Hi Anthony,

thanks for the clarification and the suggestion. As an immediate
workaround I would like to try the "cron" approach you suggested. I
have therefore experimented a little bit with the chat api command.

My jingle_profiles/client.xml looks essentially as the one provided
with the example configuration:

  <profile type="client">
    <param name="name" value="$${xmpp_client_profile}"/>
    <param name="login" value="user1 at gmail.com/talk"/>
    <param name="password" value="XXXX"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="auto-login" value="true"/>
    <!-- SASL "plain" or "md5" -->
    <param name="sasl" value="plain"/>
    <!-- if the server where the jabber is hosted is not the same as
the one in the jid -->
    <param name="server" value="talk.google.com"/>
    <!-- Enable TLS or not -->
    <param name="tls" value="true"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
    <!-- default extension (if one cannot be determined) -->
    <param name="exten" value="888"/>
    <!-- VAD choose one -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <!--<param name="vad" value="both"/>-->
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <!--<param name="candidate-acl" value="wan.auto"/>-->
    <param name="local-network-acl" value="localnet.auto"/>
  </profile>

When I try to use the chat command at the fs_cli prompt I'm unable to
form a full jid and always get an error message from the remote
server. Here an example:

chat jingle|888 at xmppc|user2 at gmail.com|test

Sent
freeswitch at ubuntu-vbox-t5500> 2011-10-08 14:22:09.804596 [NOTICE]
libdingaling.c:1373 SecSEND:
-------------------------------------------------------------------------------
<message to="user2 at gmail.com" type="chat" from="user1 at gmail.com">
  <body>test</body>
</message>

2011-10-08 14:22:09.844698 [INFO] libdingaling.c:1371 SecRECV:
-------------------------------------------------------------------------------
<message to="user1 at gmail.com" type="error" from="user2 at gmail.com">
  <body>test</body>
  <error code="400" type="modify">
    <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request>
    <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">If set, the
'from' attribute must be set to the user's full JID.</text>
  </error>
</message>

"xmppc" is the value of the variable "xmpp_client_profile" as in the
default config. I get the same error independently from what I set
before "@xmppc" in the from_jid. Differently from this, if as the
from_jid I set my explicit jid "user1 at gmail.com/talk" as in the
client.xml file I get the error:

2011-10-08 14:20:50.446696 [ERR] mod_dingaling.c:544 Invalid Profile gmail.com

Despite the fact that even a wrong message reaching the remote server
could work as a "keep alive" signal, I still would like to understand
the proper syntax.
I would be grateful if somebody could explain me what is the correct
syntax to use with chat and jingle?

Thank you!
Fede

P.S.: By the way, I'm now working on HEAD.

On Fri, Oct 7, 2011 at 7:59 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> There is no keep alive functionality in the core XMPP library we use
> in the core of libdingaling (IKSEMEL)
>
> One option would to have a NAT that doesn't disconnect active
> connections just because they are idle (which seems bad to me).
>
> Otherwise you could work on a patch, request a feature or come up with
> some cron job maybe to use the chat api command to send a message to
> your server from cron or something as a workaround.
>
>
>
> On Thu, Oct 6, 2011 at 2:18 AM, Federico Beffa <beffa at ieee.org> wrote:
>> Unfortunately I was singing victory too early. My problem still persists...
>>
>> The git version I'm running is from the 20th of August. In the mean
>> time have there been changes which could affect mod_dingaling? Looking
>> in the git commit comments I have the impression that the answer is
>> no.
>>
>> Regards,
>> Fede
>>
>> On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa at ieee.org> wrote:
>>> Just in case someone will hit the same problem...
>>>
>>> After some tests, I've found that, changing "bind_server_ip" from
>>> "auto" to the local IP address of the FS server (192.168.X.X), appears
>>> to solve my timeout problem. ?(The account did not go offline for more
>>> than 12 hours.)
>>>
>>> To summarize my current configuration: I have a fix public IP assigned
>>> to my router (NAT/Firewall). NAT is configured to forward ports 5080
>>> and 5081 to the host on which FS is running and UPnP is disabled.
>>>
>>> I'm running a FS version checked out from git the 20th of August.
>>>
>>> Regards,
>>> Fede
>>>
>>> On Thu, Sep 29, 2011 at 5:04 PM, mazilo
>>> <Nabble_01394 at slickdeals.endjunk.com> wrote:
>>>> The only thing I can say is to update your FS with the latest git commit.
>>>>
>>>> -----
>>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
>>>> --
>>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-offline-mod-dingaling-tp6839355p6844212.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From X.Liu at hw.ac.uk  Sat Oct  8 22:19:42 2011
From: X.Liu at hw.ac.uk (Liu, Xingkun)
Date: Sat, 8 Oct 2011 19:19:42 +0100
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
	work as expected
References: <4E8DC98A.8040408@hw.ac.uk><CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com><4E8DD739.5090104@hw.ac.uk><CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com><6A6B4C284AD15042B429EB9D904544AD02206ED4A1@NY1-EXMB-01.ip-soft.net><4E8EDA84.5000806@hw.ac.uk>
	<6A6B4C284AD15042B429EB9D904544AD02209F7DF3@NY1-EXMB-01.ip-soft.net>
Message-ID: <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1C76@ex6.mail.win.hw.ac.uk>


Hi Hector,

Thank you very much for sharing thoughts and ideas, very helpful!

Cheers,
Xing


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Hector Geraldino
Sent: Fri 10/7/2011 22:10
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
 
Hi,

Most of the issues you've can be addressed with a 'single instanced' application, but I understand why you designed the application to work this way. If at any moment you think about enabling this application to handle several calls, you can overtake these issues in various ways, for example:

- Controlling the number of calls: you can register all answered calls (uuids) in a data structure (list, map), remove them when disconnected, and decide when to accept a new call by getting the collection size and comparing it to your parameter. If you ever get more Nuance licenses, increasing the number of managed calls would be as easy as change a parameter in a property file.

- You can have as many MRCP profiles as you want defined in the configuration xml files on mrcp_profiles/ directory. Coming up with the algorithm to assign channels to MRCP engines (Nuance server/port) won't be that hard. I can think of one: having all mrcp profiles in a map with a flag (in use/free), when the call comes I lock the first available profile so it's attached to this call. When the call is disconnected the profile is freed.

- log mixed, sure. But, is this really a problem? The same with the publishing of topics, you can replicate the same behavior of having multiple instances in a single one. I mean, if you make an abstraction and say: one instance of my java application equals one thread of my new java app you'll see how easy is to overcome this and other problems.

Sorry for such a long message, but as many others I feel real passion about application design topics. My intention is never to criticize what you've done, as each mind is a different world.  

Anyway, good luck with that!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xl127
Sent: Friday, October 07, 2011 6:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected

Hi Hector,

Your points make sense but I have some specific issues for my 
application so I thought of binding only one call per runnning instance.

1. My outbound socket application is one component/server of our Spoken 
Dialogue System (SDS).
We have four ports Nuance ASR/TTS licenses, each port is specific to one 
machine( yeah, we have four seperate license files for four machines).
It seems to be difficult to specify which channel uses which Nuance MRCP 
profile if simultenous calls are answered by only one instance.

2. It seems to me that the log info would be mixed for simultenous calls 
if running one instance.

3. My socket app will be communicating with other components/servers of 
the SDS via ICE middleware (http://www.zeroc.com/).
It will register to the central ICE server when starting the system via 
a Topic name which is a static name given by the developer,
it will keep publishing ASR results on that topic so the other 
components listening on the topic will get the results,
and it also subscribes and listens to other topics from other components 
for TTS.

In simultenous calls to one running instance scenario, the 
distinguishable thing between channels would be the channel UUID which I 
could use
as a part of the topic name, but it is dynamic and other servers could 
not know it when registering to the central server.

I could let the all channels publish results with a same static topic 
name and channel UUID, the interested parties could parse the results
they want based on the UUID, but I worried this would cause more burdens 
to the whole system.


4. Even if running one instance, I still need some ways to check and 
control the maximum number of calls as we have limited ASR/TTS resources.

Cheers,
Xing

On 06/10/11 22:34, Hector Geraldino wrote:
> This might sound a little bit Off-Topic but, why are you willing to bind only one call per running instance? I cannot imagine any user scenario where this is desirable. Maybe as a proof of concept is Ok, but having an equivalence between calls and instances mmm... I don't know.
>
> Can you please tell me what are you willing to accomplish? I'm really curious about it.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Thursday, October 06, 2011 3:20 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
>
> don't actually hang it up just don't connect the socket or connnect it
> and disconnect it right away and the app will fail on in the
> dial-plan.
>
> again you could try mod_limit otherwise.
>
>
> On Thu, Oct 6, 2011 at 11:28 AM, xl127<x.liu at hw.ac.uk>  wrote:
>> Thanks for the suggestions!
>>
>> I thought about and tried to make my outbound socket application
>> fail e.g. by send a hangup message but it didn't work.
>>
>> What I did is that I set a public variable channelUUID with initial
>> value null in MyOutboundHandler,
>> I check if it is null in the handleConnectionResponse method. If it not
>> null, a hangup command is sent to FS.
>> But it looks like each call channel creates its own Java thread, so the
>> channelUUID is always null before the connection.
>> So my approach doesn't work.
>>
>> Any idea about checking if the remote application is answering a call?
>>
>> Thanks!
>> Xing
>>
>>
>> On 06/10/11 16:43, Anthony Minessale wrote:
>>> Really the socket app has not limit to how many simultaneous
>>> connections it can handle.
>>> You could try mod_limit, or make the remote end of you application
>>> fail when there are too many connections.
>>>
>>>
>>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk>    wrote:
>>>> Hello,
>>>>
>>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>>> socket applications.
>>>>
>>>> Basically I want to check if the first socket app is engaged then the
>>>> call should be transferred
>>>> to the second socket app and so on.
>>>>
>>>> For initial test, I tried two concurrent calls, the problem is both
>>>> calls are always answered
>>>> by the first outbound socket.
>>>>
>>>> I have following setting in the dialplan:
>>>>
>>>> <extension name="sipgateDialedIn">
>>>> <condition field="destination_number" expression="^5800$">
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="bridge" data="loopback/5801"/>
>>>> <action application="bridge" data="loopback/5802"/>
>>>>
>>>> <action application="set" data="tts_engine=flite"/>
>>>> <action application="set" data="tts_voice=kal"/>
>>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>>> try again later."/>
>>>> <action application="sleep" data="10"/>
>>>> <action application="hangup"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5801">
>>>> <condition field="destination_number" expression="^5801$">
>>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> <extension name="myOutboundESL5802">
>>>> <condition field="destination_number" expression="^5802$">
>>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>>> second concurrent call to 5800 will be
>>>> answered by extension 5802, and the third call gets a busy message.
>>>>
>>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>>> the socket is engaged?
>>>>
>>>> I've tried and googled a while but without success.
>>>>
>>>> Any advices please?
>>>>
>>>> Many thanks!
>>>> Xing
>>>>
>>>>
>>>>
>>>> --
>>>> Heriot-Watt University is a Scottish charity
>>>> registered under charity number SC000278.
>>>>
>>>> Heriot-Watt University is the Sunday Times
>>>> Scottish University of the Year 2011-2012
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>> --
>> Heriot-Watt University is a Scottish charity
>> registered under charity number SC000278.
>>
>> Heriot-Watt University is the Sunday Times
>> Scottish University of the Year 2011-2012
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012




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-- 
Heriot-Watt University is a Scottish charity
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Heriot-Watt University is the Sunday Times
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From mi.ke at null.net  Sat Oct  8 22:33:12 2011
From: mi.ke at null.net (Mi Ke)
Date: Sat, 08 Oct 2011 14:33:12 -0400
Subject: [Freeswitch-users] mod_sofia@
Message-ID: <20111008183312.309130@gmx.com>

Hi !

 I originate call from lua script by the following command:

 local ob_session = freeswitch.Session("{originate_timeout=120,return_ring_ready=true,origination_caller_id_number=" .. calling_id .. ",called_id=" .. called_id .. ",accountcode=" .. accountcode .. ",credit_time=" .. out_credit_time .. ",execute_on_answer=lua b_answer.lua}" .. out_called_id,session)

 My iface IP is 2.2.2.2, origination_caller_id_number is my calling ID (55555) and out_called_id = sofia/internal/1212121212 at 1.1.1.1

 Leg B setup and bridging goes normally, when remote party disconnects, we receive:

tport_wakeup_pri(0x81c1a0): events IN
tport_recv_event(0x81c1a0)
tport_recv_iovec(0x81c1a0) msg 0x7f28386348a0 from (udp/2.2.2.2:5060 http://78.46.44.138:5060/ ) has 425 bytes, veclen = 1
tport_deliver(0x81c1a0): msg 0x7f28386348a0 (425 bytes) from udp/1.1.1.1:5060/sip http://193.104.107.7:5060/sip  next=(nil)
nta: received BYE sip:mod_sofia at 2.2.2.2: http://sip:mod_sofia at 78.46.44.138:5060/  http://sip:mod_sofia at 78.46.44.138:5060/ 5060 http://sip:mod_sofia at 78.46.44.138:5060/  SIP/2.0 (CSeq 48765)
nta: canonizing sip:mod_sofia at 2.2.2.2:5060 http://sip:mod_sofia at 78.46.44.138:5060/  with contact
nta: BYE (48765) going to existing leg
nua: nua_stack_process_request: entering
nua(0x94c4a0): event i_bye 100 Trying
nua: nua_application_event: entering
2011-10-08 14:54:48.866309 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/ 841266482056 at 193.104.107.7 .1.1.1 [BREAK]
nua: nua_handle_magic: entering
2011-10-08 14:54:48.886307 [DEBUG] switch_channel.c:2800 (sofia/internal/ 841266482056 at 193.104.107.7 .1.1.1) Callstate Change ACTIVE -> HANGUP

 (normal bye handshake follows...)

 Should remote party see me as sip:55555 at 2.2.2.2 and not as sip:mod_sofia at 2.2.2.2 ? What info should I include into session originate to acheive that?

 Thanks / Mike
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From X.Liu at hw.ac.uk  Sat Oct  8 22:38:47 2011
From: X.Liu at hw.ac.uk (Liu, Xingkun)
Date: Sat, 8 Oct 2011 19:38:47 +0100
Subject: [Freeswitch-users] Bridge calls to Java ESL socket does not
	work as expected
References: <4E8DC98A.8040408@hw.ac.uk><CAKbxfG85GM4Sj0qeFi9_9rkGeCX-+j7iNC5q+U1Pc2uM9k9z4g@mail.gmail.com><4E8DD739.5090104@hw.ac.uk><CAKbxfG-pkFrMgzQu=kzhbyYud1J-cDs2-zttmouNrWfQVOoDCQ@mail.gmail.com><6A6B4C284AD15042B429EB9D904544AD02206ED4A1@NY1-EXMB-01.ip-soft.net><4E8EDA84.5000806@hw.ac.uk><6A6B4C284AD15042B429EB9D904544AD02209F7DF3@NY1-EXMB-01.ip-soft.net>
	<CAKbxfG-w-F+Ar-zFUr0x8AURGXjL_CHDbYvaVac5TuctP+KbeQ@mail.gmail.com>
Message-ID: <976F9C1B58DCB84EA7021B8BFB483B8E0E1A1C77@ex6.mail.win.hw.ac.uk>

Hi Anthony,

I appreciate your points, thank you very much for your helps!

Cheers,
Xing



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Anthony Minessale
Sent: Fri 10/7/2011 22:55
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
 
My only real point is that pushing subsequent calls into the dial-plan
mechanism of FreeSWITCH and failing over to another "slot" is not
going to work very well in the FreeSWITCH architecture coming from
first hand experience of developing it myself.  Rather than question
the motive I just restate that you can use one call to the socket app
and limit the connections on the other side in a forking process or
use mod_limit ....


On Fri, Oct 7, 2011 at 4:10 PM, Hector Geraldino
<Hector.Geraldino at ip-soft.net> wrote:
> Hi,
>
> Most of the issues you've can be addressed with a 'single instanced' application, but I understand why you designed the application to work this way. If at any moment you think about enabling this application to handle several calls, you can overtake these issues in various ways, for example:
>
> - Controlling the number of calls: you can register all answered calls (uuids) in a data structure (list, map), remove them when disconnected, and decide when to accept a new call by getting the collection size and comparing it to your parameter. If you ever get more Nuance licenses, increasing the number of managed calls would be as easy as change a parameter in a property file.
>
> - You can have as many MRCP profiles as you want defined in the configuration xml files on mrcp_profiles/ directory. Coming up with the algorithm to assign channels to MRCP engines (Nuance server/port) won't be that hard. I can think of one: having all mrcp profiles in a map with a flag (in use/free), when the call comes I lock the first available profile so it's attached to this call. When the call is disconnected the profile is freed.
>
> - log mixed, sure. But, is this really a problem? The same with the publishing of topics, you can replicate the same behavior of having multiple instances in a single one. I mean, if you make an abstraction and say: one instance of my java application equals one thread of my new java app you'll see how easy is to overcome this and other problems.
>
> Sorry for such a long message, but as many others I feel real passion about application design topics. My intention is never to criticize what you've done, as each mind is a different world.
>
> Anyway, good luck with that!
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xl127
> Sent: Friday, October 07, 2011 6:55 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
>
> Hi Hector,
>
> Your points make sense but I have some specific issues for my
> application so I thought of binding only one call per runnning instance.
>
> 1. My outbound socket application is one component/server of our Spoken
> Dialogue System (SDS).
> We have four ports Nuance ASR/TTS licenses, each port is specific to one
> machine( yeah, we have four seperate license files for four machines).
> It seems to be difficult to specify which channel uses which Nuance MRCP
> profile if simultenous calls are answered by only one instance.
>
> 2. It seems to me that the log info would be mixed for simultenous calls
> if running one instance.
>
> 3. My socket app will be communicating with other components/servers of
> the SDS via ICE middleware (http://www.zeroc.com/).
> It will register to the central ICE server when starting the system via
> a Topic name which is a static name given by the developer,
> it will keep publishing ASR results on that topic so the other
> components listening on the topic will get the results,
> and it also subscribes and listens to other topics from other components
> for TTS.
>
> In simultenous calls to one running instance scenario, the
> distinguishable thing between channels would be the channel UUID which I
> could use
> as a part of the topic name, but it is dynamic and other servers could
> not know it when registering to the central server.
>
> I could let the all channels publish results with a same static topic
> name and channel UUID, the interested parties could parse the results
> they want based on the UUID, but I worried this would cause more burdens
> to the whole system.
>
>
> 4. Even if running one instance, I still need some ways to check and
> control the maximum number of calls as we have limited ASR/TTS resources.
>
> Cheers,
> Xing
>
> On 06/10/11 22:34, Hector Geraldino wrote:
>> This might sound a little bit Off-Topic but, why are you willing to bind only one call per running instance? I cannot imagine any user scenario where this is desirable. Maybe as a proof of concept is Ok, but having an equivalence between calls and instances mmm... I don't know.
>>
>> Can you please tell me what are you willing to accomplish? I'm really curious about it.
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
>> Sent: Thursday, October 06, 2011 3:20 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Bridge calls to Java ESL socket does not work as expected
>>
>> don't actually hang it up just don't connect the socket or connnect it
>> and disconnect it right away and the app will fail on in the
>> dial-plan.
>>
>> again you could try mod_limit otherwise.
>>
>>
>> On Thu, Oct 6, 2011 at 11:28 AM, xl127<x.liu at hw.ac.uk> ?wrote:
>>> Thanks for the suggestions!
>>>
>>> I thought about and tried to make my outbound socket application
>>> fail e.g. by send a hangup message but it didn't work.
>>>
>>> What I did is that I set a public variable channelUUID with initial
>>> value null in MyOutboundHandler,
>>> I check if it is null in the handleConnectionResponse method. If it not
>>> null, a hangup command is sent to FS.
>>> But it looks like each call channel creates its own Java thread, so the
>>> channelUUID is always null before the connection.
>>> So my approach doesn't work.
>>>
>>> Any idea about checking if the remote application is answering a call?
>>>
>>> Thanks!
>>> Xing
>>>
>>>
>>> On 06/10/11 16:43, Anthony Minessale wrote:
>>>> Really the socket app has not limit to how many simultaneous
>>>> connections it can handle.
>>>> You could try mod_limit, or make the remote end of you application
>>>> fail when there are too many connections.
>>>>
>>>>
>>>> On Thu, Oct 6, 2011 at 10:30 AM, xl127<x.liu at hw.ac.uk> ? ?wrote:
>>>>> Hello,
>>>>>
>>>>> I am trying to bridge multiple calls to multiple Java ESL outbound
>>>>> socket applications.
>>>>>
>>>>> Basically I want to check if the first socket app is engaged then the
>>>>> call should be transferred
>>>>> to the second socket app and so on.
>>>>>
>>>>> For initial test, I tried two concurrent calls, the problem is both
>>>>> calls are always answered
>>>>> by the first outbound socket.
>>>>>
>>>>> I have following setting in the dialplan:
>>>>>
>>>>> <extension name="sipgateDialedIn">
>>>>> <condition field="destination_number" expression="^5800$">
>>>>> <action application="set" data="continue_on_fail=true"/>
>>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>>> <action application="bridge" data="loopback/5801"/>
>>>>> <action application="bridge" data="loopback/5802"/>
>>>>>
>>>>> <action application="set" data="tts_engine=flite"/>
>>>>> <action application="set" data="tts_voice=kal"/>
>>>>> <action application="speak" data="Sorry, I am busy at the moment. Please
>>>>> try again later."/>
>>>>> <action application="sleep" data="10"/>
>>>>> <action application="hangup"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> <extension name="myOutboundESL5801">
>>>>> <condition field="destination_number" expression="^5801$">
>>>>> <action application="socket" data="127.0.0.1:9981 async full"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> <extension name="myOutboundESL5802">
>>>>> <condition field="destination_number" expression="^5802$">
>>>>> <action application="socket" data="127.0.0.1:9982 async full"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> I expect the first call to 5800 would be asnwered by extension 5801, the
>>>>> second concurrent call to 5800 will be
>>>>> answered by extension 5802, and the third call gets a busy message.
>>>>>
>>>>> I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if
>>>>> the socket is engaged?
>>>>>
>>>>> I've tried and googled a while but without success.
>>>>>
>>>>> Any advices please?
>>>>>
>>>>> Many thanks!
>>>>> Xing
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Heriot-Watt University is a Scottish charity
>>>>> registered under charity number SC000278.
>>>>>
>>>>> Heriot-Watt University is the Sunday Times
>>>>> Scottish University of the Year 2011-2012
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>>
>>> --
>>> Heriot-Watt University is a Scottish charity
>>> registered under charity number SC000278.
>>>
>>> Heriot-Watt University is the Sunday Times
>>> Scottish University of the Year 2011-2012
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
> Heriot-Watt University is the Sunday Times
> Scottish University of the Year 2011-2012
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012


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From cvogel at lyonl.com  Sun Oct  9 00:08:35 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Sat, 8 Oct 2011 20:08:35 +0000
Subject: [Freeswitch-users] From param
Message-ID: <6DB33DE1-E47A-4C2E-B32D-8D7289C61921@lyonl.com>

Does anyone know of a way to add a param to the from header in a dialplan? I would like to do something like a isup-oli=6  (From: <sip:restricted at 10.100.63.137;OTG=777;isup-oli=6>)

Chad
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From brad at tech21.com  Sun Oct  9 01:14:37 2011
From: brad at tech21.com (Brad Mina)
Date: Sat, 8 Oct 2011 14:14:37 -0700
Subject: [Freeswitch-users] Half the call recording
In-Reply-To: <CAJh8XdUj=vMPZQrrBBZ6QcswStU2TUOjjV7id7LAirswmWezAQ@mail.gmail.com>
References: <CAJh8XdUj=vMPZQrrBBZ6QcswStU2TUOjjV7id7LAirswmWezAQ@mail.gmail.com>
Message-ID: <CAGRr05_fkkaXd-UqqHw6NJOx556LvKoyVD=VG=fKbT1pgDzDiw@mail.gmail.com>

Hynek,

The tool you're looking for is record_session.
See  http://wiki.freeswitch.com/wiki/Misc._Dialplan_Tools_record_session

On Sat, Oct 8, 2011 at 1:14 AM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

> I cannot find information on recording half the call, only a page which
> states, that split recording of the caller/callee is possible with
> freeswitch. Reading the documentation about recording API commands I
> couldn't find any info, whether freeswittch can record only one leg of a
> two-legged called or even how to do it.
>
> Any help is appreciated.
>
> Hynek
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct  9 01:54:29 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 8 Oct 2011 23:54:29 +0200
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <4E8F0F50.8010201@tagnet.ru>
References: <4E8F0F50.8010201@tagnet.ru>
Message-ID: <CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com>

I've been wondering how to do this.
When I asked in channel, I was basically told this couldn't be done in a
single bridge string... I hope that's not actually the case.
-Avi


On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     Have read docs but misundertood :( I use mod_lcr which return two
> or more gateways to the same destination. And when the user is busy with
> the first gateway the call continues with the second and so on. How may
> I to not continue gateways if user is busy?
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct  9 04:03:12 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 9 Oct 2011 02:03:12 +0200
Subject: [Freeswitch-users] Short Calls - how to prevent?
Message-ID: <CALNZuwn320Ur6TTnHWjAdK0S6CvPLUSmOOR64DPN4nsNY6YZoQ@mail.gmail.com>

Looking at my calling card logs for the last month, I have nearly 20% of my
calls under 6 seconds. I know a LOT of them are from hanging up after 20+
seconds, presumably because they got voicemail.
Most carriers don't like this and have heavy fees - e.g. 2cents for short
calls, or just 2c for each short call over the limit. Hmm, for my call
volume that's only a few dollars...
I wonder if anyone had any ideas?
I was thinking a transfer_on_hangup for leg B to send to a lua script - if
the duration is under 6 seconds, then sleep for a second a check again. But
I suppose only charge the caller for the first 6 seconds (since I never said
12 second minimum). (like 30cents instead of $5 fee on 900 short calls, of
5k calls).
Actually, 240 of them might be because of the ANI I was passing...

Does anyone have other ideas, or already wrote this code?

Thanks,
-Avi Marcus

718-989-9485 (USA)
054-844-3271 (Israel Kosher)
077-228-5055 (Israel Landline)
020-3519-3606 (UK)
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From jack at livecall.com  Sun Oct  9 08:31:06 2011
From: jack at livecall.com (Jack)
Date: Sat, 08 Oct 2011 21:31:06 -0700
Subject: [Freeswitch-users] Newbie question...
In-Reply-To: <CANtw_ZO+q74+bQwipQh9VYWhXXBZhN085QQUvD8GcLVhgBfhmA@mail.gmail.com>
References: <CANtw_ZNnE9v7+EHhnjE+8V9_UFoxuwWNJ=bj6Jiiu4AuMWpVdg@mail.gmail.com>
	<CAKzWOxXgOC4+0tZmZRdPJobQXXOef7voubUcuUnU-GwbKT0uiA@mail.gmail.com>
	<CANtw_ZO+q74+bQwipQh9VYWhXXBZhN085QQUvD8GcLVhgBfhmA@mail.gmail.com>
Message-ID: <4E91238A.8010104@livecall.com>

Are you sure you have access to the ports 5060 and 5080?
  It sounds like windows firewall may be blocking them.
You can set an exception in windows firewall to allow access.

hope that helps....
jack

On 10/8/2011 8:32 AM, Francisco Javier Cintr?n Olgu?n wrote:
> Thank you for your help.
>
> Well, Yes I installed Xlite in the same windows 7 rig in which I 
> installed freeswitch. And yes I am trying to learn what is VOIP, I 
>  read that freeswitch could run in windows so I bought freeswitch book 
> and here I am trying to understand it.
>
> this is the answer of the eval command:
>
> 10.10.100.31
>
> I put this IP in my xlite account, in fact I put 127.0.0.1 too, but it 
> always failing with SIP 408 code.
>
> I changed freeswitch to run with administrator account, but I am 
> getting same SIP 408 error.
>
> ?What am I doing wrong?
>
> Thanks in advance.
> Have a nice day.
>
> On Fri, Oct 7, 2011 at 4:19 PM, Michael Collins <msc at freeswitch.org 
> <mailto:msc at freeswitch.org>> wrote:
>
>     Welcome to FreeSWITCH and VoIP! I hope you don't get scared away
>     too easily. :)
>
>     First question: is your xlite on the same computer as your
>     FreeSWITCH install? Usually you don't want your softphone and FS
>     to be on the same computer, however if you are simply doing some
>     testing and trying to learn then it should be okay.
>
>     Here is where the fun begins. A SIP 408 means that the request
>     timed out, so the xlite is sending messages to the FS server and
>     is not getting anything back. Be sure that you use the correct IP
>     address. The best thing to do would be to start fs_cli.exe (which
>     connects you to the FS command line) and then type this command:
>
>     eval ${domain}
>
>     It should return an IP address - use that IP address in the xlite
>     account configuration.
>
>     Let us know how it goes!
>
>     -MC
>
>     On Fri, Oct 7, 2011 at 2:05 PM, Francisco Javier Cintr?n Olgu?n
>     <fcintrono at gmail.com <mailto:fcintrono at gmail.com>> wrote:
>
>         Ok guys, I am a newbie in VOIP stuff and in freeswitch of course.
>
>         I got the freeswitch book and installed freeswitch 1.0.6 in my
>         windows 7 pro rig.
>
>         I also installed XLite 4 but when I try to register the
>         extension is telling me: "Acount failed to enable" SIP 408 error.
>
>         I am sure I stoped my firewall and  even with that I am
>         getting same error.
>
>         I have not changed anything from the default configuration,
>         the extension I am trying to register is extension 1000, with
>         password 1234.
>
>
>         ?Could you help me please?
>
>         Thanks in advance.
>
>
>
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Sun Oct  9 12:45:42 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 Oct 2011 04:45:42 -0400
Subject: [Freeswitch-users] Host registration on shared ODBC
Message-ID: <A40AA1EB44224C4FB83D357B8223D5B0@e1705>

I got a strange behaviour recently.
if a phone is registrered on node 1
and check every 20s with "sofia contact"
so this command returns the right node.
but if run "sofia status profile myprofile"
it returns the wrong node after an incoming call.

there are nothing abnormal on logs at levevl7.
I run FS with postgresql and shared ODBC.

FreeSWITCH Version 1.0.head (git-f4fa0c0 2011-10-04 20-17-12 -0500)

Is anyone had already this problem ?

Thanks

Franck


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From fieldpeak at gmail.com  Sun Oct  9 14:32:24 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 9 Oct 2011 18:32:24 +0800
Subject: [Freeswitch-users] Failed to get callee number during the call
Message-ID: <CAOTmbMzFpHBAfnZzcx0ByLMSQbj1hdy8rC+qQQL9pTNOvzGMbA@mail.gmail.com>

Hello Friends,

i would like to get the callee number during the call, i used the
function 'switch_channel_get_variable',
however, when i display it to the console, it shows 'null', i tried
call this func before and after answering the call, both failed, could
anyone help advise if i used the correct function or how to realzize
it, thanks.

source code as below,

	const char* in_calledstation;

	in_calledstation = switch_channel_get_variable(channel,
"sip_callee_id_number");
	
	switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "in_username:
%s, in_calledstation: %s, in_uniqueid: %s, sessiontime: %d \n",
in_username, in_calledstation, in_uniqueid, sessiontime);
	
-- 
Regards,
Charles


From fieldpeak at gmail.com  Sun Oct  9 14:48:42 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 9 Oct 2011 18:48:42 +0800
Subject: [Freeswitch-users] Failed to get callee number during the call
In-Reply-To: <CAOTmbMzFpHBAfnZzcx0ByLMSQbj1hdy8rC+qQQL9pTNOvzGMbA@mail.gmail.com>
References: <CAOTmbMzFpHBAfnZzcx0ByLMSQbj1hdy8rC+qQQL9pTNOvzGMbA@mail.gmail.com>
Message-ID: <CAOTmbMzd_O1ix6E2Ng2cGTHJSGXuEi4oiMi2e7GH0Nwn0TVQ_Q@mail.gmail.com>

Hello Friends,

i have resolved this issue by using varialbe 'destination_number' to
replacing 'sip_callee_id_number', thanks.

BR,
Charles

2011/10/9, fieldpeak <fieldpeak at gmail.com>:
> Hello Friends,
>
> i would like to get the callee number during the call, i used the
> function 'switch_channel_get_variable',
> however, when i display it to the console, it shows 'null', i tried
> call this func before and after answering the call, both failed, could
> anyone help advise if i used the correct function or how to realzize
> it, thanks.
>
> source code as below,
>
> 	const char* in_calledstation;
>
> 	in_calledstation = switch_channel_get_variable(channel,
> "sip_callee_id_number");
> 	
> 	switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "in_username:
> %s, in_calledstation: %s, in_uniqueid: %s, sessiontime: %d \n",
> in_username, in_calledstation, in_uniqueid, sessiontime);
> 	
> --
> Regards,
> Charles
>


-- 
Regards,
Charles


From mi.ke at null.net  Sun Oct  9 15:24:41 2011
From: mi.ke at null.net (Mi Ke)
Date: Sun, 09 Oct 2011 07:24:41 -0400
Subject: [Freeswitch-users] mod_sofia@
Message-ID: <20111009112442.208730@gmx.com>

found it on wiki - it's sip_contact_user variable

----- Original Message -----
From: Mi Ke
Sent: 10/08/11 09:33 PM
To: Freeswitch Users
Subject: [Freeswitch-users] mod_sofia@

Hi !

 I originate call from lua script by the following command:

local ob_session = freeswitch.Session("{originate_timeout=120,return_ring_ready=true,origination_caller_id_number=" .. calling_id .. ",called_id=" .. called_id .. ",accountcode=" .. accountcode .. ",credit_time=" .. out_credit_time .. ",execute_on_answer=lua b_answer.lua}" .. out_called_id,session)

 My iface IP is 2.2.2.2, origination_caller_id_number is my calling ID (55555) and out_called_id = sofia/internal/1212121212 at 1.1.1.1

 Leg B setup and bridging goes normally, when remote party disconnects, we receive:

tport_wakeup_pri(0x81c1a0): events IN
tport_recv_event(0x81c1a0)
tport_recv_iovec(0x81c1a0) msg 0x7f28386348a0 from (udp/2.2.2.2:5060 http://78.46.44.138:5060/ ) has 425 bytes, veclen = 1
tport_deliver(0x81c1a0): msg 0x7f28386348a0 (425 bytes) from udp/1.1.1.1:5060/sip http://193.104.107.7:5060/sip  next=(nil)
nta: received BYE sip:mod_sofia at 2.2.2.2: http://sip:mod_sofia at 78.46.44.138:5060/ 5060 http://sip:mod_sofia at 78.46.44.138:5060/  SIP/2.0 (CSeq 48765)
nta: canonizing sip:mod_sofia at 2.2.2.2:5060 http://sip:mod_sofia at 78.46.44.138:5060/  with contact
nta: BYE (48765) going to existing leg
nua: nua_stack_process_request: entering
nua(0x94c4a0): event i_bye 100 Trying
nua: nua_application_event: entering
2011-10-08 14:54:48.866309 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/ 841266482056 at 193.104.107.7 .1.1.1 [BREAK]
nua: nua_handle_magic: entering
2011-10-08 14:54:48.886307 [DEBUG] switch_channel.c:2800 (sofia/internal/ 841266482056 at 193.104.107.7 .1.1.1) Callstate Change ACTIVE -> HANGUP

 (normal bye handshake follows...)

 Should remote party see me as sip:55555 at 2.2.2.2 and not as sip:mod_sofia at 2.2.2.2 ? What info should I include into session originate to acheive that?

 Thanks / Mike
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From a.afzali2003 at gmail.com  Sun Oct  9 18:47:10 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 9 Oct 2011 18:17:10 +0330
Subject: [Freeswitch-users] Snom 370 Presence Function
Message-ID: <CABK0fC7LqFNJT4yMayLeQXhd1O-KzsYn_ppcQiRNXRDx=wiH0g@mail.gmail.com>

Hi Guys,

I'll appreciate if I know someone has been experienced snom 370
presence function. As I trace sip messages, it shows that snom does
not subscribe for itself and just tries to publish its state which
replies 404 by freeswitch ! I don't know it is the snom or freeswitch
protocol issue.

BEST,
-- afshin


From Nabble_01394 at slickdeals.endjunk.com  Mon Oct 10 04:17:54 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sun, 9 Oct 2011 17:17:54 -0700 (PDT)
Subject: [Freeswitch-users] mod_unicall
Message-ID: <1318205874399-6875427.post@n2.nabble.com>

What is the current status of mod_unicall, i.e. usable, compilable, etc.?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-unicall-tp6875427p6875427.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chris at ghosttelecom.com  Mon Oct 10 14:01:01 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Mon, 10 Oct 2011 11:01:01 +0100
Subject: [Freeswitch-users] Break command not working
In-Reply-To: <CAKbxfG90QQTtnsB0DsxMipPWpkJZXemTsKJ5fiDsPwqjgz9MuA@mail.gmail.com>
References: <1D10AB188D6CCA46BB4369E3268E36EF35A179@SVR01.ghosttelecom.local>
	<CAKbxfG90QQTtnsB0DsxMipPWpkJZXemTsKJ5fiDsPwqjgz9MuA@mail.gmail.com>
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF35A1B5@SVR01.ghosttelecom.local>

Hi,

Thanks for that, I had found the api on answer and was trying to make
that work.

Trying to get the below to work but looking at the log it seems to be
ignoring ${uuid} as this is not inserted in the EXECUTE section. All I
see is..
EXECUTE ... bridge({api_on_answer='uuid_break  '}.... I assume that I
should see the uuid inserted here?

I then get a DEBUG message on answer saying 
api on answer: uuid_break()
-USAGE: <uuid> [all]

Which I assume is telling me that there is no uuid!

Any ideas

Regards

Chris

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 07 October 2011 17:47
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Break command not working

you can't execute the break because it's busy playing the file, you
can do the break on the other leg though,
This is more of what you want.

    <extension name="1235">
      <condition field="destination_number" expression="^1235$">
        <action application="set"
data="execute_on_media=playback::/path/file"/>
        <action application="pre_answer"/>
        <action application="bridge"
data="{ignore_early_media=true,api_on_answer='uuid_break
${uuid}'}sofia/internal/foo at bar.com"/>
      </condition>
    </extension>


On Fri, Oct 7, 2011 at 10:03 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi,
>
>
>
> I have the following scenario...
>
>
>
> application='preanswer'
>
> application='set' data='execute_on_answer=break'
>
> application='sched_broadcast' data='+1 /path/file'
>
> application='bridge' data='sofia...'
>
>
>
> This works fine in that it pre answers the call and plays back a
welcome
> message prior to normal early media ringtone.
>
>
>
> The problem I have is that if the call is answered quickly the
playback just
> continues until finished.
>
>
>
> I would have thought that application='set'
data='execute_on_answer=break'
> should stop the playback on answer but the break command doesn't have
any
> effect. I have tried break all to no effect.
>
>
>
> Should this work?
>
>
>
> Any ideas on how to make this work?
>
>
>
> Many thanks
>
>
>
> Chris
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


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


From kerem.erciyes at gmail.com  Mon Oct 10 17:27:21 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Mon, 10 Oct 2011 16:27:21 +0300
Subject: [Freeswitch-users] please remove me from the mailing list
In-Reply-To: <CAB23310.1ED98%krice@freeswitch.org>
References: <4E8CC7D8.4040406@gosilverplus.com>
	<CAB23310.1ED98%krice@freeswitch.org>
Message-ID: <CAJ6wjQGsaZ3brnX6rZrQ2Q-wWU7_D6rF+5jxxLS871yUkRLLog@mail.gmail.com>

Maybe he's gone to the page and pressed the "unsubscribe" button many times
and can't figure out what is happening.

[:wink:]

On Thu, Oct 6, 2011 at 12:15 AM, Ken Rice <krice at freeswitch.org> wrote:

>  Ok, try reading at the bottom of every email that goes out on the
> lists... Quoted below so it can be easily found
>
>
>
> On 10/5/11 4:10 PM, "ran zhang" <rzhang at gosilverplus.com> wrote:
>
> FreeSWITCH-users mailing list
> *FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> *UNSUBSCRIBE:*http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> *
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com
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From chris at ghosttelecom.com  Mon Oct 10 17:43:04 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Mon, 10 Oct 2011 14:43:04 +0100
Subject: [Freeswitch-users] Break command not working
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF35A1B5@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF35A179@SVR01.ghosttelecom.local><CAKbxfG90QQTtnsB0DsxMipPWpkJZXemTsKJ5fiDsPwqjgz9MuA@mail.gmail.com>
	<1D10AB188D6CCA46BB4369E3268E36EF35A1B5@SVR01.ghosttelecom.local>
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF35A1F5@SVR01.ghosttelecom.local>

Sorry my bad, fixed this, just a stray quote.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Chris Martineau
Sent: 10 October 2011 11:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Break command not working

Hi,

Thanks for that, I had found the api on answer and was trying to make
that work.

Trying to get the below to work but looking at the log it seems to be
ignoring ${uuid} as this is not inserted in the EXECUTE section. All I
see is..
EXECUTE ... bridge({api_on_answer='uuid_break  '}.... I assume that I
should see the uuid inserted here?

I then get a DEBUG message on answer saying 
api on answer: uuid_break()
-USAGE: <uuid> [all]

Which I assume is telling me that there is no uuid!

Any ideas

Regards

Chris

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 07 October 2011 17:47
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Break command not working

you can't execute the break because it's busy playing the file, you
can do the break on the other leg though,
This is more of what you want.

    <extension name="1235">
      <condition field="destination_number" expression="^1235$">
        <action application="set"
data="execute_on_media=playback::/path/file"/>
        <action application="pre_answer"/>
        <action application="bridge"
data="{ignore_early_media=true,api_on_answer='uuid_break
${uuid}'}sofia/internal/foo at bar.com"/>
      </condition>
    </extension>


On Fri, Oct 7, 2011 at 10:03 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi,
>
>
>
> I have the following scenario...
>
>
>
> application='preanswer'
>
> application='set' data='execute_on_answer=break'
>
> application='sched_broadcast' data='+1 /path/file'
>
> application='bridge' data='sofia...'
>
>
>
> This works fine in that it pre answers the call and plays back a
welcome
> message prior to normal early media ringtone.
>
>
>
> The problem I have is that if the call is answered quickly the
playback just
> continues until finished.
>
>
>
> I would have thought that application='set'
data='execute_on_answer=break'
> should stop the playback on answer but the break command doesn't have
any
> effect. I have tried break all to no effect.
>
>
>
> Should this work?
>
>
>
> Any ideas on how to make this work?
>
>
>
> Many thanks
>
>
>
> Chris
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


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


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


From sirvan.paraste at gmail.com  Mon Oct 10 09:38:19 2011
From: sirvan.paraste at gmail.com (sirvan paraste)
Date: Mon, 10 Oct 2011 09:08:19 +0330
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <CAAAG0NVAnacuknJTVWD+4fXCSpNf9BMaonVfp5e_ENOvQ+11-g@mail.gmail.com>
References: <CAAAG0NVAnacuknJTVWD+4fXCSpNf9BMaonVfp5e_ENOvQ+11-g@mail.gmail.com>
Message-ID: <CAAAG0NXmyse-uSKudTo1T--7DsMLgGNGhvCC1WhBZhDiu7CR=Q@mail.gmail.com>

Dear all
After regard for your attention to me
I am student how want work on freeswitch. I have read freeSwitch
documents (introduction, how to install, use default
confuguration,...).
for start of using freeswitch, I make three computer network together
with a hub, after that,I install freeswitch on one of them and X-Lite
on another computers, network checked with sharing file and ping
issue.
After registering of two X-lite hone with No. 1001,1002, they can call
each other but there is no voice !!! :(
i have  checked and did some work but the problem is not solved.

Can you help me?
thank you.


From hynek.cihlar at gmail.com  Mon Oct 10 14:04:12 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Mon, 10 Oct 2011 12:04:12 +0200
Subject: [Freeswitch-users] Half the call recording
In-Reply-To: <CAGRr05_fkkaXd-UqqHw6NJOx556LvKoyVD=VG=fKbT1pgDzDiw@mail.gmail.com>
References: <CAJh8XdUj=vMPZQrrBBZ6QcswStU2TUOjjV7id7LAirswmWezAQ@mail.gmail.com>
	<CAGRr05_fkkaXd-UqqHw6NJOx556LvKoyVD=VG=fKbT1pgDzDiw@mail.gmail.com>
Message-ID: <CAJh8XdWqYjgSj1BprgS+KD797RvbUHcgLwzzs5qM7bJyBpuutg@mail.gmail.com>

Brad, thanks for the info. I'm gonna give it a try.

Hynek



On Sat, Oct 8, 2011 at 11:14 PM, Brad Mina <brad at tech21.com> wrote:

> Hynek,
>
> The tool you're looking for is record_session.
> See  http://wiki.freeswitch.com/wiki/Misc._Dialplan_Tools_record_session
>
> On Sat, Oct 8, 2011 at 1:14 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> I cannot find information on recording half the call, only a page which
>> states, that split recording of the caller/callee is possible with
>> freeswitch. Reading the documentation about recording API commands I
>> couldn't find any info, whether freeswittch can record only one leg of a
>> two-legged called or even how to do it.
>>
>> Any help is appreciated.
>>
>> Hynek
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jovemark at gmail.com  Mon Oct 10 14:15:27 2011
From: jovemark at gmail.com (=?GB2312?B?tPPT8C6358/y?=)
Date: Mon, 10 Oct 2011 18:15:27 +0800
Subject: [Freeswitch-users] How can make mod_rmtp support text message?
Message-ID: <CAAsEjANudiRxFsE+EH4BPZ3=4h7ajGz=+5X=6ppq4XNTqsnwQQ@mail.gmail.com>

We can find that there are two mods can bind to work together. One is
mod_rmtp , another is mod_sms.

How can make mod_rmtp work with mod_sms to support text message on flash
client ?

Anybody have suggestions?

Regards.
Martin.
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From msc at freeswitch.org  Mon Oct 10 19:12:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Oct 2011 08:12:30 -0700
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com>
References: <4E8F0F50.8010201@tagnet.ru>
	<CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com>
Message-ID: <CAKzWOxUCVsx1LCa1mO8AXhQ=sPKtANOCgvVu-xAmx1Py9z68EQ@mail.gmail.com>

You will need to specify the error conditions on which to continue. See the
example here:
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

-MC

P.S. - Check out the "enumeration" column on this page to get an idea of the
kinds of failures you may not be aware of:
http://wiki.freeswitch.org/wiki/Hangup_causes

On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I've been wondering how to do this.
> When I asked in channel, I was basically told this couldn't be done in a
> single bridge string... I hope that's not actually the case.
> -Avi
>
>
> On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     Have read docs but misundertood :( I use mod_lcr which return two
>> or more gateways to the same destination. And when the user is busy with
>> the first gateway the call continues with the second and so on. How may
>> I to not continue gateways if user is busy?
>>
>> --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From djbinter at gmail.com  Mon Oct 10 19:39:42 2011
From: djbinter at gmail.com (DJB International)
Date: Mon, 10 Oct 2011 08:39:42 -0700
Subject: [Freeswitch-users] From param
In-Reply-To: <6DB33DE1-E47A-4C2E-B32D-8D7289C61921@lyonl.com>
References: <6DB33DE1-E47A-4C2E-B32D-8D7289C61921@lyonl.com>
Message-ID: <CAAB3jWvshvngMa44JQ+k=qR0a_xHEjc9MonjtQzHyRZ-Hi1EZQ@mail.gmail.com>

You can do in the bridge like: sofia/external/18005551212 at 1.2.3.4:5060;*
isup-oli=6*

or you can use sip_from_params

-djbinter



On Sat, Oct 8, 2011 at 1:08 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  Does anyone know of a way to add a param to the from header in a dialplan?
> I would like to do something like a isup-oli=6  (From: <
> sip:restricted at 10.100.63.137;OTG=777;*isup-oli=6*>)
>
> ****
> Chad
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Mon Oct 10 19:46:50 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 10 Oct 2011 15:46:50 +0000
Subject: [Freeswitch-users] From param
In-Reply-To: <CAAB3jWvshvngMa44JQ+k=qR0a_xHEjc9MonjtQzHyRZ-Hi1EZQ@mail.gmail.com>
References: <6DB33DE1-E47A-4C2E-B32D-8D7289C61921@lyonl.com>
	<CAAB3jWvshvngMa44JQ+k=qR0a_xHEjc9MonjtQzHyRZ-Hi1EZQ@mail.gmail.com>
Message-ID: <D575168E-501F-4C11-9DEE-C4DB6F519B7D@lyonl.com>

Is there a way to send it on a respond?

On Oct 10, 2011, at 10:39 AM, DJB International wrote:

You can do in the bridge like: sofia/external/18005551212 at 1.2.3.4:5060<http://18005551212 at 1.2.3.4:5060/>;isup-oli=6

or you can use sip_from_params

-djbinter



On Sat, Oct 8, 2011 at 1:08 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Does anyone know of a way to add a param to the from header in a dialplan? I would like to do something like a isup-oli=6  (From: <sip:restricted at 10.100.63.137;OTG=777;isup-oli=6>)

Chad


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



FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From avi at avimarcus.net  Mon Oct 10 19:54:48 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 10 Oct 2011 17:54:48 +0200
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <CAKzWOxUCVsx1LCa1mO8AXhQ=sPKtANOCgvVu-xAmx1Py9z68EQ@mail.gmail.com>
References: <4E8F0F50.8010201@tagnet.ru>
	<CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com>
	<CAKzWOxUCVsx1LCa1mO8AXhQ=sPKtANOCgvVu-xAmx1Py9z68EQ@mail.gmail.com>
Message-ID: <CALNZuw=sBYwWGcQmT+g1B_DeT5KdZqP49sFXZ2MAYaNYvOX71g@mail.gmail.com>

It seems from testing, and I was told on channel:  continue_on_fail is per
bridge string, not per leg. So it doesn't actually help with the question..
Unless I'm mistaken.
-Avi


On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org> wrote:

> You will need to specify the error conditions on which to continue. See the
> example here:
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
> -MC
>
> P.S. - Check out the "enumeration" column on this page to get an idea of
> the kinds of failures you may not be aware of:
> http://wiki.freeswitch.org/wiki/Hangup_causes
>
>
> On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I've been wondering how to do this.
>> When I asked in channel, I was basically told this couldn't be done in a
>> single bridge string... I hope that's not actually the case.
>> -Avi
>>
>>
>> On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>>> Hello!
>>>
>>>     Have read docs but misundertood :( I use mod_lcr which return two
>>> or more gateways to the same destination. And when the user is busy with
>>> the first gateway the call continues with the second and so on. How may
>>> I to not continue gateways if user is busy?
>>>
>>> --
>>> ? ?????????,
>>>   ????? ?????????
>>>   ??? "??????"
>>>   ???. +7 (3435) 230001
>>>   ???? +7 (3435) 230005
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 10 20:40:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Oct 2011 09:40:57 -0700
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <CALNZuw=sBYwWGcQmT+g1B_DeT5KdZqP49sFXZ2MAYaNYvOX71g@mail.gmail.com>
References: <4E8F0F50.8010201@tagnet.ru>
	<CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com>
	<CAKzWOxUCVsx1LCa1mO8AXhQ=sPKtANOCgvVu-xAmx1Py9z68EQ@mail.gmail.com>
	<CALNZuw=sBYwWGcQmT+g1B_DeT5KdZqP49sFXZ2MAYaNYvOX71g@mail.gmail.com>
Message-ID: <CAKzWOxVB=pMyWoQ_-Q5+DyV3E5in53Lgo3+kzo4oZRR9BUTFBQ@mail.gmail.com>

Well, now we're talking about something more than just continuing_on_fail.
We'll need to look at the actual dialstring to answer any more specific
questions.

-MC

On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:

> It seems from testing, and I was told on channel:  continue_on_fail is per
> bridge string, not per leg. So it doesn't actually help with the question..
> Unless I'm mistaken.
> -Avi
>
>
> On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> You will need to specify the error conditions on which to continue. See
>> the example here:
>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>
>> -MC
>>
>> P.S. - Check out the "enumeration" column on this page to get an idea of
>> the kinds of failures you may not be aware of:
>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>
>>
>> On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I've been wondering how to do this.
>>> When I asked in channel, I was basically told this couldn't be done in a
>>> single bridge string... I hope that's not actually the case.
>>> -Avi
>>>
>>>
>>> On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>
>>>> Hello!
>>>>
>>>>     Have read docs but misundertood :( I use mod_lcr which return two
>>>> or more gateways to the same destination. And when the user is busy with
>>>> the first gateway the call continues with the second and so on. How may
>>>> I to not continue gateways if user is busy?
>>>>
>>>> --
>>>> ? ?????????,
>>>>   ????? ?????????
>>>>   ??? "??????"
>>>>   ???. +7 (3435) 230001
>>>>   ???? +7 (3435) 230005
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Oct 10 20:48:30 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 10 Oct 2011 18:48:30 +0200
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <CAKzWOxVB=pMyWoQ_-Q5+DyV3E5in53Lgo3+kzo4oZRR9BUTFBQ@mail.gmail.com>
References: <4E8F0F50.8010201@tagnet.ru>
	<CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com>
	<CAKzWOxUCVsx1LCa1mO8AXhQ=sPKtANOCgvVu-xAmx1Py9z68EQ@mail.gmail.com>
	<CALNZuw=sBYwWGcQmT+g1B_DeT5KdZqP49sFXZ2MAYaNYvOX71g@mail.gmail.com>
	<CAKzWOxVB=pMyWoQ_-Q5+DyV3E5in53Lgo3+kzo4oZRR9BUTFBQ@mail.gmail.com>
Message-ID: <CALNZuwnBQixaweDEa6YFPpx4zW5_c5+zn9SFtLSEtGWvQ3CyNA@mail.gmail.com>

the question was about mod_lcr.
it returns something like:

sofia/gateway/carrier1/14105780999|sofia/gateway/
carrier2/14105780999|sofia/gateway/carrier3/14105780999

The question was how to make it when the first returns "user_busy" to
not failover and try the other ones.

-Avi


On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org> wrote:

> Well, now we're talking about something more than just continuing_on_fail.
> We'll need to look at the actual dialstring to answer any more specific
> questions.
>
> -MC
>
>
> On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> It seems from testing, and I was told on channel:  continue_on_fail is per
>> bridge string, not per leg. So it doesn't actually help with the question..
>> Unless I'm mistaken.
>> -Avi
>>
>>
>> On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> You will need to specify the error conditions on which to continue. See
>>> the example here:
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>
>>> -MC
>>>
>>> P.S. - Check out the "enumeration" column on this page to get an idea of
>>> the kinds of failures you may not be aware of:
>>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>>
>>>
>>> On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> I've been wondering how to do this.
>>>> When I asked in channel, I was basically told this couldn't be done in a
>>>> single bridge string... I hope that's not actually the case.
>>>> -Avi
>>>>
>>>>
>>>> On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru>wrote:
>>>>
>>>>> Hello!
>>>>>
>>>>>     Have read docs but misundertood :( I use mod_lcr which return two
>>>>> or more gateways to the same destination. And when the user is busy
>>>>> with
>>>>> the first gateway the call continues with the second and so on. How may
>>>>> I to not continue gateways if user is busy?
>>>>>
>>>>> --
>>>>> ? ?????????,
>>>>>   ????? ?????????
>>>>>   ??? "??????"
>>>>>   ???. +7 (3435) 230001
>>>>>   ???? +7 (3435) 230005
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From spencer at 5ninesolutions.com  Mon Oct 10 21:27:46 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 10 Oct 2011 10:27:46 -0700
Subject: [Freeswitch-users] SCA Hunt Group
Message-ID: <7EE40111-38CC-4E93-B1CC-38A9771E837D@5ninesolutions.com>

Hello all,
I'm working on replacing a legacy key system with FreeSWITCH and a few Cisco SPA-509G phones.  I have SCA/SLA working correctly but I was wondering if anyone has a suggestion for a good way to make the inbound calls to the shared extensions "roll over" like a traditional phone system.  I have a simple lua script that does a sequential dial and checks for a busy response from the phones but the problem is that these Cisco phones have a minimum of 2 calls per line.  This results in the second call going to the same "line" as the first.  What I'd like to do is have an inbound call pick the next available shared extension.  Does anyone have any suggestions?  Is there some variable I could check with a script that would show whether the shared extension is in use?

Thanks,
Spencer




From Stefan.Weigel at allianz-warranty.com  Mon Oct 10 23:00:19 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Mon, 10 Oct 2011 21:00:19 +0200
Subject: [Freeswitch-users] Antw.:  SCA Hunt Group
Message-ID: <A8CDC463-70A6-4AE1-8267-7C21A41CFC71@allianz-warranty.com>

Hi Spencer,

just use mod_limit to limit one call per extension.

http://wiki.freeswitch.org/wiki/Mod_limit


Best regards,

Stefan

----- Reply message -----
Von: "Spencer Thomason" <spencer at 5ninesolutions.com>
Datum: Mo., Okt. 10, 2011 19:29
Betreff: [Freeswitch-users] SCA Hunt Group
An: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Hello all,
I'm working on replacing a legacy key system with FreeSWITCH and a few Cisco SPA-509G phones.  I have SCA/SLA working correctly but I was wondering if anyone has a suggestion for a good way to make the inbound calls to the shared extensions "roll over" like a traditional phone system.  I have a simple lua script that does a sequential dial and checks for a busy response from the phones but the problem is that these Cisco phones have a minimum of 2 calls per line.  This results in the second call going to the same "line" as the first.  What I'd like to do is have an inbound call pick the next available shared extension.  Does anyone have any suggestions?  Is there some variable I could check with a script that would show whether the shared extension is in use?

Thanks,
Spencer




FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From adrottenberg at gmail.com  Mon Oct 10 23:38:20 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Mon, 10 Oct 2011 15:38:20 -0400
Subject: [Freeswitch-users] Conference floor given to muted member
Message-ID: <CAC4JBEAV13j4Mjn1FYX3A2U_PBv6VD7BPFVtfFADTdEeF_+PdQ@mail.gmail.com>

I am setting up a conference to use for paging similar to
http://wiki.freeswitch.org/wiki/Lua_Intercom and I noticed that the
first 2 seconds of audio are not being sent. After reviewing the logs
it appears that the first member to enter the conference is
automatically given the floor event thought it was entered with the
mute flag. I set it up so that the originator enters the conference
with the moderator flag and is also unmuted, however it appears that
he doesn't get the floor right away, even though he's the only unmuted
member. It takes about 2 seconds of speech for the moderator to get
the floor. I have the following settings for the conference profile I
use:

<param name="energy-level" value="0"/>
<param name="conference-flags" value="wait-mod"/>
<param name="moh-sound" value="silence"/>
<param name="moh-sound" value="silence"/>

I tried setting <param name="member-flags" value="waste"/> but it
didn't help, for now I changed it so that the moderator enters the
conference with the wast flag.

Are there any other settings I can use to either avoid giving a muted
member the floor, or to speed up the time it takes to give the floor
to the moderator?

Thank You,
Duvid Rottenberg


From anthony.minessale at gmail.com  Mon Oct 10 23:51:53 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 Oct 2011 14:51:53 -0500
Subject: [Freeswitch-users] Conference floor given to muted member
In-Reply-To: <CAC4JBEAV13j4Mjn1FYX3A2U_PBv6VD7BPFVtfFADTdEeF_+PdQ@mail.gmail.com>
References: <CAC4JBEAV13j4Mjn1FYX3A2U_PBv6VD7BPFVtfFADTdEeF_+PdQ@mail.gmail.com>
Message-ID: <CAKbxfG-7FUfdmw4M-sqLtr=zR4BxEMn7tc4=r68MKYi7m5zJow@mail.gmail.com>

The floor is really an aesthetic moniker.  It's not controlling the audio etc.
If you are not getting audio in the beginning of the call it's
probably a NAT translation auto adjusting.
Try adding something like answer followed by sleep 2000 in your
dialplan before entering the conference.


On Mon, Oct 10, 2011 at 2:38 PM, Duvid Rottenberg
<adrottenberg at gmail.com> wrote:
> I am setting up a conference to use for paging similar to
> http://wiki.freeswitch.org/wiki/Lua_Intercom and I noticed that the
> first 2 seconds of audio are not being sent. After reviewing the logs
> it appears that the first member to enter the conference is
> automatically given the floor event thought it was entered with the
> mute flag. I set it up so that the originator enters the conference
> with the moderator flag and is also unmuted, however it appears that
> he doesn't get the floor right away, even though he's the only unmuted
> member. It takes about 2 seconds of speech for the moderator to get
> the floor. I have the following settings for the conference profile I
> use:
>
> <param name="energy-level" value="0"/>
> <param name="conference-flags" value="wait-mod"/>
> <param name="moh-sound" value="silence"/>
> <param name="moh-sound" value="silence"/>
>
> I tried setting <param name="member-flags" value="waste"/> but it
> didn't help, for now I changed it so that the moderator enters the
> conference with the wast flag.
>
> Are there any other settings I can use to either avoid giving a muted
> member the floor, or to speed up the time it takes to give the floor
> to the moderator?
>
> Thank You,
> Duvid Rottenberg
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From sirvan.paraste at gmail.com  Tue Oct 11 00:18:51 2011
From: sirvan.paraste at gmail.com (sirvan paraste)
Date: Mon, 10 Oct 2011 23:48:51 +0330
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <CAAAG0NVAnacuknJTVWD+4fXCSpNf9BMaonVfp5e_ENOvQ+11-g@mail.gmail.com>
References: <CAAAG0NVAnacuknJTVWD+4fXCSpNf9BMaonVfp5e_ENOvQ+11-g@mail.gmail.com>
Message-ID: <CAAAG0NXXHkH3tvZmmX2FTeemXAse=LAMKDmLbZo+yYsTpn=BSg@mail.gmail.com>

Hi

I am a student and I want work on freeswitch. I have read freeSwitch
documents (introduction, how to install, use default
confuguration,...).
for start of using freeswitch, I make three computer network together
with a hub, after that,I install freeswitch on one of them and X-Lite
on another computers, network checked with sharing file and ping
issue.
After registering of two X-lite phone with No. 1001,1002, they can call
each other but there is no voice !!! :(
 I am not able to find what's the problem. it is very important to me
please help me.
thank you.
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From jmesquita at freeswitch.org  Tue Oct 11 00:41:35 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 10 Oct 2011 17:41:35 -0300
Subject: [Freeswitch-users] How can make mod_rmtp support text message?
In-Reply-To: <CAAsEjANudiRxFsE+EH4BPZ3=4h7ajGz=+5X=6ppq4XNTqsnwQQ@mail.gmail.com>
References: <CAAsEjANudiRxFsE+EH4BPZ3=4h7ajGz=+5X=6ppq4XNTqsnwQQ@mail.gmail.com>
Message-ID: <CABH8QX2e=Drbg71w4ihYwKX-xS0UgjuMBnk748cqnB_3vpOTKg@mail.gmail.com>

I guess this is more of a feature request. Maybe a bounty would get it done
by someone?

Regards,
Jo?o Mesquita



2011/10/10 ??.?? <jovemark at gmail.com>

> We can find that there are two mods can bind to work together. One is
> mod_rmtp , another is mod_sms.
>
> How can make mod_rmtp work with mod_sms to support text message on flash
> client ?
>
> Anybody have suggestions?
>
> Regards.
> Martin.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hrqiang at gmail.com  Mon Oct 10 23:49:13 2011
From: hrqiang at gmail.com (RuiQiang Huang)
Date: Mon, 10 Oct 2011 12:49:13 -0700
Subject: [Freeswitch-users] Inband SPS/PPS missing because of private/public
	IP.
Message-ID: <CANN-RPStRqH71RnTXds053y9UFp1=nOwNBgw+6yhh8_cgpqdPA@mail.gmail.com>

Hi,

I tried to get video working between one Grandstream and Polycom IP phone
with H.264.
I'm planning to make freeswitch as SBC for my internal network.
However, both proxy media a d pass-thru media are getting one way video.
That is Polycom VVX can see Grandstream, not the other way.
I sniffer and found out that Polycom is doing inband SPS/PPS, so the first
packet or so is for SPS/PPS, without this, the other-side cannot decode the
video.
But freeswitch media proxy doesn't forward packets to right public address
port until it receives packets from the other stream, which will trigger a
RTP auto adjustment.
I just wondering is there any config to fix this or I have to modify the
code? Thanks.
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From edpimentl at gmail.com  Tue Oct 11 01:31:37 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Mon, 10 Oct 2011 17:31:37 -0400
Subject: [Freeswitch-users] How can make mod_rmtp support text message?
In-Reply-To: <CABH8QX2e=Drbg71w4ihYwKX-xS0UgjuMBnk748cqnB_3vpOTKg@mail.gmail.com>
References: <CAAsEjANudiRxFsE+EH4BPZ3=4h7ajGz=+5X=6ppq4XNTqsnwQQ@mail.gmail.com>
	<CABH8QX2e=Drbg71w4ihYwKX-xS0UgjuMBnk748cqnB_3vpOTKg@mail.gmail.com>
Message-ID: <CANMSB1ZNnfKyEUxPNZp=AZwtOSqvzq8wzX7TDddeOmEhDcYnqw@mail.gmail.com>

Have consider using an external RMTP? Such as the one from
http://www.themidnightcoders.com/products.html


From spencer at 5ninesolutions.com  Tue Oct 11 01:40:51 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 10 Oct 2011 14:40:51 -0700
Subject: [Freeswitch-users] Antw.:  SCA Hunt Group
In-Reply-To: <A8CDC463-70A6-4AE1-8267-7C21A41CFC71@allianz-warranty.com>
References: <A8CDC463-70A6-4AE1-8267-7C21A41CFC71@allianz-warranty.com>
Message-ID: <E842385B-FC68-45BF-8898-F7EBE5E10FA3@5ninesolutions.com>

Hi Stefan,
Thanks for your suggestion, that did the trick.

Spencer

On Oct 10, 2011, at 12:00 PM, Weigel, Stefan wrote:

> Hi Spencer,
> 
> just use mod_limit to limit one call per extension.
> 
> http://wiki.freeswitch.org/wiki/Mod_limit
> 
> 
> Best regards,
> 
> Stefan
> 
> ----- Reply message -----
> Von: "Spencer Thomason" <spencer at 5ninesolutions.com>
> Datum: Mo., Okt. 10, 2011 19:29
> Betreff: [Freeswitch-users] SCA Hunt Group
> An: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> 
> Hello all,
> I'm working on replacing a legacy key system with FreeSWITCH and a few Cisco SPA-509G phones.  I have SCA/SLA working correctly but I was wondering if anyone has a suggestion for a good way to make the inbound calls to the shared extensions "roll over" like a traditional phone system.  I have a simple lua script that does a sequential dial and checks for a busy response from the phones but the problem is that these Cisco phones have a minimum of 2 calls per line.  This results in the second call going to the same "line" as the first.  What I'd like to do is have an inbound call pick the next available shared extension.  Does anyone have any suggestions?  Is there some variable I could check with a script that would show whether the shared extension is in use?
> 
> Thanks,
> Spencer
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From adrottenberg at gmail.com  Tue Oct 11 01:57:36 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Mon, 10 Oct 2011 17:57:36 -0400
Subject: [Freeswitch-users] Conference floor given to muted member
In-Reply-To: <CAKbxfG-7FUfdmw4M-sqLtr=zR4BxEMn7tc4=r68MKYi7m5zJow@mail.gmail.com>
References: <CAC4JBEAV13j4Mjn1FYX3A2U_PBv6VD7BPFVtfFADTdEeF_+PdQ@mail.gmail.com>
	<CAKbxfG-7FUfdmw4M-sqLtr=zR4BxEMn7tc4=r68MKYi7m5zJow@mail.gmail.com>
Message-ID: <CAC4JBEAaT7vGcYHbHpqrKNAAJoi9SEJLMO_RmuXPS2k2xjT_jQ@mail.gmail.com>

Thanks for the quick response. Turns out the phones that were being
paged were configured to have a 2 second delay before answering a
page.

On Mon, Oct 10, 2011 at 3:51 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> The floor is really an aesthetic moniker. ?It's not controlling the audio etc.
> If you are not getting audio in the beginning of the call it's
> probably a NAT translation auto adjusting.
> Try adding something like answer followed by sleep 2000 in your
> dialplan before entering the conference.
>
>
> On Mon, Oct 10, 2011 at 2:38 PM, Duvid Rottenberg
> <adrottenberg at gmail.com> wrote:
>> I am setting up a conference to use for paging similar to
>> http://wiki.freeswitch.org/wiki/Lua_Intercom and I noticed that the
>> first 2 seconds of audio are not being sent. After reviewing the logs
>> it appears that the first member to enter the conference is
>> automatically given the floor event thought it was entered with the
>> mute flag. I set it up so that the originator enters the conference
>> with the moderator flag and is also unmuted, however it appears that
>> he doesn't get the floor right away, even though he's the only unmuted
>> member. It takes about 2 seconds of speech for the moderator to get
>> the floor. I have the following settings for the conference profile I
>> use:
>>
>> <param name="energy-level" value="0"/>
>> <param name="conference-flags" value="wait-mod"/>
>> <param name="moh-sound" value="silence"/>
>> <param name="moh-sound" value="silence"/>
>>
>> I tried setting <param name="member-flags" value="waste"/> but it
>> didn't help, for now I changed it so that the moderator enters the
>> conference with the wast flag.
>>
>> Are there any other settings I can use to either avoid giving a muted
>> member the floor, or to speed up the time it takes to give the floor
>> to the moderator?
>>
>> Thank You,
>> Duvid Rottenberg
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Tue Oct 11 03:30:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Oct 2011 16:30:28 -0700
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <CAAAG0NXXHkH3tvZmmX2FTeemXAse=LAMKDmLbZo+yYsTpn=BSg@mail.gmail.com>
References: <CAAAG0NVAnacuknJTVWD+4fXCSpNf9BMaonVfp5e_ENOvQ+11-g@mail.gmail.com>
	<CAAAG0NXXHkH3tvZmmX2FTeemXAse=LAMKDmLbZo+yYsTpn=BSg@mail.gmail.com>
Message-ID: <CAKzWOxU1Aqy4RhxdngcJN=oP7V3p6sUXn5xw3m5nVs9rXezSNg@mail.gmail.com>

Most likely this is a network configuration issue. The best thing to do
would be to get a hold of someone who knows basic network configuration and
debugging. Typically, the kinds of things that will cause no-way or one-way
audio are firewalls and NAT devices.

The best way to make sure that you have a working server is to set up
FreeSWITCH on a server and then connect to it from one or two other phones
that are also on the same LAN. That helps to bypass firewall/NAT issues.

Also, if you are using xlite on Windows machines you will need to open up
various ports on the Windows firewall. (For testing you can even disable the
firewall, make a test call, then re-enable the firewall.) The FS wiki
Firewall page has some specific information:

http://wiki.freeswitch.org/wiki/Firewall

Hope this helps!

-Michael

On Mon, Oct 10, 2011 at 1:18 PM, sirvan paraste <sirvan.paraste at gmail.com>wrote:

>
>
> Hi
>
> I am a student and I want work on freeswitch. I have read freeSwitch
>
> documents (introduction, how to install, use default
> confuguration,...).
> for start of using freeswitch, I make three computer network together
> with a hub, after that,I install freeswitch on one of them and X-Lite
> on another computers, network checked with sharing file and ping
> issue.
> After registering of two X-lite phone with No. 1001,1002, they can call
>
> each other but there is no voice !!! :(
>  I am not able to find what's the problem. it is very important to me
> please help me.
> thank you.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Oct 11 06:46:11 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 10 Oct 2011 22:46:11 -0400
Subject: [Freeswitch-users] How can make mod_rmtp support text message?
References: <CAAsEjANudiRxFsE+EH4BPZ3=4h7ajGz=+5X=6ppq4XNTqsnwQQ@mail.gmail.com>
	<CABH8QX2e=Drbg71w4ihYwKX-xS0UgjuMBnk748cqnB_3vpOTKg@mail.gmail.com>
Message-ID: <91326682D6DE41C3BC81C0827AB7E3E8@e1705>

maybe translate your message from Flash to SIP IM,
so you have to code a script for that
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: FreeSWITCH Users Help 
  Sent: Monday, October 10, 2011 4:41 PM
  Subject: Re: [Freeswitch-users] How can make mod_rmtp support text message?


  I guess this is more of a feature request. Maybe a bounty would get it done by someone?


  Regards,
  Jo?o Mesquita




  2011/10/10 ??.?? <jovemark at gmail.com>

    We can find that there are two mods can bind to work together. One is mod_rmtp , another is mod_sms.


    How can make mod_rmtp work with mod_sms to support text message on flash client ?


    Anybody have suggestions?


    Regards.
    Martin.


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






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



  FreeSWITCH-users mailing 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 boris at tagnet.ru  Tue Oct 11 07:12:09 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 11 Oct 2011 09:12:09 +0600
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <CALNZuwnBQixaweDEa6YFPpx4zW5_c5+zn9SFtLSEtGWvQ3CyNA@mail.gmail.com>
References: <4E8F0F50.8010201@tagnet.ru>
	<CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com>
	<CAKzWOxUCVsx1LCa1mO8AXhQ=sPKtANOCgvVu-xAmx1Py9z68EQ@mail.gmail.com>
	<CALNZuw=sBYwWGcQmT+g1B_DeT5KdZqP49sFXZ2MAYaNYvOX71g@mail.gmail.com>
	<CAKzWOxVB=pMyWoQ_-Q5+DyV3E5in53Lgo3+kzo4oZRR9BUTFBQ@mail.gmail.com>
	<CALNZuwnBQixaweDEa6YFPpx4zW5_c5+zn9SFtLSEtGWvQ3CyNA@mail.gmail.com>
Message-ID: <4E93B409.3010908@tagnet.ru>

Hello!

     No, the question was how to FAILOVER :) I need to stop processing 
dialstring if user at carrier1 is busy.

> the question was about mod_lcr.
> it returns something like:
>
> sofia/gateway/carrier1/14105780999|sofia/gateway/ 
> carrier2/14105780999|sofia/gateway/ carrier3/14105780999
> The question was how to make it when the first returns "user_busy" to not failover and try the other ones.
> -Avi
>
>
> On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org 
> <mailto:msc at freeswitch.org>> wrote:
>
>     Well, now we're talking about something more than just
>     continuing_on_fail. We'll need to look at the actual dialstring to
>     answer any more specific questions.
>
>     -MC
>
>
>     On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net
>     <mailto:avi at avimarcus.net>> wrote:
>
>         It seems from testing, and I was told on channel:
>          continue_on_fail is per bridge string, not per leg. So it
>         doesn't actually help with the question..
>         Unless I'm mistaken.
>         -Avi
>
>
>         On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins
>         <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>             You will need to specify the error conditions on which to
>             continue. See the example here:
>             http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
>             -MC
>
>             P.S. - Check out the "enumeration" column on this page to
>             get an idea of the kinds of failures you may not be aware
>             of: http://wiki.freeswitch.org/wiki/Hangup_causes
>
>
>             On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus
>             <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>
>                 I've been wondering how to do this.
>                 When I asked in channel, I was basically told this
>                 couldn't be done in a single bridge string... I hope
>                 that's not actually the case.
>                 -Avi
>
>
>                 On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko
>                 <boris at tagnet.ru <mailto:boris at tagnet.ru>> wrote:
>
>                     Hello!
>
>                         Have read docs but misundertood :( I use
>                     mod_lcr which return two
>                     or more gateways to the same destination. And when
>                     the user is busy with
>                     the first gateway the call continues with the
>                     second and so on. How may
>                     I to not continue gateways if user is busy?
>
>                     --
>                     ? ?????????,
>                       ????? ?????????
>                       ??? "??????"
>                       ???. +7 (3435) 230001
>                     <tel:%2B7%20%283435%29%20230001>
>                       ???? +7 (3435) 230005
>                     <tel:%2B7%20%283435%29%20230005>
>
>
>
>                     FreeSWITCH-users mailing list
>                     FreeSWITCH-users at lists.freeswitch.org
>                     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                     http://www.freeswitch.org
>
>
>
>
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- 
Regards,
Boris


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From infos at madovsky.org  Tue Oct 11 07:28:40 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 10 Oct 2011 23:28:40 -0400
Subject: [Freeswitch-users] continue_on_fail
References: <4E8F0F50.8010201@tagnet.ru><CALNZuwkubmbXd5eThJjhmYM7nUQNOF5BY+oiWpNQJBZN3Hn9xw@mail.gmail.com><CAKzWOxUCVsx1LCa1mO8AXhQ=sPKtANOCgvVu-xAmx1Py9z68EQ@mail.gmail.com><CALNZuw=sBYwWGcQmT+g1B_DeT5KdZqP49sFXZ2MAYaNYvOX71g@mail.gmail.com><CAKzWOxVB=pMyWoQ_-Q5+DyV3E5in53Lgo3+kzo4oZRR9BUTFBQ@mail.gmail.com><CALNZuwnBQixaweDEa6YFPpx4zW5_c5+zn9SFtLSEtGWvQ3CyNA@mail.gmail.com>
	<4E93B409.3010908@tagnet.ru>
Message-ID: <6F24574DDE5B4A8D8DC8E927FD57A265@e1705>

maybe modify continue_on_fail as a leg var on the first bridge
  ----- Original Message ----- 
  From: Boris Kovalenko 
  To: FreeSWITCH Users Help 
  Sent: Monday, October 10, 2011 11:12 PM
  Subject: Re: [Freeswitch-users] continue_on_fail


  Hello!

      No, the question was how to FAILOVER :) I need to stop processing dialstring if user at carrier1 is busy.


    the question was about mod_lcr. 
    it returns something like:


    sofia/gateway/carrier1/14105780999|sofia/gateway/ carrier2/14105780999|sofia/gateway/ carrier3/14105780999
The question was how to make it when the first returns "user_busy" to not failover and try the other ones.-Avi



    On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org> wrote:

      Well, now we're talking about something more than just continuing_on_fail. We'll need to look at the actual dialstring to answer any more specific questions. 


      -MC 



      On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:

        It seems from testing, and I was told on channel:  continue_on_fail is per bridge string, not per leg. So it doesn't actually help with the question.. 
        Unless I'm mistaken.

        -Avi



        On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org> wrote:

          You will need to specify the error conditions on which to continue. See the example here: 
          http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail


          -MC


          P.S. - Check out the "enumeration" column on this page to get an idea of the kinds of failures you may not be aware of: http://wiki.freeswitch.org/wiki/Hangup_causes 



          On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

            I've been wondering how to do this. 
            When I asked in channel, I was basically told this couldn't be done in a single bridge string... I hope that's not actually the case.

            -Avi



            On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

              Hello!

                  Have read docs but misundertood :( I use mod_lcr which return two
              or more gateways to the same destination. And when the user is busy with
              the first gateway the call continues with the second and so on. How may
              I to not continue gateways if user is busy?

              --
              ? ?????????,
                ????? ?????????
                ??? "??????"
                ???. +7 (3435) 230001
                ???? +7 (3435) 230005



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-- 
Regards,
Boris



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From joe.jflemmings at gmail.com  Tue Oct 11 12:09:59 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Tue, 11 Oct 2011 01:09:59 -0700
Subject: [Freeswitch-users] Attendant transfer Fails
Message-ID: <CALH5Zrbj77y_CnequW6Xb1SwOMPkvo9n42ozKJqJP0o38y-b0Q@mail.gmail.com>

My attendant transfer and can not seam to see eny errors. It just hangs up.
See log bellow.
Please help


2011-10-11 01:04:47.844969 [DEBUG] sofia.c:5649 Replaces: [
11968503-b9ad5388-5ff2c6b1 at 192.168.1.72
;to-tag=gU5125FNXNQ1K;from-tag=DAF258EB-4AB67790]
2011-10-11 01:04:47.844969 [NOTICE] sofia.c:5766 Attended Transfer
[e4c3df1b-f06c-41bb-972c-8bca0c454b55][21baf737-8e51-4409-9d39-3c0b1ed654cb]
2011-10-11 01:04:47.844969 [DEBUG] switch_ivr_bridge.c:1487 (sofia/internal/
sip:df45234 at 99.65.217.24:52783) State Change CS_EXCHANGE_MEDIA ->
CS_HIBERNATE
2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:1114 Send signal
sofia/internal/sip:df45234 at 99.65.217.24:52783 [BREAK]
2011-10-11 01:04:47.844969 [DEBUG] switch_ivr_bridge.c:1489 (sofia/external/
12132568954 at 10.10.58.26) State Change CS_EXCHANGE_MEDIA -> CS_HIBERNATE
2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:1114 Send signal
sofia/external/12132568954 at 10.10.58.26 [BREAK]
2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:769 Send signal
sofia/external/12132568954 at 10.10.58.26 [BREAK]
2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:769 Send signal
sofia/internal/sip:df45234 at 99.65.217.24:52783 [BREAK]
2011-10-11 01:04:47.844969 [DEBUG] sofia.c:5796 (sofia/internal/
pbx_70222_2564 at 66.113.78.36) State Change CS_EXECUTE -> CS_PARK
2011-10-11 01:04:47.844969 [DEBUG] switch_core_session.c:1114 Send signal
sofia/internal/pbx_70222_2564 at 66.113.78.36 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_play_say.c:1649 done playing
file
2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:707 Send signal
sofia/external/12132568954 at 10.10.58.26 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:329 Send signal
sofia/internal/pbx_70222_2564 at 66.113.78.36 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD
DONE [sofia/internal/pbx_70222_2564 at 66.113.78.36]
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:604 Send signal
sofia/internal/sip:df45234 at 99.65.217.24:52783 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:707 Send signal
sofia/internal/sip:df45234 at 99.65.217.24:52783 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:707 Send signal
sofia/internal/pbx_70222_2564 at 66.113.78.36 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_cpp.cpp:988 sofia/internal/
pbx_70222_2564 at 66.113.78.36 destroy/unlink session from object
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:371
(sofia/internal/pbx_70222_2564 at 66.113.78.36) State EXECUTE going to sleep
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/pbx_70222_2564 at 66.113.78.36) Running State Change CS_PARK
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:380
(sofia/internal/pbx_70222_2564 at 66.113.78.36) State PARK
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:210
sofia/internal/pbx_70222_2564 at 66.113.78.36 Standard PARK
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD
DONE [sofia/internal/sip:df45234 at 99.65.217.24:52783]
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:604 Send signal
sofia/internal/pbx_70222_2564 at 66.113.78.36 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:374
(sofia/internal/sip:df45234 at 99.65.217.24:52783) State EXCHANGE_MEDIA going
to sleep
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/sip:df45234 at 99.65.217.24:52783) Running State Change
CS_HIBERNATE
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:386
(sofia/internal/sip:df45234 at 99.65.217.24:52783) State HIBERNATE
2011-10-11 01:04:47.854974 [DEBUG] mod_sofia.c:223 sofia/internal/
sip:df45234 at 99.65.217.24:52783 SOFIA HIBERNATE
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:734 (sofia/internal/
sip:df45234 at 99.65.217.24:52783) State Change CS_HIBERNATE -> CS_RESET
2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:1114 Send signal
sofia/internal/sip:df45234 at 99.65.217.24:52783 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:386
(sofia/internal/sip:df45234 at 99.65.217.24:52783) State HIBERNATE going to
sleep
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/sip:df45234 at 99.65.217.24:52783) Running State Change
CS_RESET
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:367
(sofia/internal/sip:df45234 at 99.65.217.24:52783) State RESET
2011-10-11 01:04:47.854974 [DEBUG] mod_sofia.c:167 sofia/internal/
sip:df45234 at 99.65.217.24:52783 SOFIA RESET
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:719 sofia/internal/
sip:df45234 at 99.65.217.24:52783 CUSTOM RESET
2011-10-11 01:04:47.854974 [DEBUG] switch_ivr_bridge.c:726 (sofia/internal/
sip:df45234 at 99.65.217.24:52783) State Change CS_RESET -> CS_SOFT_EXECUTE
2011-10-11 01:04:47.854974 [DEBUG] switch_core_session.c:1114 Send signal
sofia/internal/sip:df45234 at 99.65.217.24:52783 [BREAK]
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:367
(sofia/internal/sip:df45234 at 99.65.217.24:52783) State RESET going to sleep
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/sip:df45234 at 99.65.217.24:52783) Running State Change
CS_SOFT_EXECUTE
2011-10-11 01:04:47.854974 [DEBUG] switch_core_state_machine.c:377
(sofia/internal/sip:df45234 at 99.65.217.24:52783) State SOFT_EXECUTE
2011-10-11 01:04:47.854974 [DEBUG] mod_sofia.c:560 SOFIA SOFT_EXECUTE
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From chrisbware at interfree.it  Tue Oct 11 13:17:00 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 11 Oct 2011 09:17:00 -0000
Subject: [Freeswitch-users] There is no voice in Two party call
Message-ID: <20111011091700.16181.qmail@community22.interfree.it>

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From sirvan.paraste at gmail.com  Tue Oct 11 13:35:27 2011
From: sirvan.paraste at gmail.com (sirvan paraste)
Date: Tue, 11 Oct 2011 13:05:27 +0330
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <CAKzWOxU1Aqy4RhxdngcJN=oP7V3p6sUXn5xw3m5nVs9rXezSNg@mail.gmail.com>
References: <CAAAG0NVAnacuknJTVWD+4fXCSpNf9BMaonVfp5e_ENOvQ+11-g@mail.gmail.com>
	<CAAAG0NXXHkH3tvZmmX2FTeemXAse=LAMKDmLbZo+yYsTpn=BSg@mail.gmail.com>
	<CAKzWOxU1Aqy4RhxdngcJN=oP7V3p6sUXn5xw3m5nVs9rXezSNg@mail.gmail.com>
Message-ID: <CAAAG0NW7W9TJzXdGgMUm79Uuo4DFBwHe7U6WUF8NFNFdg8b6rQ@mail.gmail.com>

Hi
I thing there is no problem with network configuration. I change firewall to
off mode but problem not solved. even i give allow to x-lite on client and
it didn't solved.

lets consider the network that I had configured it :
In the same LAN network an just using a hub-switch :

compter                          ip                     application
A                           192.168.0.1                FreeSwitch
 (server) windows
B                           192.168.0.2                X-Lite
 (client )windows
C                            192.168.0.3               X-Lite
 (client)

some available scenarios :
1- calling IVR menu (5000) and voicemail  work true.
2- calling 3000 and starting conference work true, two way voice after
connecting B and C. ( !!!)
3- B can call C and inverse C can call B .

problems  :
1- after answering call by C ( B ) , there is no-way audio .
2- the voicemail dont work some times. it says the message is to short and
then  finished. for some call it finished after many seconds.
 I think there is a problem with starting RTP , is it true?
how i can fixed it ?

thank you.


On Tue, Oct 11, 2011 at 3:00 AM, Michael Collins <msc at freeswitch.org> wrote:

> Most likely this is a network configuration issue. The best thing to do
> would be to get a hold of someone who knows basic network configuration and
> debugging. Typically, the kinds of things that will cause no-way or one-way
> audio are firewalls and NAT devices.
>
> The best way to make sure that you have a working server is to set up
> FreeSWITCH on a server and then connect to it from one or two other phones
> that are also on the same LAN. That helps to bypass firewall/NAT issues.
>
> Also, if you are using xlite on Windows machines you will need to open up
> various ports on the Windows firewall. (For testing you can even disable the
> firewall, make a test call, then re-enable the firewall.) The FS wiki
> Firewall page has some specific information:
>
> http://wiki.freeswitch.org/wiki/Firewall
>
> Hope this helps!
>
> -Michael
>
> On Mon, Oct 10, 2011 at 1:18 PM, sirvan paraste <sirvan.paraste at gmail.com>wrote:
>
>>
>>
>> Hi
>>
>> I am a student and I want work on freeswitch. I have read freeSwitch
>>
>> documents (introduction, how to install, use default
>> confuguration,...).
>> for start of using freeswitch, I make three computer network together
>> with a hub, after that,I install freeswitch on one of them and X-Lite
>> on another computers, network checked with sharing file and ping
>> issue.
>> After registering of two X-lite phone with No. 1001,1002, they can call
>>
>> each other but there is no voice !!! :(
>>  I am not able to find what's the problem. it is very important to me
>> please help me.
>> thank you.
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michaelt at callall.co.za  Tue Oct 11 15:20:10 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Tue, 11 Oct 2011 13:20:10 +0200
Subject: [Freeswitch-users] hangup error missing remote port
In-Reply-To: <CAFmdODgsp9z5FOtsXiHoE-g5vxpjhrKH_fEZBs=fwfDDSbrbkQ@mail.gmail.com>
References: <CAK=jKZJ3cvdya8i-2A-ygRx12VK3d8ZSbLkxLDxPsA3uUxj2gg@mail.gmail.com>
	<CAFmdODgsp9z5FOtsXiHoE-g5vxpjhrKH_fEZBs=fwfDDSbrbkQ@mail.gmail.com>
Message-ID: <CAFmdODi3Sjgj5qwJRvL1WJuaK7Vri1WLgasPcK0AupKxBYQm5A@mail.gmail.com>

Hi,

 Here is an example of the SDP that Freeswitch sends, I am sure it is
incorrect:

send 1119 bytes to udp/[41.76.99.7]:266 at 10:58:07.812104:
   ------------------------------------------------------------------------

   v=0
   o=FreeSWITCH 1655224844 1655224845 IN IP4 XXX.YYY.XXX
   s=FreeSWITCH
   c=IN IP4 XXX.YYY.XXX.AAA
   t=0 0
   m=audio 0 RTP/AVP 96 101
   a=rtpmap:96 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15

 ------------------------------------------------------------------------..

 What else can I send to assist, should I log a formal bug?

Thanks,

Michael

On Tue, Oct 4, 2011 at 12:51 PM, Michael Toop <michaelt at callall.co.za>wrote:

> Hi Allan,
>
>  I am having the same problem with back to back FS boxes, I am also getting
> port '0' sent in the m= SDP, in my case it is sent in the 183 SDP.
>
>  Did you get to the bottom of the issue?
>
> Thanks,
>
> Michael
>
> On Sun, Sep 4, 2011 at 6:44 AM, Allan Piske <zyryznet at gmail.com> wrote:
>
>> Hi,
>>
>> I'm having some trouble discovering the source and solution to one
>> problem.
>>
>> I setup a FS box for receiving faxes, and a already have another FS box as
>> SBC.
>>
>>
>> SPA3102   ---> opensips ---> FS_SBC --> opensips --> FS_FAX
>> this works ok, call gets answered, i hear the fax tone, the SPA3102
>> changes to T38 and everyone is happy.
>>
>>
>> Nortel CS2000   ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc)
>> ---> FS_FAX
>> but his doen't.
>>
>> This little devil CS2000 sends every invite with m=image header along with
>> m=audio. With aparently FS doesn't like.
>>
>>
>> like that:
>>
>> freeswitch at internal> recv 1215 bytes from udp/[10.143.82.250]:5060 at
>> 08:26:08.226858:
>>
>>  ------------------------------------------------------------------------
>>    INVITE sip:4730365302 at 10.143.82.253:5060;transport=UDP;user=phone
>> SIP/2.0
>>    Record-Route:
>> <sip:10.143.82.250;lr=on;ftag=-45026-10d9cdb-2668a317-10d9cdb>
>>    f: <sip:4784064435 at 10.150.65.16:5060
>> ;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb
>>    t: <sip:4730365302 at 10.143.82.250:5060;user=phone>
>>    i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481
>>    CSeq: 1 INVITE
>>    User-agent: CS2000_NGSS/9.0
>>    P-Asserted-Identity: <sip:4784064435 at 10.150.65.16;user=phone>
>>    Allow:
>> ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE
>>    Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0
>>    v: SIP/2.0/UDP
>> PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d
>>    Max-Forwards: 69
>>    m: <sip:10.150.65.16:5060;transport=UDP>
>>    k: 100rel
>>    c: application/sdp
>>    l: 420
>>
>>    v=0
>>    o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202
>>    s=-
>>    p=+1 6135555555
>>    c=IN IP4 10.152.204.202
>>    t=0 0
>>    m=audio 55920 RTP/AVP 18 8 101
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:20
>>    a=fmtp:18 annexb=no
>>    m=image 64112 udptl t38
>>    a=T38FaxVersion:0
>>    a=T38FaxMaxBuffer:1100
>>    a=T38FaxMaxDatagram:612
>>    a=T38MaxBitRate:14400
>>    a=T38FaxRateManagement:transferredTCF
>>    a=T38FaxUdpEC:t38UDPRedundancy
>>
>> But then we have a problem ...
>>
>>
>>
>>
>> My beloved FS_SBC changes the SDP before sending it to the next hop .. as
>> it should becase they are on diferent subnets and set to proxy_media
>>
>> 2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302
>> Patched SDP
>> ---
>> v=0
>> o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202
>> s=-
>> p=+1 6135555555
>> c=IN IP4 10.152.204.202
>> t=0 0
>> m=audio 55920 RTP/AVP 18 8 101
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>> m=image 64112 udptl t38
>> a=T38FaxVersion:0
>> a=T38FaxMaxBuffer:1100
>> a=T38FaxMaxDatagram:612
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> +++
>> v=0
>> o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D
>> s=FreeSWITCH
>> p=+1 6135555555
>> c=IN IP4 A.B.C.D
>> t=0 0
>> m=audio 19262 RTP/AVP 18 8 101
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>> m=image 19262 udptl t38
>> a=T38FaxVersion:0
>> a=T38FaxMaxBuffer:1100
>> a=T38FaxMaxDatagram:612
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>>
>> so we have now  2  m=   fields with the same port !!!!!
>>
>> then when it gets answered on the FS_FAX box .. we got disconnection on
>> the FS_SBC box
>>
>>
>> 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel
>> sofia/tpa/4730365302 entering state [completing][200]
>> 2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP:
>> v=0
>> o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19
>> s=FreeSWITCH
>> c=IN IP4 189.45.192.19
>> t=0 0
>> m=audio 31436 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> m=image 0 udptl 19
>>
>> 2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel
>> sofia/tpa/4730365302 entering state [ready][200]
>> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830
>> (sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE
>> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal
>> sofia/voxip/4784064435 at 10.150.65.16:5060 [BREAK]
>> 2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel
>> [sofia/tpa/4730365302] has been answered
>> 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec
>> sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits
>> 2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP
>> [sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20
>> 2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS
>> ERROR: [Missing remote port]
>> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563
>> (sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP
>> 2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup
>> sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
>> 2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal
>> sofia/tpa/4730365302 [KILL]
>> 2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/tpa/4730365302 [BREAK]
>>  2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate
>> Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed.
>>  Cause: DESTINATION_OUT_OF_ORDER
>>
>>
>> port = 0 !!!
>>
>> is that maybe a fault in opensips in the middle because the both m= fields
>> with the same port ? , or one of the FS boxes ?
>> something is broken here, sadly :(
>>
>> Appreciate any help.
>>
>> Allan
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Tue Oct 11 17:01:37 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 11 Oct 2011 15:01:37 +0200
Subject: [Freeswitch-users] Problem with Sangoma D100 Transconding card
In-Reply-To: <1317371069.8959.15.camel@marces.madrid.commsmundi.com>
References: <1317116551.12197.21.camel@marces.madrid.commsmundi.com>
	<CAA4nhyBM0cRr472xe+-XZ_2fsZieP1cWWGPGgnRbPc_qHoscsA@mail.gmail.com>
	<1317289483.4860.20.camel@marces.madrid.commsmundi.com>
	<CAA4nhyCdR9mbXS=_q3UN2Hnrm+M=eySBwN3+31-4epTbjHYbmw@mail.gmail.com>
	<1317371069.8959.15.camel@marces.madrid.commsmundi.com>
Message-ID: <1318338097.9263.4.camel@marces.madrid.commsmundi.com>

Hi,

i couldn't reproduce the error using the last freeswitch git version.

I try to kill freeswitch, stop/start / restart , i did a complete
restart of the daemon sngtc_server, and every time that i call it work
nice.

For now is perfect!



On Fri, 2011-09-30 at 10:24 +0200, Antonio wrote:

> Hi Moises,
> 
> the card is in a produce system and i can execute as many tests as i
> wish....
> 
> I'm suspecting that is when i freeswitch stops, and then start it.
> Once you place a call with g729 it doesn't work.
> 
> Next monday i will have the opportunity to do more extensive tests in
> this server.
> 
> I post here the configuration and results.
> 
> 
> 
> On Fri, 2011-09-30 at 01:15 -0400, Moises Silva wrote: 
> 
> > On Thu, Sep 29, 2011 at 5:44 AM, Antonio <asilva at wirelessmundi.com>
> > wrote:
> > 
> >         Hi,
> >         
> >         The log when this error occurs can be found in
> >         http://pastebin.freeswitch.org/17431 . 
> >         
> >         
> > 
> > 
> > 
> > Can you reproduce this problem easily?
> > 
> > Moises Silva
> > Senior Software Engineer, Software Development Manager
> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> > L3R 9R6 Canada
> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> > 
> > 
> > 
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> -- 
> 
> Un cordial saludo / Best regards, 
> 
>  _________________________
> 
> Ant?nio Silva
> 
> E-mail:asilva at wirelessmundi.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 coolshoulder at gmail.com  Tue Oct 11 18:59:22 2011
From: coolshoulder at gmail.com (Alex Chan)
Date: Tue, 11 Oct 2011 22:59:22 +0800
Subject: [Freeswitch-users] outbound call with spa-3000 behind nat how?
Message-ID: <CAAo94TWs4Fz39qJTk0L4_wCDQZY7gFr7ms13COEgvUh=PtQrPg@mail.gmail.com>

Hi folks,

I want to use a spa-3000 with freeswitch setup. The freeswitch box is
on the internet, and the spa-3000 is used at home which access the
internet via ADSL connection behind nat.

I have follow the instruction as link below:

http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo

Inbound call from PSTN just works. The problem is make a outbound call
to PSTN via spa-3000.

This line did not make any sense in my scenarios

<action application="bridge"
data="sofia/internal/${destination_number}@[IP address of
SPA3102]:5061" />

as:

1. the WAN ip address is dynamic (ADSL)
2. the port is dynamic too (behind nat)

so please advice.

Thanks.


-- 

Alex Chan


From max.clark at gmail.com  Tue Oct 11 23:36:43 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 11 Oct 2011 12:36:43 -0700
Subject: [Freeswitch-users] Multiple Customer Accountcode
Message-ID: <CAAaeqWcs_5r1m9Be4O53ZcFXDgxV05zC=eoOpJj4biC6YEXb4A@mail.gmail.com>

FreeSWITCH uses accountcode to specify the billing customer for a
given call. This is simple enough when there is only one customer in
the equation:

Customer -> A-leg -> FS -> B-leg -> PSTN
PSTN -> A-leg -> FS -> B-leg -> Customer

I'm curious on thoughts when two customers are involved:

Customer 1 -> A-leg -> FS -> B-leg -> Customer 2

Who do you set the accountcode to and why?

Thanks in advance,
-Max


From masood24 at gmail.com  Tue Oct 11 23:41:33 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Tue, 11 Oct 2011 22:41:33 +0300
Subject: [Freeswitch-users] How to Register using radiusclient digest
	authentication
Message-ID: <CAMd3iJJLCGtCmYoArSzfv2oySnWEVgUksAWRWhx8Mk5MgK3bjQ@mail.gmail.com>

Dear ,

I had successfully installed and configured radiusclient  but now as I am
newbie I really do not know that what next should i do to get users
registered on my freeswitch using digest authentication using my radius
server,
Like I have my radsius server and billing installed on other and working
with other system. Now from where should i begin shouldd i change something
in directory or in dial plans or in sofia.conf that whenever any user hits
to get register freeswitch should trigger the radius client to send request
to the server and if gets a return code 0 it should register the client on
the basis of his username and password.
I know i may be asking too much but I really am confused at this point that
from where should i beging and which file to edit or start editing.
Help would be appreciated.

-- 
*Regards,
Masood Ahmed
*
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From max.clark at gmail.com  Tue Oct 11 23:41:43 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 11 Oct 2011 12:41:43 -0700
Subject: [Freeswitch-users] MOD_LCR: IntraState and IntraLata Confusion.
In-Reply-To: <CAFoaFX6aaHJKNsfM48OhgRBp5n8jLLiZ2wzNQrCP3ZAChj_gdA@mail.gmail.com>
References: <CAFoaFX6aaHJKNsfM48OhgRBp5n8jLLiZ2wzNQrCP3ZAChj_gdA@mail.gmail.com>
Message-ID: <CAAaeqWeWx-KCzSUkK2-JSHPbZyAYRJAcYJt_jVHAS9dZv63REA@mail.gmail.com>

Alan,

mod_lcr performs a series of tests against the database when you load
the module. I worked around this by dropping the intralata column from
the database. Granted this is with a much older version of the module,
but you might want to try the same for your installation.

-Max

On Sun, Oct 2, 2011 at 9:02 PM, Alan Frisch <frisch.alan at gmail.com> wrote:
> Been cutting my teeth on Mod_LCR today with inter/intra state costs and have
> run into a little issue that I am sure one of the refined experts can help
> with.
>
> Now if I have a rate list that is like:
>
> NPANXX,INTERSTATE,INTRASTATE
> 212203,0.00738,0.00716
> ...
> 203302,0.00604,0.00716
>
> I assume I should load the INTRASTATE into the intrastate_rate MySQL field
> (of course).... and would I then load the INTERSTATE into the rate field?
> What do I do about the intralata_rate field?
>
> If I leave intralata_rate 0 or NULL, then LCR fails.? I only want to rate on
> inter/intra-state and not on LATA.? If tried setting the intralata_rate to
> be the same as the interstate_rate which works fine in most examples, but
> with the above data (one number being in NY and the other in CT) the LCR
> selects an intralata rate instead of an interstate rate, due to the common
> lata (though the states are different).
>
> How can I get Mod_LCR to rate only by state?? I am just playing with the LCR
> command in CLI and not in the dialplan yet.
>
> Thanks.
>
> AF.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Oct 11 23:46:08 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 11 Oct 2011 21:46:08 +0200
Subject: [Freeswitch-users] Multiple Customer Accountcode
In-Reply-To: <CAAaeqWcs_5r1m9Be4O53ZcFXDgxV05zC=eoOpJj4biC6YEXb4A@mail.gmail.com>
References: <CAAaeqWcs_5r1m9Be4O53ZcFXDgxV05zC=eoOpJj4biC6YEXb4A@mail.gmail.com>
Message-ID: <CALNZuwkZwA=4UxjbeMnRnnh+mQ5QMK7FLqw=T5pSSjs_fY=QJg@mail.gmail.com>

You set the account code on the relevant leg. And then bill based on each
leg, rather than the call.

-Avi

On Tue, Oct 11, 2011 at 9:36 PM, Max Clark <max.clark at gmail.com> wrote:

> FreeSWITCH uses accountcode to specify the billing customer for a
> given call. This is simple enough when there is only one customer in
> the equation:
>
> Customer -> A-leg -> FS -> B-leg -> PSTN
> PSTN -> A-leg -> FS -> B-leg -> Customer
>
> I'm curious on thoughts when two customers are involved:
>
> Customer 1 -> A-leg -> FS -> B-leg -> Customer 2
>
> Who do you set the accountcode to and why?
>
> Thanks in advance,
> -Max
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From spencer at 5ninesolutions.com  Tue Oct 11 23:47:55 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 11 Oct 2011 12:47:55 -0700
Subject: [Freeswitch-users] Cisco SPA509G Blf Issues
Message-ID: <C8A8B32F-2955-4AF6-9199-19B904D899ED@5ninesolutions.com>

Hello all,
I have several Cisco SPA509G phones, most without the SPA500 expansion module, but a few with.  I'm having trouble getting the BLFs to work reliably.  Specifically when a user hangs up the BLF continues to show them on the phone.  Currently the server type is set to RFC3265_4235.  The phones have the option for Asterisk, Broadsoft, and Sylantro as well.  What's the difference?  The phones are all running firmware v 7.4.9a.  Freeswitch is on a public ip and all of the phones are behind NAT, but all other signaling is working very reliably.  Any help would be greatly appreciated!

Thanks,
Spencer




From max.clark at gmail.com  Wed Oct 12 00:28:50 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 11 Oct 2011 13:28:50 -0700
Subject: [Freeswitch-users] Multiple Customer Accountcode
In-Reply-To: <CALNZuwkZwA=4UxjbeMnRnnh+mQ5QMK7FLqw=T5pSSjs_fY=QJg@mail.gmail.com>
References: <CAAaeqWcs_5r1m9Be4O53ZcFXDgxV05zC=eoOpJj4biC6YEXb4A@mail.gmail.com>
	<CALNZuwkZwA=4UxjbeMnRnnh+mQ5QMK7FLqw=T5pSSjs_fY=QJg@mail.gmail.com>
Message-ID: <CAAaeqWdO3ZHDYefE5wMMCEJUD7mdfuqoit6yG=M1391MFswNdg@mail.gmail.com>

With mod_log_csv how would I split these calls?

On Tue, Oct 11, 2011 at 12:46 PM, Avi Marcus <avi at avimarcus.net> wrote:
> You set the account code on the relevant leg. And then bill based on each
> leg, rather than the call.
> -Avi
> On Tue, Oct 11, 2011 at 9:36 PM, Max Clark <max.clark at gmail.com> wrote:
>>
>> FreeSWITCH uses accountcode to specify the billing customer for a
>> given call. This is simple enough when there is only one customer in
>> the equation:
>>
>> Customer -> A-leg -> FS -> B-leg -> PSTN
>> PSTN -> A-leg -> FS -> B-leg -> Customer
>>
>> I'm curious on thoughts when two customers are involved:
>>
>> Customer 1 -> A-leg -> FS -> B-leg -> Customer 2
>>
>> Who do you set the accountcode to and why?
>>
>> Thanks in advance,
>> -Max
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From joe.jflemmings at gmail.com  Wed Oct 12 01:29:33 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Tue, 11 Oct 2011 14:29:33 -0700
Subject: [Freeswitch-users] Call/Transaction Does Not Exist
Message-ID: <CALH5ZrbFUBNMrBEwN+uJMO+ee0R=X0=g00rUi6+jPuOT=Y17SQ@mail.gmail.com>

When i call from one extension and hangup, the other call does not hangup
and continues ringing. Please see log bellow.

Any ideas?



U 10.10.78.24:55479 -> 10.10.78.36:5060
  INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
  <sip:213 at 10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: <
sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MES
  SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported:
100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards:
  70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN IP4
10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
  101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..
#
U 10.10.78.36:5060 -> 10.10.78.24:52732
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From:
"Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <
sip:213 at 10.10.78.36;user=phone>
  ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1
INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
22-51-43 -0500..Content-Length: 0....
#
U 10.10.78.36:5060 -> 10.10.78.24:52732
  SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 6
  6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 INVITE..User-Agent:
FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Accept:
appl
  ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
precondition, path, replaces..Allow-Events: talk, hold, prese
  nce, dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer..Proxy-Authenticate: Digest
realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac
  ", algorithm=MD5, qop="auth"..Content-Length: 0....
#
U 10.10.78.24:55479 -> 10.10.78.36:5060
  ACK sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <si
  p:213 at 10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 ACK..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: <
sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL, OPTION
  S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
en..Max-Forwards: 70..Content-Length: 0....
#
U 10.10.78.24:55479 -> 10.10.78.36:5060
  INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
  <sip:213 at 10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: <
sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MES
  SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported:
100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz
  ation: Digest username="pbxjker_2564", realm="10.10.78.36",
nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth,
cnonce="KnfhF00Oi0GfJCi", nc=00000001, uri="sip:213 at 10.10.78.36:5060
;user=phone",
  response="82a75edb7ba97bfe7dea585684701d0c", algorithm=MD5..Max-Forwards:
70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone
  ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
  telephone-event/8000..
#
U 10.10.78.36:5060 -> 10.10.78.24:52732
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From:
"Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <
sip:213 at 10.10.78.36;user=phone>
  ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
22-51-43 -0500..Content-Length: 0....
#
U 10.10.78.36:5060 -> 10.10.78.24:52783
  INVITE sip:pbxjker_213 at 10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP
10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: 69..From:
"Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81
  rv8Fp..To: <sip:pbxjker_213 at 10.10.78.24:52783>..Call-ID:
4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact: <
sip:mod_sofia at 10.10.78.36:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.
  head-git-1086cba 2011-05-23 22-51-43 -0500..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-Type: application/sdp..Content-Disposition: session..
  Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID:
"Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
1318333156 1318333157 IN IP4
   10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio 29124
RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98
bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320
  00..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=ptime:20..
#
U 10.10.78.24:55516 -> 10.10.78.36:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
<sip:pbxjker_213 at 99.65.2
  17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: <
sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
PolycomSoundPointIP-SPIP_320-UA/3.2
  .2.0477..Accept-Language: en..Content-Length: 0....
#
U 10.10.78.24:55516 -> 10.10.78.36:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
<sip:pbxjker_213 at 99.65.
  217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: <
sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
PolycomSoundPointIP-SPIP_320-UA/3.
  2.2.0477..Allow-Events: talk,hold,conference..Accept-Language:
en..Content-Length: 0....
#
U 10.10.78.36:5060 -> 10.10.78.24:52732
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From:
"Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <
sip:213 at 10.10.78.36;user=phone
  >;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq:
2 INVITE..Contact: <sip:213 at 10.10.78.36:5060;transport=udp>..User-Agent:
FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2
  3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE,
CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE..Supported: timer, precondition, path, replaces..A
  llow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary,
refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker
  _213>;party=calling;privacy=off;screen=no....
#
U 10.10.78.24:55479 -> 10.10.78.36:5060
  CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
  13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: <
sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
INFO, MESSAGE, SUBSCR
  IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
username="pbxjker_2564", realm="10.10.78.36",
nonce="ab4a9421-fbb9-41fd-943d-462
  89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
uri="sip:213 at 10.10.78.36:5060;user=phone",
response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
70..Content-Length: 0..
  ..
#
U 10.10.78.36:5060 -> 10.10.78.24:5060
  SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
  3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
0....
#
U 10.10.78.24:55479 -> 10.10.78.36:5060
  CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
  13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: <
sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
INFO, MESSAGE, SUBSCR
  IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
username="pbxjker_2564", realm="10.10.78.36",
nonce="ab4a9421-fbb9-41fd-943d-462
  89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
uri="sip:213 at 10.10.78.36:5060;user=phone",
response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
70..Content-Length: 0..
  ..
#
U 10.10.78.36:5060 -> 10.10.78.24:5060
  SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
  3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
0....
#
U 10.10.78.24:55479 -> 10.10.78.36:5060
  CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
  13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: <
sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
INFO, MESSAGE, SUBSCR
  IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
username="pbxjker_2564", realm="10.10.78.36",
nonce="ab4a9421-fbb9-41fd-943d-462
  89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
uri="sip:213 at 10.10.78.36:5060;user=phone",
response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
70..Content-Length: 0..
  ..
#
U 10.10.78.36:5060 -> 10.10.78.24:5060
  SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <
sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
  3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
0....
#
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From sirvan.paraste at gmail.com  Wed Oct 12 01:50:25 2011
From: sirvan.paraste at gmail.com (sirvan paraste)
Date: Wed, 12 Oct 2011 01:20:25 +0330
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <CAAAG0NW7W9TJzXdGgMUm79Uuo4DFBwHe7U6WUF8NFNFdg8b6rQ@mail.gmail.com>
References: <CAAAG0NVAnacuknJTVWD+4fXCSpNf9BMaonVfp5e_ENOvQ+11-g@mail.gmail.com>
	<CAAAG0NXXHkH3tvZmmX2FTeemXAse=LAMKDmLbZo+yYsTpn=BSg@mail.gmail.com>
	<CAKzWOxU1Aqy4RhxdngcJN=oP7V3p6sUXn5xw3m5nVs9rXezSNg@mail.gmail.com>
	<CAAAG0NW7W9TJzXdGgMUm79Uuo4DFBwHe7U6WUF8NFNFdg8b6rQ@mail.gmail.com>
Message-ID: <CAAAG0NV5ifFWOE3psag8Zr4yZ3JrGOkFzQE9zhY1vj1Mf7GCfg@mail.gmail.com>

Hi
I thing there is no problem with network configuration. I change firewall to
off mode but problem not solved. even i give allow to x-lite on client and
it didn't solved.

lets consider the network that I had configured it :
In the same LAN network an just using a hub-switch :

compter                          ip                     application
A                           192.168.0.1                FreeSwitch
 (server) windows
B                           192.168.0.2                X-Lite
 (client )windows
C                            192.168.0.3               X-Lite
 (client)

some available scenarios :
1- calling IVR menu (5000) and voicemail  work true.
2- calling 3000 and starting conference work true, two way voice after
connecting B and C. ( !!!)
3- B can call C and inverse C can call B .

problems  :
1- after answering call by C ( B ) , there is no-way audio .
2- the voicemail dont work some times. it says the message is to short and
then  finished. for some call it finished after many seconds.
 I think there is a problem with starting RTP , is it true?
how i can fixed it ?

thank you.


On Tue, Oct 11, 2011 at 3:00 AM, Michael Collins <msc at freeswitch.org> wrote:

> Most likely this is a network configuration issue. The best thing to do
> would be to get a hold of someone who knows basic network configuration and
> debugging. Typically, the kinds of things that will cause no-way or one-way
> audio are firewalls and NAT devices.
>
> The best way to make sure that you have a working server is to set up
> FreeSWITCH on a server and then connect to it from one or two other phones
> that are also on the same LAN. That helps to bypass firewall/NAT issues.
>
> Also, if you are using xlite on Windows machines you will need to open up
> various ports on the Windows firewall. (For testing you can even disable the
> firewall, make a test call, then re-enable the firewall.) The FS wiki
> Firewall page has some specific information:
>
> http://wiki.freeswitch.org/wiki/Firewall
>
> Hope this helps!
>
> -Michael
>
>  On Mon, Oct 10, 2011 at 1:18 PM, sirvan paraste <sirvan.paraste at gmail.com
> > wrote:
>
>>
>>
>> Hi
>>
>> I am a student and I want work on freeswitch. I have read freeSwitch
>>
>> documents (introduction, how to install, use default
>> confuguration,...).
>> for start of using freeswitch, I make three computer network together
>> with a hub, after that,I install freeswitch on one of them and X-Lite
>> on another computers, network checked with sharing file and ping
>> issue.
>> After registering of two X-lite phone with No. 1001,1002, they can call
>>
>> each other but there is no voice !!! :(
>>  I am not able to find what's the problem. it is very important to me
>> please help me.
>> thank you.
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nmenoni at glotweb.com  Tue Oct 11 16:59:31 2011
From: nmenoni at glotweb.com (=?ISO-8859-1?Q?Nicol=E1s_Menoni?=)
Date: Tue, 11 Oct 2011 10:59:31 -0200
Subject: [Freeswitch-users] portaudio with 5.1 sound card
Message-ID: <CALS_U7jHe3M_UzK5siWJK+jB9meBs9QTMw6qxgbPyo0sZFZGUA@mail.gmail.com>

Hello!

I have a 5.1 sound card. The question is: Can I use the Front, Rear and
Center channels for independents streams with mod_portaudio?
I have been trying but I couldn't find the way!

Thank you from Uruguay!
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From fssupport at highergear.com  Tue Oct 11 23:26:53 2011
From: fssupport at highergear.com (fssupport at highergear.com)
Date: Tue, 11 Oct 2011 14:26:53 -0500 (CDT)
Subject: [Freeswitch-users] PRI Passthrough
Message-ID: <64552.75.27.110.246.1318361213.squirrel@webmail.highergear.com>

I'm in the process of building a freeswitch box to replace our existing
Mitel PBX setup.  I was hoping I could configure the freeswitch box with a
dual port PRI card, such Sangoma A102, plug my PRI into it, then use the
other port to connect to the Mite's PRI jack.  That way I can test the new
system, but still have the Mitel online.  Does anyone have a link to how
to do this type of configuration?



From olimonkey at gmail.com  Tue Oct 11 18:23:01 2011
From: olimonkey at gmail.com (Oliver Schenk)
Date: Tue, 11 Oct 2011 22:23:01 +0800
Subject: [Freeswitch-users] CISCO FXO - IVR
Message-ID: <CADVbj_hA1xJNU77tqnbO3DdgNDmKB1_v_e1ODgBGjg1SJuvR7A@mail.gmail.com>

Hi All,


I have posted here some months back regarding interfacing CISCO with
FreeSWITCH. I have had basic success, however there is quite a lot of
fine tuning to do which is beyond my general knowledge or what I can
find via Google.


Currently I can do the following:


1. Make a call from any phone to a specific phone number. This phone
number ends up at the CISCO FXO card. This then redirects to a
specific extension at the FreeSwitch box. The dialing plan then causes
a managed C# .NET application to start playing.

2. In the other direction, I have a windows service that monitors a
given database. If any alarm conditions occur it uses ESL and the
?Originate? API call to create a connection to the CISCO. The CISCO is
then configured to redirect the given number back through the FXO card
via plain old telephone (POTS).

So, I can successfully make incoming and outgoing connnections on demand.




PROBLEMS:

1. Using ESL for outgoing call, I don?t know how to determine whether
a) the phone on the far end was picked up,   b) whether the phone on
the far end just rang out   c) whether it was busy. In all cases the
IVR just starts playing right away even if I don?t even pick up the
phone. (I THINK THAT HAS SOMETHING TO DO WITH CISCO CONFIG). It seems
that FreeSwitch simply triggers the CISCO to make a call, but then has
no ability to track progress and assumes a connection was made. How do
I get POTS to talk to VOIP/SIP to talk to FreeSwitch?

2. Incoming calls simply come in as ?Processing unknown
<0000000000>->1024 in context public. The actual incoming phone number
does not seem to be getting from the CISCO to the FreeSwitch box.

3. I have 2 lines. I was simply going to assign extension 1024 for
line 1 and 1025 for line 2. But for outgoing calls from my managed
code how can I use ESL to say:

a. Is line 1 busy? Then use Line 2 else make call.

b. Is line 2 busy? Then wait for next scan else make call.






My Code in the outgoing service:

ESLconnection eslConection = new ESLconnection(?x.x.x.x?, ?8021?, ?password?);

If (eslConnection.Connected() == ESL_SUCCESS)
{
    eslConnection.Api(?originate?, ?sofia/internal/0? + phoneNumber +
?@? + CISCO_IP + ? ?&managed(xNetIVR)??);

    // determine if call is ringing, busy or picked up here
    // no idea how to do that?
}






Relevant CISCO config:


version 12.4
!
!
!
voice service pots
fax rate disable
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
sip
registrar server expires max 3600 min 3600
  no update-callerid
  no call service stop
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice register global
mode cme
source-address <CISCO IP> port 5060
max-dn 10
max-pool 10
authenticate register
hold-alert
tftp-path flash:
!
!
!
voice-card 0
dsp services dspfarm
!
!
!
voice-port 0/3/0
output attenuation -3
no comfort-noise
cptone AU
connection plar opx 1024
impedance complex1
caller-id enable
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
!
!
!
!
dial-peer voice 10 pots
incoming called-number .T
no digit-strip
direct-inward-dial
port 0/3/0
forward-digits all
!
dial-peer voice 100 pots
preference 1
destination-pattern 0T
no digit-strip
port 0/3/0
forward-digits all
!
dial-peer voice 1 voip
destination-pattern 1024
session protocol sipv2
session target ipv4:<FreeSwitch IP>
session transport tcp
dtmf-relay rtp-nte
codec g711alaw
no vad
!
!
sip-ua
presence enable
!
!




Relevant FreeSwitch config:


conf\dialplan\public\00_xNetIVR.xml and conf\dialplan\default\00_xNetIVR.xml:

<include>
  <extension name="testapp">
  <condition field="destination_number" expression="^(1024)">
     <action application="managed" data="xNetIVR"/>
  </condition>
</extension>
</include>





conf\directory\default\1024.xml

<include>
  <user id="1024">
    <params>
      <param name="password" value="xxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1024"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1024"/>
      <variable name="effective_caller_id_number" value="1024"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
    </variables>
  </user>
</include>




Public.xml and default.xml dial plans are basically the same as
default installation.







Cheers,



Oliver Schenk


From curriegrad2004 at gmail.com  Wed Oct 12 04:37:46 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 11 Oct 2011 17:37:46 -0700
Subject: [Freeswitch-users] Multiple Customer Accountcode
In-Reply-To: <CAAaeqWdO3ZHDYefE5wMMCEJUD7mdfuqoit6yG=M1391MFswNdg@mail.gmail.com>
References: <CAAaeqWcs_5r1m9Be4O53ZcFXDgxV05zC=eoOpJj4biC6YEXb4A@mail.gmail.com>
	<CALNZuwkZwA=4UxjbeMnRnnh+mQ5QMK7FLqw=T5pSSjs_fY=QJg@mail.gmail.com>
	<CAAaeqWdO3ZHDYefE5wMMCEJUD7mdfuqoit6yG=M1391MFswNdg@mail.gmail.com>
Message-ID: <CAMWiMy9hoesExtNfaBeBk-+r5kE_bOFZGg22ujh-QGdSYWZEpw@mail.gmail.com>

I personally would not reccommend using mod_log_csv, but instead would
reccomend using a SQL like solution where mod_xml_cdr parses all the
relevant information and stores it on a database so it can be queried
later to retrieve those details

On Tue, Oct 11, 2011 at 1:28 PM, Max Clark <max.clark at gmail.com> wrote:
> With mod_log_csv how would I split these calls?
>
> On Tue, Oct 11, 2011 at 12:46 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> You set the account code on the relevant leg. And then bill based on each
>> leg, rather than the call.
>> -Avi
>> On Tue, Oct 11, 2011 at 9:36 PM, Max Clark <max.clark at gmail.com> wrote:
>>>
>>> FreeSWITCH uses accountcode to specify the billing customer for a
>>> given call. This is simple enough when there is only one customer in
>>> the equation:
>>>
>>> Customer -> A-leg -> FS -> B-leg -> PSTN
>>> PSTN -> A-leg -> FS -> B-leg -> Customer
>>>
>>> I'm curious on thoughts when two customers are involved:
>>>
>>> Customer 1 -> A-leg -> FS -> B-leg -> Customer 2
>>>
>>> Who do you set the accountcode to and why?
>>>
>>> Thanks in advance,
>>> -Max
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 hrqiang at gmail.com  Wed Oct 12 05:47:19 2011
From: hrqiang at gmail.com (RuiQiang Huang)
Date: Tue, 11 Oct 2011 18:47:19 -0700
Subject: [Freeswitch-users] How to define extension capability in Media
	Proxy mode.
Message-ID: <CANN-RPTDHmg8wK60jO2GR6sAV4=dQS0OFXq2sPCAvOrAR0cLzg@mail.gmail.com>

Hi,

Some of my endpoint support G.722, while others doesn't.
My freeswitch is working at media proxy mode. Is there anyway to define
codec capability for a particular extension? Thanks.
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From spencer at 5ninesolutions.com  Wed Oct 12 06:51:36 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 11 Oct 2011 19:51:36 -0700
Subject: [Freeswitch-users] Cisco SPA509G Blf Issues
In-Reply-To: <C8A8B32F-2955-4AF6-9199-19B904D899ED@5ninesolutions.com>
References: <C8A8B32F-2955-4AF6-9199-19B904D899ED@5ninesolutions.com>
Message-ID: <4842967D-527B-406A-BAF4-32F3185C6179@5ninesolutions.com>

I apologize for including an except of the log here, but in troubleshooting this I noticed there is a slightly more than 50 second delay between the time the channel is hung up and the time the NOTIFY is sent to the phone.  Anyone have any ideas as to why this isn't sent when the channel is hung up?

If need be I can pastebin complete logs but there is a lot to sift through.

2011-10-11 21:01:51.446883 [NOTICE] switch_ivr_bridge.c:658 Hangup sofia/external/18004359792 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]

2011-10-11 21:02:41.760065 [WARNING] sofia_presence.c:1834 send payload:
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="246" state="partial" entity="sip:2001 at 206.125.40.179">
</dialog-info>


On Oct 11, 2011, at 12:47 PM, Spencer Thomason wrote:

> Hello all,
> I have several Cisco SPA509G phones, most without the SPA500 expansion module, but a few with.  I'm having trouble getting the BLFs to work reliably.  Specifically when a user hangs up the BLF continues to show them on the phone.  Currently the server type is set to RFC3265_4235.  The phones have the option for Asterisk, Broadsoft, and Sylantro as well.  What's the difference?  The phones are all running firmware v 7.4.9a.  Freeswitch is on a public ip and all of the phones are behind NAT, but all other signaling is working very reliably.  Any help would be greatly appreciated!
> 
> Thanks,
> Spencer
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 hrqiang at gmail.com  Wed Oct 12 08:52:50 2011
From: hrqiang at gmail.com (RuiQiang Huang)
Date: Tue, 11 Oct 2011 21:52:50 -0700
Subject: [Freeswitch-users] bypass media and media proxy
Message-ID: <CANN-RPTw=J=sCJjSGL8hsGxvgAGkAM60FVLnDN3Ptdk_Qz10EQ@mail.gmail.com>

Hi,

I want my server sometimes does media proxy and sometimes bypass media.
For example, I have A, B and C.
A and B always in same internal network. C is outside.
When A call B, I want to bypass media, so that media peer to peer.
A call C will be media proxy.
I know ICE can do that. But unfortunately, all my endpoint doesn't support
ICE.
I'm wondering is there anyway to configure this. Thanks
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From hrqiang at gmail.com  Wed Oct 12 08:53:57 2011
From: hrqiang at gmail.com (RuiQiang Huang)
Date: Tue, 11 Oct 2011 21:53:57 -0700
Subject: [Freeswitch-users] bypass media and media proxy
In-Reply-To: <CANN-RPTw=J=sCJjSGL8hsGxvgAGkAM60FVLnDN3Ptdk_Qz10EQ@mail.gmail.com>
References: <CANN-RPTw=J=sCJjSGL8hsGxvgAGkAM60FVLnDN3Ptdk_Qz10EQ@mail.gmail.com>
Message-ID: <CANN-RPSvm0GxMC3i375mH3u5jgH=Z=+tLiJGq0Uj3g3c8gn7ug@mail.gmail.com>

The reason, I need this is I'm building a video app with freeswitch, so
bandwidth is very expensive for me.

On Tue, Oct 11, 2011 at 9:52 PM, RuiQiang Huang <hrqiang at gmail.com> wrote:

> Hi,
>
> I want my server sometimes does media proxy and sometimes bypass media.
> For example, I have A, B and C.
> A and B always in same internal network. C is outside.
> When A call B, I want to bypass media, so that media peer to peer.
> A call C will be media proxy.
> I know ICE can do that. But unfortunately, all my endpoint doesn't support
> ICE.
> I'm wondering is there anyway to configure this. Thanks
>
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From krice at freeswitch.org  Wed Oct 12 08:54:58 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 11 Oct 2011 23:54:58 -0500
Subject: [Freeswitch-users] bypass media and media proxy
In-Reply-To: <CANN-RPTw=J=sCJjSGL8hsGxvgAGkAM60FVLnDN3Ptdk_Qz10EQ@mail.gmail.com>
Message-ID: <CABA87D2.1F182%krice@freeswitch.org>

Late negotiation... Its on the wiki


On 10/11/11 11:52 PM, "RuiQiang Huang" <hrqiang at gmail.com> wrote:

> Hi,?
> 
> I want my server sometimes does media proxy and sometimes bypass media.
> For example, I have A, B and C.
> A and B always in same internal network. C is outside.
> When A call B, I want to bypass media, so that media peer to peer.
> A call C will be media proxy.
> I know ICE can do that. But?unfortunately, all my endpoint doesn't support
> ICE.
> I'm wondering is there anyway to configure this. Thanks
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sirvan.paraste at gmail.com  Wed Oct 12 16:15:28 2011
From: sirvan.paraste at gmail.com (sirvan paraste)
Date: Wed, 12 Oct 2011 15:45:28 +0330
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <20111011091700.16181.qmail@community22.interfree.it>
References: <20111011091700.16181.qmail@community22.interfree.it>
Message-ID: <CAAAG0NVaYJbqUCRV76rTkEucWybZ1RzGpRanCqmSnV7E6VP_dA@mail.gmail.com>

Dear

Iused this commands to get log :
sodia loglevel all 9
sofia global siptrace on
sofia tracelevel info

now, the resulted log file is attached to this mail. thank you for your
regard
another thing that is strange is when B call C party and C answering the
call this message  be shown on softphone at the start and then there is no
way voice : "on hold by another party" ,

please help mo to find its solution. I am overwhelmed with it about 3 weeks
:(((((

thank you


On Tue, Oct 11, 2011 at 12:47 PM, <chrisbware at interfree.it> wrote:

> Send a SIP trace of a call.
> Use freeswitch sofia debug (see on wiki) or install ngrep (if you are on
> linux) and use the following command
>
> ngrep port 5060 -Wbyline -q -d any
>
>
> Attach the output on this list.
>
>
>
>
>
> Chrisbware
>
> -----Messaggio originale-----
> Da: sirvan paraste <sirvan.paraste at gmail.com>
> Inviato il: 9 Ott 2011 - 22:38
> A: freeswitch-users at lists.freeswitch.org
>
>
> Dear all
> After regard for your attention to me
> I am student how want work on freeswitch. I have read freeSwitch
>
> documents (introduction, how to install, use default
> confuguration,...).
> for start of using freeswitch, I make three computer network together
> with a hub, after that,I install freeswitch on one of them and X-Lite
> on another computers, network checked with sharing file and ping
> issue.
> After registering of two X-lite hone with No. 1001,1002, they can call
>
> each other but there is no voice !!! :(
> i have checked and did some work but the problem is not solved.
>
> Can you help me?
>
> thank you.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -------------------------------------------------------------------------------
> Valore legale alle tue mail
> InterfreePEC - la tua Posta Elettronica Certificata
> http://pec.interfree.it
>
> -------------------------------------------------------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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2011-10-12 13:01:57.343750 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'enum'
2011-10-12 13:01:57.343750 [NOTICE] switch_loadable_module.c:254 Adding Application 'enum'
2011-10-12 13:01:57.343750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum'
2011-10-12 13:01:57.343750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum_auto'
2011-10-12 13:01:57.343750 [DEBUG] mod_cdr_csv.c:339 Adding default template.
2011-10-12 13:01:57.343750 [DEBUG] mod_cdr_csv.c:386 Adding template sql.
2011-10-12 13:01:57.343750 [DEBUG] mod_cdr_csv.c:386 Adding template example.
2011-10-12 13:01:57.343750 [DEBUG] mod_cdr_csv.c:386 Adding template snom.
2011-10-12 13:01:57.343750 [DEBUG] mod_cdr_csv.c:386 Adding template linksys.
2011-10-12 13:01:57.343750 [DEBUG] mod_cdr_csv.c:386 Adding template asterisk.
2011-10-12 13:01:57.359375 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cdr_csv'
2011-10-12 13:01:57.359375 [NOTICE] switch_loadable_module.c:254 Adding Application 'socket'
2011-10-12 13:01:57.359375 [NOTICE] switch_loadable_module.c:298 Adding API Function 'event_sink'
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 debug [0]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-trace [no]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-capture [no]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rfc2833-pt [101]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-port [5080]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 dialplan [XML]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 context [public]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 dtmf-duration [2000]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 outbound-codec-prefs [PCMU,PCMA,GSM]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 hold-music [local_stream://moh]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-timer-name [soft]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 local-network-acl [localnet.auto]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 manage-presence [false]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 inbound-codec-negotiation [generous]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 nonce-ttl [60]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 auth-calls [false]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-ip [0.0.0.0]
2011-10-12 13:01:57.359375 [WARNING] sofia.c:3844 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-ip [0.0.0.0]
2011-10-12 13:01:57.359375 [WARNING] sofia.c:3857 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 ext-rtp-ip [auto-nat]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 ext-sip-ip [auto-nat]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-timeout-sec [300]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls [false]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-bind-params [transport=tls]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-sip-port [5081]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-cert-dir [C:\Program Files\FreeSWITCH/conf/ssl]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-version [tlsv1]
2011-10-12 13:01:57.359375 [NOTICE] sofia.c:4372 Started Profile external [sofia_reg_external]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 debug [0]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-trace [no]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 context [public]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rfc2833-pt [101]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-port [5060]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 dialplan [XML]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 dtmf-duration [2000]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 outbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 use-rtp-timer [true]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-timer-name [soft]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-ip [::1]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-ip [::1]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 hold-music [local_stream://moh]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 apply-inbound-acl [domains]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 record-template [C:\Program Files\FreeSWITCH\recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 manage-presence [true]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:1747 Creating agent for external
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 inbound-codec-negotiation [generous]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls [false]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-bind-params [transport=tls]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-sip-port [5061]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-cert-dir [C:\Program Files\FreeSWITCH/conf/ssl]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 tls-version [tlsv1]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 nonce-ttl [60]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 auth-calls [true]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 auth-all-packets [false]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-timeout-sec [300]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 force-register-domain [0.0.0.0]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 force-register-db-domain [0.0.0.0]
2011-10-12 13:01:57.359375 [NOTICE] sofia.c:4372 Started Profile internal-ipv6 [sofia_reg_internal-ipv6]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 debug [0]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-trace [no]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-capture [no]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 watchdog-enabled [no]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 watchdog-step-timeout [30000]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 watchdog-event-timeout [30000]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 log-auth-failures [true]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 forward-unsolicited-mwi-notify [false]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 context [public]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rfc2833-pt [101]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:1747 Creating agent for internal-ipv6
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-port [5060]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 dialplan [XML]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 dtmf-duration [2000]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 outbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-timer-name [soft]
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 rtp-ip [0.0.0.0]
2011-10-12 13:01:57.359375 [WARNING] sofia.c:3844 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 sip-ip [0.0.0.0]
2011-10-12 13:01:57.359375 [WARNING] sofia.c:3857 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:01:57.359375 [DEBUG] sofia.c:3527 hold-music [local_stream://moh]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 apply-nat-acl [nat.auto]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 apply-inbound-acl [domains]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 local-network-acl [localnet.auto]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 record-path [C:\Program Files\FreeSWITCH\recordings]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 manage-presence [true]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 presence-hosts [0.0.0.0,0.0.0.0]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 inbound-codec-negotiation [generous]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 tls [false]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 tls-bind-params [transport=tls]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 tls-sip-port [5061]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 tls-cert-dir [C:\Program Files\FreeSWITCH/conf/ssl]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 tls-version [tlsv1]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 nonce-ttl [60]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 auth-calls [true]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 inbound-reg-force-matching-username [true]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 auth-all-packets [false]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 ext-rtp-ip [auto-nat]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 ext-sip-ip [auto-nat]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 rtp-timeout-sec [300]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 force-register-domain [0.0.0.0]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 force-subscription-domain [0.0.0.0]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 force-register-db-domain [0.0.0.0]
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:3527 challenge-realm [auto_from]
2011-10-12 13:01:57.375000 [NOTICE] sofia.c:4372 Started Profile internal [sofia_reg_internal]
2011-10-12 13:01:57.375000 [DEBUG] mod_sofia.c:5193 Waiting for profiles to start
2011-10-12 13:01:57.375000 [DEBUG] sofia.c:1747 Creating agent for internal
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1819 Created agent for internal
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1856 Set params for internal
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1899 Activated db for internal
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1819 Created agent for external
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1856 Set params for external
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1899 Activated db for external
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1926 Starting thread for external
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1649 Launching worker thread for external
2011-10-12 13:01:57.515625 [NOTICE] sofia_reg.c:2894 Added gateway 'example.com' to profile 'external'
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1926 Starting thread for internal
2011-10-12 13:01:57.515625 [DEBUG] sofia.c:1649 Launching worker thread for internal
2011-10-12 13:01:57.515625 [NOTICE] sofia.c:2571 Adding Alias [0.0.0.0] for profile [internal]
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'sofia'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_gateway_data'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_username_of'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_contact'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_count_reg'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_dig'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sip'
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:447 Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1001]
2011-10-12 13:01:58.906250 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'loopback'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'acl'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'alias'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'banner'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bgapi'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bg_system'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'break'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'complete'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cond'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete_xml'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'create_uuid'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db_cache'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'domain_exists'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'echo'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'escape'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'eval'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expand'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'find_user_xml'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fsctl'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function '...'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'shutdown'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'version'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_getvar'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_setvar'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group_call'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'help'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'host_lookup'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hostname'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'switchname'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hupall'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'in_group'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'is_lan_addr'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_usage'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_hash_usage'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_status'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_reset'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_interval_reset'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'load'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'log'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'md5'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'module_exists'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'msleep'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'nat_map'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'originate'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'pause'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'regex'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadacl'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reload'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadxml'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'replace'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'say_string'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_api'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_broadcast'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_del'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_hangup'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_transfer'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sql_escape'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'status'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime_tz'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stun'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'system'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'time_test'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'timer_test'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'tone_detect'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unload'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unsched_api'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reg_url'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_decode'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_encode'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_data'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_exists'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_audio'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_break'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_bridge'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_broadcast'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_buglist'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_chat'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_debug_audio'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_deflect'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_displace'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_display'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dump'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_exists'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_fileman'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_flush_dtmf'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_getvar'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_hold'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_kill'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_outgoing_answer'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit_release'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_loglevel'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_media'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_park'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_phone_event'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_preprocess'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_record'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recovery_refresh'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recv_dtmf'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_dtmf'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_session_heartbeat'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar_multi'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_transfer'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dual_transfer'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_simplify'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_jitterbuffer'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_zombie_exec'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_flush_cache'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_locate'
2011-10-12 13:01:59.093750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_wrap'
2011-10-12 13:01:59.140625 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference'
2011-10-12 13:01:59.140625 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference_set_auto_outcall'
2011-10-12 13:01:59.140625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'conference'
2011-10-12 13:01:59.140625 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'conf'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'db'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'group'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'db'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'error'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'group'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'user'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'inline'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_digit_action'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'capture'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_digit_action'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'digit_action_set_realm'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'privacy'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_audio_level'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_mute'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'flush_dtmf'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'hold'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'unhold'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'transfer'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'check_acl'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'verbose_events'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'cng_plc'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'early_hangup'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'sleep'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'delay_echo'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'strftime'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'phrase'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'eval'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_zombie_exec'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'pre_answer'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'answer'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'hangup'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_name'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'presence'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'log'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'info'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'event'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'sound_test'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'export'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge_export'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'set'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'push'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'unshift'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_global'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_profile_var'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'unset'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'ring_ready'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'remove_bugs'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'break'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'detect_speech'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'ivr'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'redirect'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'jitterbuffer'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_display'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'respond'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'deflect'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'recovery_refresh'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'queue_dtmf'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_dtmf'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_cencel'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_hangup'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_broadcast'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_transfer'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'execute_extension'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_heartbeat'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'enable_heartbeat'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'mkdir'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'soft_hold'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_meta_app'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'unbind_meta_app'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'block_dfmf'
2011-10-12 13:01:59.156250 [NOTICE] switch_loadable_module.c:254 Adding Application 'unblock_dtmf'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'intercept'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'eavesdrop'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'three_way'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_user'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf_generate'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf_generate'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'fax_detect'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'tone_detect'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'echo'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'park'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'park_state'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'gentones'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'playback'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'endless_playback'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'att_xfer'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'read'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_and_get_digits'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_record_session'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_session'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'record'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'preprocess'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_displace_session'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'displace_session'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'speak'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_speech_cache'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'system'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'say'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'wait_for_silence'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'session_loglevel'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_execute'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash_execute'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strepoch'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'chat'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'presence'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'file_string'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'event'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'api'
2011-10-12 13:01:59.171875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expr'
2011-10-12 13:01:59.171875 [INFO] mod_fifo.c:4158 cool_fifo at 0.0.0.0 configured
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo_track_call'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_member'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_add_outbound'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_check_bridge'
2011-10-12 13:01:59.187500 [DEBUG] switch_scheduler.c:214 Added task 3 limit_hash_cleanup (mod_hash) to run at 1318412819
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:254 Adding Application 'hash'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_dump'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_remote'
2011-10-12 13:01:59.187500 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'hash'
2011-10-12 13:01:59.203125 [INFO] mod_voicemail.c:746 Added Profile default
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'voicemail'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail_inject'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_inject'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_boxcount'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_prefs'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_delete'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_read'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_list'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_auth_login'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_count'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_list'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_get'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_delete'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_undelete'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_purge'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_save'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_forward'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_greeting_set'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_recname_set'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_password_set'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'esf_page_group'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_fsv'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_fsv'
2011-10-12 13:01:59.203125 [CRIT] switch_loadable_module.c:1271 Error Loading module C:\Program Files\FreeSWITCH\mod\mod_cluechoo.dll
**dll open error [126l]
**
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'valet_park'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'valet_info'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'XML'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'SIP'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'IAX2'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'asterisk'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'Dial'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'Goto'
2011-10-12 13:01:59.203125 [NOTICE] switch_loadable_module.c:254 Adding Application 'AvoidingDeadlock'
2011-10-12 13:01:59.203125 [DEBUG] mod_spandsp_fax.c:202 FAX timer thread started.
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:541 Adding tone_descriptor: 1
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: CED_TONE(0)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: SIT(1)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:541 Adding tone_descriptor: 44
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: CED_TONE(0)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: SIT(1)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:541 Adding tone_descriptor: 49
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: CED_TONE(0)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: SIT(1)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
2011-10-12 13:01:59.234375 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2011-10-12 13:01:59.234375 [DEBUG] mod_spandsp.c:295 mod_spandsp loaded, using spandsp library version [20100724 163333]
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 10ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 20ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 30ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 40ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 50ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 60ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 10ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 20ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 30ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 40ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 50ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 60ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 70ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 80ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 90ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 100ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 110ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 120ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 10ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 20ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 30ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 40ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 50ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 60ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 70ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 80ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 90ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 100ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 110ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 120ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 10ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 20ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 30ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 40ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 50ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 60ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 70ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 80ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 90ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 100ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 110ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 120ms 16000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 10ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 20ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 30ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 40ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 50ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 60ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 70ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 80ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 90ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 100ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 110ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 120ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 10ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 20ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 30ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 40ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 50ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 60ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 70ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 80ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 90ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 100ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 110ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 120ms 24000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 10ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 20ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 30ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 40ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 50ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 60ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 70ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 80ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 90ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 100ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 110ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 120ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 10ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 20ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 30ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 40ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 50ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 60ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 70ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 80ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 90ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 100ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 110ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 120ms 32000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 10ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 20ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 30ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 40ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 50ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 60ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 70ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 80ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 90ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 100ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 110ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 120ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 10ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 20ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 30ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 40ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 50ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 60ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 70ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 80ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 90ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 100ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 110ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 120ms 40000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 10ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 20ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 30ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 40ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 50ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 60ms 64000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 20ms 13200bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 40ms 13200bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 60ms 13200bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 80ms 13200bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 100ms 13200bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 120ms 13200bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec LPC 7 LPC-10 8000hz 90ms 2400bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 't38_gateway'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'rxfax'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'txfax'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_dtmf'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_dtmf'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_fax_detect'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_fax_detect'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_tone_detect'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_tone_detect'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_tone_detect'
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 120ms 6300bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 90ms 6300bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 60ms 6300bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 30ms 6300bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 10ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 20ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 30ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 40ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 50ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 60ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 70ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 80ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 90ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 100ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 110ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 120ms 8000bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec AMR 96 AMR 8000hz 20ms 12200bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec iLBC 97 iLBC 8000hz 30ms 13330bps
2011-10-12 13:01:59.234375 [NOTICE] switch_loadable_module.c:187 Adding Codec iLBC 98 iLBC 8000hz 20ms 15200bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 32000hz 20ms 44000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 16000hz 20ms 42200bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 8000hz 20ms 24600bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec H264 99 H.264 Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec H263 34 H.263 Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-1998 115 H.263+ Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-2000 121 H.263++ Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec H261 31 H.261 Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:01:59.250000 [INFO] mod_siren.c:155 Audio coding: ITU-T Rec. G.722.1, licensed from Polycom(R)
2011-10-12 13:01:59.250000 [INFO] mod_siren.c:156 Audio coding: ITU-T Rec. G.722.1 Annex C, licensed from Polycom(R)
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 115 Polycom(R) G722.1/G722.1C 32000hz 20ms 48000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 115 Polycom(R) G722.1/G722.1C 32000hz 40ms 48000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 115 Polycom(R) G722.1/G722.1C 32000hz 60ms 48000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 20ms 32000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 40ms 32000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 60ms 32000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 20ms 24000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 40ms 24000bps
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 60ms 24000bps
2011-10-12 13:01:59.250000 [INFO] mod_sndfile.c:355 
LibSndFile Version : libsndfile-1.0.12 Supported Formats
================================================================================
AIFF (Apple/SGI)  (extension "aiff")
AU (Sun/NeXT)  (extension "au")
AVR (Audio Visual Research)  (extension "avr")
CAF (Apple Core Audio File)  (extension "caf")
HTK (HMM Tool Kit)  (extension "htk")
IFF (Amiga IFF/SVX8/SV16)  (extension "iff")
MAT4 (GNU Octave 2.0 / Matlab 4.2)  (extension "mat")
MAT5 (GNU Octave 2.1 / Matlab 5.0)  (extension "mat")
PAF (Ensoniq PARIS)  (extension "paf")
PVF (Portable Voice Format)  (extension "pvf")
RAW (header-less)  (extension "raw")
SD2 (Sound Designer II)  (extension "sd2")
SDS (Midi Sample Dump Standard)  (extension "sds")
SF (Berkeley/IRCAM/CARL)  (extension "sf")
VOC (Creative Labs)  (extension "voc")
W64 (SoundFoundry WAVE 64)  (extension "w64")
WAV (Microsoft)  (extension "wav")
WAV (NIST Sphere)  (extension "wav")
WAVEX (Microsoft)  (extension "wav")
WVE (Psion Series 3)  (extension "wve")
XI (FastTracker 2)  (extension "xi")
================================================================================
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'aiff'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'au'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'avr'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'caf'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'htk'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'iff'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'mat'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'paf'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'pvf'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'raw'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sd2'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sds'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sf'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'voc'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'w64'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wav'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wve'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'xi'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r8'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r16'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r24'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r32'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'gsm'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ul'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ulaw'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'al'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'alaw'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'adpcm'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G7221'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AMR'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'SPEEX'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G729'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-16'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-16'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMA'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'DVI4'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-24'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMU'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'L16'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-24'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-32'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-32'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G722'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-40'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-40'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'GSM'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'LPC'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-2000'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H264'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H261'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-1998'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'iLBC'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G723'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY-VID'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_local_stream'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_local_stream'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show_local_stream'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'local_stream'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'tone_stream'
2011-10-12 13:01:59.250000 [NOTICE] switch_loadable_module.c:322 Adding File Format 'silence_stream'
2011-10-12 13:01:59.265625 [DEBUG] switch_loadable_module.c:1232 Loading module with global namespace at request of module
2011-10-12 13:01:59.265625 [NOTICE] switch_loadable_module.c:254 Adding Application 'javascript'
2011-10-12 13:01:59.265625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsrun'
2011-10-12 13:01:59.265625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsapi'
2011-10-12 13:01:59.281250 [DEBUG] switch_loadable_module.c:1232 Loading module with global namespace at request of module
2011-10-12 13:01:59.281250 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'LUA'
2011-10-12 13:01:59.281250 [NOTICE] switch_loadable_module.c:254 Adding Application 'lua'
2011-10-12 13:01:59.281250 [NOTICE] switch_loadable_module.c:276 Adding Chat Application 'lua'
2011-10-12 13:01:59.281250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'luarun'
2011-10-12 13:01:59.281250 [NOTICE] switch_loadable_module.c:298 Adding API Function 'lua'
2011-10-12 13:01:59.281250 [NOTICE] switch_loadable_module.c:423 Adding Say interface 'en'
2011-10-12 13:01:59.281250 [DEBUG] switch_loadable_module.c:607 Chat Thread Started
2011-10-12 13:01:59.281250 [DEBUG] switch_loadable_module.c:607 Chat Thread Started
2011-10-12 13:01:59.281250 [DEBUG] mod_event_socket.c:2745 Socket up listening on 127.0.0.1:8021
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1101 Created ip list rfc1918.auto default (deny)
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list rfc1918.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1109 Created ip list wan.auto default (allow)
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (deny) [] to list wan.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (deny) [] to list wan.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (deny) [] to list wan.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1117 Created ip list nat.auto default (deny)
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1119 Adding 0.0.0.0/0.0.0.0 (deny) to list nat.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list nat.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list nat.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list nat.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1128 Created ip list loopback.auto default (deny)
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 127.0.0.0/8 (allow) [] to list loopback.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1134 Created ip list localnet.auto default (deny)
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1137 Adding 0.0.0.0/0.0.0.0 (allow) to list localnet.auto
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 192.168.42.0/24 (deny) [] to list lan
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1233 Adding 192.168.42.0/24 (deny) to list lan
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 192.168.42.42/32 (allow) [] to list lan
2011-10-12 13:01:59.281250 [NOTICE] switch_core.c:1233 Adding 192.168.42.42/32 (allow) to list lan
2011-10-12 13:01:59.281250 [NOTICE] switch_utils.c:248 Adding 192.0.2.0/24 (allow) [brian at 0.0.0.0] to list domains
2011-10-12 13:01:59.625000 [ERR] sofia.c:1813 Error Creating SIP UA for profile: internal-ipv6
2011-10-12 13:03:07.812500 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'enum'
2011-10-12 13:03:07.812500 [NOTICE] switch_loadable_module.c:254 Adding Application 'enum'
2011-10-12 13:03:07.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum'
2011-10-12 13:03:07.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'enum_auto'
2011-10-12 13:03:07.812500 [DEBUG] mod_cdr_csv.c:339 Adding default template.
2011-10-12 13:03:07.812500 [DEBUG] mod_cdr_csv.c:386 Adding template sql.
2011-10-12 13:03:07.812500 [DEBUG] mod_cdr_csv.c:386 Adding template example.
2011-10-12 13:03:07.812500 [DEBUG] mod_cdr_csv.c:386 Adding template snom.
2011-10-12 13:03:07.812500 [DEBUG] mod_cdr_csv.c:386 Adding template linksys.
2011-10-12 13:03:07.812500 [DEBUG] mod_cdr_csv.c:386 Adding template asterisk.
2011-10-12 13:03:07.828125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cdr_csv'
2011-10-12 13:03:07.828125 [NOTICE] switch_loadable_module.c:254 Adding Application 'socket'
2011-10-12 13:03:07.828125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'event_sink'
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 debug [0]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 sip-trace [no]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 sip-capture [no]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rfc2833-pt [101]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 sip-port [5080]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 dialplan [XML]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 context [public]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 dtmf-duration [2000]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 outbound-codec-prefs [PCMU,PCMA,GSM]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 hold-music [local_stream://moh]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-timer-name [soft]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 local-network-acl [localnet.auto]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 manage-presence [false]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 inbound-codec-negotiation [generous]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 nonce-ttl [60]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 auth-calls [false]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-ip [0.0.0.0]
2011-10-12 13:03:07.828125 [WARNING] sofia.c:3844 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 sip-ip [0.0.0.0]
2011-10-12 13:03:07.828125 [WARNING] sofia.c:3857 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 ext-rtp-ip [auto-nat]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 ext-sip-ip [auto-nat]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-timeout-sec [300]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls [false]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-bind-params [transport=tls]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-sip-port [5081]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-cert-dir [C:\Program Files\FreeSWITCH/conf/ssl]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-version [tlsv1]
2011-10-12 13:03:07.828125 [NOTICE] sofia.c:4372 Started Profile external [sofia_reg_external]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 debug [0]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 sip-trace [no]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 context [public]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:1747 Creating agent for external
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rfc2833-pt [101]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 sip-port [5060]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 dialplan [XML]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 dtmf-duration [2000]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 outbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 use-rtp-timer [true]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-timer-name [soft]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-ip [::1]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 sip-ip [::1]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 hold-music [local_stream://moh]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 apply-inbound-acl [domains]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 record-template [C:\Program Files\FreeSWITCH\recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 manage-presence [true]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 inbound-codec-negotiation [generous]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls [false]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-bind-params [transport=tls]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-sip-port [5061]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-cert-dir [C:\Program Files\FreeSWITCH/conf/ssl]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 tls-version [tlsv1]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 nonce-ttl [60]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 auth-calls [true]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 auth-all-packets [false]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-timeout-sec [300]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800]
2011-10-12 13:03:07.828125 [DEBUG] sofia.c:3527 force-register-domain [0.0.0.0]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 force-register-db-domain [0.0.0.0]
2011-10-12 13:03:07.843750 [NOTICE] sofia.c:4372 Started Profile internal-ipv6 [sofia_reg_internal-ipv6]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 debug [0]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 sip-trace [no]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 sip-capture [no]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 watchdog-enabled [no]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 watchdog-step-timeout [30000]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 watchdog-event-timeout [30000]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 log-auth-failures [true]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 forward-unsolicited-mwi-notify [false]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 context [public]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 rfc2833-pt [101]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 sip-port [5060]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 dialplan [XML]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 dtmf-duration [2000]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:1747 Creating agent for internal-ipv6
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 outbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 rtp-timer-name [soft]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 rtp-ip [0.0.0.0]
2011-10-12 13:03:07.843750 [WARNING] sofia.c:3844 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 sip-ip [0.0.0.0]
2011-10-12 13:03:07.843750 [WARNING] sofia.c:3857 Invalid IP 0.0.0.0 replaced with 0.0.0.0
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 hold-music [local_stream://moh]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 apply-nat-acl [nat.auto]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 apply-inbound-acl [domains]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 local-network-acl [localnet.auto]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 record-path [C:\Program Files\FreeSWITCH\recordings]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 manage-presence [true]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 presence-hosts [0.0.0.0,0.0.0.0]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 inbound-codec-negotiation [generous]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 tls [false]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 tls-bind-params [transport=tls]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 tls-sip-port [5061]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 tls-cert-dir [C:\Program Files\FreeSWITCH/conf/ssl]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 tls-version [tlsv1]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 nonce-ttl [60]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 auth-calls [true]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 inbound-reg-force-matching-username [true]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 auth-all-packets [false]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 ext-rtp-ip [auto-nat]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 ext-sip-ip [auto-nat]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 rtp-timeout-sec [300]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 force-register-domain [0.0.0.0]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 force-subscription-domain [0.0.0.0]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 force-register-db-domain [0.0.0.0]
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:3527 challenge-realm [auto_from]
2011-10-12 13:03:07.843750 [NOTICE] sofia.c:4372 Started Profile internal [sofia_reg_internal]
2011-10-12 13:03:07.843750 [DEBUG] mod_sofia.c:5193 Waiting for profiles to start
2011-10-12 13:03:07.843750 [DEBUG] sofia.c:1747 Creating agent for internal
2011-10-12 13:03:08.015625 [ERR] sofia.c:1813 Error Creating SIP UA for profile: internal
2011-10-12 13:03:08.015625 [ERR] sofia.c:1813 Error Creating SIP UA for profile: external
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'sofia'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_gateway_data'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_username_of'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_contact'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_count_reg'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sofia_dig'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sip'
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:447 Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1001]
2011-10-12 13:03:09.390625 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'loopback'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'acl'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'alias'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'banner'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bgapi'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'bg_system'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'break'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'complete'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'cond'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'console_complete_xml'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'create_uuid'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db_cache'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'domain_exists'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'echo'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'escape'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'eval'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expand'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'find_user_xml'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fsctl'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function '...'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'shutdown'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'version'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_getvar'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'global_setvar'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group_call'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'help'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'host_lookup'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hostname'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'switchname'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hupall'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'in_group'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'is_lan_addr'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_usage'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_hash_usage'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_status'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_reset'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'limit_interval_reset'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'load'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'log'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'md5'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'module_exists'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'msleep'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'nat_map'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'originate'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'pause'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'regex'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadacl'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reload'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reloadxml'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'replace'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'say_string'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_api'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_broadcast'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_del'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_hangup'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sched_transfer'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'sql_escape'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'status'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime_tz'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stun'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'system'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'time_test'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'timer_test'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'tone_detect'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unload'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'unsched_api'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'reg_url'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_decode'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'url_encode'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_data'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'user_exists'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_audio'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_break'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_bridge'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_broadcast'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_buglist'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_chat'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_debug_audio'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_deflect'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_displace'
2011-10-12 13:03:09.578125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_display'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dump'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_exists'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_fileman'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_flush_dtmf'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_getvar'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_hold'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_kill'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_outgoing_answer'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_limit_release'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_loglevel'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_media'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_park'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_phone_event'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_preprocess'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_record'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recovery_refresh'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_recv_dtmf'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_send_dtmf'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_session_heartbeat'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar_multi'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_setvar'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_transfer'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_dual_transfer'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_simplify'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_jitterbuffer'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'uuid_zombie_exec'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_flush_cache'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_locate'
2011-10-12 13:03:09.593750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'xml_wrap'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:254 Adding Application 'conference_set_auto_outcall'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:298 Adding API Function 'conference'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'conf'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:254 Adding Application 'db'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:254 Adding Application 'group'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'db'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'error'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'group'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'user'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'inline'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_digit_action'
2011-10-12 13:03:09.750000 [NOTICE] switch_loadable_module.c:254 Adding Application 'capture'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_digit_action'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'digit_action_set_realm'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'privacy'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_audio_level'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_mute'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'flush_dtmf'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'hold'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'unhold'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'transfer'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'check_acl'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'verbose_events'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'cng_plc'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'early_hangup'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'sleep'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'delay_echo'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'strftime'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'phrase'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'eval'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_zombie_exec'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'pre_answer'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'answer'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'hangup'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_name'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'presence'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'log'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'info'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'event'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'sound_test'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'export'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge_export'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'push'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'unshift'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_global'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_profile_var'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'unset'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'ring_ready'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'remove_bugs'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'break'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'detect_speech'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'ivr'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'redirect'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'jitterbuffer'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_display'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'respond'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'deflect'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'recovery_refresh'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'queue_dtmf'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'send_dtmf'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_cencel'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_hangup'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_broadcast'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_transfer'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'execute_extension'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'sched_heartbeat'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'enable_heartbeat'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'mkdir'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'soft_hold'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'bind_meta_app'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'unbind_meta_app'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'block_dfmf'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'unblock_dtmf'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'intercept'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'eavesdrop'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'three_way'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'set_user'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_dtmf_generate'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_dtmf_generate'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'fax_detect'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'tone_detect'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'echo'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'park'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'park_state'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'gentones'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'playback'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'endless_playback'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'att_xfer'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'read'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_and_get_digits'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_record_session'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_session'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'record'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'preprocess'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_displace_session'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'displace_session'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'speak'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'clear_speech_cache'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'bridge'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'system'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'say'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'wait_for_silence'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'session_loglevel'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_execute'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:254 Adding Application 'limit_hash_execute'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strepoch'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'chat'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'strftime'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'presence'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:322 Adding File Format 'file_string'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'event'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'api'
2011-10-12 13:03:09.765625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'expr'
2011-10-12 13:03:09.781250 [INFO] mod_fifo.c:4158 cool_fifo at 0.0.0.0 configured
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:254 Adding Application 'fifo_track_call'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_member'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_add_outbound'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'fifo_check_bridge'
2011-10-12 13:03:09.796875 [DEBUG] switch_scheduler.c:214 Added task 3 limit_hash_cleanup (mod_hash) to run at 1318412889
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:254 Adding Application 'hash'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_dump'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'hash_remote'
2011-10-12 13:03:09.796875 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'hash'
2011-10-12 13:03:09.812500 [INFO] mod_voicemail.c:746 Added Profile default
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:254 Adding Application 'voicemail'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'voicemail_inject'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_inject'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_boxcount'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_prefs'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_delete'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_read'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_list'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_auth_login'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_count'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_list'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_get'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_delete'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_undelete'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_purge'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_save'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_msg_forward'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_greeting_set'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_recname_set'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:298 Adding API Function 'vm_fsdb_pref_password_set'
2011-10-12 13:03:09.812500 [NOTICE] switch_loadable_module.c:254 Adding Application 'esf_page_group'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:254 Adding Application 'play_fsv'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:254 Adding Application 'record_fsv'
2011-10-12 13:03:09.828125 [CRIT] switch_loadable_module.c:1271 Error Loading module C:\Program Files\FreeSWITCH\mod\mod_cluechoo.dll
**dll open error [126l]
**
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:254 Adding Application 'valet_park'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:298 Adding API Function 'valet_info'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'XML'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'SIP'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'IAX2'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'asterisk'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:254 Adding Application 'Dial'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:254 Adding Application 'Goto'
2011-10-12 13:03:09.828125 [NOTICE] switch_loadable_module.c:254 Adding Application 'AvoidingDeadlock'
2011-10-12 13:03:09.875000 [DEBUG] mod_spandsp_fax.c:202 FAX timer thread started.
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:541 Adding tone_descriptor: 1
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: CED_TONE(0)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: SIT(1)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:541 Adding tone_descriptor: 44
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: CED_TONE(0)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: SIT(1)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:541 Adding tone_descriptor: 49
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: CED_TONE(0)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: SIT(1)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:561 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
2011-10-12 13:03:09.890625 [INFO] mod_spandsp_dsp.c:594 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2011-10-12 13:03:09.890625 [DEBUG] mod_spandsp.c:295 mod_spandsp loaded, using spandsp library version [20100724 163333]
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 10ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 20ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 30ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 40ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 50ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 6 ADPCM (IMA) 16000hz 60ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 10ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 20ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 30ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 40ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 50ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 60ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 70ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 80ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 90ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 100ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 110ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec DVI4 5 ADPCM (IMA) 8000hz 120ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 10ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 20ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 30ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 40ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 50ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 60ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 70ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 80ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 90ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 100ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 110ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-16 124 G.726 16k (AAL2) 8000hz 120ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 10ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 20ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 30ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 40ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 50ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 60ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 70ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 80ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 90ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 100ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 110ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-16 124 G.726 16k 8000hz 120ms 16000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 10ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 20ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 30ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 40ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 50ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 60ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 70ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 80ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 90ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 100ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 110ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-24 123 G.726 24k (AAL2) 8000hz 120ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 10ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 20ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 30ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 40ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 50ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 60ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 70ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 80ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 90ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 100ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 110ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-24 123 G.726 24k 8000hz 120ms 24000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 10ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 20ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 30ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 40ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 50ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 60ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 70ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 80ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 90ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 100ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 110ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-32 122 G.726 32k (AAL2) 8000hz 120ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 10ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 20ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 30ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 40ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 50ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 60ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 70ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 80ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 90ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 100ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 110ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-32 122 G.726 32k 8000hz 120ms 32000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 10ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 20ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 30ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 40ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 50ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 60ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 70ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 80ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 90ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 100ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 110ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec AAL2-G726-40 121 G.726 40k (AAL2) 8000hz 120ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 10ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 20ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 30ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 40ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 50ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 60ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 70ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 80ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 90ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 100ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 110ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G726-40 121 G.726 40k 8000hz 120ms 40000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 10ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 20ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 30ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 40ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 50ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G722 9 G.722 16000hz 60ms 64000bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 20ms 13200bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 40ms 13200bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 60ms 13200bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 80ms 13200bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 100ms 13200bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec GSM 3 GSM 8000hz 120ms 13200bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec LPC 7 LPC-10 8000hz 90ms 2400bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 't38_gateway'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'rxfax'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'txfax'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_dtmf'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_dtmf'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_start_fax_detect'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'spandsp_stop_fax_detect'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'start_tone_detect'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:254 Adding Application 'stop_tone_detect'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_tone_detect'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_tone_detect'
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 120ms 6300bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 90ms 6300bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 60ms 6300bps
2011-10-12 13:03:09.890625 [NOTICE] switch_loadable_module.c:187 Adding Codec G723 4 G.723.1 6.3k 8000hz 30ms 6300bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 10ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 20ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 30ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 40ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 50ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 60ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 70ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 80ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 90ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 100ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 110ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G729 18 G.729 8000hz 120ms 8000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec AMR 96 AMR 8000hz 20ms 12200bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec iLBC 97 iLBC 8000hz 30ms 13330bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec iLBC 98 iLBC 8000hz 20ms 15200bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 32000hz 20ms 44000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 16000hz 20ms 42200bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec SPEEX 99 Speex 8000hz 20ms 24600bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec H264 99 H.264 Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec H263 34 H.263 Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-1998 115 H.263+ Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec H263-2000 121 H.263++ Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec H261 31 H.261 Video (passthru) 90000hz 0ms 0bps
2011-10-12 13:03:09.906250 [INFO] mod_siren.c:155 Audio coding: ITU-T Rec. G.722.1, licensed from Polycom(R)
2011-10-12 13:03:09.906250 [INFO] mod_siren.c:156 Audio coding: ITU-T Rec. G.722.1 Annex C, licensed from Polycom(R)
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 115 Polycom(R) G722.1/G722.1C 32000hz 20ms 48000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 115 Polycom(R) G722.1/G722.1C 32000hz 40ms 48000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 115 Polycom(R) G722.1/G722.1C 32000hz 60ms 48000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 20ms 32000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 40ms 32000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 60ms 32000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 20ms 24000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 40ms 24000bps
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:187 Adding Codec G7221 107 Polycom(R) G722.1/G722.1C 16000hz 60ms 24000bps
2011-10-12 13:03:09.906250 [INFO] mod_sndfile.c:355 
LibSndFile Version : libsndfile-1.0.12 Supported Formats
================================================================================
AIFF (Apple/SGI)  (extension "aiff")
AU (Sun/NeXT)  (extension "au")
AVR (Audio Visual Research)  (extension "avr")
CAF (Apple Core Audio File)  (extension "caf")
HTK (HMM Tool Kit)  (extension "htk")
IFF (Amiga IFF/SVX8/SV16)  (extension "iff")
MAT4 (GNU Octave 2.0 / Matlab 4.2)  (extension "mat")
MAT5 (GNU Octave 2.1 / Matlab 5.0)  (extension "mat")
PAF (Ensoniq PARIS)  (extension "paf")
PVF (Portable Voice Format)  (extension "pvf")
RAW (header-less)  (extension "raw")
SD2 (Sound Designer II)  (extension "sd2")
SDS (Midi Sample Dump Standard)  (extension "sds")
SF (Berkeley/IRCAM/CARL)  (extension "sf")
VOC (Creative Labs)  (extension "voc")
W64 (SoundFoundry WAVE 64)  (extension "w64")
WAV (Microsoft)  (extension "wav")
WAV (NIST Sphere)  (extension "wav")
WAVEX (Microsoft)  (extension "wav")
WVE (Psion Series 3)  (extension "wve")
XI (FastTracker 2)  (extension "xi")
================================================================================
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'aiff'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'au'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'avr'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'caf'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'htk'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'iff'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'mat'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'paf'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'pvf'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'raw'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sd2'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sds'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'sf'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'voc'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'w64'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wav'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'wve'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'xi'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r8'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r16'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r24'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'r32'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'gsm'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ul'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'ulaw'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'al'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'alaw'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'adpcm'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G7221'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AMR'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'SPEEX'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G729'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-16'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-16'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMA'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'DVI4'
2011-10-12 13:03:09.906250 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-24'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PCMU'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'L16'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-24'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-32'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-32'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G722'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'AAL2-G726-40'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G726-40'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'GSM'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'LPC'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-2000'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H264'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H261'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'H263-1998'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'iLBC'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'G723'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'PROXY-VID'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'stop_local_stream'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'start_local_stream'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:298 Adding API Function 'show_local_stream'
2011-10-12 13:03:09.921875 [NOTICE] switch_loadable_module.c:322 Adding File Format 'local_stream'
2011-10-12 13:03:10.703125 [ERR] sofia.c:1813 Error Creating SIP UA for profile: internal-ipv6
2011-10-12 13:03:10.703125 [NOTICE] switch_loadable_module.c:322 Adding File Format 'tone_stream'
2011-10-12 13:03:10.703125 [NOTICE] switch_loadable_module.c:322 Adding File Format 'silence_stream'
2011-10-12 13:03:10.718750 [DEBUG] switch_loadable_module.c:1232 Loading module with global namespace at request of module
2011-10-12 13:03:10.718750 [NOTICE] switch_loadable_module.c:254 Adding Application 'javascript'
2011-10-12 13:03:10.718750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsrun'
2011-10-12 13:03:10.718750 [NOTICE] switch_loadable_module.c:298 Adding API Function 'jsapi'
2011-10-12 13:03:10.734375 [DEBUG] switch_loadable_module.c:1232 Loading module with global namespace at request of module
2011-10-12 13:03:10.734375 [NOTICE] switch_loadable_module.c:214 Adding Dialplan 'LUA'
2011-10-12 13:03:10.734375 [NOTICE] switch_loadable_module.c:254 Adding Application 'lua'
2011-10-12 13:03:10.734375 [NOTICE] switch_loadable_module.c:276 Adding Chat Application 'lua'
2011-10-12 13:03:10.734375 [NOTICE] switch_loadable_module.c:298 Adding API Function 'luarun'
2011-10-12 13:03:10.734375 [NOTICE] switch_loadable_module.c:298 Adding API Function 'lua'
2011-10-12 13:03:10.734375 [NOTICE] switch_loadable_module.c:423 Adding Say interface 'en'
2011-10-12 13:03:10.734375 [DEBUG] switch_loadable_module.c:607 Chat Thread Started
2011-10-12 13:03:10.734375 [DEBUG] switch_loadable_module.c:607 Chat Thread Started
2011-10-12 13:03:10.734375 [DEBUG] mod_event_socket.c:2745 Socket up listening on 127.0.0.1:8021
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1101 Created ip list rfc1918.auto default (deny)
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list rfc1918.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1109 Created ip list wan.auto default (allow)
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (deny) [] to list wan.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (deny) [] to list wan.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (deny) [] to list wan.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1117 Created ip list nat.auto default (deny)
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1119 Adding 0.0.0.0/0.0.0.0 (deny) to list nat.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list nat.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list nat.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list nat.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1128 Created ip list loopback.auto default (deny)
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 127.0.0.0/8 (allow) [] to list loopback.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1134 Created ip list localnet.auto default (deny)
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1137 Adding 0.0.0.0/0.0.0.0 (allow) to list localnet.auto
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 192.168.42.0/24 (deny) [] to list lan
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1233 Adding 192.168.42.0/24 (deny) to list lan
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 192.168.42.42/32 (allow) [] to list lan
2011-10-12 13:03:10.734375 [NOTICE] switch_core.c:1233 Adding 192.168.42.42/32 (allow) to list lan
2011-10-12 13:03:10.734375 [NOTICE] switch_utils.c:248 Adding 192.0.2.0/24 (allow) [brian at 0.0.0.0] to list domains
2011-10-12 13:04:33.546875 [DEBUG] sofia.c:7124 IP 192.168.0.2 Rejected by acl "domains". Falling back to Digest auth.
2011-10-12 13:04:33.546875 [DEBUG] sofia.c:7124 IP 192.168.0.2 Rejected by acl "domains". Falling back to Digest auth.
2011-10-12 13:04:33.546875 [NOTICE] switch_channel.c:911 New Channel sofia/internal/1010 at 192.168.0.1 [867b9cf6-5cee-40b8-982b-4485f570ecdc]
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1010 at 192.168.0.1) Running State Change CS_NEW
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/1010 at 192.168.0.1) State NEW
2011-10-12 13:04:33.546875 [DEBUG] sofia.c:5156 Channel sofia/internal/1010 at 192.168.0.1 entering state [received][100]
2011-10-12 13:04:33.546875 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=- 6 2 IN IP4 192.168.0.2
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.0.2
t=0 0
m=audio 39320 RTP/AVP 107 0 8 18 101
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : O+hwJhCe Mh9bQNAG 192.168.0.2 39320

2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:115:32000:20:48000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:107:16000:20:32000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [BV32:107:16000:20:0]/[G722:9:8000:20:64000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [BV32:107:16000:20:0]/[GSM:3:8000:20:13200]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:2841 Set Codec sofia/internal/1010 at 192.168.0.1 PCMU/8000 20 ms 160 samples 64000 bits
2011-10-12 13:04:33.546875 [DEBUG] sofia_glue.c:4835 Set 2833 dtmf send/recv payload to 101
2011-10-12 13:04:33.546875 [DEBUG] sofia.c:5366 (sofia/internal/1010 at 192.168.0.1) State Change CS_NEW -> CS_INIT
2011-10-12 13:04:33.546875 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1010 at 192.168.0.1) Running State Change CS_INIT
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1010 at 192.168.0.1) State INIT
2011-10-12 13:04:33.546875 [DEBUG] mod_sofia.c:85 sofia/internal/1010 at 192.168.0.1 SOFIA INIT
2011-10-12 13:04:33.546875 [DEBUG] mod_sofia.c:125 (sofia/internal/1010 at 192.168.0.1) State Change CS_INIT -> CS_ROUTING
2011-10-12 13:04:33.546875 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1010 at 192.168.0.1) State INIT going to sleep
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1010 at 192.168.0.1) Running State Change CS_ROUTING
2011-10-12 13:04:33.546875 [DEBUG] switch_channel.c:1840 (sofia/internal/1010 at 192.168.0.1) Callstate Change DOWN -> RINGING
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1010 at 192.168.0.1) State ROUTING
2011-10-12 13:04:33.546875 [DEBUG] mod_sofia.c:148 sofia/internal/1010 at 192.168.0.1 SOFIA ROUTING
2011-10-12 13:04:33.546875 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1010 at 192.168.0.1 Standard ROUTING
2011-10-12 13:04:33.546875 [INFO] mod_dialplan_xml.c:336 Processing ali <1010>->1005 in context default
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1010 at 192.168.0.1 Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(open=true) 
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1010 at 192.168.0.1 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [global-intercept] destination_number(1005) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [group-intercept] destination_number(1005) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [intercept-ext] destination_number(1005) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->redial] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [redial] destination_number(1005) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->global] continue=true
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1010 at 192.168.0.1 Absolute Condition [global]
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [snom-demo-2] destination_number(1005) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [snom-demo-1] destination_number(1005) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [eavesdrop] destination_number(1005) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [eavesdrop] destination_number(1005) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [call_return] destination_number(1005) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [del-group] destination_number(1005) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->add-group] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [add-group] destination_number(1005) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->call-group-simo] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [call-group-simo] destination_number(1005) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->call-group-order] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [call-group-order] destination_number(1005) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->extension-intercom] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (FAIL) [extension-intercom] destination_number(1005) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1010 at 192.168.0.1 Regex (PASS) [Local_Extension] destination_number(1005) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(dialed_extension=1005) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action export(dialed_extension=1005) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action bind_meta_app(1 b s execute_extension::dx XML features) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action bind_meta_app(2 b s record_session::C:\Program Files\FreeSWITCH\recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action bind_meta_app(3 b s execute_extension::cf XML features) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(ringback=${us-ring}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(transfer_ringback=local_stream://moh) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(call_timeout=30) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(hangup_after_bridge=true) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(continue_on_fail=true) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action bridge(user/${dialed_extension}@${domain_name}) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action answer() 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action sleep(1000) 
Dialplan: sofia/internal/1010 at 192.168.0.1 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1010 at 192.168.0.1) State Change CS_ROUTING -> CS_EXECUTE
2011-10-12 13:04:33.562500 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1010 at 192.168.0.1) State ROUTING going to sleep
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1010 at 192.168.0.1) Running State Change CS_EXECUTE
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1010 at 192.168.0.1) State EXECUTE
2011-10-12 13:04:33.562500 [DEBUG] mod_sofia.c:241 sofia/internal/1010 at 192.168.0.1 SOFIA EXECUTE
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1010 at 192.168.0.1 Standard EXECUTE
EXECUTE sofia/internal/1010 at 192.168.0.1 set(open=true)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [open]=[true]
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-spymap/1010/867b9cf6-5cee-40b8-982b-4485f570ecdc)
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-last_dial/1010/1005)
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-last_dial/global/867b9cf6-5cee-40b8-982b-4485f570ecdc)
EXECUTE sofia/internal/1010 at 192.168.0.1 set(RFC2822_DATE=Wed, 12 Oct 2011 13:04:33 Iran Standard Time)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [RFC2822_DATE]=[Wed, 12 Oct 2011 13:04:33 Iran Standard Time]
EXECUTE sofia/internal/1010 at 192.168.0.1 set(dialed_extension=1005)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [dialed_extension]=[1005]
EXECUTE sofia/internal/1010 at 192.168.0.1 export(dialed_extension=1005)
2011-10-12 13:04:33.562500 [DEBUG] switch_channel.c:1078 EXPORT (export_vars) [dialed_extension]=[1005]
EXECUTE sofia/internal/1010 at 192.168.0.1 bind_meta_app(1 b s execute_extension::dx XML features)
2011-10-12 13:04:33.562500 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *1 execute_extension::dx XML features
EXECUTE sofia/internal/1010 at 192.168.0.1 bind_meta_app(2 b s record_session::C:\Program Files\FreeSWITCH\recordings/1010.2011-10-12-13-04-33.wav)
2011-10-12 13:04:33.562500 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *2 record_session::C:\Program Files\FreeSWITCH
ecordings/1010.2011-10-12-13-04-33.wav
EXECUTE sofia/internal/1010 at 192.168.0.1 bind_meta_app(3 b s execute_extension::cf XML features)
2011-10-12 13:04:33.562500 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *3 execute_extension::cf XML features
EXECUTE sofia/internal/1010 at 192.168.0.1 bind_meta_app(4 b s execute_extension::att_xfer XML features)
2011-10-12 13:04:33.562500 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *4 execute_extension::att_xfer XML features
EXECUTE sofia/internal/1010 at 192.168.0.1 set(ringback=%(2000,4000,440,480))
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [ringback]=[%(2000,4000,440,480)]
EXECUTE sofia/internal/1010 at 192.168.0.1 set(transfer_ringback=local_stream://moh)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1010 at 192.168.0.1 set(call_timeout=30)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [call_timeout]=[30]
EXECUTE sofia/internal/1010 at 192.168.0.1 set(hangup_after_bridge=true)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1010 at 192.168.0.1 set(continue_on_fail=true)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-call_return/1005/1010)
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-last_dial_ext/1005/867b9cf6-5cee-40b8-982b-4485f570ecdc)
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-last_dial_ext//867b9cf6-5cee-40b8-982b-4485f570ecdc)
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-last_dial_ext/global/867b9cf6-5cee-40b8-982b-4485f570ecdc)
EXECUTE sofia/internal/1010 at 192.168.0.1 set(called_party_callgroup=techsupport)
2011-10-12 13:04:33.562500 [DEBUG] mod_dptools.c:1176 sofia/internal/1010 at 192.168.0.1 SET [called_party_callgroup]=[techsupport]
EXECUTE sofia/internal/1010 at 192.168.0.1 hash(insert/0.0.0.0-last_dial/techsupport/867b9cf6-5cee-40b8-982b-4485f570ecdc)
EXECUTE sofia/internal/1010 at 192.168.0.1 bridge(user/1005 at 0.0.0.0)
2011-10-12 13:04:33.562500 [DEBUG] switch_channel.c:1032 sofia/internal/1010 at 192.168.0.1 EXPORTING[export_vars] [dialed_extension]=[1005] to event
2011-10-12 13:04:33.562500 [DEBUG] switch_ivr_originate.c:1882 Parsing global variables
2011-10-12 13:04:33.562500 [DEBUG] switch_channel.c:1032 sofia/internal/1010 at 192.168.0.1 EXPORTING[export_vars] [dialed_extension]=[1005] to event
2011-10-12 13:04:33.562500 [DEBUG] switch_ivr_originate.c:1882 Parsing global variables
2011-10-12 13:04:33.562500 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[0.0.0.0]
2011-10-12 13:04:33.562500 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1005 at 0.0.0.0]
2011-10-12 13:04:33.562500 [NOTICE] switch_channel.c:911 New Channel sofia/internal/sip:1005 at 192.168.0.3:38232 [9033357c-0204-4b10-96ce-1c092025291b]
2011-10-12 13:04:33.562500 [DEBUG] mod_sofia.c:4494 (sofia/internal/sip:1005 at 192.168.0.3:38232) State Change CS_NEW -> CS_INIT
2011-10-12 13:04:33.562500 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1005 at 192.168.0.3:38232) Running State Change CS_INIT
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1005 at 192.168.0.3:38232) State INIT
2011-10-12 13:04:33.562500 [DEBUG] mod_sofia.c:85 sofia/internal/sip:1005 at 192.168.0.3:38232 SOFIA INIT
2011-10-12 13:04:33.562500 [DEBUG] sofia_glue.c:2401 sip:1005 at 192.168.0.3:38232;rinstance=1d3f14faf631d3f1 Setting proxy route to sofia/internal/sip:1005 at 192.168.0.3:38232
2011-10-12 13:04:33.562500 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:1005 at 192.168.0.3:38232) State Change CS_INIT -> CS_ROUTING
2011-10-12 13:04:33.562500 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1005 at 192.168.0.3:38232) State INIT going to sleep
2011-10-12 13:04:33.562500 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1005 at 192.168.0.3:38232) Running State Change CS_ROUTING
2011-10-12 13:04:33.593750 [DEBUG] switch_channel.c:1840 (sofia/internal/sip:1005 at 192.168.0.3:38232) Callstate Change DOWN -> RINGING
2011-10-12 13:04:33.593750 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1005 at 192.168.0.3:38232) State ROUTING
2011-10-12 13:04:33.593750 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1005 at 192.168.0.3:38232 SOFIA ROUTING
2011-10-12 13:04:33.593750 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1005 at 192.168.0.3:38232) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-10-12 13:04:33.593750 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:33.593750 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1005 at 192.168.0.3:38232) State ROUTING going to sleep
2011-10-12 13:04:33.593750 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1005 at 192.168.0.3:38232) Running State Change CS_CONSUME_MEDIA
2011-10-12 13:04:33.593750 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1005 at 192.168.0.3:38232) State CONSUME_MEDIA
2011-10-12 13:04:33.593750 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1005 at 192.168.0.3:38232) State CONSUME_MEDIA going to sleep
2011-10-12 13:04:33.593750 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:33.593750 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1005 at 192.168.0.3:38232 entering state [calling][0]
2011-10-12 13:04:33.609375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:33.609375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:33.609375 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1005 at 192.168.0.3:38232 entering state [proceeding][180]
2011-10-12 13:04:33.609375 [NOTICE] sofia.c:5235 Ring-Ready sofia/internal/sip:1005 at 192.168.0.3:38232!
2011-10-12 13:04:33.609375 [INFO] switch_ivr_originate.c:1113 Sending early media
2011-10-12 13:04:33.609375 [DEBUG] sofia_glue.c:3093 AUDIO RTP [sofia/internal/1010 at 192.168.0.1] 0.0.0.0 port 31606 -> 192.168.0.2 port 39320 codec: 0 ms: 20
2011-10-12 13:04:33.609375 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms
2011-10-12 13:04:33.609375 [DEBUG] sofia_glue.c:3356 Set 2833 dtmf send payload to 101
2011-10-12 13:04:33.609375 [DEBUG] sofia_glue.c:3362 Set 2833 dtmf receive payload to 101
2011-10-12 13:04:33.609375 [DEBUG] mod_sofia.c:2461 Ring SDP:
v=0
o=FreeSWITCH 1318380467 1318380468 IN IP4 0.0.0.0
s=FreeSWITCH
c=IN IP4 0.0.0.0
t=0 0
m=audio 31606 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

2011-10-12 13:04:33.609375 [NOTICE] mod_sofia.c:2464 Pre-Answer sofia/internal/1010 at 192.168.0.1!
2011-10-12 13:04:33.609375 [DEBUG] switch_channel.c:2877 (sofia/internal/1010 at 192.168.0.1) Callstate Change RINGING -> EARLY
2011-10-12 13:04:33.609375 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:33.609375 [DEBUG] switch_ivr_originate.c:1162 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2011-10-12 13:04:33.609375 [DEBUG] switch_core_codec.c:116 sofia/internal/1010 at 192.168.0.1 Push codec L16:70
2011-10-12 13:04:33.609375 [DEBUG] switch_ivr_originate.c:1225 Play Ringback Tone [%(2000,4000,440,480)]
2011-10-12 13:04:33.609375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:33.609375 [DEBUG] sofia.c:5149 Channel sofia/internal/1010 at 192.168.0.1 skipping state [early][183]
2011-10-12 13:04:41.984375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:41.984375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:41.984375 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1005 at 192.168.0.3:38232 entering state [completing][200]
2011-10-12 13:04:41.984375 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=- 12962927269687500 1 IN IP4 192.168.0.3
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.0.3
t=0 0
a=ice-ufrag:206352
a=ice-pwd:2d300d1fd1728ec2eb2ada8e1c5a97a7
m=audio 62482 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=recvonly
a=candidate:1 1 UDP 659136 192.168.0.3 62482 typ host
a=candidate:1 2 UDP 659134 192.168.0.3 62483 typ host

2011-10-12 13:04:41.984375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:41.984375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:41.984375 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1005 at 192.168.0.3:38232 entering state [ready][200]
2011-10-12 13:04:41.984375 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-10-12 13:04:41.984375 [DEBUG] sofia_glue.c:2841 Set Codec sofia/internal/sip:1005 at 192.168.0.3:38232 PCMU/8000 20 ms 160 samples 64000 bits
2011-10-12 13:04:41.984375 [DEBUG] sofia_glue.c:4828 Set 2833 dtmf send payload to 101
2011-10-12 13:04:41.984375 [DEBUG] sofia_glue.c:3093 AUDIO RTP [sofia/internal/sip:1005 at 192.168.0.3:38232] 0.0.0.0 port 16664 -> 192.168.0.3 port 62482 codec: 0 ms: 20
2011-10-12 13:04:41.984375 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms
2011-10-12 13:04:41.984375 [DEBUG] sofia_glue.c:3356 Set 2833 dtmf send payload to 101
2011-10-12 13:04:41.984375 [DEBUG] sofia_glue.c:3362 Set 2833 dtmf receive payload to 101
2011-10-12 13:04:41.984375 [DEBUG] switch_channel.c:3135 (sofia/internal/sip:1005 at 192.168.0.3:38232) Callstate Change RINGING -> ACTIVE
2011-10-12 13:04:41.984375 [DEBUG] switch_channel.c:3147 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:41.984375 [NOTICE] sofia.c:5841 Channel [sofia/internal/sip:1005 at 192.168.0.3:38232] has been answered
2011-10-12 13:04:42.015625 [DEBUG] switch_core_codec.c:141 sofia/internal/1010 at 192.168.0.1 Restore previous codec PCMU:0.
2011-10-12 13:04:42.015625 [DEBUG] mod_sofia.c:738 Local SDP sofia/internal/1010 at 192.168.0.1:
v=0
o=FreeSWITCH 1318380467 1318380469 IN IP4 0.0.0.0
s=FreeSWITCH
c=IN IP4 0.0.0.0
t=0 0
m=audio 31606 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

2011-10-12 13:04:42.015625 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:42.015625 [DEBUG] switch_channel.c:3135 (sofia/internal/1010 at 192.168.0.1) Callstate Change EARLY -> ACTIVE
2011-10-12 13:04:42.015625 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:42.015625 [NOTICE] switch_ivr_originate.c:3192 Channel [sofia/internal/1010 at 192.168.0.1] has been answered
2011-10-12 13:04:42.015625 [DEBUG] sofia.c:5156 Channel sofia/internal/1010 at 192.168.0.1 entering state [completed][200]
2011-10-12 13:04:42.015625 [DEBUG] switch_ivr_originate.c:3252 Originate Resulted in Success: [sofia/internal/sip:1005 at 192.168.0.3:38232]
2011-10-12 13:04:42.015625 [DEBUG] switch_ivr_originate.c:3252 Originate Resulted in Success: [sofia/internal/sip:1005 at 192.168.0.3:38232]
2011-10-12 13:04:42.015625 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:42.015625 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:04:42.015625 [DEBUG] switch_ivr_bridge.c:1257 (sofia/internal/sip:1005 at 192.168.0.3:38232) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2011-10-12 13:04:42.015625 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:04:42.015625 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1005 at 192.168.0.3:38232) Running State Change CS_EXCHANGE_MEDIA
2011-10-12 13:04:42.015625 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:1005 at 192.168.0.3:38232) State EXCHANGE_MEDIA
2011-10-12 13:04:42.015625 [DEBUG] mod_sofia.c:566 SOFIA EXCHANGE_MEDIA
2011-10-12 13:05:14.015625 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:05:14.015625 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:05:14.031250 [DEBUG] sofia.c:5156 Channel sofia/internal/1010 at 192.168.0.1 entering state [terminating][0]
2011-10-12 13:05:14.031250 [DEBUG] switch_channel.c:2800 (sofia/internal/1010 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-10-12 13:05:14.031250 [NOTICE] sofia.c:5897 Hangup sofia/internal/1010 at 192.168.0.1 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2011-10-12 13:05:14.031250 [DEBUG] switch_channel.c:2816 Send signal sofia/internal/1010 at 192.168.0.1 [KILL]
2011-10-12 13:05:14.031250 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:05:14.031250 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/1010 at 192.168.0.1]
2011-10-12 13:05:14.031250 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:05:14.046875 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:1005 at 192.168.0.3:38232]
2011-10-12 13:05:14.046875 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:05:14.046875 [DEBUG] switch_channel.c:2800 (sofia/internal/sip:1005 at 192.168.0.3:38232) Callstate Change ACTIVE -> HANGUP
2011-10-12 13:05:14.046875 [NOTICE] switch_ivr_bridge.c:658 Hangup sofia/internal/sip:1005 at 192.168.0.3:38232 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2011-10-12 13:05:14.046875 [DEBUG] switch_channel.c:2816 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [KILL]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:1005 at 192.168.0.3:38232) State EXCHANGE_MEDIA going to sleep
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1005 at 192.168.0.3:38232) Running State Change CS_HANGUP
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1005 at 192.168.0.3:38232) State HANGUP
2011-10-12 13:05:14.046875 [DEBUG] mod_sofia.c:458 Channel sofia/internal/sip:1005 at 192.168.0.3:38232 hanging up, cause: NORMAL_CLEARING
2011-10-12 13:05:14.046875 [DEBUG] switch_ivr_bridge.c:1332 sofia/internal/sip:1005 at 192.168.0.3:38232 skip receive message [UNBRIDGE] (channel is hungup already)
2011-10-12 13:05:14.046875 [DEBUG] mod_sofia.c:501 Sending BYE to sofia/internal/sip:1005 at 192.168.0.3:38232
2011-10-12 13:05:14.046875 [DEBUG] switch_ivr_bridge.c:1335 sofia/internal/1010 at 192.168.0.1 skip receive message [UNBRIDGE] (channel is hungup already)
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1005 at 192.168.0.3:38232 Standard HANGUP, cause: NORMAL_CLEARING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1005 at 192.168.0.3:38232) State HANGUP going to sleep
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1005 at 192.168.0.3:38232) State Change CS_HANGUP -> CS_REPORTING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1005 at 192.168.0.3:38232) Running State Change CS_REPORTING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:2233 sofia/internal/1010 at 192.168.0.1 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1010 at 192.168.0.1) State EXECUTE going to sleep
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1010 at 192.168.0.1) Running State Change CS_HANGUP
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1005 at 192.168.0.3:38232) State REPORTING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1005 at 192.168.0.3:38232 Standard REPORTING, cause: NORMAL_CLEARING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1005 at 192.168.0.3:38232) State REPORTING going to sleep
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1005 at 192.168.0.3:38232) State Change CS_REPORTING -> CS_DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1005 at 192.168.0.3:38232 [BREAK]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:1349 Session 2 (sofia/internal/sip:1005 at 192.168.0.3:38232) Locked, Waiting on external entities
2011-10-12 13:05:14.046875 [NOTICE] switch_core_session.c:1367 Session 2 (sofia/internal/sip:1005 at 192.168.0.3:38232) Ended
2011-10-12 13:05:14.046875 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/sip:1005 at 192.168.0.3:38232 [CS_DESTROY]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1005 at 192.168.0.3:38232) Callstate Change HANGUP -> DOWN
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1010 at 192.168.0.1) State HANGUP
2011-10-12 13:05:14.046875 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1010 at 192.168.0.1 hanging up, cause: NORMAL_UNSPECIFIED
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1005 at 192.168.0.3:38232) Running State Change CS_DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1005 at 192.168.0.3:38232) State DESTROY
2011-10-12 13:05:14.046875 [DEBUG] mod_sofia.c:363 sofia/internal/sip:1005 at 192.168.0.3:38232 SOFIA DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1005 at 192.168.0.3:38232 Standard DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1005 at 192.168.0.3:38232) State DESTROY going to sleep
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1010 at 192.168.0.1 Standard HANGUP, cause: NORMAL_UNSPECIFIED
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1010 at 192.168.0.1) State HANGUP going to sleep
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1010 at 192.168.0.1) State Change CS_HANGUP -> CS_REPORTING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1010 at 192.168.0.1) Running State Change CS_REPORTING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1010 at 192.168.0.1) State REPORTING
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1010 at 192.168.0.1 Standard REPORTING, cause: NORMAL_UNSPECIFIED
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1010 at 192.168.0.1) State REPORTING going to sleep
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1010 at 192.168.0.1) State Change CS_REPORTING -> CS_DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1010 at 192.168.0.1 [BREAK]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_session.c:1349 Session 1 (sofia/internal/1010 at 192.168.0.1) Locked, Waiting on external entities
2011-10-12 13:05:14.046875 [NOTICE] switch_core_session.c:1367 Session 1 (sofia/internal/1010 at 192.168.0.1) Ended
2011-10-12 13:05:14.046875 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/1010 at 192.168.0.1 [CS_DESTROY]
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1010 at 192.168.0.1) Callstate Change HANGUP -> DOWN
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1010 at 192.168.0.1) Running State Change CS_DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1010 at 192.168.0.1) State DESTROY
2011-10-12 13:05:14.046875 [DEBUG] mod_sofia.c:363 sofia/internal/1010 at 192.168.0.1 SOFIA DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1010 at 192.168.0.1 Standard DESTROY
2011-10-12 13:05:14.046875 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1010 at 192.168.0.1) State DESTROY going to sleep
2011-10-12 13:06:27.203125 [DEBUG] sofia.c:7124 IP 192.168.0.3 Rejected by acl "domains". Falling back to Digest auth.
2011-10-12 13:06:27.203125 [DEBUG] sofia.c:7124 IP 192.168.0.3 Rejected by acl "domains". Falling back to Digest auth.
2011-10-12 13:06:27.203125 [NOTICE] switch_channel.c:911 New Channel sofia/internal/1005 at 192.168.0.1 [d220a657-910f-42a1-b653-f002d52bf88b]
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1005 at 192.168.0.1) Running State Change CS_NEW
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/1005 at 192.168.0.1) State NEW
2011-10-12 13:06:27.203125 [DEBUG] sofia.c:5156 Channel sofia/internal/1005 at 192.168.0.1 entering state [received][100]
2011-10-12 13:06:27.203125 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=- 12962927374968750 1 IN IP4 192.168.0.3
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.0.3
t=0 0
a=ice-ufrag:1e9c35
a=ice-pwd:df95dbe29c949038f96f335f3d36658c
m=audio 51282 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=candidate:1 1 UDP 659136 192.168.0.3 51282 typ host
a=candidate:1 2 UDP 659134 192.168.0.3 51283 typ host

2011-10-12 13:06:27.203125 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2011-10-12 13:06:27.203125 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2011-10-12 13:06:27.203125 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2011-10-12 13:06:27.203125 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-10-12 13:06:27.203125 [DEBUG] sofia_glue.c:2841 Set Codec sofia/internal/1005 at 192.168.0.1 PCMU/8000 20 ms 160 samples 64000 bits
2011-10-12 13:06:27.203125 [DEBUG] sofia_glue.c:4835 Set 2833 dtmf send/recv payload to 101
2011-10-12 13:06:27.203125 [DEBUG] sofia.c:5366 (sofia/internal/1005 at 192.168.0.1) State Change CS_NEW -> CS_INIT
2011-10-12 13:06:27.203125 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1005 at 192.168.0.1) Running State Change CS_INIT
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1005 at 192.168.0.1) State INIT
2011-10-12 13:06:27.203125 [DEBUG] mod_sofia.c:85 sofia/internal/1005 at 192.168.0.1 SOFIA INIT
2011-10-12 13:06:27.203125 [DEBUG] mod_sofia.c:125 (sofia/internal/1005 at 192.168.0.1) State Change CS_INIT -> CS_ROUTING
2011-10-12 13:06:27.203125 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1005 at 192.168.0.1) State INIT going to sleep
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1005 at 192.168.0.1) Running State Change CS_ROUTING
2011-10-12 13:06:27.203125 [DEBUG] switch_channel.c:1840 (sofia/internal/1005 at 192.168.0.1) Callstate Change DOWN -> RINGING
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1005 at 192.168.0.1) State ROUTING
2011-10-12 13:06:27.203125 [DEBUG] mod_sofia.c:148 sofia/internal/1005 at 192.168.0.1 SOFIA ROUTING
2011-10-12 13:06:27.203125 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1005 at 192.168.0.1 Standard ROUTING
2011-10-12 13:06:27.203125 [INFO] mod_dialplan_xml.c:336 Processing clien2 <1005>->1010 in context default
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1005 at 192.168.0.1 Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(open=true) 
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1005 at 192.168.0.1 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [global-intercept] destination_number(1010) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [group-intercept] destination_number(1010) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [intercept-ext] destination_number(1010) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->redial] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [redial] destination_number(1010) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->global] continue=true
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1005 at 192.168.0.1 Absolute Condition [global]
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [snom-demo-2] destination_number(1010) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [snom-demo-1] destination_number(1010) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [eavesdrop] destination_number(1010) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [eavesdrop] destination_number(1010) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [call_return] destination_number(1010) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [del-group] destination_number(1010) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->add-group] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [add-group] destination_number(1010) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->call-group-simo] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [call-group-simo] destination_number(1010) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->call-group-order] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [call-group-order] destination_number(1010) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->extension-intercom] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (FAIL) [extension-intercom] destination_number(1010) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1005 at 192.168.0.1 Regex (PASS) [Local_Extension] destination_number(1010) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(dialed_extension=1010) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action export(dialed_extension=1010) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action bind_meta_app(1 b s execute_extension::dx XML features) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action bind_meta_app(2 b s record_session::C:\Program Files\FreeSWITCH\recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action bind_meta_app(3 b s execute_extension::cf XML features) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(ringback=${us-ring}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(transfer_ringback=local_stream://moh) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(call_timeout=30) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(hangup_after_bridge=true) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(continue_on_fail=true) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action bridge(user/${dialed_extension}@${domain_name}) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action answer() 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action sleep(1000) 
Dialplan: sofia/internal/1005 at 192.168.0.1 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 
2011-10-12 13:06:27.234375 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1005 at 192.168.0.1) State Change CS_ROUTING -> CS_EXECUTE
2011-10-12 13:06:27.234375 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:27.234375 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1005 at 192.168.0.1) State ROUTING going to sleep
2011-10-12 13:06:27.234375 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1005 at 192.168.0.1) Running State Change CS_EXECUTE
2011-10-12 13:06:27.234375 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1005 at 192.168.0.1) State EXECUTE
2011-10-12 13:06:27.234375 [DEBUG] mod_sofia.c:241 sofia/internal/1005 at 192.168.0.1 SOFIA EXECUTE
2011-10-12 13:06:27.234375 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1005 at 192.168.0.1 Standard EXECUTE
EXECUTE sofia/internal/1005 at 192.168.0.1 set(open=true)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [open]=[true]
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-spymap/1005/d220a657-910f-42a1-b653-f002d52bf88b)
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-last_dial/1005/1010)
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-last_dial/global/d220a657-910f-42a1-b653-f002d52bf88b)
EXECUTE sofia/internal/1005 at 192.168.0.1 set(RFC2822_DATE=Wed, 12 Oct 2011 13:06:27 Iran Standard Time)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [RFC2822_DATE]=[Wed, 12 Oct 2011 13:06:27 Iran Standard Time]
EXECUTE sofia/internal/1005 at 192.168.0.1 set(dialed_extension=1010)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [dialed_extension]=[1010]
EXECUTE sofia/internal/1005 at 192.168.0.1 export(dialed_extension=1010)
2011-10-12 13:06:27.234375 [DEBUG] switch_channel.c:1078 EXPORT (export_vars) [dialed_extension]=[1010]
EXECUTE sofia/internal/1005 at 192.168.0.1 bind_meta_app(1 b s execute_extension::dx XML features)
2011-10-12 13:06:27.234375 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *1 execute_extension::dx XML features
EXECUTE sofia/internal/1005 at 192.168.0.1 bind_meta_app(2 b s record_session::C:\Program Files\FreeSWITCH\recordings/1005.2011-10-12-13-06-27.wav)
2011-10-12 13:06:27.234375 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *2 record_session::C:\Program Files\FreeSWITCH
ecordings/1005.2011-10-12-13-06-27.wav
EXECUTE sofia/internal/1005 at 192.168.0.1 bind_meta_app(3 b s execute_extension::cf XML features)
2011-10-12 13:06:27.234375 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *3 execute_extension::cf XML features
EXECUTE sofia/internal/1005 at 192.168.0.1 bind_meta_app(4 b s execute_extension::att_xfer XML features)
2011-10-12 13:06:27.234375 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *4 execute_extension::att_xfer XML features
EXECUTE sofia/internal/1005 at 192.168.0.1 set(ringback=%(2000,4000,440,480))
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [ringback]=[%(2000,4000,440,480)]
EXECUTE sofia/internal/1005 at 192.168.0.1 set(transfer_ringback=local_stream://moh)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1005 at 192.168.0.1 set(call_timeout=30)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [call_timeout]=[30]
EXECUTE sofia/internal/1005 at 192.168.0.1 set(hangup_after_bridge=true)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1005 at 192.168.0.1 set(continue_on_fail=true)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-call_return/1010/1005)
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-last_dial_ext/1010/d220a657-910f-42a1-b653-f002d52bf88b)
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-last_dial_ext//d220a657-910f-42a1-b653-f002d52bf88b)
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-last_dial_ext/global/d220a657-910f-42a1-b653-f002d52bf88b)
EXECUTE sofia/internal/1005 at 192.168.0.1 set(called_party_callgroup=techsupport)
2011-10-12 13:06:27.234375 [DEBUG] mod_dptools.c:1176 sofia/internal/1005 at 192.168.0.1 SET [called_party_callgroup]=[techsupport]
EXECUTE sofia/internal/1005 at 192.168.0.1 hash(insert/0.0.0.0-last_dial/techsupport/d220a657-910f-42a1-b653-f002d52bf88b)
EXECUTE sofia/internal/1005 at 192.168.0.1 bridge(user/1010 at 0.0.0.0)
2011-10-12 13:06:27.234375 [DEBUG] switch_channel.c:1032 sofia/internal/1005 at 192.168.0.1 EXPORTING[export_vars] [dialed_extension]=[1010] to event
2011-10-12 13:06:27.234375 [DEBUG] switch_ivr_originate.c:1882 Parsing global variables
2011-10-12 13:06:27.234375 [DEBUG] switch_channel.c:1032 sofia/internal/1005 at 192.168.0.1 EXPORTING[export_vars] [dialed_extension]=[1010] to event
2011-10-12 13:06:27.234375 [DEBUG] switch_ivr_originate.c:1882 Parsing global variables
2011-10-12 13:06:27.234375 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[0.0.0.0]
2011-10-12 13:06:27.234375 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1010 at 0.0.0.0]
2011-10-12 13:06:27.234375 [NOTICE] switch_channel.c:911 New Channel sofia/internal/sip:1010 at 192.168.0.2:59286 [b2f7ab69-aecf-48a4-8488-934d9e1645b7]
2011-10-12 13:06:27.234375 [DEBUG] mod_sofia.c:4494 (sofia/internal/sip:1010 at 192.168.0.2:59286) State Change CS_NEW -> CS_INIT
2011-10-12 13:06:27.234375 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:27.234375 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1010 at 192.168.0.2:59286) Running State Change CS_INIT
2011-10-12 13:06:27.234375 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1010 at 192.168.0.2:59286) State INIT
2011-10-12 13:06:27.234375 [DEBUG] mod_sofia.c:85 sofia/internal/sip:1010 at 192.168.0.2:59286 SOFIA INIT
2011-10-12 13:06:27.250000 [DEBUG] sofia_glue.c:2401 sip:1010 at 192.168.0.2:59286;rinstance=0314dd65b25b2e38 Setting proxy route to sofia/internal/sip:1010 at 192.168.0.2:59286
2011-10-12 13:06:27.250000 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:1010 at 192.168.0.2:59286) State Change CS_INIT -> CS_ROUTING
2011-10-12 13:06:27.250000 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:27.250000 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1010 at 192.168.0.2:59286) State INIT going to sleep
2011-10-12 13:06:27.250000 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1010 at 192.168.0.2:59286) Running State Change CS_ROUTING
2011-10-12 13:06:27.250000 [DEBUG] switch_channel.c:1840 (sofia/internal/sip:1010 at 192.168.0.2:59286) Callstate Change DOWN -> RINGING
2011-10-12 13:06:27.250000 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1010 at 192.168.0.2:59286) State ROUTING
2011-10-12 13:06:27.250000 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1010 at 192.168.0.2:59286 SOFIA ROUTING
2011-10-12 13:06:27.250000 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1010 at 192.168.0.2:59286) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-10-12 13:06:27.250000 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:27.250000 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1010 at 192.168.0.2:59286) State ROUTING going to sleep
2011-10-12 13:06:27.250000 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1010 at 192.168.0.2:59286) Running State Change CS_CONSUME_MEDIA
2011-10-12 13:06:27.250000 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1010 at 192.168.0.2:59286) State CONSUME_MEDIA
2011-10-12 13:06:27.250000 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1010 at 192.168.0.2:59286) State CONSUME_MEDIA going to sleep
2011-10-12 13:06:27.250000 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:27.250000 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1010 at 192.168.0.2:59286 entering state [calling][0]
2011-10-12 13:06:27.359375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:27.359375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:27.359375 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1010 at 192.168.0.2:59286 entering state [proceeding][180]
2011-10-12 13:06:27.359375 [NOTICE] sofia.c:5235 Ring-Ready sofia/internal/sip:1010 at 192.168.0.2:59286!
2011-10-12 13:06:27.359375 [INFO] switch_ivr_originate.c:1113 Sending early media
2011-10-12 13:06:27.359375 [DEBUG] sofia_glue.c:3093 AUDIO RTP [sofia/internal/1005 at 192.168.0.1] 0.0.0.0 port 28414 -> 192.168.0.3 port 51282 codec: 0 ms: 20
2011-10-12 13:06:27.359375 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms
2011-10-12 13:06:27.359375 [DEBUG] sofia_glue.c:3356 Set 2833 dtmf send payload to 101
2011-10-12 13:06:27.359375 [DEBUG] sofia_glue.c:3362 Set 2833 dtmf receive payload to 101
2011-10-12 13:06:27.359375 [DEBUG] mod_sofia.c:2461 Ring SDP:
v=0
o=FreeSWITCH 1318383773 1318383774 IN IP4 0.0.0.0
s=FreeSWITCH
c=IN IP4 0.0.0.0
t=0 0
m=audio 28414 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

2011-10-12 13:06:27.359375 [NOTICE] mod_sofia.c:2464 Pre-Answer sofia/internal/1005 at 192.168.0.1!
2011-10-12 13:06:27.359375 [DEBUG] switch_channel.c:2877 (sofia/internal/1005 at 192.168.0.1) Callstate Change RINGING -> EARLY
2011-10-12 13:06:27.359375 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:27.359375 [DEBUG] switch_ivr_originate.c:1162 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2011-10-12 13:06:27.359375 [DEBUG] switch_core_codec.c:116 sofia/internal/1005 at 192.168.0.1 Push codec L16:70
2011-10-12 13:06:27.359375 [DEBUG] switch_ivr_originate.c:1225 Play Ringback Tone [%(2000,4000,440,480)]
2011-10-12 13:06:27.359375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:27.359375 [DEBUG] sofia.c:5149 Channel sofia/internal/1005 at 192.168.0.1 skipping state [early][183]
2011-10-12 13:06:34.812500 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:34.812500 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:34.812500 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1010 at 192.168.0.2:59286 entering state [completing][200]
2011-10-12 13:06:34.812500 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=- 2 2 IN IP4 192.168.0.2
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.0.2
t=0 0
m=audio 21326 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2011-10-12 13:06:34.812500 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:34.812500 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:34.812500 [DEBUG] sofia.c:5156 Channel sofia/internal/sip:1010 at 192.168.0.2:59286 entering state [ready][200]
2011-10-12 13:06:34.812500 [DEBUG] sofia_glue.c:4714 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-10-12 13:06:34.812500 [DEBUG] sofia_glue.c:2841 Set Codec sofia/internal/sip:1010 at 192.168.0.2:59286 PCMU/8000 20 ms 160 samples 64000 bits
2011-10-12 13:06:34.812500 [DEBUG] sofia_glue.c:4828 Set 2833 dtmf send payload to 101
2011-10-12 13:06:34.812500 [DEBUG] sofia_glue.c:3093 AUDIO RTP [sofia/internal/sip:1010 at 192.168.0.2:59286] 0.0.0.0 port 20326 -> 192.168.0.2 port 21326 codec: 0 ms: 20
2011-10-12 13:06:34.812500 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms
2011-10-12 13:06:34.812500 [DEBUG] sofia_glue.c:3356 Set 2833 dtmf send payload to 101
2011-10-12 13:06:34.812500 [DEBUG] sofia_glue.c:3362 Set 2833 dtmf receive payload to 101
2011-10-12 13:06:34.812500 [DEBUG] switch_channel.c:3135 (sofia/internal/sip:1010 at 192.168.0.2:59286) Callstate Change RINGING -> ACTIVE
2011-10-12 13:06:34.812500 [DEBUG] switch_channel.c:3147 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:34.812500 [NOTICE] sofia.c:5841 Channel [sofia/internal/sip:1010 at 192.168.0.2:59286] has been answered
2011-10-12 13:06:34.843750 [DEBUG] switch_core_codec.c:141 sofia/internal/1005 at 192.168.0.1 Restore previous codec PCMU:0.
2011-10-12 13:06:34.843750 [DEBUG] mod_sofia.c:738 Local SDP sofia/internal/1005 at 192.168.0.1:
v=0
o=FreeSWITCH 1318383773 1318383775 IN IP4 0.0.0.0
s=FreeSWITCH
c=IN IP4 0.0.0.0
t=0 0
m=audio 28414 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

2011-10-12 13:06:34.843750 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:34.843750 [DEBUG] switch_channel.c:3135 (sofia/internal/1005 at 192.168.0.1) Callstate Change EARLY -> ACTIVE
2011-10-12 13:06:34.843750 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:34.843750 [NOTICE] switch_ivr_originate.c:3192 Channel [sofia/internal/1005 at 192.168.0.1] has been answered
2011-10-12 13:06:34.843750 [DEBUG] sofia.c:5156 Channel sofia/internal/1005 at 192.168.0.1 entering state [completed][200]
2011-10-12 13:06:34.843750 [DEBUG] switch_ivr_originate.c:3252 Originate Resulted in Success: [sofia/internal/sip:1010 at 192.168.0.2:59286]
2011-10-12 13:06:34.843750 [DEBUG] switch_ivr_originate.c:3252 Originate Resulted in Success: [sofia/internal/sip:1010 at 192.168.0.2:59286]
2011-10-12 13:06:34.843750 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:34.843750 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:06:34.843750 [DEBUG] switch_ivr_bridge.c:1257 (sofia/internal/sip:1010 at 192.168.0.2:59286) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2011-10-12 13:06:34.843750 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:06:34.843750 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1010 at 192.168.0.2:59286) Running State Change CS_EXCHANGE_MEDIA
2011-10-12 13:06:34.843750 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:1010 at 192.168.0.2:59286) State EXCHANGE_MEDIA
2011-10-12 13:06:34.843750 [DEBUG] mod_sofia.c:566 SOFIA EXCHANGE_MEDIA
2011-10-12 13:07:06.859375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.859375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.859375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.859375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.859375 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.875000 [DEBUG] sofia.c:5156 Channel sofia/internal/1005 at 192.168.0.1 entering state [terminating][0]
2011-10-12 13:07:06.875000 [DEBUG] switch_channel.c:2800 (sofia/internal/1005 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP
2011-10-12 13:07:06.875000 [NOTICE] sofia.c:5897 Hangup sofia/internal/1005 at 192.168.0.1 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2011-10-12 13:07:06.875000 [DEBUG] switch_channel.c:2816 Send signal sofia/internal/1005 at 192.168.0.1 [KILL]
2011-10-12 13:07:06.875000 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.875000 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/1005 at 192.168.0.1]
2011-10-12 13:07:06.875000 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:07:06.890625 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:1010 at 192.168.0.2:59286]
2011-10-12 13:07:06.890625 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.890625 [DEBUG] switch_channel.c:2800 (sofia/internal/sip:1010 at 192.168.0.2:59286) Callstate Change ACTIVE -> HANGUP
2011-10-12 13:07:06.890625 [NOTICE] switch_ivr_bridge.c:658 Hangup sofia/internal/sip:1010 at 192.168.0.2:59286 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2011-10-12 13:07:06.890625 [DEBUG] switch_channel.c:2816 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [KILL]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:1010 at 192.168.0.2:59286) State EXCHANGE_MEDIA going to sleep
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1010 at 192.168.0.2:59286) Running State Change CS_HANGUP
2011-10-12 13:07:06.890625 [DEBUG] switch_ivr_bridge.c:1332 sofia/internal/sip:1010 at 192.168.0.2:59286 skip receive message [UNBRIDGE] (channel is hungup already)
2011-10-12 13:07:06.890625 [DEBUG] switch_ivr_bridge.c:1335 sofia/internal/1005 at 192.168.0.1 skip receive message [UNBRIDGE] (channel is hungup already)
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:2233 sofia/internal/1005 at 192.168.0.1 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1005 at 192.168.0.1) State EXECUTE going to sleep
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1005 at 192.168.0.1) Running State Change CS_HANGUP
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1010 at 192.168.0.2:59286) State HANGUP
2011-10-12 13:07:06.890625 [DEBUG] mod_sofia.c:458 Channel sofia/internal/sip:1010 at 192.168.0.2:59286 hanging up, cause: NORMAL_CLEARING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1005 at 192.168.0.1) State HANGUP
2011-10-12 13:07:06.890625 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1005 at 192.168.0.1 hanging up, cause: NORMAL_UNSPECIFIED
2011-10-12 13:07:06.890625 [DEBUG] mod_sofia.c:501 Sending BYE to sofia/internal/sip:1010 at 192.168.0.2:59286
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1010 at 192.168.0.2:59286 Standard HANGUP, cause: NORMAL_CLEARING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1010 at 192.168.0.2:59286) State HANGUP going to sleep
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1010 at 192.168.0.2:59286) State Change CS_HANGUP -> CS_REPORTING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1010 at 192.168.0.2:59286) Running State Change CS_REPORTING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1010 at 192.168.0.2:59286) State REPORTING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1010 at 192.168.0.2:59286 Standard REPORTING, cause: NORMAL_CLEARING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1010 at 192.168.0.2:59286) State REPORTING going to sleep
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1005 at 192.168.0.1 Standard HANGUP, cause: NORMAL_UNSPECIFIED
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1005 at 192.168.0.1) State HANGUP going to sleep
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1010 at 192.168.0.2:59286) State Change CS_REPORTING -> CS_DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1010 at 192.168.0.2:59286 [BREAK]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:1349 Session 4 (sofia/internal/sip:1010 at 192.168.0.2:59286) Locked, Waiting on external entities
2011-10-12 13:07:06.890625 [NOTICE] switch_core_session.c:1367 Session 4 (sofia/internal/sip:1010 at 192.168.0.2:59286) Ended
2011-10-12 13:07:06.890625 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/sip:1010 at 192.168.0.2:59286 [CS_DESTROY]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1005 at 192.168.0.1) State Change CS_HANGUP -> CS_REPORTING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1005 at 192.168.0.1) Running State Change CS_REPORTING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1010 at 192.168.0.2:59286) Callstate Change HANGUP -> DOWN
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1005 at 192.168.0.1) State REPORTING
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1010 at 192.168.0.2:59286) Running State Change CS_DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1005 at 192.168.0.1 Standard REPORTING, cause: NORMAL_UNSPECIFIED
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1005 at 192.168.0.1) State REPORTING going to sleep
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1010 at 192.168.0.2:59286) State DESTROY
2011-10-12 13:07:06.890625 [DEBUG] mod_sofia.c:363 sofia/internal/sip:1010 at 192.168.0.2:59286 SOFIA DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1010 at 192.168.0.2:59286 Standard DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1010 at 192.168.0.2:59286) State DESTROY going to sleep
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1005 at 192.168.0.1) State Change CS_REPORTING -> CS_DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/1005 at 192.168.0.1 [BREAK]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_session.c:1349 Session 3 (sofia/internal/1005 at 192.168.0.1) Locked, Waiting on external entities
2011-10-12 13:07:06.890625 [NOTICE] switch_core_session.c:1367 Session 3 (sofia/internal/1005 at 192.168.0.1) Ended
2011-10-12 13:07:06.890625 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/1005 at 192.168.0.1 [CS_DESTROY]
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1005 at 192.168.0.1) Callstate Change HANGUP -> DOWN
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1005 at 192.168.0.1) Running State Change CS_DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1005 at 192.168.0.1) State DESTROY
2011-10-12 13:07:06.890625 [DEBUG] mod_sofia.c:363 sofia/internal/1005 at 192.168.0.1 SOFIA DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1005 at 192.168.0.1 Standard DESTROY
2011-10-12 13:07:06.890625 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1005 at 192.168.0.1) State DESTROY going to sleep

From anthony.minessale at gmail.com  Wed Oct 12 19:51:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Oct 2011 10:51:10 -0500
Subject: [Freeswitch-users] Call/Transaction Does Not Exist
In-Reply-To: <CALH5ZrbFUBNMrBEwN+uJMO+ee0R=X0=g00rUi6+jPuOT=Y17SQ@mail.gmail.com>
References: <CALH5ZrbFUBNMrBEwN+uJMO+ee0R=X0=g00rUi6+jPuOT=Y17SQ@mail.gmail.com>
Message-ID: <CAKbxfG9zcByP3un6RJNRUuw4hJhSd+KZ+tYmLbQd+WmFPwxc8A@mail.gmail.com>

collect a full trace and put it on pastebin.freeswitch.org

console loglevel debug (/log debug on fs_cli)
sofia global siptrace on




On Tue, Oct 11, 2011 at 4:29 PM, Joe Flemmings <joe.jflemmings at gmail.com> wrote:
> When i call from one extension and hangup, the other call does not hangup
> and continues ringing. Please see log bellow.
>
> Any ideas?
>
>
>
> U 10.10.78.24:55479 -> 10.10.78.36:5060
> ? INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> ? <sip:213 at 10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
> OPTIONS, INFO, MES
> ? SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported:
> 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards:
> ? 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
> 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN IP4
> 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
> ? 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
> telephone-event/8000..
> #
> U 10.10.78.36:5060 -> 10.10.78.24:52732
> ? SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> <sip:213 at 10.10.78.36;user=phone>
> ? ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1
> INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
> 22-51-43 -0500..Content-Length: 0....
> #
> U 10.10.78.36:5060 -> 10.10.78.24:52732
> ? SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
> 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 6
> ? 6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 INVITE..User-Agent:
> FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Accept:
> appl
> ? ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
> precondition, path, replaces..Allow-Events: talk, hold, prese
> ? nce, dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer..Proxy-Authenticate: Digest
> realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac
> ? ", algorithm=MD5, qop="auth"..Content-Length: 0....
> #
> U 10.10.78.24:55479 -> 10.10.78.36:5060
> ? ACK sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <si
> ? p:213 at 10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 ACK..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
> OPTION
> ? S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
> en..Max-Forwards: 70..Content-Length: 0....
> #
> U 10.10.78.24:55479 -> 10.10.78.36:5060
> ? INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> ? <sip:213 at 10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
> OPTIONS, INFO, MES
> ? SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported:
> 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz
> ? ation: Digest username="pbxjker_2564", realm="10.10.78.36",
> nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth,
> cnonce="KnfhF00Oi0GfJCi", nc=00000001,
> uri="sip:213 at 10.10.78.36:5060;user=phone",
> ? response="82a75edb7ba97bfe7dea585684701d0c", algorithm=MD5..Max-Forwards:
> 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
> 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone
> ? ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
> 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
> ? telephone-event/8000..
> #
> U 10.10.78.36:5060 -> 10.10.78.24:52732
> ? SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> <sip:213 at 10.10.78.36;user=phone>
> ? ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
> INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
> 22-51-43 -0500..Content-Length: 0....
> #
> U 10.10.78.36:5060 -> 10.10.78.24:52783
> ? INVITE sip:pbxjker_213 at 10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP
> 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: 69..From:
> "Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81
> ? rv8Fp..To: <sip:pbxjker_213 at 10.10.78.24:52783>..Call-ID:
> 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact:
> <sip:mod_sofia at 10.10.78.36:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.
> ? head-git-1086cba 2011-05-23 22-51-43 -0500..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-Type: application/sdp..Content-Disposition: session..
> ? Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID:
> "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
> 1318333156 1318333157 IN IP4
> ?? 10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio 29124
> RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98
> bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320
> ? 00..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
> 0-16..a=ptime:20..
> #
> U 10.10.78.24:55516 -> 10.10.78.36:5060
> ? SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
> <sip:pbxjker_213 at 99.65.2
> ? 17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
> 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
> <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
> PolycomSoundPointIP-SPIP_320-UA/3.2
> ? .2.0477..Accept-Language: en..Content-Length: 0....
> #
> U 10.10.78.24:55516 -> 10.10.78.36:5060
> ? SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
> <sip:pbxjker_213 at 99.65.
> ? 217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
> 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
> <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
> PolycomSoundPointIP-SPIP_320-UA/3.
> ? 2.2.0477..Allow-Events: talk,hold,conference..Accept-Language:
> en..Content-Length: 0....
> #
> U 10.10.78.36:5060 -> 10.10.78.24:52732
> ? SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> <sip:213 at 10.10.78.36;user=phone
> ? >;tag=8aQNK2QyUKjXa..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 INVITE..Contact:
> <sip:213 at 10.10.78.36:5060;transport=udp>..User-Agent:
> FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2
> ? 3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE,
> CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
> SUBSCRIBE..Supported: timer, precondition, path, replaces..A
> ? llow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary,
> refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker
> ? _213>;party=calling;privacy=off;screen=no....
> #
> U 10.10.78.24:55479 -> 10.10.78.36:5060
> ? CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> ? 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
> INFO, MESSAGE, SUBSCR
> ? IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
> username="pbxjker_2564", realm="10.10.78.36",
> nonce="ab4a9421-fbb9-41fd-943d-462
> ? 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> uri="sip:213 at 10.10.78.36:5060;user=phone",
> response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
> 70..Content-Length: 0..
> ? ..
> #
> U 10.10.78.36:5060 -> 10.10.78.24:5060
> ? SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
> ? 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
> 0....
> #
> U 10.10.78.24:55479 -> 10.10.78.36:5060
> ? CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> ? 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
> INFO, MESSAGE, SUBSCR
> ? IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
> username="pbxjker_2564", realm="10.10.78.36",
> nonce="ab4a9421-fbb9-41fd-943d-462
> ? 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> uri="sip:213 at 10.10.78.36:5060;user=phone",
> response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
> 70..Content-Length: 0..
> ? ..
> #
> U 10.10.78.36:5060 -> 10.10.78.24:5060
> ? SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
> ? 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
> 0....
> #
> U 10.10.78.24:55479 -> 10.10.78.36:5060
> ? CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> ? 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
> INFO, MESSAGE, SUBSCR
> ? IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
> username="pbxjker_2564", realm="10.10.78.36",
> nonce="ab4a9421-fbb9-41fd-943d-462
> ? 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> uri="sip:213 at 10.10.78.36:5060;user=phone",
> response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
> 70..Content-Length: 0..
> ? ..
> #
> U 10.10.78.36:5060 -> 10.10.78.24:5060
> ? SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
> ? 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
> 0....
> #
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Wed Oct 12 21:05:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Oct 2011 10:05:05 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Right Now! - Phil
	Zimmermann
Message-ID: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>

Hey all,

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

-Michael
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From alec.taylor6 at gmail.com  Wed Oct 12 21:24:54 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Thu, 13 Oct 2011 04:24:54 +1100
Subject: [Freeswitch-users] FreeSWITCH Conference Call Right Now! - Phil
	Zimmermann
In-Reply-To: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>
References: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>
Message-ID: <CAO+9iGfQDHSygLaXUC3+AC5mE0qHFSzd_PENke_Gmp7Xi58rkQ@mail.gmail.com>

link?

(4:24 am here, arghh)

On Thu, Oct 13, 2011 at 4:05 AM, Michael Collins <msc at freeswitch.org> wrote:
> Hey all,
> It seems my first email to the list didn't make it. Just a reminder that
> Philip Zimmermann of PGP fame is coming in to talk about VoIP security, ZRTP
> and Zfone. Please join us now!
> -Michael
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cmrienzo at gmail.com  Wed Oct 12 21:29:19 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 12 Oct 2011 13:29:19 -0400
Subject: [Freeswitch-users] FreeSWITCH Conference Call Right Now! - Phil
	Zimmermann
In-Reply-To: <CAO+9iGfQDHSygLaXUC3+AC5mE0qHFSzd_PENke_Gmp7Xi58rkQ@mail.gmail.com>
References: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>
	<CAO+9iGfQDHSygLaXUC3+AC5mE0qHFSzd_PENke_Gmp7Xi58rkQ@mail.gmail.com>
Message-ID: <CAD-qRki0TsK07sZ_ut3zFJ2JpUL+-CRj-VC1SqPxecRZwA6K+w@mail.gmail.com>

http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_12

On Wed, Oct 12, 2011 at 1:24 PM, Alec Taylor <alec.taylor6 at gmail.com> wrote:

> link?
>
> (4:24 am here, arghh)
>
> On Thu, Oct 13, 2011 at 4:05 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Hey all,
> > It seems my first email to the list didn't make it. Just a reminder that
> > Philip Zimmermann of PGP fame is coming in to talk about VoIP security,
> ZRTP
> > and Zfone. Please join us now!
> > -Michael
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From johnrose at google.hm  Wed Oct 12 22:06:29 2011
From: johnrose at google.hm (John Rose)
Date: Wed, 12 Oct 2011 14:06:29 -0400
Subject: [Freeswitch-users] Is Bgapi threadsafe?
Message-ID: <001401cc8909$ab2e48f0$018adad0$@google.hm>

Does anyone know if calling Bgapi on an ESL connection is threadsafe? ?I?m
calling it from ManagedESL via C# and have a lock around it but would like
to remove the lock if possible


John



From fs-list at communicatefreely.net  Wed Oct 12 22:10:46 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 12 Oct 2011 14:10:46 -0400
Subject: [Freeswitch-users] User management
In-Reply-To: <0EDCF8FF-EAD8-4F3B-B974-0A19FD953379@lyonl.com>
References: <ACCA0268-D0F0-42A0-A755-43745D9E1570@lyonl.com>	<4E85440B.4020203@livecall.com>
	<0EDCF8FF-EAD8-4F3B-B974-0A19FD953379@lyonl.com>
Message-ID: <4E95D826.2060307@communicatefreely.net>

You can still do that with mod_xml_curl

Freeswitch will give your server side script all the details about the 
authentication so you can decide how to handle them.  This may include 
checking them against a database table, or tracking statistics about 
them before deciding whether or not to return credentials and allow that 
client to register or authenticate.

If you really don't want to use curl and insist that it runs on the same 
box, you can do something very similar with lua.  There is a way to bind 
a lua script to directory lookups, so whenever a SIP auth happens, the 
xml is fed to your lua script and you can make similar decisions.

I just like curl because it offloads all that onto another machine, and 
it's really easy to implement in a clustered environment.

-Tim

Chad Vogel wrote:
> I'm not looking to use curl, I would like to use some kind of scripting so i can apply custom rules to track and reject authentication based on rate and origin of authentication from source ip address
>
> On Sep 29, 2011, at 11:22 PM, Jack wrote:
>
>   
>> Hi Chad,
>>  mod_curl works great for that....
>> Jack
>>
>> On 9/29/2011 5:47 PM, Chad Vogel wrote:
>>     
>>> Hello,
>>>
>>> I am wondering if anyone has created any scripts to manage user authentication using a database?
>>>
>>> Chad
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From fs-list at communicatefreely.net  Wed Oct 12 22:13:23 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 12 Oct 2011 14:13:23 -0400
Subject: [Freeswitch-users] Voicemail counts saved messages wrong.
In-Reply-To: <CAGH8SeY5roEsVXhfqozMuiTS8HOt9t2-Js64YLG8v32Y6tvogA@mail.gmail.com>
References: <CAGH8SeY5roEsVXhfqozMuiTS8HOt9t2-Js64YLG8v32Y6tvogA@mail.gmail.com>
Message-ID: <4E95D8C3.2040704@communicatefreely.net>

Check the database.

The message data is stored in sql - the file system is only used for audio.

Try vm_list from the cli and see what messages are shown.

-Tim

Henrik Aagaard S?rensen wrote:
> I'm running FreeSwitch 1.0.head on Ubuntu 10.04 LTS.
>
> Everything seems to work pretty well, but I'm experience some problems 
> with the voicemail and how it counts saved messages.
>
> Currently I have 2 saved messages, but when it announce the number of 
> messages in the voicemail it says "You have 3 saved messages".
> Looking in the directory storage/... there's also only 2 messages.
>
> So how come it counts that as 3 messages?
> ------------------------------------------------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 hrqiang at gmail.com  Wed Oct 12 22:28:23 2011
From: hrqiang at gmail.com (RuiQiang Huang)
Date: Wed, 12 Oct 2011 11:28:23 -0700
Subject: [Freeswitch-users] bypass media and media proxy
In-Reply-To: <CABA87D2.1F182%krice@freeswitch.org>
References: <CANN-RPTw=J=sCJjSGL8hsGxvgAGkAM60FVLnDN3Ptdk_Qz10EQ@mail.gmail.com>
	<CABA87D2.1F182%krice@freeswitch.org>
Message-ID: <CANN-RPRWoV8L1AuPhrTsEWV_R0kNi1ugBysMXEAUPtVNT9=RCg@mail.gmail.com>

Hi, Ken

I have
    <param name="inbound-proxy-media" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
in internal.xml. However, when my two endpoints in the same network call
each other, instead of peer to peer, audio still goes to freeswitch. I know
that I may be able to achieve this by this action in my dialplan
<action application="set" data="bypass_media=true"/>
For that I have to statically configure which endpoint calls which goes
bypass media, but I want it more intelligent, automatically
detect two endpoint in the same network or not and alternative between proxy
media and bypass media. Anything I'm missing or suggestion? Thanks.


On Tue, Oct 11, 2011 at 9:54 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Late negotiation... Its on the wiki
>
>
>
> On 10/11/11 11:52 PM, "RuiQiang Huang" <hrqiang at gmail.com> wrote:
>
> Hi,
>
> I want my server sometimes does media proxy and sometimes bypass media.
> For example, I have A, B and C.
> A and B always in same internal network. C is outside.
> When A call B, I want to bypass media, so that media peer to peer.
> A call C will be media proxy.
> I know ICE can do that. But unfortunately, all my endpoint doesn't support
> ICE.
> I'm wondering is there anyway to configure this. Thanks
>
> ------------------------------
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From admharris at gmx.com  Wed Oct 12 21:24:55 2011
From: admharris at gmx.com (adam harris)
Date: Wed, 12 Oct 2011 19:24:55 +0200
Subject: [Freeswitch-users] setEventLock in c
Message-ID: <20111012172456.199130@gmx.com>

What is the equivilent fuction in c to setEventLock


From fcintrono at gmail.com  Wed Oct 12 22:50:19 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Wed, 12 Oct 2011 13:50:19 -0500
Subject: [Freeswitch-users] Newbie trouble with default config extension
	9198...
Message-ID: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>

Hi there. :)
After spending hours trying to register Xlite 4.1 in freeswitch 1.0.6 both
installed in my Windows 7 pro rig I gave up.

Now I installed Freeswtich 1.0.6 in a centos 5.7 x86 virtual machine with
the help of freeswitch book.

Xlite is registering fine but now the problem is this:

When  I dial 9198 extension Xlite tells me "Failed to establish call" and
This log messages are written in freeswitch console:

freeswitch at pbx> 2011-10-12 09:01:17.469489 [NOTICE] switch_channel.c:669 New
Channel sofia/internal/1000 at 10.10.100.35[e3f7697a-74ff-46a6-a3ac-c5b95f58d103]
2011-10-12 09:01:17.474982 [INFO] mod_dialplan_xml.c:418 Processing
1000->9198 in context default
2011-10-12 09:01:17.480554 [NOTICE] switch_ivr.c:1447 Transfer
sofia/internal/1000 at 10.10.100.35 to enum[9198 at default]
2011-10-12 09:01:17.497178 [INFO] switch_core_state_machine.c:142 No Route,
Aborting
2011-10-12 09:01:17.497178 [NOTICE] switch_core_state_machine.c:143 Hangup
sofia/internal/1000 at 10.10.100.35 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2011-10-12 09:01:17.502715 [NOTICE] switch_core_session.c:1182 Session 6
(sofia/internal/1000 at 10.10.100.35) Ended
2011-10-12 09:01:17.504038 [NOTICE] switch_core_session.c:1184 Close Channel
sofia/internal/1000 at 10.10.100.35 [CS_DESTROY]


It looks that 9198 extension is not defined.
This is the default configuration I haven't change anything.

If it helps extension 5000 is working.

What am I doing wrong?

Thanks in advance for your  help and have a good day.
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From anthony.minessale at gmail.com  Thu Oct 13 00:40:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Oct 2011 15:40:56 -0500
Subject: [Freeswitch-users] setEventLock in c
In-Reply-To: <20111012172456.199130@gmx.com>
References: <20111012172456.199130@gmx.com>
Message-ID: <CAKbxfG9wCYLGdw7DNhi86L7aEyMhF03w9pTWVeeXwjyYTRc3Pw@mail.gmail.com>

you just work

event-lock: true

into the headers of any sendmsg


On Wed, Oct 12, 2011 at 12:24 PM, adam harris <admharris at gmx.com> wrote:
> What is the equivilent fuction in c to setEventLock
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Oct 13 01:06:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Oct 2011 16:06:47 -0500
Subject: [Freeswitch-users] Is Bgapi threadsafe?
In-Reply-To: <001401cc8909$ab2e48f0$018adad0$@google.hm>
References: <001401cc8909$ab2e48f0$018adad0$@google.hm>
Message-ID: <CAKbxfG9aNd-3Jmims5gmCiZA_UOcm9giG8Zo3F9QPZTWhEh6Zw@mail.gmail.com>

yes since its a socket and the other side is properly threaded.  If
you are using the in-tree ESL lib the client also has race condition
protection

On Wed, Oct 12, 2011 at 1:06 PM, John Rose <johnrose at google.hm> wrote:
> Does anyone know if calling Bgapi on an ESL connection is threadsafe? ?I?m
> calling it from ManagedESL via C# and have a lock around it but would like
> to remove the lock if possible?
>
> John
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Oct 13 01:07:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Oct 2011 16:07:42 -0500
Subject: [Freeswitch-users] Is Bgapi threadsafe?
In-Reply-To: <CAKbxfG9aNd-3Jmims5gmCiZA_UOcm9giG8Zo3F9QPZTWhEh6Zw@mail.gmail.com>
References: <001401cc8909$ab2e48f0$018adad0$@google.hm>
	<CAKbxfG9aNd-3Jmims5gmCiZA_UOcm9giG8Zo3F9QPZTWhEh6Zw@mail.gmail.com>
Message-ID: <CAKbxfG_64fBccqjf6kddF6bsPnyy=VzGWGM9zOuq6jmhLT8=VQ@mail.gmail.com>

oh but wait...

if you mean you want threads on your side to share it, I'd either give
them all their own socket or share one on a mutex cos it's a socket
and you can do i/o on it from 2 threads at once naturally.


On Wed, Oct 12, 2011 at 4:06 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> yes since its a socket and the other side is properly threaded. ?If
> you are using the in-tree ESL lib the client also has race condition
> protection
>
> On Wed, Oct 12, 2011 at 1:06 PM, John Rose <johnrose at google.hm> wrote:
>> Does anyone know if calling Bgapi on an ESL connection is threadsafe? ?I?m
>> calling it from ManagedESL via C# and have a lock around it but would like
>> to remove the lock if possible?
>>
>> John
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Oct 13 01:08:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Oct 2011 16:08:47 -0500
Subject: [Freeswitch-users] Is Bgapi threadsafe?
In-Reply-To: <CAKbxfG_64fBccqjf6kddF6bsPnyy=VzGWGM9zOuq6jmhLT8=VQ@mail.gmail.com>
References: <001401cc8909$ab2e48f0$018adad0$@google.hm>
	<CAKbxfG9aNd-3Jmims5gmCiZA_UOcm9giG8Zo3F9QPZTWhEh6Zw@mail.gmail.com>
	<CAKbxfG_64fBccqjf6kddF6bsPnyy=VzGWGM9zOuq6jmhLT8=VQ@mail.gmail.com>
Message-ID: <CAKbxfG9A0dnxcYioFDTjYf1ire7ALiKmmFbpghkjOxBWooi1=g@mail.gmail.com>

wrong week to quit sniffing glue....

correction to last sentence of previous email:

and you **can't** do i/o on it from 2 threads at once naturally.

On Wed, Oct 12, 2011 at 4:07 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> oh but wait...
>
> if you mean you want threads on your side to share it, I'd either give
> them all their own socket or share one on a mutex cos it's a socket
> and you can do i/o on it from 2 threads at once naturally.
>
>
> On Wed, Oct 12, 2011 at 4:06 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> yes since its a socket and the other side is properly threaded. ?If
>> you are using the in-tree ESL lib the client also has race condition
>> protection
>>
>> On Wed, Oct 12, 2011 at 1:06 PM, John Rose <johnrose at google.hm> wrote:
>>> Does anyone know if calling Bgapi on an ESL connection is threadsafe? ?I?m
>>> calling it from ManagedESL via C# and have a lock around it but would like
>>> to remove the lock if possible?
>>>
>>> John
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
>



-- 
Anthony Minessale II

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

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

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


From cmrienzo at gmail.com  Thu Oct 13 03:54:32 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 12 Oct 2011 19:54:32 -0400
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <CAAAG0NVaYJbqUCRV76rTkEucWybZ1RzGpRanCqmSnV7E6VP_dA@mail.gmail.com>
References: <20111011091700.16181.qmail@community22.interfree.it>
	<CAAAG0NVaYJbqUCRV76rTkEucWybZ1RzGpRanCqmSnV7E6VP_dA@mail.gmail.com>
Message-ID: <CAD-qRkjApPOappx4CFaqn1=0qpFbXU-Di=WPCMBK1LKM963_Dw@mail.gmail.com>

Sirvan resolved this issue on irc.  local_ip_v4 was 0.0.0.0, so SDP looked
like a hold SDP to x-lite.  Setting rtp-ip and sip-ip to 192.168.0.1 fixed
it.



On Wed, Oct 12, 2011 at 8:15 AM, sirvan paraste <sirvan.paraste at gmail.com>wrote:

> Dear
>
> Iused this commands to get log :
> sodia loglevel all 9
> sofia global siptrace on
> sofia tracelevel info
>
> now, the resulted log file is attached to this mail. thank you for your
> regard
> another thing that is strange is when B call C party and C answering the
> call this message  be shown on softphone at the start and then there is no
> way voice : "on hold by another party" ,
>
> please help mo to find its solution. I am overwhelmed with it about 3 weeks
> :(((((
>
> thank you
>
>
> On Tue, Oct 11, 2011 at 12:47 PM, <chrisbware at interfree.it> wrote:
>
>> Send a SIP trace of a call.
>> Use freeswitch sofia debug (see on wiki) or install ngrep (if you are on
>> linux) and use the following command
>>
>> ngrep port 5060 -Wbyline -q -d any
>>
>>
>> Attach the output on this list.
>>
>>
>>
>>
>>
>> Chrisbware
>>
>> -----Messaggio originale-----
>> Da: sirvan paraste <sirvan.paraste at gmail.com>
>> Inviato il: 9 Ott 2011 - 22:38
>> A: freeswitch-users at lists.freeswitch.org
>>
>>
>> Dear all
>> After regard for your attention to me
>> I am student how want work on freeswitch. I have read freeSwitch
>>
>> documents (introduction, how to install, use default
>> confuguration,...).
>> for start of using freeswitch, I make three computer network together
>> with a hub, after that,I install freeswitch on one of them and X-Lite
>> on another computers, network checked with sharing file and ping
>> issue.
>> After registering of two X-lite hone with No. 1001,1002, they can call
>>
>> each other but there is no voice !!! :(
>> i have checked and did some work but the problem is not solved.
>>
>> Can you help me?
>>
>> thank you.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> -------------------------------------------------------------------------------
>> Valore legale alle tue mail
>> InterfreePEC - la tua Posta Elettronica Certificata
>> http://pec.interfree.it
>>
>> -------------------------------------------------------------------------------
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chenzhanping at gmail.com  Thu Oct 13 04:04:46 2011
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Thu, 13 Oct 2011 08:04:46 +0800
Subject: [Freeswitch-users] Audio is not flowing ,
 log is mod_sofia.c:1117 Asynchronous PTIME not supported,
 changing our end from 30 to 20
Message-ID: <CALuT7K8aPSwOg8HNjsGHPrV2JdYjnJYGjYS7iDYJTCFhmqFM0w@mail.gmail.com>

I use FS ,codec is ulaw ,when other softswitch call in,the audio is not
flowing and distortion.

the log is :

http://pastebin.freeswitch.org/17517

I change my codec ulaw packet time to 30 ms (PCMU at 30i) in external.xml,but
the audio is not flowing also.

I add  <param name="rtp-autofix-timing" value="false"/> to external.xml, it
no useful.

Someone can tell me ,Why is it?
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From gcd at i.ph  Thu Oct 13 04:23:13 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 13 Oct 2011 08:23:13 +0800
Subject: [Freeswitch-users] Newbie trouble with default config extension
	9198...
In-Reply-To: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
References: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
Message-ID: <CAHKqMb6axtRCGfNXe3Cr11=yKigfcb_ELuHt6dnpkeD2w3pbkg@mail.gmail.com>

9198 is not defined so it goes to the default destination enum. try dialing
music on hold or echo test.  check the default.xml entries.


On Thu, Oct 13, 2011 at 2:50 AM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Hi there. :)
> After spending hours trying to register Xlite 4.1 in freeswitch 1.0.6 both
> installed in my Windows 7 pro rig I gave up.
>
> Now I installed Freeswtich 1.0.6 in a centos 5.7 x86 virtual machine with
> the help of freeswitch book.
>
> Xlite is registering fine but now the problem is this:
>
> When  I dial 9198 extension Xlite tells me "Failed to establish call" and
> This log messages are written in freeswitch console:
>
> freeswitch at pbx> 2011-10-12 09:01:17.469489 [NOTICE] switch_channel.c:669
> New Channel sofia/internal/1000 at 10.10.100.35[e3f7697a-74ff-46a6-a3ac-c5b95f58d103]
> 2011-10-12 09:01:17.474982 [INFO] mod_dialplan_xml.c:418 Processing
> 1000->9198 in context default
> 2011-10-12 09:01:17.480554 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/internal/1000 at 10.10.100.35 to enum[9198 at default]
> 2011-10-12 09:01:17.497178 [INFO] switch_core_state_machine.c:142 No
> Route, Aborting
> 2011-10-12 09:01:17.497178 [NOTICE] switch_core_state_machine.c:143 Hangup
> sofia/internal/1000 at 10.10.100.35 [CS_ROUTING] [NO_ROUTE_DESTINATION]
> 2011-10-12 09:01:17.502715 [NOTICE] switch_core_session.c:1182 Session 6
> (sofia/internal/1000 at 10.10.100.35) Ended
> 2011-10-12 09:01:17.504038 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1000 at 10.10.100.35 [CS_DESTROY]
>
>
> It looks that 9198 extension is not defined.
> This is the default configuration I haven't change anything.
>
> If it helps extension 5000 is working.
>
> What am I doing wrong?
>
> Thanks in advance for your  help and have a good day.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From joe.jflemmings at gmail.com  Thu Oct 13 05:34:43 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Wed, 12 Oct 2011 18:34:43 -0700
Subject: [Freeswitch-users] Call/Transaction Does Not Exist
In-Reply-To: <CAKbxfG9zcByP3un6RJNRUuw4hJhSd+KZ+tYmLbQd+WmFPwxc8A@mail.gmail.com>
References: <CALH5ZrbFUBNMrBEwN+uJMO+ee0R=X0=g00rUi6+jPuOT=Y17SQ@mail.gmail.com>
	<CAKbxfG9zcByP3un6RJNRUuw4hJhSd+KZ+tYmLbQd+WmFPwxc8A@mail.gmail.com>
Message-ID: <CALH5ZrbxMHUFdBzv_du4t+ji_Hcp9y5qiNs4mvGm86uCsq4wkw@mail.gmail.com>

Thanks for the help Anthony,

Bellow is the pastebin URL

http://pastebin.freeswitch.org/17518

Joe

On Wed, Oct 12, 2011 at 8:51 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> collect a full trace and put it on pastebin.freeswitch.org
>
> console loglevel debug (/log debug on fs_cli)
> sofia global siptrace on
>
>
>
>
> On Tue, Oct 11, 2011 at 4:29 PM, Joe Flemmings <joe.jflemmings at gmail.com>
> wrote:
> > When i call from one extension and hangup, the other call does not hangup
> > and continues ringing. Please see log bellow.
> >
> > Any ideas?
> >
> >
> >
> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >   INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> >   <sip:213 at 10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
> > OPTIONS, INFO, MES
> >   SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
> en..Supported:
> > 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards:
> >   70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN IP4
> > 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
> >   101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
> > telephone-event/8000..
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> > <sip:213 at 10.10.78.36;user=phone>
> >   ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1
> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
> > 22-51-43 -0500..Content-Length: 0....
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >   SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 6
> >   6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 INVITE..User-Agent:
> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43
> -0500..Accept:
> > appl
> >   ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> > INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
> > precondition, path, replaces..Allow-Events: talk, hold, prese
> >   nce, dialog, line-seize, call-info, sla, include-session-description,
> > presence.winfo, message-summary, refer..Proxy-Authenticate: Digest
> > realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac
> >   ", algorithm=MD5, qop="auth"..Content-Length: 0....
> > #
> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >   ACK sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <si
> >   p:213 at 10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 ACK..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
> > OPTION
> >   S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
> > en..Max-Forwards: 70..Content-Length: 0....
> > #
> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >   INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> >   <sip:213 at 10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
> > OPTIONS, INFO, MES
> >   SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
> en..Supported:
> > 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz
> >   ation: Digest username="pbxjker_2564", realm="10.10.78.36",
> > nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth,
> > cnonce="KnfhF00Oi0GfJCi", nc=00000001,
> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> >   response="82a75edb7ba97bfe7dea585684701d0c",
> algorithm=MD5..Max-Forwards:
> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone
> >   ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8
> 18
> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
> >   telephone-event/8000..
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> > <sip:213 at 10.10.78.36;user=phone>
> >   ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
> > 22-51-43 -0500..Content-Length: 0....
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:52783
> >   INVITE sip:pbxjker_213 at 10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP
> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: 69..From:
> > "Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81
> >   rv8Fp..To: <sip:pbxjker_213 at 10.10.78.24:52783>..Call-ID:
> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact:
> > <sip:mod_sofia at 10.10.78.36:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.
> >   head-git-1086cba 2011-05-23 22-51-43 -0500..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-Type: application/sdp..Content-Disposition: session..
> >   Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID:
> > "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36
> >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
> > 1318333156 1318333157 IN IP4
> >    10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio 29124
> > RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98
> > bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320
> >   00..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
> > 0-16..a=ptime:20..
> > #
> > U 10.10.78.24:55516 -> 10.10.78.36:5060
> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
> > <sip:pbxjker_213 at 99.65.2
> >   17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
> > PolycomSoundPointIP-SPIP_320-UA/3.2
> >   .2.0477..Accept-Language: en..Content-Length: 0....
> > #
> > U 10.10.78.24:55516 -> 10.10.78.36:5060
> >   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
> > <sip:pbxjker_213 at 99.65.
> >   217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
> > PolycomSoundPointIP-SPIP_320-UA/3.
> >   2.2.0477..Allow-Events: talk,hold,conference..Accept-Language:
> > en..Content-Length: 0....
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> > <sip:213 at 10.10.78.36;user=phone
> >   >;tag=8aQNK2QyUKjXa..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 INVITE..Contact:
> > <sip:213 at 10.10.78.36:5060;transport=udp>..User-Agent:
> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2
> >   3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE,
> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
> > SUBSCRIBE..Supported: timer, precondition, path, replaces..A
> >   llow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> > include-session-description, presence.winfo, message-summary,
> > refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker
> >   _213>;party=calling;privacy=off;screen=no....
> > #
> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
> OPTIONS,
> > INFO, MESSAGE, SUBSCR
> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
> > username="pbxjker_2564", realm="10.10.78.36",
> > nonce="ab4a9421-fbb9-41fd-943d-462
> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> > response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
> > 70..Content-Length: 0..
> >   ..
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:5060
> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
> > 0....
> > #
> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
> OPTIONS,
> > INFO, MESSAGE, SUBSCR
> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
> > username="pbxjker_2564", realm="10.10.78.36",
> > nonce="ab4a9421-fbb9-41fd-943d-462
> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> > response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
> > 70..Content-Length: 0..
> >   ..
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:5060
> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
> > 0....
> > #
> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
> OPTIONS,
> > INFO, MESSAGE, SUBSCR
> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
> > username="pbxjker_2564", realm="10.10.78.36",
> > nonce="ab4a9421-fbb9-41fd-943d-462
> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> > response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards:
> > 70..Content-Length: 0..
> >   ..
> > #
> > U 10.10.78.36:5060 -> 10.10.78.24:5060
> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
> > 0....
> > #
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
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>
>
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From infos at madovsky.org  Thu Oct 13 05:53:31 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Oct 2011 21:53:31 -0400
Subject: [Freeswitch-users] continue_on_fail
Message-ID: <73233639A2EE4ACC9DDBC958E457099B@e1705>

did you resolve it ?

after some test I'm facing the same problem
if for example the callee decides to send a user busy so FS tries with the second bridge and
makes a second call which can be interpreted as harrassment ;)

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Monday, October 10, 2011 11:28 PM
  Subject: Re: [Freeswitch-users] continue_on_fail


  maybe modify continue_on_fail as a leg var on the first bridge
    ----- Original Message ----- 
    From: Boris Kovalenko 
    To: FreeSWITCH Users Help 
    Sent: Monday, October 10, 2011 11:12 PM
    Subject: Re: [Freeswitch-users] continue_on_fail


    Hello!

        No, the question was how to FAILOVER :) I need to stop processing dialstring if user at carrier1 is busy.


      the question was about mod_lcr. 
      it returns something like:


      sofia/gateway/carrier1/14105780999|sofia/gateway/ carrier2/14105780999|sofia/gateway/ carrier3/14105780999
The question was how to make it when the first returns "user_busy" to not failover and try the other ones.-Avi



      On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org> wrote:

        Well, now we're talking about something more than just continuing_on_fail. We'll need to look at the actual dialstring to answer any more specific questions. 


        -MC 



        On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:

          It seems from testing, and I was told on channel:  continue_on_fail is per bridge string, not per leg. So it doesn't actually help with the question.. 
          Unless I'm mistaken.

          -Avi



          On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org> wrote:

            You will need to specify the error conditions on which to continue. See the example here: 
            http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail


            -MC


            P.S. - Check out the "enumeration" column on this page to get an idea of the kinds of failures you may not be aware of: http://wiki.freeswitch.org/wiki/Hangup_causes 



            On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

              I've been wondering how to do this. 
              When I asked in channel, I was basically told this couldn't be done in a single bridge string... I hope that's not actually the case.

              -Avi



              On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

                Hello!

                    Have read docs but misundertood :( I use mod_lcr which return two
                or more gateways to the same destination. And when the user is busy with
                the first gateway the call continues with the second and so on. How may
                I to not continue gateways if user is busy?

                --
                ? ?????????,
                  ????? ?????????
                  ??? "??????"
                  ???. +7 (3435) 230001
                  ???? +7 (3435) 230005



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-- 
Regards,
Boris



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



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From jeff at jefflenk.com  Thu Oct 13 06:33:06 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 12 Oct 2011 19:33:06 -0700 (PDT)
Subject: [Freeswitch-users] Newbie trouble with default config extension
	9198...
In-Reply-To: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
References: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
Message-ID: <1318473186527-6887414.post@n2.nabble.com>

Trying to use a client and FS on the same box is not a configuration that is
recommended especially for new users. You will have much better success
using two different IP's(multiple machines). Also make sure that you allow
exceptions for FS and the client(soft phone) though the windows firewall. 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-trouble-with-default-config-extension-9198-tp6886283p6887414.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From boris at tagnet.ru  Thu Oct 13 07:43:51 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 13 Oct 2011 09:43:51 +0600
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <73233639A2EE4ACC9DDBC958E457099B@e1705>
References: <73233639A2EE4ACC9DDBC958E457099B@e1705>
Message-ID: <4E965E77.90704@tagnet.ru>

Hello!

     No. I even tried continue_on_fail=false and hangup_after_bridge=true
> did you resolve it ?
> after some test I'm facing the same problem
> if for example the callee decides to send a user busy so FS tries with 
> the second bridge and
> makes a second call which can be interpreted as harrassment ;)
>
>     ----- Original Message -----
>     *From:* Madovsky <mailto:infos at madovsky.org>
>     *To:* FreeSWITCH Users Help
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Monday, October 10, 2011 11:28 PM
>     *Subject:* Re: [Freeswitch-users] continue_on_fail
>
>     maybe modify continue_on_fail as a leg var on the first bridge
>
>         ----- Original Message -----
>         *From:* Boris Kovalenko <mailto:boris at tagnet.ru>
>         *To:* FreeSWITCH Users Help
>         <mailto:freeswitch-users at lists.freeswitch.org>
>         *Sent:* Monday, October 10, 2011 11:12 PM
>         *Subject:* Re: [Freeswitch-users] continue_on_fail
>
>         Hello!
>
>             No, the question was how to FAILOVER :) I need to stop
>         processing dialstring if user at carrier1 is busy.
>
>>         the question was about mod_lcr.
>>         it returns something like:
>>
>>         sofia/gateway/carrier1/14105780999|sofia/gateway/
>>         carrier2/14105780999|sofia/gateway/ carrier3/14105780999
>>         The question was how to make it when the first returns "user_busy" to not failover and try the other ones.
>>         -Avi
>>
>>
>>         On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins
>>         <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>
>>             Well, now we're talking about something more than just
>>             continuing_on_fail. We'll need to look at the actual
>>             dialstring to answer any more specific questions.
>>
>>             -MC
>>
>>
>>             On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus
>>             <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>>
>>                 It seems from testing, and I was told on channel:
>>                  continue_on_fail is per bridge string, not per leg.
>>                 So it doesn't actually help with the question..
>>                 Unless I'm mistaken.
>>                 -Avi
>>
>>
>>                 On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins
>>                 <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>
>>                     You will need to specify the error conditions on
>>                     which to continue. See the example here:
>>                     http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>
>>                     -MC
>>
>>                     P.S. - Check out the "enumeration" column on this
>>                     page to get an idea of the kinds of failures you
>>                     may not be aware of:
>>                     http://wiki.freeswitch.org/wiki/Hangup_causes
>>
>>
>>                     On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus
>>                     <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>>
>>                         I've been wondering how to do this.
>>                         When I asked in channel, I was basically told
>>                         this couldn't be done in a single bridge
>>                         string... I hope that's not actually the case.
>>                         -Avi
>>
>>
>>                         On Fri, Oct 7, 2011 at 4:40 PM, Boris
>>                         Kovalenko <boris at tagnet.ru
>>                         <mailto:boris at tagnet.ru>> wrote:
>>
>>                             Hello!
>>
>>                                 Have read docs but misundertood :( I
>>                             use mod_lcr which return two
>>                             or more gateways to the same destination.
>>                             And when the user is busy with
>>                             the first gateway the call continues with
>>                             the second and so on. How may
>>                             I to not continue gateways if user is busy?
>>
>>                             --
>>                             ? ?????????,
>>                               ????? ?????????
>>                               ??? "??????"
>>                               ???. +7 (3435) 230001
>>                             <tel:%2B7%20%283435%29%20230001>
>>                               ???? +7 (3435) 230005
>>                             <tel:%2B7%20%283435%29%20230005>
>>
>>
>>
>>                             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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>>
>>
>>
>>
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>>
>>
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>>
>>
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>>
>>
>>
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>
>
>         -- 
>         Regards,
>         Boris
>
>
>         ------------------------------------------------------------------------
>
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Boris


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From infos at madovsky.org  Thu Oct 13 08:21:59 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 13 Oct 2011 00:21:59 -0400
Subject: [Freeswitch-users] continue_on_fail
References: <73233639A2EE4ACC9DDBC958E457099B@e1705> <4E965E77.90704@tagnet.ru>
Message-ID: <709ED34B0EB94BE0B95D0C2D97ED3D8A@e1705>

try to use bridge witn single line
and put condition with hangup cause between....

hope it helps
  ----- Original Message ----- 
  From: Boris Kovalenko 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, October 12, 2011 11:43 PM
  Subject: Re: [Freeswitch-users] continue_on_fail


  Hello!

      No. I even tried continue_on_fail=false and hangup_after_bridge=true

    did you resolve it ?

    after some test I'm facing the same problem
    if for example the callee decides to send a user busy so FS tries with the second bridge and
    makes a second call which can be interpreted as harrassment ;)

      ----- Original Message ----- 
      From: Madovsky 
      To: FreeSWITCH Users Help 
      Sent: Monday, October 10, 2011 11:28 PM
      Subject: Re: [Freeswitch-users] continue_on_fail


      maybe modify continue_on_fail as a leg var on the first bridge
        ----- Original Message ----- 
        From: Boris Kovalenko 
        To: FreeSWITCH Users Help 
        Sent: Monday, October 10, 2011 11:12 PM
        Subject: Re: [Freeswitch-users] continue_on_fail


        Hello!

            No, the question was how to FAILOVER :) I need to stop processing dialstring if user at carrier1 is busy.


          the question was about mod_lcr. 
          it returns something like:


          sofia/gateway/carrier1/14105780999|sofia/gateway/ carrier2/14105780999|sofia/gateway/ carrier3/14105780999
The question was how to make it when the first returns "user_busy" to not failover and try the other ones.-Avi



          On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org> wrote:

            Well, now we're talking about something more than just continuing_on_fail. We'll need to look at the actual dialstring to answer any more specific questions. 


            -MC 



            On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:

              It seems from testing, and I was told on channel:  continue_on_fail is per bridge string, not per leg. So it doesn't actually help with the question.. 
              Unless I'm mistaken.

              -Avi



              On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org> wrote:

                You will need to specify the error conditions on which to continue. See the example here: 
                http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail


                -MC


                P.S. - Check out the "enumeration" column on this page to get an idea of the kinds of failures you may not be aware of: http://wiki.freeswitch.org/wiki/Hangup_causes 



                On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

                  I've been wondering how to do this. 
                  When I asked in channel, I was basically told this couldn't be done in a single bridge string... I hope that's not actually the case.

                  -Avi



                  On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

                    Hello!

                        Have read docs but misundertood :( I use mod_lcr which return two
                    or more gateways to the same destination. And when the user is busy with
                    the first gateway the call continues with the second and so on. How may
                    I to not continue gateways if user is busy?

                    --
                    ? ?????????,
                      ????? ?????????
                      ??? "??????"
                      ???. +7 (3435) 230001
                      ???? +7 (3435) 230005



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From boris at tagnet.ru  Thu Oct 13 08:32:33 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 13 Oct 2011 10:32:33 +0600
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <709ED34B0EB94BE0B95D0C2D97ED3D8A@e1705>
References: <73233639A2EE4ACC9DDBC958E457099B@e1705> <4E965E77.90704@tagnet.ru>
	<709ED34B0EB94BE0B95D0C2D97ED3D8A@e1705>
Message-ID: <4E9669E1.7060509@tagnet.ru>

Hello!

     May You give an example? I use mod_lcr

> try to use bridge witn single line
> and put condition with hangup cause between....
> hope it helps
>
>     ----- Original Message -----
>     *From:* Boris Kovalenko <mailto:boris at tagnet.ru>
>     *To:* FreeSWITCH Users Help
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Wednesday, October 12, 2011 11:43 PM
>     *Subject:* Re: [Freeswitch-users] continue_on_fail
>
>     Hello!
>
>         No. I even tried continue_on_fail=false and
>     hangup_after_bridge=true
>>     did you resolve it ?
>>     after some test I'm facing the same problem
>>     if for example the callee decides to send a user busy so FS tries
>>     with the second bridge and
>>     makes a second call which can be interpreted as harrassment ;)
>>
>>         ----- Original Message -----
>>         *From:* Madovsky <mailto:infos at madovsky.org>
>>         *To:* FreeSWITCH Users Help
>>         <mailto:freeswitch-users at lists.freeswitch.org>
>>         *Sent:* Monday, October 10, 2011 11:28 PM
>>         *Subject:* Re: [Freeswitch-users] continue_on_fail
>>
>>         maybe modify continue_on_fail as a leg var on the first bridge
>>
>>             ----- Original Message -----
>>             *From:* Boris Kovalenko <mailto:boris at tagnet.ru>
>>             *To:* FreeSWITCH Users Help
>>             <mailto:freeswitch-users at lists.freeswitch.org>
>>             *Sent:* Monday, October 10, 2011 11:12 PM
>>             *Subject:* Re: [Freeswitch-users] continue_on_fail
>>
>>             Hello!
>>
>>                 No, the question was how to FAILOVER :) I need to
>>             stop processing dialstring if user at carrier1 is busy.
>>
>>>             the question was about mod_lcr.
>>>             it returns something like:
>>>
>>>             sofia/gateway/carrier1/14105780999|sofia/gateway/
>>>             carrier2/14105780999|sofia/gateway/ carrier3/14105780999
>>>             The question was how to make it when the first returns "user_busy" to not failover and try the other ones.
>>>             -Avi
>>>
>>>
>>>             On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins
>>>             <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>
>>>                 Well, now we're talking about something more than
>>>                 just continuing_on_fail. We'll need to look at the
>>>                 actual dialstring to answer any more specific
>>>                 questions.
>>>
>>>                 -MC
>>>
>>>
>>>                 On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus
>>>                 <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>>>
>>>                     It seems from testing, and I was told on
>>>                     channel:  continue_on_fail is per bridge string,
>>>                     not per leg. So it doesn't actually help with
>>>                     the question..
>>>                     Unless I'm mistaken.
>>>                     -Avi
>>>
>>>
>>>                     On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins
>>>                     <msc at freeswitch.org <mailto:msc at freeswitch.org>>
>>>                     wrote:
>>>
>>>                         You will need to specify the error
>>>                         conditions on which to continue. See the
>>>                         example here:
>>>                         http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>
>>>                         -MC
>>>
>>>                         P.S. - Check out the "enumeration" column on
>>>                         this page to get an idea of the kinds of
>>>                         failures you may not be aware of:
>>>                         http://wiki.freeswitch.org/wiki/Hangup_causes
>>>
>>>
>>>                         On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus
>>>                         <avi at avimarcus.net
>>>                         <mailto:avi at avimarcus.net>> wrote:
>>>
>>>                             I've been wondering how to do this.
>>>                             When I asked in channel, I was basically
>>>                             told this couldn't be done in a single
>>>                             bridge string... I hope that's not
>>>                             actually the case.
>>>                             -Avi
>>>
>>>
>>>                             On Fri, Oct 7, 2011 at 4:40 PM, Boris
>>>                             Kovalenko <boris at tagnet.ru
>>>                             <mailto:boris at tagnet.ru>> wrote:
>>>
>>>                                 Hello!
>>>
>>>                                     Have read docs but misundertood
>>>                                 :( I use mod_lcr which return two
>>>                                 or more gateways to the same
>>>                                 destination. And when the user is
>>>                                 busy with
>>>                                 the first gateway the call continues
>>>                                 with the second and so on. How may
>>>                                 I to not continue gateways if user
>>>                                 is busy?
>>>
>>>                                 --
>>>                                 ? ?????????,
>>>                                   ????? ?????????
>>>                                   ??? "??????"
>>>                                   ???. +7 (3435) 230001
>>>                                 <tel:%2B7%20%283435%29%20230001>
>>>                                   ???? +7 (3435) 230005
>>>                                 <tel:%2B7%20%283435%29%20230005>
>>>
>>>
>>>
>>>                                 FreeSWITCH-users mailing list
>>>                                 FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>                                 http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>                             FreeSWITCH-users mailing list
>>>                             FreeSWITCH-users at lists.freeswitch.org
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>>>
>>>
>>>
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>>>
>>>
>>>
>>>
>>>                     FreeSWITCH-users mailing list
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>>>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>
>>>
>>>
>>>
>>>                 FreeSWITCH-users mailing list
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>>>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>
>>>
>>>
>>>
>>>             FreeSWITCH-users mailing list
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>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>
>>
>>             -- 
>>             Regards,
>>             Boris
>>
>>
>>             ------------------------------------------------------------------------
>>
>>             FreeSWITCH-users mailing list
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>>
>>
>>
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>>     http://www.freeswitch.org
>
>
>     -- 
>     Regards,
>     Boris
>
>
>     ------------------------------------------------------------------------
>
>     FreeSWITCH-users mailing list
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From infos at madovsky.org  Thu Oct 13 09:07:33 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 13 Oct 2011 01:07:33 -0400
Subject: [Freeswitch-users] continue_on_fail
References: <73233639A2EE4ACC9DDBC958E457099B@e1705>
	<4E965E77.90704@tagnet.ru><709ED34B0EB94BE0B95D0C2D97ED3D8A@e1705>
	<4E9669E1.7060509@tagnet.ru>
Message-ID: <C6693088B9934A72B6A893751427F42C@e1705>

no ex sorry.
try to hash you mod_lcr result in different variables
and create multiline bridge with continue_on_fail and hangup_cause conidtions between them.

  ----- Original Message ----- 
  From: Boris Kovalenko 
  To: FreeSWITCH Users Help 
  Sent: Thursday, October 13, 2011 12:32 AM
  Subject: Re: [Freeswitch-users] continue_on_fail


  Hello!

      May You give an example? I use mod_lcr 


    try to use bridge witn single line
    and put condition with hangup cause between....

    hope it helps
      ----- Original Message ----- 
      From: Boris Kovalenko 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, October 12, 2011 11:43 PM
      Subject: Re: [Freeswitch-users] continue_on_fail


      Hello!

          No. I even tried continue_on_fail=false and hangup_after_bridge=true

        did you resolve it ?

        after some test I'm facing the same problem
        if for example the callee decides to send a user busy so FS tries with the second bridge and
        makes a second call which can be interpreted as harrassment ;)

          ----- Original Message ----- 
          From: Madovsky 
          To: FreeSWITCH Users Help 
          Sent: Monday, October 10, 2011 11:28 PM
          Subject: Re: [Freeswitch-users] continue_on_fail


          maybe modify continue_on_fail as a leg var on the first bridge
            ----- Original Message ----- 
            From: Boris Kovalenko 
            To: FreeSWITCH Users Help 
            Sent: Monday, October 10, 2011 11:12 PM
            Subject: Re: [Freeswitch-users] continue_on_fail


            Hello!

                No, the question was how to FAILOVER :) I need to stop processing dialstring if user at carrier1 is busy.


              the question was about mod_lcr. 
              it returns something like:


              sofia/gateway/carrier1/14105780999|sofia/gateway/ carrier2/14105780999|sofia/gateway/ carrier3/14105780999
The question was how to make it when the first returns "user_busy" to not failover and try the other ones.-Avi



              On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org> wrote:

                Well, now we're talking about something more than just continuing_on_fail. We'll need to look at the actual dialstring to answer any more specific questions. 


                -MC 



                On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:

                  It seems from testing, and I was told on channel:  continue_on_fail is per bridge string, not per leg. So it doesn't actually help with the question.. 
                  Unless I'm mistaken.

                  -Avi



                  On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org> wrote:

                    You will need to specify the error conditions on which to continue. See the example here: 
                    http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail


                    -MC


                    P.S. - Check out the "enumeration" column on this page to get an idea of the kinds of failures you may not be aware of: http://wiki.freeswitch.org/wiki/Hangup_causes 



                    On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

                      I've been wondering how to do this. 
                      When I asked in channel, I was basically told this couldn't be done in a single bridge string... I hope that's not actually the case.

                      -Avi



                      On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

                        Hello!

                            Have read docs but misundertood :( I use mod_lcr which return two
                        or more gateways to the same destination. And when the user is busy with
                        the first gateway the call continues with the second and so on. How may
                        I to not continue gateways if user is busy?

                        --
                        ? ?????????,
                          ????? ?????????
                          ??? "??????"
                          ???. +7 (3435) 230001
                          ???? +7 (3435) 230005



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From masood24 at gmail.com  Thu Oct 13 15:53:24 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Thu, 13 Oct 2011 14:53:24 +0300
Subject: [Freeswitch-users] Digest Authentcation with radius server
Message-ID: <CAMd3iJJnvCq8xM_gTLDyqgG0B1WLxYrZWjMwSQW1fNKPK0QcSg@mail.gmail.com>

Dear ,

I had successfully installed and configured radiusclient. I do not know that
what next lines should i do to get users registered on my freeswitch using
digest authentication with my radius server,
Like I have my radsius server and billing system. Now what lines should i
add in sofit conf or in dialplans that every time client hits server request
should be sent to radius server and what lines should i add to register them
if i get proper return code and how to do that.


-- 
*Regards,
Masood Ahmed
*
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From bnaylor at sirran.com  Thu Oct 13 16:12:29 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Thu, 13 Oct 2011 13:12:29 +0100
Subject: [Freeswitch-users] TLS problems
Message-ID: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>

Hi all

 

I was wondering if anyone can help me with a problem I am having with using
TLS SIP when NAT routers are involved?

 

The scenario is as follows -

 

SIP UA --- > NAT firewall  --- > Internet --- > Freeswitch --- > Internet
--- > NAT Firewall --- > SIP UA

 

When I use UDP as the transport protocol, everything works fine.  This is
down to (correct me if I am wrong) the NAT router being able to inspect the
SIP traffic, and changing the SIP INVITE message so that the public IP is
used for the RTP stream.  This is seen in the following ( 1.2.3.4 = SIP UA,
50.50.50.50 = Freeswitch server) -

 

  recv 1042 bytes from udp/[1.2.3.4]:34019 at 08:40:24.302496:

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

   INVITE sip:1004 at 50.50.50.50 SIP/2.0

   Via: SIP/2.0/UDP 1.2.3.4:34019;branch=z9hG4bKD8xhgo0smbsOQMUy;rport

   Contact: <sip:1002 at 1.2.3.4:34019>

   Max-Forwards: 70

   From: "1002" <sip:1002 at 50.50.50.50>;tag=A6E182EBB96E21919A44C50399F1BF27

   Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY

   Supported: replaces, path

   User-Agent: Acrobits Softphone Business/1.8.8

   To: <sip:1004 at 50.50.50.50>

   Content-Type: application/sdp

   Call-ID: 2926496CCA1C6189C90A52D821B06B5E9DBA2B32

   CSeq: 1 INVITE

   Proxy-Authorization: Digest
username="1002",realm="50.50.50.50",algorithm=MD5,uri="sip:1004 at 50.50.50.50"
,nonce="09195369-6474-4310-b3a6-60e7d9c780f1",qop=auth,cnonce="1b774f833fd01
b5b29954c2ddc2b5457",nc=00000002,response="07bda5b4c5fa5e17f653a3e9c7a2f05e"

   Content-Length: 241

   

   v=0

   o=- 10805 6353 IN IP4 1.2.3.4  < --- Public IP address used so that the
other party knows to send the RTP stream to this address.

   s=xynwwjx

   c=IN IP4 1.2.3.4

   t=0 0

   m=audio 34962 RTP/AVP 102 3 0 8 9 101

   a=rtpmap:101 telephone-event/8000

   a=rtpmap:102 ILBC/8000

   a=fmtp:102 mode=30

   a=fmtp:101 0-15

   a=ptime:30

   a=sendrecv

 

This was obtained from the Siptrace on the Freeswitch server.  The
Freeswitch server then forwards these details to the other party.

 

The problem occurs when TLS is used, as the NAT firewalls are unable to
inspect the SIP packets.  Here is another SIPtrace on the Freeswitch server
for TLS example (again I have changed the public IPs to the same as the
above example) -

 

recv 1048 bytes from tls/[1.2.3.4]:64403 at 16:00:42.271063:

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

   INVITE sip:1002 at 50.50.50.50 SIP/2.0

   Via: SIP/2.0/TLS 1.2.3.4:64403;branch=z9hG4bKz6qAkQj3XsYuJ5Os;rport

   Contact: <sip:1004 at 1.2.3.4:64403;transport=tls>

   Max-Forwards: 70

   From: "Ben" <sip:1004 at 50.50.50.50>;tag=F28D08D5E039D8FE021D08022A72F982

   Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY

   Supported: replaces, path

   User-Agent: Acrobits Softphone Business/1.8.8

   To: <sip:1002 at 50.50.50.50>

   Content-Type: application/sdp

   Call-ID: 32166133760BDDAD823725B563F6A1B3989090D6

   CSeq: 1 INVITE

   Authorization: Digest
username="1004",realm="50.50.50.50",algorithm=MD5,uri="sip:1002 at 50.50.50.50"
,nonce="5af9829e-716d-4472-924f-84eb41d9e78e",qop=auth,cnonce="d641702f981e1
0f3a41e074061bdb5b3",nc=00000003,response="a9f577ac37f8f367b2f839a58d43e1e0"

   Content-Length: 240

   

   v=0

   o=- 71766 47529 IN IP4 172.16.234.176 < --- Using Private IP for the RTP
details

   s=fscmvnc

   c=IN IP4 172.16.234.176

   t=0 0

   m=audio 64628 RTP/AVP 3 102 101

   a=rtpmap:101 telephone-event/8000

   a=rtpmap:102 ILBC/8000

   a=fmtp:102 mode=30

   a=fmtp:101 0-15

   a=ptime:30

   a=sendrecv

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

 

As you can see, the SIP UAs are going to try and use a non-existent private
IP address to send the RTP stream to.

I can also confirm that the UAs are trying to do this, by looking at the SIP
logs built in to the SIP UA client software.

 

My (Freeswitch related) question is, am I able to set something in
Freeswitch to change the details to be the public IP of the firewall?  The
SIP messages include the correct IP, can the Freeswitch server use this
address and replace the Private IP with this one?

 

I have heard about TLS proxies/STUn servers etc etc, but would much rather
use Freeswitch to do this.  

 

Any suggestions are welcome, also please let me know if I have misunderstood
any of the concepts behind SIP/VOIP and firewalls - I am rather new to the
VOIP game!

 

Kind regards

 

Ben 

 

 

 

Ben Naylor

Network Support Engineer

 <http://www.sirran.com/> Description: Description:
cid:image001.png at 01CC4312.3962A460

 

T: +44 (0)207 096 1648`

M: +44 (0)7924 349 113

F:  +44 (0)207 150 595

E:  <mailto:bnaylor at sirran.com> bnaylor at sirran.com W:
<http://www.sirran.com/> www.sirran.com

 

A: SiRRAN Communications Ltd, The Hawk Business Park, Easingwold, YO61 3FE

Registered Office: SiRRAN Communications Ltd,. 6-9 Trinity Street Dublin 2,
Ireland

 

_____________________________________________________________

The information contained in this e-mail and any attached documents may be
privileged, confidential and protected from disclosure. If you are not the
intended recipient you may not read, copy, distribute or use this
information. If you have received this communication in error, please notify
the sender immediately by replying to this message and then delete it from
your system.

 

 

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From johnrose at google.hm  Thu Oct 13 18:08:19 2011
From: johnrose at google.hm (John Rose)
Date: Thu, 13 Oct 2011 10:08:19 -0400
Subject: [Freeswitch-users] Is Bgapi threadsafe?
In-Reply-To: <CAKbxfG9A0dnxcYioFDTjYf1ire7ALiKmmFbpghkjOxBWooi1=g@mail.gmail.com>
References: <001401cc8909$ab2e48f0$018adad0$@google.hm>	<CAKbxfG9aNd-3Jmims5gmCiZA_UOcm9giG8Zo3F9QPZTWhEh6Zw@mail.gmail.com>	<CAKbxfG_64fBccqjf6kddF6bsPnyy=VzGWGM9zOuq6jmhLT8=VQ@mail.gmail.com>
	<CAKbxfG9A0dnxcYioFDTjYf1ire7ALiKmmFbpghkjOxBWooi1=g@mail.gmail.com>
Message-ID: <001401cc89b1$9008bc10$b01a3430$@google.hm>

OK thanks, you've answered my question. So basically calling the methods on
an ESL connection, Bgapi for example, from a client is not threadsafe, the
method calls should be exclusively locked when using multiple threads since
it's utilizing one socket. That makes sense for the one connection scenario.

John

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> Subject: Re: [Freeswitch-users] Is Bgapi threadsafe?
> 
> wrong week to quit sniffing glue....
> 
> correction to last sentence of previous email:
> 
> and you **can't** do i/o on it from 2 threads at once naturally.
> 
> On Wed, Oct 12, 2011 at 4:07 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > oh but wait...
> >
> > if you mean you want threads on your side to share it, I'd either give
> > them all their own socket or share one on a mutex cos it's a socket
> > and you can do i/o on it from 2 threads at once naturally.
> >
> >
> > On Wed, Oct 12, 2011 at 4:06 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> yes since its a socket and the other side is properly threaded. ?If
> >> you are using the in-tree ESL lib the client also has race condition
> >> protection
> >>
> >> On Wed, Oct 12, 2011 at 1:06 PM, John Rose <johnrose at google.hm>
> wrote:
> >>> Does anyone know if calling Bgapi on an ESL connection is
> >>> threadsafe? ?I?m calling it from ManagedESL via C# and have a lock
> >>> around it but would like to remove the lock if possible

> >>>
> >>> John
> >>>
> >>>
> >>>



From kris at kriskinc.com  Thu Oct 13 18:39:15 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 13 Oct 2011 10:39:15 -0400
Subject: [Freeswitch-users] TLS problems
In-Reply-To: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>
References: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>
Message-ID: <CAKDfjgcm1ADmsY4Bdj4Kpr7WcvuFrnjCPiP=agADH81n0ge03A@mail.gmail.com>

Hi Ben,

  FreeSWITCH handles NAT rather well and should automatically ignore
the natted SDP info; using the actual send/receive RTP IP/port once
media arrives.  It will then leave itself in the media path (like it
usually does anyway) and magically fix SDP/RTP data to other clients.

  Are you doing anything funny like bypass_media?

  Please include a SIP trace and FS console logs so we can see if what
FreeSWITCH is doing.

On Thu, Oct 13, 2011 at 8:12 AM, Ben Naylor <bnaylor at sirran.com> wrote:
>
> Hi all
>
>
>
> I was wondering if anyone can help me with a problem I am having with using TLS SIP when NAT routers are involved?
>
>
>
> The scenario is as follows ?
>
>
>
> SIP UA --- > NAT firewall? --- > Internet --- > Freeswitch --- > Internet --- > NAT Firewall --- > SIP UA
>
>
>
> When I use UDP as the transport protocol, everything works fine.? This is down to (correct me if I am wrong) the NAT router being able to inspect the SIP traffic, and changing the SIP INVITE message so that the public IP is used for the RTP stream.? This is seen in the following ( 1.2.3.4 = SIP UA, ??50.50.50.50 = Freeswitch server) ?
>
>
>
> ? recv 1042 bytes from udp/[1.2.3.4]:34019 at 08:40:24.302496:
>
> ?? ------------------------------------------------------------------------
>
> ?? INVITE sip:1004 at 50.50.50.50 SIP/2.0
>
> ?? Via: SIP/2.0/UDP 1.2.3.4:34019;branch=z9hG4bKD8xhgo0smbsOQMUy;rport
>
> ?? Contact: <sip:1002 at 1.2.3.4:34019>
>
> ?? Max-Forwards: 70
>
> ?? From: "1002" <sip:1002 at 50.50.50.50>;tag=A6E182EBB96E21919A44C50399F1BF27
>
> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>
> ?? Supported: replaces, path
>
> ?? User-Agent: Acrobits Softphone Business/1.8.8
>
> ?? To: <sip:1004 at 50.50.50.50>
>
> ?? Content-Type: application/sdp
>
> ?? Call-ID: 2926496CCA1C6189C90A52D821B06B5E9DBA2B32
>
> ?? CSeq: 1 INVITE
>
> ?? Proxy-Authorization: Digest username="1002",realm="50.50.50.50",algorithm=MD5,uri="sip:1004 at 50.50.50.50",nonce="09195369-6474-4310-b3a6-60e7d9c780f1",qop=auth,cnonce="1b774f833fd01b5b29954c2ddc2b5457",nc=00000002,response="07bda5b4c5fa5e17f653a3e9c7a2f05e"
>
> ?? Content-Length: 241
>
>
>
> ???v=0
>
> ?? o=- 10805 6353 IN IP4 1.2.3.4? < --- Public IP address used so that the other party knows to send the RTP stream to this address.
>
> ?? s=xynwwjx
>
> ?? c=IN IP4 1.2.3.4
>
> ?? t=0 0
>
> ?? m=audio 34962 RTP/AVP 102 3 0 8 9 101
>
> ?? a=rtpmap:101 telephone-event/8000
>
> ?? a=rtpmap:102 ILBC/8000
>
> ?? a=fmtp:102 mode=30
>
> ?? a=fmtp:101 0-15
>
> ?? a=ptime:30
>
> ?? a=sendrecv
>
>
>
> This was obtained from the Siptrace on the Freeswitch server.? The Freeswitch server then forwards these details to the other party.
>
>
>
> The problem occurs when TLS is used, as the NAT firewalls are unable to inspect the SIP packets.? Here is another SIPtrace on the Freeswitch server for TLS example (again I have changed the public IPs to the same as the above example) ?
>
>
>
> recv 1048 bytes from tls/[1.2.3.4]:64403 at 16:00:42.271063:
>
> ?? ------------------------------------------------------------------------
>
> ?? INVITE sip:1002 at 50.50.50.50 SIP/2.0
>
> ?? Via: SIP/2.0/TLS 1.2.3.4:64403;branch=z9hG4bKz6qAkQj3XsYuJ5Os;rport
>
> ?? Contact: <sip:1004 at 1.2.3.4:64403;transport=tls>
>
> ?? Max-Forwards: 70
>
> ?? From: "Ben" <sip:1004 at 50.50.50.50>;tag=F28D08D5E039D8FE021D08022A72F982
>
> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>
> ?? Supported: replaces, path
>
> ?? User-Agent: Acrobits Softphone Business/1.8.8
>
> ?? To: <sip:1002 at 50.50.50.50>
>
> ?? Content-Type: application/sdp
>
> ?? Call-ID: 32166133760BDDAD823725B563F6A1B3989090D6
>
> ?? CSeq: 1 INVITE
>
> ?? Authorization: Digest username="1004",realm="50.50.50.50",algorithm=MD5,uri="sip:1002 at 50.50.50.50",nonce="5af9829e-716d-4472-924f-84eb41d9e78e",qop=auth,cnonce="d641702f981e10f3a41e074061bdb5b3",nc=00000003,response="a9f577ac37f8f367b2f839a58d43e1e0"
>
> ?? Content-Length: 240
>
>
>
> ???v=0
>
> ?? o=- 71766 47529 IN IP4 172.16.234.176 < --- Using Private IP for the RTP details
>
> ?? s=fscmvnc
>
> ?? c=IN IP4 172.16.234.176
>
> ?? t=0 0
>
> ?? m=audio 64628 RTP/AVP 3 102 101
>
> ?? a=rtpmap:101 telephone-event/8000
>
> ?? a=rtpmap:102 ILBC/8000
>
> ?? a=fmtp:102 mode=30
>
> ?? a=fmtp:101 0-15
>
> ?? a=ptime:30
>
> ?? a=sendrecv
>
> ?? ------------------------------------------------------------------------
>
>
>
> As you can see, the SIP UAs are going to try and use a non-existent private IP address to send the RTP stream to.
>
> I can also confirm that the UAs are trying to do this, by looking at the SIP logs built in to the SIP UA client software.
>
>
>
> My (Freeswitch related) question is, am I able to set something in Freeswitch to change the details to be the public IP of the firewall?? The SIP messages include the correct IP, can the Freeswitch server use this address and replace the Private IP with this one?
>
>
>
> I have heard about TLS proxies/STUn servers etc etc, but would much rather use Freeswitch to do this.
>
>
>
> Any suggestions are welcome, also please let me know if I have misunderstood any of the concepts behind SIP/VOIP and firewalls ? I am rather new to the VOIP game!
>
>
>
> Kind regards
>
>
>
> Ben
>
>
>
>
>
>
>
> Ben Naylor
>
> Network Support Engineer
>
>
>
> T: +44 (0)207 096 1648`
>
> M: +44 (0)7924 349 113
>
> F:? +44 (0)207 150 595
>
> E: bnaylor at sirran.com W: www.sirran.com
>
>
>
> A: SiRRAN Communications Ltd, The Hawk Business Park, Easingwold, YO61 3FE
>
> Registered Office: SiRRAN Communications Ltd,. 6-9 Trinity Street Dublin 2, Ireland
>
>
>
> _____________________________________________________________
>
> The information contained in this e-mail and any attached documents may be privileged, confidential and protected from disclosure. If you are not the intended recipient you may not read, copy, distribute or use this information. If you have received this communication in error, please notify the sender immediately by replying to this message and then delete it from your system.
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bnaylor at sirran.com  Thu Oct 13 18:46:06 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Thu, 13 Oct 2011 15:46:06 +0100
Subject: [Freeswitch-users] TLS problems
In-Reply-To: <CAKDfjgcm1ADmsY4Bdj4Kpr7WcvuFrnjCPiP=agADH81n0ge03A@mail.gmail.com>
References: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>
	<CAKDfjgcm1ADmsY4Bdj4Kpr7WcvuFrnjCPiP=agADH81n0ge03A@mail.gmail.com>
Message-ID: <005f01cc89b6$d76a1a90$863e4fb0$@sirran.com>

Ahh yes, I forgot to mention that I am using bypass_media, apologies for not
mentioning this before!

Logs to follow shortly......

Thanks for the help,

Ben

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: 13 October 2011 15:39
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TLS problems

Hi Ben,

  FreeSWITCH handles NAT rather well and should automatically ignore the
natted SDP info; using the actual send/receive RTP IP/port once media
arrives.  It will then leave itself in the media path (like it usually does
anyway) and magically fix SDP/RTP data to other clients.

  Are you doing anything funny like bypass_media?

  Please include a SIP trace and FS console logs so we can see if what
FreeSWITCH is doing.

On Thu, Oct 13, 2011 at 8:12 AM, Ben Naylor <bnaylor at sirran.com> wrote:
>
> Hi all
>
>
>
> I was wondering if anyone can help me with a problem I am having with
using TLS SIP when NAT routers are involved?
>
>
>
> The scenario is as follows ?
>
>
>
> SIP UA --- > NAT firewall? --- > Internet --- > Freeswitch --- > 
> Internet --- > NAT Firewall --- > SIP UA
>
>
>
> When I use UDP as the transport protocol, everything works fine.? This 
> is down to (correct me if I am wrong) the NAT router being able to 
> inspect the SIP traffic, and changing the SIP INVITE message so that 
> the public IP is used for the RTP stream.? This is seen in the 
> following ( 1.2.3.4 = SIP UA, ??50.50.50.50 = Freeswitch server) ?
>
>
>
> ? recv 1042 bytes from udp/[1.2.3.4]:34019 at 08:40:24.302496:
>
> ?? 
> ----------------------------------------------------------------------
> --
>
> ?? INVITE sip:1004 at 50.50.50.50 SIP/2.0
>
> ?? Via: SIP/2.0/UDP 1.2.3.4:34019;branch=z9hG4bKD8xhgo0smbsOQMUy;rport
>
> ?? Contact: <sip:1002 at 1.2.3.4:34019>
>
> ?? Max-Forwards: 70
>
> ?? From: "1002" 
> <sip:1002 at 50.50.50.50>;tag=A6E182EBB96E21919A44C50399F1BF27
>
> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>
> ?? Supported: replaces, path
>
> ?? User-Agent: Acrobits Softphone Business/1.8.8
>
> ?? To: <sip:1004 at 50.50.50.50>
>
> ?? Content-Type: application/sdp
>
> ?? Call-ID: 2926496CCA1C6189C90A52D821B06B5E9DBA2B32
>
> ?? CSeq: 1 INVITE
>
> ?? Proxy-Authorization: Digest
username="1002",realm="50.50.50.50",algorithm=MD5,uri="sip:1004 at 50.50.50.50"
,nonce="09195369-6474-4310-b3a6-60e7d9c780f1",qop=auth,cnonce="1b774f833fd01
b5b29954c2ddc2b5457",nc=00000002,response="07bda5b4c5fa5e17f653a3e9c7a2f05e"
>
> ?? Content-Length: 241
>
>
>
> ???v=0
>
> ?? o=- 10805 6353 IN IP4 1.2.3.4? < --- Public IP address used so that the
other party knows to send the RTP stream to this address.
>
> ?? s=xynwwjx
>
> ?? c=IN IP4 1.2.3.4
>
> ?? t=0 0
>
> ?? m=audio 34962 RTP/AVP 102 3 0 8 9 101
>
> ?? a=rtpmap:101 telephone-event/8000
>
> ?? a=rtpmap:102 ILBC/8000
>
> ?? a=fmtp:102 mode=30
>
> ?? a=fmtp:101 0-15
>
> ?? a=ptime:30
>
> ?? a=sendrecv
>
>
>
> This was obtained from the Siptrace on the Freeswitch server.? The
Freeswitch server then forwards these details to the other party.
>
>
>
> The problem occurs when TLS is used, as the NAT firewalls are unable 
> to inspect the SIP packets.? Here is another SIPtrace on the 
> Freeswitch server for TLS example (again I have changed the public IPs 
> to the same as the above example) ?
>
>
>
> recv 1048 bytes from tls/[1.2.3.4]:64403 at 16:00:42.271063:
>
> ?? 
> ----------------------------------------------------------------------
> --
>
> ?? INVITE sip:1002 at 50.50.50.50 SIP/2.0
>
> ?? Via: SIP/2.0/TLS 1.2.3.4:64403;branch=z9hG4bKz6qAkQj3XsYuJ5Os;rport
>
> ?? Contact: <sip:1004 at 1.2.3.4:64403;transport=tls>
>
> ?? Max-Forwards: 70
>
> ?? From: "Ben" 
> <sip:1004 at 50.50.50.50>;tag=F28D08D5E039D8FE021D08022A72F982
>
> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>
> ?? Supported: replaces, path
>
> ?? User-Agent: Acrobits Softphone Business/1.8.8
>
> ?? To: <sip:1002 at 50.50.50.50>
>
> ?? Content-Type: application/sdp
>
> ?? Call-ID: 32166133760BDDAD823725B563F6A1B3989090D6
>
> ?? CSeq: 1 INVITE
>
> ?? Authorization: Digest
username="1004",realm="50.50.50.50",algorithm=MD5,uri="sip:1002 at 50.50.50.50"
,nonce="5af9829e-716d-4472-924f-84eb41d9e78e",qop=auth,cnonce="d641702f981e1
0f3a41e074061bdb5b3",nc=00000003,response="a9f577ac37f8f367b2f839a58d43e1e0"

>
> ?? Content-Length: 240
>
>
>
> ???v=0
>
> ?? o=- 71766 47529 IN IP4 172.16.234.176 < --- Using Private IP for 
> the RTP details
>
> ?? s=fscmvnc
>
> ?? c=IN IP4 172.16.234.176
>
> ?? t=0 0
>
> ?? m=audio 64628 RTP/AVP 3 102 101
>
> ?? a=rtpmap:101 telephone-event/8000
>
> ?? a=rtpmap:102 ILBC/8000
>
> ?? a=fmtp:102 mode=30
>
> ?? a=fmtp:101 0-15
>
> ?? a=ptime:30
>
> ?? a=sendrecv
>
> ?? 
> ----------------------------------------------------------------------
> --
>
>
>
> As you can see, the SIP UAs are going to try and use a non-existent
private IP address to send the RTP stream to.
>
> I can also confirm that the UAs are trying to do this, by looking at the
SIP logs built in to the SIP UA client software.
>
>
>
> My (Freeswitch related) question is, am I able to set something in
Freeswitch to change the details to be the public IP of the firewall?? The
SIP messages include the correct IP, can the Freeswitch server use this
address and replace the Private IP with this one?
>
>
>
> I have heard about TLS proxies/STUn servers etc etc, but would much rather
use Freeswitch to do this.
>
>
>
> Any suggestions are welcome, also please let me know if I have
misunderstood any of the concepts behind SIP/VOIP and firewalls ? I am
rather new to the VOIP game!
>
>
>
> Kind regards
>
>
>
> Ben
>
>
>
>
>
>
>
> Ben Naylor
>
> Network Support Engineer
>
>
>
> T: +44 (0)207 096 1648`
>
> M: +44 (0)7924 349 113
>
> F:? +44 (0)207 150 595
>
> E: bnaylor at sirran.com W: www.sirran.com
>
>
>
> A: SiRRAN Communications Ltd, The Hawk Business Park, Easingwold, YO61 
> 3FE
>
> Registered Office: SiRRAN Communications Ltd,. 6-9 Trinity Street 
> Dublin 2, Ireland
>
>
>
> _____________________________________________________________
>
> The information contained in this e-mail and any attached documents may be
privileged, confidential and protected from disclosure. If you are not the
intended recipient you may not read, copy, distribute or use this
information. If you have received this communication in error, please notify
the sender immediately by replying to this message and then delete it from
your system.
>
>
>
>
>
> FreeSWITCH-users 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
>



--
Kristian Kielhofner


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



From kris at kriskinc.com  Thu Oct 13 18:54:08 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 13 Oct 2011 10:54:08 -0400
Subject: [Freeswitch-users] TLS problems
In-Reply-To: <005f01cc89b6$d76a1a90$863e4fb0$@sirran.com>
References: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>
	<CAKDfjgcm1ADmsY4Bdj4Kpr7WcvuFrnjCPiP=agADH81n0ge03A@mail.gmail.com>
	<005f01cc89b6$d76a1a90$863e4fb0$@sirran.com>
Message-ID: <CAKDfjgdFZah8YWOCAU13EdDjqX6tuAA1isMKRNzx1TZ2hQS4WA@mail.gmail.com>

Ben,

  SIP+NAT, where do I even start ;)...

Local/near handling (STUN, TURN, ICE, etc)
"Middle" handling (SIP ALG, SBC)
Far end handling (SIP latching/RTP proxy)

  Your natted devices are currently sitting behind a NAT device with a
SIP ALG (application layer gateway).  As you (correctly) guessed this
device sits and watches for SIP packets.  When it sees them it
rewrites various fields (c=, Contact:, etc) to effectively insert
itself into the dialog.  It will then (transparently) rewrite these
fields between both sides of the NAT while also handling RTP.  This
would be "middle" handling.

  This works with bypass_media because your endpoints see the "fixed"
SDP fields from the ALG and know where to send media.  Because the ALG
device is unable to inspect TLS packets it can't rewrite these
addresses.  Normally FreeSWITCH would insert itself in the media path
(Far end handling), proxy the media stream, and rewrite the packets to
match.  With bypass_media enabled FreeSWITCH can't do this.

  With TLS + NAT your options are:

- Configure STUN/TURN/ICE on your client (Acrobits) and cross your fingers
- Disable bypass_media in FreeSWITCH

On Thu, Oct 13, 2011 at 10:46 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Ahh yes, I forgot to mention that I am using bypass_media, apologies for not
> mentioning this before!
>
> Logs to follow shortly......
>
> Thanks for the help,
>
> Ben
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
> Kielhofner
> Sent: 13 October 2011 15:39
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] TLS problems
>
> Hi Ben,
>
> ?FreeSWITCH handles NAT rather well and should automatically ignore the
> natted SDP info; using the actual send/receive RTP IP/port once media
> arrives. ?It will then leave itself in the media path (like it usually does
> anyway) and magically fix SDP/RTP data to other clients.
>
> ?Are you doing anything funny like bypass_media?
>
> ?Please include a SIP trace and FS console logs so we can see if what
> FreeSWITCH is doing.
>
> On Thu, Oct 13, 2011 at 8:12 AM, Ben Naylor <bnaylor at sirran.com> wrote:
>>
>> Hi all
>>
>>
>>
>> I was wondering if anyone can help me with a problem I am having with
> using TLS SIP when NAT routers are involved?
>>
>>
>>
>> The scenario is as follows ?
>>
>>
>>
>> SIP UA --- > NAT firewall? --- > Internet --- > Freeswitch --- >
>> Internet --- > NAT Firewall --- > SIP UA
>>
>>
>>
>> When I use UDP as the transport protocol, everything works fine.? This
>> is down to (correct me if I am wrong) the NAT router being able to
>> inspect the SIP traffic, and changing the SIP INVITE message so that
>> the public IP is used for the RTP stream.? This is seen in the
>> following ( 1.2.3.4 = SIP UA, ??50.50.50.50 = Freeswitch server) ?
>>
>>
>>
>> ? recv 1042 bytes from udp/[1.2.3.4]:34019 at 08:40:24.302496:
>>
>>
>> ----------------------------------------------------------------------
>> --
>>
>> ?? INVITE sip:1004 at 50.50.50.50 SIP/2.0
>>
>> ?? Via: SIP/2.0/UDP 1.2.3.4:34019;branch=z9hG4bKD8xhgo0smbsOQMUy;rport
>>
>> ?? Contact: <sip:1002 at 1.2.3.4:34019>
>>
>> ?? Max-Forwards: 70
>>
>> ?? From: "1002"
>> <sip:1002 at 50.50.50.50>;tag=A6E182EBB96E21919A44C50399F1BF27
>>
>> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>>
>> ?? Supported: replaces, path
>>
>> ?? User-Agent: Acrobits Softphone Business/1.8.8
>>
>> ?? To: <sip:1004 at 50.50.50.50>
>>
>> ?? Content-Type: application/sdp
>>
>> ?? Call-ID: 2926496CCA1C6189C90A52D821B06B5E9DBA2B32
>>
>> ?? CSeq: 1 INVITE
>>
>> ?? Proxy-Authorization: Digest
> username="1002",realm="50.50.50.50",algorithm=MD5,uri="sip:1004 at 50.50.50.50"
> ,nonce="09195369-6474-4310-b3a6-60e7d9c780f1",qop=auth,cnonce="1b774f833fd01
> b5b29954c2ddc2b5457",nc=00000002,response="07bda5b4c5fa5e17f653a3e9c7a2f05e"
>>
>> ?? Content-Length: 241
>>
>>
>>
>> ???v=0
>>
>> ?? o=- 10805 6353 IN IP4 1.2.3.4? < --- Public IP address used so that the
> other party knows to send the RTP stream to this address.
>>
>> ?? s=xynwwjx
>>
>> ?? c=IN IP4 1.2.3.4
>>
>> ?? t=0 0
>>
>> ?? m=audio 34962 RTP/AVP 102 3 0 8 9 101
>>
>> ?? a=rtpmap:101 telephone-event/8000
>>
>> ?? a=rtpmap:102 ILBC/8000
>>
>> ?? a=fmtp:102 mode=30
>>
>> ?? a=fmtp:101 0-15
>>
>> ?? a=ptime:30
>>
>> ?? a=sendrecv
>>
>>
>>
>> This was obtained from the Siptrace on the Freeswitch server.? The
> Freeswitch server then forwards these details to the other party.
>>
>>
>>
>> The problem occurs when TLS is used, as the NAT firewalls are unable
>> to inspect the SIP packets.? Here is another SIPtrace on the
>> Freeswitch server for TLS example (again I have changed the public IPs
>> to the same as the above example) ?
>>
>>
>>
>> recv 1048 bytes from tls/[1.2.3.4]:64403 at 16:00:42.271063:
>>
>>
>> ----------------------------------------------------------------------
>> --
>>
>> ?? INVITE sip:1002 at 50.50.50.50 SIP/2.0
>>
>> ?? Via: SIP/2.0/TLS 1.2.3.4:64403;branch=z9hG4bKz6qAkQj3XsYuJ5Os;rport
>>
>> ?? Contact: <sip:1004 at 1.2.3.4:64403;transport=tls>
>>
>> ?? Max-Forwards: 70
>>
>> ?? From: "Ben"
>> <sip:1004 at 50.50.50.50>;tag=F28D08D5E039D8FE021D08022A72F982
>>
>> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>>
>> ?? Supported: replaces, path
>>
>> ?? User-Agent: Acrobits Softphone Business/1.8.8
>>
>> ?? To: <sip:1002 at 50.50.50.50>
>>
>> ?? Content-Type: application/sdp
>>
>> ?? Call-ID: 32166133760BDDAD823725B563F6A1B3989090D6
>>
>> ?? CSeq: 1 INVITE
>>
>> ?? Authorization: Digest
> username="1004",realm="50.50.50.50",algorithm=MD5,uri="sip:1002 at 50.50.50.50"
> ,nonce="5af9829e-716d-4472-924f-84eb41d9e78e",qop=auth,cnonce="d641702f981e1
> 0f3a41e074061bdb5b3",nc=00000003,response="a9f577ac37f8f367b2f839a58d43e1e0"
>
>>
>> ?? Content-Length: 240
>>
>>
>>
>> ???v=0
>>
>> ?? o=- 71766 47529 IN IP4 172.16.234.176 < --- Using Private IP for
>> the RTP details
>>
>> ?? s=fscmvnc
>>
>> ?? c=IN IP4 172.16.234.176
>>
>> ?? t=0 0
>>
>> ?? m=audio 64628 RTP/AVP 3 102 101
>>
>> ?? a=rtpmap:101 telephone-event/8000
>>
>> ?? a=rtpmap:102 ILBC/8000
>>
>> ?? a=fmtp:102 mode=30
>>
>> ?? a=fmtp:101 0-15
>>
>> ?? a=ptime:30
>>
>> ?? a=sendrecv
>>
>>
>> ----------------------------------------------------------------------
>> --
>>
>>
>>
>> As you can see, the SIP UAs are going to try and use a non-existent
> private IP address to send the RTP stream to.
>>
>> I can also confirm that the UAs are trying to do this, by looking at the
> SIP logs built in to the SIP UA client software.
>>
>>
>>
>> My (Freeswitch related) question is, am I able to set something in
> Freeswitch to change the details to be the public IP of the firewall?? The
> SIP messages include the correct IP, can the Freeswitch server use this
> address and replace the Private IP with this one?
>>
>>
>>
>> I have heard about TLS proxies/STUn servers etc etc, but would much rather
> use Freeswitch to do this.
>>
>>
>>
>> Any suggestions are welcome, also please let me know if I have
> misunderstood any of the concepts behind SIP/VOIP and firewalls ? I am
> rather new to the VOIP game!
>>
>>
>>
>> Kind regards
>>
>>
>>
>> Ben
>>
>>
>>
>>
>>
>>
>>
>> Ben Naylor
>>
>> Network Support Engineer
>>
>>
>>
>> T: +44 (0)207 096 1648`
>>
>> M: +44 (0)7924 349 113
>>
>> F:? +44 (0)207 150 595
>>
>> E: bnaylor at sirran.com W: www.sirran.com
>>
>>
>>
>> A: SiRRAN Communications Ltd, The Hawk Business Park, Easingwold, YO61
>> 3FE
>>
>> Registered Office: SiRRAN Communications Ltd,. 6-9 Trinity Street
>> Dublin 2, Ireland
>>
>>
>>
>> _____________________________________________________________
>>
>> The information contained in this e-mail and any attached documents may be
> privileged, confidential and protected from disclosure. If you are not the
> intended recipient you may not read, copy, distribute or use this
> information. If you have received this communication in error, please notify
> the sender immediately by replying to this message and then delete it from
> your system.
>>
>>
>>
>>
>>
>> FreeSWITCH-users 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
>>
>
>
>
> --
> Kristian Kielhofner
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 claudio at clickfono.com  Thu Oct 13 19:03:04 2011
From: claudio at clickfono.com (=?ISO-8859-1?Q?Claudio_=C1lvarez?=)
Date: Thu, 13 Oct 2011 12:03:04 -0300
Subject: [Freeswitch-users] ESL Ruby - sched_api / task id issue
Message-ID: <4E96FDA8.8@clickfono.com>

Hi all,

I'm currently experimenting with sched_api to schedule a ringback sound 
from a Ruby script. My code looks like this:

ringback_result = con.bgapi('sched_api +8 none uuid_broadcast ' + 
leg1_uuid + ' /var/www/mi/audio/ringback.mp3 aleg')
puts "Serialized result:"
puts ringback_result.serialize()

When I run the script, my output looks like:

Serialized result:
Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK%20Job-UUID%3A%20cec047cc-97d7-4320-811a-e15e454843f9
Job-UUID: cec047cc-97d7-4320-811a-e15e454843f9

The problem here is that I'm not getting the scheduled task ID returned 
by sched_api I expect. I need the task ID so as to run sched_del should 
I need to abort the task. Trying the same api_sched command on fs_cli I 
get an output confirming that the task has been added, together with the 
assigned task ID.

sched_api +8 none uuid_broadcast 41b482a9-8595-47d4-a3fd-3c6a0d32e942 
/var/www/mi/audio/ringback.mp3 aleg
+OK Added: 34

Can anyone help me out with this one?

Thanks!

-- 
Claudio ?lvarez, Ph.D.
Arquitecto de Sistemas / Systems Architect
Medularis S.p.A. - ClickFono
Tel:  56 2 934 40 74
Cel: 56 9 9079 4951
claudio at clickfono.com

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From infos at madovsky.org  Thu Oct 13 19:29:56 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 13 Oct 2011 11:29:56 -0400
Subject: [Freeswitch-users] Is Bgapi threadsafe?
References: <001401cc8909$ab2e48f0$018adad0$@google.hm>	<CAKbxfG9aNd-3Jmims5gmCiZA_UOcm9giG8Zo3F9QPZTWhEh6Zw@mail.gmail.com>	<CAKbxfG_64fBccqjf6kddF6bsPnyy=VzGWGM9zOuq6jmhLT8=VQ@mail.gmail.com><CAKbxfG9A0dnxcYioFDTjYf1ire7ALiKmmFbpghkjOxBWooi1=g@mail.gmail.com>
	<001401cc89b1$9008bc10$b01a3430$@google.hm>
Message-ID: <1B5C5EF34A4A4D81905FBB4D7E7FB97B@e1705>

and don't forget to sniffing glue.. 8o)

----- Original Message ----- 
From: "John Rose" <johnrose at google.hm>
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, October 13, 2011 10:08 AM
Subject: Re: [Freeswitch-users] Is Bgapi threadsafe?


OK thanks, you've answered my question. So basically calling the methods on
an ESL connection, Bgapi for example, from a client is not threadsafe, the
method calls should be exclusively locked when using multiple threads since
it's utilizing one socket. That makes sense for the one connection scenario.

John

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> Subject: Re: [Freeswitch-users] Is Bgapi threadsafe?
>
> wrong week to quit sniffing glue....
>
> correction to last sentence of previous email:
>
> and you **can't** do i/o on it from 2 threads at once naturally.
>
> On Wed, Oct 12, 2011 at 4:07 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > oh but wait...
> >
> > if you mean you want threads on your side to share it, I'd either give
> > them all their own socket or share one on a mutex cos it's a socket
> > and you can do i/o on it from 2 threads at once naturally.
> >
> >
> > On Wed, Oct 12, 2011 at 4:06 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> yes since its a socket and the other side is properly threaded. If
> >> you are using the in-tree ESL lib the client also has race condition
> >> protection
> >>
> >> On Wed, Oct 12, 2011 at 1:06 PM, John Rose <johnrose at google.hm>
> wrote:
> >>> Does anyone know if calling Bgapi on an ESL connection is
> >>> threadsafe? I'm calling it from ManagedESL via C# and have a lock
> >>> around it but would like to remove the lock if possible.
> >>>
> >>> John
> >>>
> >>>
> >>>



FreeSWITCH-users mailing list
FreeSWITCH-users at 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 bnaylor at sirran.com  Thu Oct 13 19:48:18 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Thu, 13 Oct 2011 16:48:18 +0100
Subject: [Freeswitch-users] TLS problems
In-Reply-To: <CAKDfjgdFZah8YWOCAU13EdDjqX6tuAA1isMKRNzx1TZ2hQS4WA@mail.gmail.com>
References: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>	<CAKDfjgcm1ADmsY4Bdj4Kpr7WcvuFrnjCPiP=agADH81n0ge03A@mail.gmail.com>	<005f01cc89b6$d76a1a90$863e4fb0$@sirran.com>
	<CAKDfjgdFZah8YWOCAU13EdDjqX6tuAA1isMKRNzx1TZ2hQS4WA@mail.gmail.com>
Message-ID: <006901cc89bf$87b66c20$97234460$@sirran.com>

Many thanks for the info Kristian, it has definetly been a bit of a
nightmare understanding how it all works!

Bypass_media is unfortunately unavoidable, so it looks like it's going to be
a STUN server.  You mention SIP latching/RTP proxy, would this replace the
need for a STUN server, or work alongside it?

The benefit of a STUN server is that it supports TLS, which is an absolute
requirement in our case.

Thanks again for your help.

Ben.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: 13 October 2011 15:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TLS problems

Ben,

  SIP+NAT, where do I even start ;)...

Local/near handling (STUN, TURN, ICE, etc) "Middle" handling (SIP ALG, SBC)
Far end handling (SIP latching/RTP proxy)

  Your natted devices are currently sitting behind a NAT device with a SIP
ALG (application layer gateway).  As you (correctly) guessed this device
sits and watches for SIP packets.  When it sees them it rewrites various
fields (c=, Contact:, etc) to effectively insert itself into the dialog.  It
will then (transparently) rewrite these fields between both sides of the NAT
while also handling RTP.  This would be "middle" handling.

  This works with bypass_media because your endpoints see the "fixed"
SDP fields from the ALG and know where to send media.  Because the ALG
device is unable to inspect TLS packets it can't rewrite these addresses.
Normally FreeSWITCH would insert itself in the media path (Far end
handling), proxy the media stream, and rewrite the packets to match.  With
bypass_media enabled FreeSWITCH can't do this.

  With TLS + NAT your options are:

- Configure STUN/TURN/ICE on your client (Acrobits) and cross your fingers
- Disable bypass_media in FreeSWITCH

On Thu, Oct 13, 2011 at 10:46 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Ahh yes, I forgot to mention that I am using bypass_media, apologies 
> for not mentioning this before!
>
> Logs to follow shortly......
>
> Thanks for the help,
>
> Ben
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Kristian Kielhofner
> Sent: 13 October 2011 15:39
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] TLS problems
>
> Hi Ben,
>
> ?FreeSWITCH handles NAT rather well and should automatically ignore 
> the natted SDP info; using the actual send/receive RTP IP/port once 
> media arrives. ?It will then leave itself in the media path (like it 
> usually does
> anyway) and magically fix SDP/RTP data to other clients.
>
> ?Are you doing anything funny like bypass_media?
>
> ?Please include a SIP trace and FS console logs so we can see if what 
> FreeSWITCH is doing.
>
> On Thu, Oct 13, 2011 at 8:12 AM, Ben Naylor <bnaylor at sirran.com> wrote:
>>
>> Hi all
>>
>>
>>
>> I was wondering if anyone can help me with a problem I am having with
> using TLS SIP when NAT routers are involved?
>>
>>
>>
>> The scenario is as follows ?
>>
>>
>>
>> SIP UA --- > NAT firewall? --- > Internet --- > Freeswitch --- > 
>> Internet --- > NAT Firewall --- > SIP UA
>>
>>
>>
>> When I use UDP as the transport protocol, everything works fine.? 
>> This is down to (correct me if I am wrong) the NAT router being able 
>> to inspect the SIP traffic, and changing the SIP INVITE message so 
>> that the public IP is used for the RTP stream.? This is seen in the 
>> following ( 1.2.3.4 = SIP UA, ??50.50.50.50 = Freeswitch server) ?
>>
>>
>>
>> ? recv 1042 bytes from udp/[1.2.3.4]:34019 at 08:40:24.302496:
>>
>>
>> ---------------------------------------------------------------------
>> -
>> --
>>
>> ?? INVITE sip:1004 at 50.50.50.50 SIP/2.0
>>
>> ?? Via: SIP/2.0/UDP 
>> 1.2.3.4:34019;branch=z9hG4bKD8xhgo0smbsOQMUy;rport
>>
>> ?? Contact: <sip:1002 at 1.2.3.4:34019>
>>
>> ?? Max-Forwards: 70
>>
>> ?? From: "1002"
>> <sip:1002 at 50.50.50.50>;tag=A6E182EBB96E21919A44C50399F1BF27
>>
>> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>>
>> ?? Supported: replaces, path
>>
>> ?? User-Agent: Acrobits Softphone Business/1.8.8
>>
>> ?? To: <sip:1004 at 50.50.50.50>
>>
>> ?? Content-Type: application/sdp
>>
>> ?? Call-ID: 2926496CCA1C6189C90A52D821B06B5E9DBA2B32
>>
>> ?? CSeq: 1 INVITE
>>
>> ?? Proxy-Authorization: Digest
>
username="1002",realm="50.50.50.50",algorithm=MD5,uri="sip:1004 at 50.50.50.50"
> ,nonce="09195369-6474-4310-b3a6-60e7d9c780f1",qop=auth,cnonce="1b774f8
> 33fd01 
>
b5b29954c2ddc2b5457",nc=00000002,response="07bda5b4c5fa5e17f653a3e9c7a2f05e"
>>
>> ?? Content-Length: 241
>>
>>
>>
>> ???v=0
>>
>> ?? o=- 10805 6353 IN IP4 1.2.3.4? < --- Public IP address used so 
>> that the
> other party knows to send the RTP stream to this address.
>>
>> ?? s=xynwwjx
>>
>> ?? c=IN IP4 1.2.3.4
>>
>> ?? t=0 0
>>
>> ?? m=audio 34962 RTP/AVP 102 3 0 8 9 101
>>
>> ?? a=rtpmap:101 telephone-event/8000
>>
>> ?? a=rtpmap:102 ILBC/8000
>>
>> ?? a=fmtp:102 mode=30
>>
>> ?? a=fmtp:101 0-15
>>
>> ?? a=ptime:30
>>
>> ?? a=sendrecv
>>
>>
>>
>> This was obtained from the Siptrace on the Freeswitch server.? The
> Freeswitch server then forwards these details to the other party.
>>
>>
>>
>> The problem occurs when TLS is used, as the NAT firewalls are unable 
>> to inspect the SIP packets.? Here is another SIPtrace on the 
>> Freeswitch server for TLS example (again I have changed the public 
>> IPs to the same as the above example) ?
>>
>>
>>
>> recv 1048 bytes from tls/[1.2.3.4]:64403 at 16:00:42.271063:
>>
>>
>> ---------------------------------------------------------------------
>> -
>> --
>>
>> ?? INVITE sip:1002 at 50.50.50.50 SIP/2.0
>>
>> ?? Via: SIP/2.0/TLS 
>> 1.2.3.4:64403;branch=z9hG4bKz6qAkQj3XsYuJ5Os;rport
>>
>> ?? Contact: <sip:1004 at 1.2.3.4:64403;transport=tls>
>>
>> ?? Max-Forwards: 70
>>
>> ?? From: "Ben"
>> <sip:1004 at 50.50.50.50>;tag=F28D08D5E039D8FE021D08022A72F982
>>
>> ?? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>>
>> ?? Supported: replaces, path
>>
>> ?? User-Agent: Acrobits Softphone Business/1.8.8
>>
>> ?? To: <sip:1002 at 50.50.50.50>
>>
>> ?? Content-Type: application/sdp
>>
>> ?? Call-ID: 32166133760BDDAD823725B563F6A1B3989090D6
>>
>> ?? CSeq: 1 INVITE
>>
>> ?? Authorization: Digest
>
username="1004",realm="50.50.50.50",algorithm=MD5,uri="sip:1002 at 50.50.50.50"
> ,nonce="5af9829e-716d-4472-924f-84eb41d9e78e",qop=auth,cnonce="d641702
> f981e1 
>
0f3a41e074061bdb5b3",nc=00000003,response="a9f577ac37f8f367b2f839a58d43e1e0"
>
>>
>> ?? Content-Length: 240
>>
>>
>>
>> ???v=0
>>
>> ?? o=- 71766 47529 IN IP4 172.16.234.176 < --- Using Private IP for 
>> the RTP details
>>
>> ?? s=fscmvnc
>>
>> ?? c=IN IP4 172.16.234.176
>>
>> ?? t=0 0
>>
>> ?? m=audio 64628 RTP/AVP 3 102 101
>>
>> ?? a=rtpmap:101 telephone-event/8000
>>
>> ?? a=rtpmap:102 ILBC/8000
>>
>> ?? a=fmtp:102 mode=30
>>
>> ?? a=fmtp:101 0-15
>>
>> ?? a=ptime:30
>>
>> ?? a=sendrecv
>>
>>
>> ---------------------------------------------------------------------
>> -
>> --
>>
>>
>>
>> As you can see, the SIP UAs are going to try and use a non-existent
> private IP address to send the RTP stream to.
>>
>> I can also confirm that the UAs are trying to do this, by looking at 
>> the
> SIP logs built in to the SIP UA client software.
>>
>>
>>
>> My (Freeswitch related) question is, am I able to set something in
> Freeswitch to change the details to be the public IP of the firewall?? 
> The SIP messages include the correct IP, can the Freeswitch server use 
> this address and replace the Private IP with this one?
>>
>>
>>
>> I have heard about TLS proxies/STUn servers etc etc, but would much 
>> rather
> use Freeswitch to do this.
>>
>>
>>
>> Any suggestions are welcome, also please let me know if I have
> misunderstood any of the concepts behind SIP/VOIP and firewalls ? I am 
> rather new to the VOIP game!
>>
>>
>>
>> Kind regards
>>
>>
>>
>> Ben
>>
>>
>>
>>
>>
>>
>>
>> Ben Naylor
>>
>> Network Support Engineer
>>
>>
>>
>> T: +44 (0)207 096 1648`
>>
>> M: +44 (0)7924 349 113
>>
>> F:? +44 (0)207 150 595
>>
>> E: bnaylor at sirran.com W: www.sirran.com
>>
>>
>>
>> A: SiRRAN Communications Ltd, The Hawk Business Park, Easingwold, 
>> YO61 3FE
>>
>> Registered Office: SiRRAN Communications Ltd,. 6-9 Trinity Street 
>> Dublin 2, Ireland
>>
>>
>>
>> _____________________________________________________________
>>
>> The information contained in this e-mail and any attached documents 
>> may be
> privileged, confidential and protected from disclosure. If you are not 
> the intended recipient you may not read, copy, distribute or use this 
> information. If you have received this communication in error, please 
> notify the sender immediately by replying to this message and then 
> delete it from your system.
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> e
>> rs
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kristian Kielhofner
>
>
> FreeSWITCH-users 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
>
>
>
> FreeSWITCH-users 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
>



--
Kristian Kielhofner


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



From kris at kriskinc.com  Thu Oct 13 20:03:34 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 13 Oct 2011 12:03:34 -0400
Subject: [Freeswitch-users] TLS problems
In-Reply-To: <006901cc89bf$87b66c20$97234460$@sirran.com>
References: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>
	<CAKDfjgcm1ADmsY4Bdj4Kpr7WcvuFrnjCPiP=agADH81n0ge03A@mail.gmail.com>
	<005f01cc89b6$d76a1a90$863e4fb0$@sirran.com>
	<CAKDfjgdFZah8YWOCAU13EdDjqX6tuAA1isMKRNzx1TZ2hQS4WA@mail.gmail.com>
	<006901cc89bf$87b66c20$97234460$@sirran.com>
Message-ID: <CAKDfjgcYgyt_O7VAbAhYZP5qCv-eJtntgJwcXn8N3uvh2WX6mQ@mail.gmail.com>

Ben,

  If your endpoints support STUN and everything works correctly you
won't need an RTP proxy or anything else (you're using near end NAT
traversal).  If you absolutely cannot disable bypass_media mode an RTP
proxy is out of the question.  However - there is a mode of TURN/ICE
that effectively is an RTP proxy...

  Clear as mud, right?

On Thu, Oct 13, 2011 at 11:48 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Many thanks for the info Kristian, it has definetly been a bit of a
> nightmare understanding how it all works!
>
> Bypass_media is unfortunately unavoidable, so it looks like it's going to be
> a STUN server. ?You mention SIP latching/RTP proxy, would this replace the
> need for a STUN server, or work alongside it?
>
> The benefit of a STUN server is that it supports TLS, which is an absolute
> requirement in our case.
>
> Thanks again for your help.
>
> Ben.

-- 
Kristian Kielhofner


From claudio at clickfono.com  Thu Oct 13 20:13:34 2011
From: claudio at clickfono.com (=?ISO-8859-1?Q?Claudio_=C1lvarez?=)
Date: Thu, 13 Oct 2011 13:13:34 -0300
Subject: [Freeswitch-users] ESL Ruby - sched_api / task id issue
In-Reply-To: <4E96FDA8.8@clickfono.com>
References: <4E96FDA8.8@clickfono.com>
Message-ID: <4E970E2E.5090202@clickfono.com>

Hi all,

I finally nailed it. I was calling bgapi (api call in the background), 
whereas I should have just called the synchronous 'api' method. The 
synchronous api call returns the scheduled task inmediatly:

result = con.api('sched_api +8 none uuid_broadcast ' + leg1_uuid + ' 
/var/www/mi/audio/ringback.mp3 aleg')
result_body = result.getBody().to_s.gsub(/\n/,'')
result_elements = result_body.scan(/(.+\:)\s+(\d+)/).collect { 
|result_msg, id| { :result_msg => result_msg, :id => id }
job_id = result_elements[0][:id]
puts "job_id: #{job_id}"
puts "uuid_setvar #{leg1_uuid} ringback_job_id #{job_id}"
con.bgapi("uuid_setvar #{leg1_uuid} ringback_job_id #{job_id}")

Output:

job_id: 46
uuid_setvar 2e41916f-8f33-4c4e-9583-91b9953a61e1 ringback_job_id 46

Thanks anyway.

Best,

Claudio

On 10/13/11 12:03 PM, Claudio ?lvarez wrote:
> Hi all,
>
> I'm currently experimenting with sched_api to schedule a ringback 
> sound from a Ruby script. My code looks like this:
>
> ringback_result = con.bgapi('sched_api +8 none uuid_broadcast ' + 
> leg1_uuid + ' /var/www/mi/audio/ringback.mp3 aleg')
> puts "Serialized result:"
> puts ringback_result.serialize()
>
> When I run the script, my output looks like:
>
> Serialized result:
> Event-Name: SOCKET_DATA
> Content-Type: command/reply
> Reply-Text: %2BOK%20Job-UUID%3A%20cec047cc-97d7-4320-811a-e15e454843f9
> Job-UUID: cec047cc-97d7-4320-811a-e15e454843f9
>
> The problem here is that I'm not getting the scheduled task ID 
> returned by sched_api I expect. I need the task ID so as to run 
> sched_del should I need to abort the task. Trying the same api_sched 
> command on fs_cli I get an output confirming that the task has been 
> added, together with the assigned task ID.
>
> sched_api +8 none uuid_broadcast 41b482a9-8595-47d4-a3fd-3c6a0d32e942 
> /var/www/mi/audio/ringback.mp3 aleg
> +OK Added: 34
>
> Can anyone help me out with this one?
>
> Thanks!
> -- 
> Claudio ?lvarez, Ph.D.
> Arquitecto de Sistemas / Systems Architect
> Medularis S.p.A. - ClickFono
> Tel:  56 2 934 40 74
> Cel: 56 9 9079 4951
> claudio at clickfono.com


-- 
Claudio ?lvarez, Ph.D.
Arquitecto de Sistemas / Systems Architect
Medularis S.p.A. - ClickFono
Tel:  56 2 934 40 74
Cel: 56 9 9079 4951
claudio at clickfono.com

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From avi at avimarcus.net  Thu Oct 13 20:33:07 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 13 Oct 2011 18:33:07 +0200
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <C6693088B9934A72B6A893751427F42C@e1705>
References: <73233639A2EE4ACC9DDBC958E457099B@e1705> <4E965E77.90704@tagnet.ru>
	<709ED34B0EB94BE0B95D0C2D97ED3D8A@e1705> <4E9669E1.7060509@tagnet.ru>
	<C6693088B9934A72B6A893751427F42C@e1705>
Message-ID: <CALNZuwn1t9qa7SJsnNOCPccLWeSrFUJjEm2x=BNjAxiG_eViVw@mail.gmail.com>

Wiki says:

> It will return a value suitable for use with the bridge application in the
> variable ${lcr_auto_route}. It will also populate the variables lcr_route_N
> (where N is 1 to route count) and lcr_route_count for more flexible use of
> the lcr information in scripts.

You can bridge to ${lcr_route_1} and then ${lcr_route_2}, etc. but I don't
know how many carriers it will end up returning..

-Avi


On Thu, Oct 13, 2011 at 7:07 AM, Madovsky <infos at madovsky.org> wrote:

> **
> no ex sorry.
> try to hash you mod_lcr result in different variables
> and create multiline bridge with continue_on_fail and hangup_cause
> conidtions between them.
>
>
> ----- Original Message -----
> *From:* Boris Kovalenko <boris at tagnet.ru>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, October 13, 2011 12:32 AM
> *Subject:* Re: [Freeswitch-users] continue_on_fail
>
> Hello!
>
>     May You give an example? I use mod_lcr
>
>  try to use bridge witn single line
> and put condition with hangup cause between....
>
> hope it helps
>
> ----- Original Message -----
> *From:* Boris Kovalenko <boris at tagnet.ru>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, October 12, 2011 11:43 PM
> *Subject:* Re: [Freeswitch-users] continue_on_fail
>
> Hello!
>
>     No. I even tried continue_on_fail=false and hangup_after_bridge=true
>
> did you resolve it ?
>
> after some test I'm facing the same problem
> if for example the callee decides to send a user busy so FS tries with the
> second bridge and
> makes a second call which can be interpreted as harrassment ;)
>
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, October 10, 2011 11:28 PM
> *Subject:* Re: [Freeswitch-users] continue_on_fail
>
> maybe modify continue_on_fail as a leg var on the first bridge
>
> ----- Original Message -----
> *From:* Boris Kovalenko <boris at tagnet.ru>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, October 10, 2011 11:12 PM
> *Subject:* Re: [Freeswitch-users] continue_on_fail
>
> Hello!
>
>     No, the question was how to FAILOVER :) I need to stop processing
> dialstring if user at carrier1 is busy.
>
>  the question was about mod_lcr.
> it returns something like:
>
> sofia/gateway/carrier1/14105780999|sofia/gateway/
> carrier2/14105780999|sofia/gateway/ carrier3/14105780999
>
> The question was how to make it when the first returns "user_busy" to not failover and try the other ones.
>
> -Avi
>
>
> On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Well, now we're talking about something more than just continuing_on_fail.
>> We'll need to look at the actual dialstring to answer any more specific
>> questions.
>>
>> -MC
>>
>>
>> On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> It seems from testing, and I was told on channel:  continue_on_fail is
>>> per bridge string, not per leg. So it doesn't actually help with the
>>> question..
>>> Unless I'm mistaken.
>>> -Avi
>>>
>>>
>>> On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> You will need to specify the error conditions on which to continue. See
>>>> the example here:
>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>>
>>>> -MC
>>>>
>>>> P.S. - Check out the "enumeration" column on this page to get an idea of
>>>> the kinds of failures you may not be aware of:
>>>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>>>
>>>>
>>>> On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>
>>>>> I've been wondering how to do this.
>>>>> When I asked in channel, I was basically told this couldn't be done in
>>>>> a single bridge string... I hope that's not actually the case.
>>>>> -Avi
>>>>>
>>>>>
>>>>> On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru>wrote:
>>>>>
>>>>>> Hello!
>>>>>>
>>>>>>     Have read docs but misundertood :( I use mod_lcr which return two
>>>>>> or more gateways to the same destination. And when the user is busy
>>>>>> with
>>>>>> the first gateway the call continues with the second and so on. How
>>>>>> may
>>>>>> I to not continue gateways if user is busy?
>>>>>>
>>>>>> --
>>>>>> ? ?????????,
>>>>>>   ????? ?????????
>>>>>>   ??? "??????"
>>>>>>   ???. +7 (3435) 230001 <%2B7%20%283435%29%20230001>
>>>>>>   ???? +7 (3435) 230005 <%2B7%20%283435%29%20230005>
>>>>>>
>>>>>>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
> ------------------------------
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>  ------------------------------
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>  ------------------------------
>
>
> FreeSWITCH-users mailing 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 Hector.Geraldino at ip-soft.net  Thu Oct 13 20:35:38 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Thu, 13 Oct 2011 12:35:38 -0400
Subject: [Freeswitch-users] ESL Ruby - sched_api / task id issue
In-Reply-To: <4E96FDA8.8@clickfono.com>
References: <4E96FDA8.8@clickfono.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02209F82A2@NY1-EXMB-01.ip-soft.net>

Hi,

I don't know if that's the intended behavior or not, but I can cancel a scheduled task by calling sched_del and passing the uuid as a parameter (no task_id whatsoever)

e.g.
Schedule a hangup:
"sched_hangup +10 " + uuid;

Cancel scheduled hangup:
sched_del uuid

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Claudio ?lvarez
Sent: Thursday, October 13, 2011 11:03 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] ESL Ruby - sched_api / task id issue

Hi all,

I'm currently experimenting with sched_api to schedule a ringback sound from a Ruby script. My code looks like this:

ringback_result = con.bgapi('sched_api +8 none uuid_broadcast ' + leg1_uuid + ' /var/www/mi/audio/ringback.mp3 aleg')
puts "Serialized result:"
puts ringback_result.serialize()

When I run the script, my output looks like:

Serialized result:
Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK%20Job-UUID%3A%20cec047cc-97d7-4320-811a-e15e454843f9
Job-UUID: cec047cc-97d7-4320-811a-e15e454843f9

The problem here is that I'm not getting the scheduled task ID returned by sched_api I expect. I need the task ID so as to run sched_del should I need to abort the task. Trying the same api_sched command on fs_cli I get an output confirming that the task has been added, together with the assigned task ID.

sched_api +8 none uuid_broadcast 41b482a9-8595-47d4-a3fd-3c6a0d32e942 /var/www/mi/audio/ringback.mp3 aleg
+OK Added: 34

Can anyone help me out with this one?

Thanks!


--

Claudio ?lvarez, Ph.D.

Arquitecto de Sistemas / Systems Architect

Medularis S.p.A. - ClickFono

Tel:  56 2 934 40 74

Cel: 56 9 9079 4951

claudio at clickfono.com<mailto:claudio at clickfono.com>
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From boris at tagnet.ru  Thu Oct 13 20:38:44 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 13 Oct 2011 22:38:44 +0600
Subject: [Freeswitch-users] continue_on_fail
In-Reply-To: <CALNZuwn1t9qa7SJsnNOCPccLWeSrFUJjEm2x=BNjAxiG_eViVw@mail.gmail.com>
References: <73233639A2EE4ACC9DDBC958E457099B@e1705> <4E965E77.90704@tagnet.ru>
	<709ED34B0EB94BE0B95D0C2D97ED3D8A@e1705>
	<4E9669E1.7060509@tagnet.ru>
	<C6693088B9934A72B6A893751427F42C@e1705>
	<CALNZuwn1t9qa7SJsnNOCPccLWeSrFUJjEm2x=BNjAxiG_eViVw@mail.gmail.com>
Message-ID: <4E971414.1060507@tagnet.ru>

Hello!

     Yes, in scripts it may be usefull, but I'm useing XML.

> Wiki says:
>
>     It will return a value suitable for use with the bridge
>     application in the variable ${lcr_auto_route}. It will also
>     populate the variables lcr_route_N (where N is 1 to route count)
>     and lcr_route_count for more flexible use of the lcr information
>     in scripts.
>
> You can bridge to ${lcr_route_1} and then ${lcr_route_2}, etc. but I 
> don't know how many carriers it will end up returning..
>
> -Avi
>
>
> On Thu, Oct 13, 2011 at 7:07 AM, Madovsky <infos at madovsky.org 
> <mailto:infos at madovsky.org>> wrote:
>
>     no ex sorry.
>     try to hash you mod_lcr result in different variables
>     and create multiline bridge with continue_on_fail and hangup_cause
>     conidtions between them.
>
>         ----- Original Message -----
>         *From:* Boris Kovalenko <mailto:boris at tagnet.ru>
>         *To:* FreeSWITCH Users Help
>         <mailto:freeswitch-users at lists.freeswitch.org>
>         *Sent:* Thursday, October 13, 2011 12:32 AM
>         *Subject:* Re: [Freeswitch-users] continue_on_fail
>
>         Hello!
>
>             May You give an example? I use mod_lcr
>
>>         try to use bridge witn single line
>>         and put condition with hangup cause between....
>>         hope it helps
>>
>>             ----- Original Message -----
>>             *From:* Boris Kovalenko <mailto:boris at tagnet.ru>
>>             *To:* FreeSWITCH Users Help
>>             <mailto:freeswitch-users at lists.freeswitch.org>
>>             *Sent:* Wednesday, October 12, 2011 11:43 PM
>>             *Subject:* Re: [Freeswitch-users] continue_on_fail
>>
>>             Hello!
>>
>>                 No. I even tried continue_on_fail=false and
>>             hangup_after_bridge=true
>>>             did you resolve it ?
>>>             after some test I'm facing the same problem
>>>             if for example the callee decides to send a user busy so
>>>             FS tries with the second bridge and
>>>             makes a second call which can be interpreted as
>>>             harrassment ;)
>>>
>>>                 ----- Original Message -----
>>>                 *From:* Madovsky <mailto:infos at madovsky.org>
>>>                 *To:* FreeSWITCH Users Help
>>>                 <mailto:freeswitch-users at lists.freeswitch.org>
>>>                 *Sent:* Monday, October 10, 2011 11:28 PM
>>>                 *Subject:* Re: [Freeswitch-users] continue_on_fail
>>>
>>>                 maybe modify continue_on_fail as a leg var on the
>>>                 first bridge
>>>
>>>                     ----- Original Message -----
>>>                     *From:* Boris Kovalenko <mailto:boris at tagnet.ru>
>>>                     *To:* FreeSWITCH Users Help
>>>                     <mailto:freeswitch-users at lists.freeswitch.org>
>>>                     *Sent:* Monday, October 10, 2011 11:12 PM
>>>                     *Subject:* Re: [Freeswitch-users] continue_on_fail
>>>
>>>                     Hello!
>>>
>>>                         No, the question was how to FAILOVER :) I
>>>                     need to stop processing dialstring if user at
>>>                     carrier1 is busy.
>>>
>>>>                     the question was about mod_lcr.
>>>>                     it returns something like:
>>>>
>>>>                     sofia/gateway/carrier1/14105780999|sofia/gateway/
>>>>                     carrier2/14105780999|sofia/gateway/
>>>>                     carrier3/14105780999
>>>>                     The question was how to make it when the first returns "user_busy" to not failover and try the other ones.
>>>>                     -Avi
>>>>
>>>>
>>>>                     On Mon, Oct 10, 2011 at 6:40 PM, Michael
>>>>                     Collins <msc at freeswitch.org
>>>>                     <mailto:msc at freeswitch.org>> wrote:
>>>>
>>>>                         Well, now we're talking about something
>>>>                         more than just continuing_on_fail. We'll
>>>>                         need to look at the actual dialstring to
>>>>                         answer any more specific questions.
>>>>
>>>>                         -MC
>>>>
>>>>
>>>>                         On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus
>>>>                         <avi at avimarcus.net
>>>>                         <mailto:avi at avimarcus.net>> wrote:
>>>>
>>>>                             It seems from testing, and I was told
>>>>                             on channel:  continue_on_fail is per
>>>>                             bridge string, not per leg. So it
>>>>                             doesn't actually help with the question..
>>>>                             Unless I'm mistaken.
>>>>                             -Avi
>>>>
>>>>
>>>>                             On Mon, Oct 10, 2011 at 5:12 PM,
>>>>                             Michael Collins <msc at freeswitch.org
>>>>                             <mailto:msc at freeswitch.org>> wrote:
>>>>
>>>>                                 You will need to specify the error
>>>>                                 conditions on which to continue.
>>>>                                 See the example here:
>>>>                                 http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>>
>>>>                                 -MC
>>>>
>>>>                                 P.S. - Check out the "enumeration"
>>>>                                 column on this page to get an idea
>>>>                                 of the kinds of failures you may
>>>>                                 not be aware of:
>>>>                                 http://wiki.freeswitch.org/wiki/Hangup_causes
>>>>
>>>>
>>>>
>>>>                                 On Sat, Oct 8, 2011 at 2:54 PM, Avi
>>>>                                 Marcus <avi at avimarcus.net
>>>>                                 <mailto:avi at avimarcus.net>> wrote:
>>>>
>>>>                                     I've been wondering how to do
>>>>                                     this.
>>>>                                     When I asked in channel, I was
>>>>                                     basically told this couldn't be
>>>>                                     done in a single bridge
>>>>                                     string... I hope that's not
>>>>                                     actually the case.
>>>>                                     -Avi
>>>>
>>>>
>>>>                                     On Fri, Oct 7, 2011 at 4:40 PM,
>>>>                                     Boris Kovalenko
>>>>                                     <boris at tagnet.ru
>>>>                                     <mailto:boris at tagnet.ru>> wrote:
>>>>
>>>>                                         Hello!
>>>>
>>>>                                             Have read docs but
>>>>                                         misundertood :( I use
>>>>                                         mod_lcr which return two
>>>>                                         or more gateways to the
>>>>                                         same destination. And when
>>>>                                         the user is busy with
>>>>                                         the first gateway the call
>>>>                                         continues with the second
>>>>                                         and so on. How may
>>>>                                         I to not continue gateways
>>>>                                         if user is busy?
>>>>
>>>>                                         --
>>>>                                         ? ?????????,
>>>>                                           ????? ?????????
>>>>                                           ??? "??????"
>>>>                                           ???. +7 (3435) 230001
>>>>                                         <tel:%2B7%20%283435%29%20230001>
>>>>                                           ???? +7 (3435) 230005
>>>>                                         <tel:%2B7%20%283435%29%20230005>
>>>>
>>>>
>>>>
>>>>                                         FreeSWITCH-users mailing list
>>>>                                         FreeSWITCH-users at lists.freeswitch.org
>>>>                                         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>                                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>                                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>                                         http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>                                     FreeSWITCH-users mailing list
>>>>                                     FreeSWITCH-users at lists.freeswitch.org
>>>>                                     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>                                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>                                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>                                     http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>                                 FreeSWITCH-users mailing list
>>>>                                 FreeSWITCH-users at lists.freeswitch.org
>>>>                                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>
>>>>
>>>>
>>>>
>>>>                             FreeSWITCH-users mailing list
>>>>                             FreeSWITCH-users at lists.freeswitch.org
>>>>                             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>                             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>                             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>                             http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>                         FreeSWITCH-users mailing list
>>>>                         FreeSWITCH-users at lists.freeswitch.org
>>>>                         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>                         http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>                     FreeSWITCH-users mailing list
>>>>                     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>                     http://www.freeswitch.org
>>>
>>>
>>>                     -- 
>>>                     Regards,
>>>                     Boris
>>>
>>>
>>>                     ------------------------------------------------------------------------
>>>
>>>                     FreeSWITCH-users mailing list
>>>                     FreeSWITCH-users at lists.freeswitch.org
>>>                     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>                     http://www.freeswitch.org
>>>
>>>
>>>
>>>             FreeSWITCH-users mailing list
>>>             FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>
>>
>>             -- 
>>             Regards,
>>             Boris
>>
>>
>>             ------------------------------------------------------------------------
>>
>>             FreeSWITCH-users mailing list
>>             FreeSWITCH-users at lists.freeswitch.org
>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>             http://www.freeswitch.org
>>
>>
>>
>>         FreeSWITCH-users mailing list
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>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>
>
>         -- 
>         Regards,
>         Boris
>
>
>         ------------------------------------------------------------------------
>
>         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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>
>
>
>     FreeSWITCH-users mailing list
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>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
>
>
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-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From johnrose at google.hm  Thu Oct 13 20:53:12 2011
From: johnrose at google.hm (John Rose)
Date: Thu, 13 Oct 2011 12:53:12 -0400
Subject: [Freeswitch-users] Is Bgapi threadsafe?
In-Reply-To: <1B5C5EF34A4A4D81905FBB4D7E7FB97B@e1705>
References: <001401cc8909$ab2e48f0$018adad0$@google.hm>	<CAKbxfG9aNd-3Jmims5gmCiZA_UOcm9giG8Zo3F9QPZTWhEh6Zw@mail.gmail.com>	<CAKbxfG_64fBccqjf6kddF6bsPnyy=VzGWGM9zOuq6jmhLT8=VQ@mail.gmail.com><CAKbxfG9A0dnxcYioFDTjYf1ire7ALiKmmFbpghkjOxBWooi1=g@mail.gmail.com>	<001401cc89b1$9008bc10$b01a3430$@google.hm>
	<1B5C5EF34A4A4D81905FBB4D7E7FB97B@e1705>
Message-ID: <001801cc89c8$991645e0$cb42d1a0$@google.hm>

Make sure there is an ample supply of coffee and Elmer's BEFORE checking in
the code.


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Madovsky
> Sent: Thursday, October 13, 2011 11:30 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Is Bgapi threadsafe?
> 
> and don't forget to sniffing glue.. 8o)
> 
> ----- Original Message -----
> From: "John Rose" <johnrose at google.hm>
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, October 13, 2011 10:08 AM
> Subject: Re: [Freeswitch-users] Is Bgapi threadsafe?
> 
> 
> OK thanks, you've answered my question. So basically calling the methods
on
> an ESL connection, Bgapi for example, from a client is not threadsafe, the
> method calls should be exclusively locked when using multiple threads
since
> it's utilizing one socket. That makes sense for the one connection
scenario.
> 
> John
> 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > Subject: Re: [Freeswitch-users] Is Bgapi threadsafe?
> >
> > wrong week to quit sniffing glue....
> >
> > correction to last sentence of previous email:
> >
> > and you **can't** do i/o on it from 2 threads at once naturally.
> >
> > On Wed, Oct 12, 2011 at 4:07 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> > > oh but wait...
> > >
> > > if you mean you want threads on your side to share it, I'd either give
> > > them all their own socket or share one on a mutex cos it's a socket
> > > and you can do i/o on it from 2 threads at once naturally.
> > >
> > >
> > > On Wed, Oct 12, 2011 at 4:06 PM, Anthony Minessale
> > > <anthony.minessale at gmail.com> wrote:
> > >> yes since its a socket and the other side is properly threaded. If
> > >> you are using the in-tree ESL lib the client also has race condition
> > >> protection
> > >>
> > >> On Wed, Oct 12, 2011 at 1:06 PM, John Rose <johnrose at google.hm>
> > wrote:
> > >>> Does anyone know if calling Bgapi on an ESL connection is
> > >>> threadsafe? I'm calling it from ManagedESL via C# and have a lock
> > >>> around it but would like to remove the lock if possible.
> > >>>
> > >>> John
> > >>>
> > >>>
> > >>>
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Thu Oct 13 20:57:08 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 13 Oct 2011 12:57:08 -0400
Subject: [Freeswitch-users] continue_on_fail
References: <73233639A2EE4ACC9DDBC958E457099B@e1705>
	<4E965E77.90704@tagnet.ru><709ED34B0EB94BE0B95D0C2D97ED3D8A@e1705><4E9669E1.7060509@tagnet.ru><C6693088B9934A72B6A893751427F42C@e1705><CALNZuwn1t9qa7SJsnNOCPccLWeSrFUJjEm2x=BNjAxiG_eViVw@mail.gmail.com>
	<4E971414.1060507@tagnet.ru>
Message-ID: <26A32CDAB7E543FAB808BD06E91D80CE@e1705>

it can be done also in XML I think..
  ----- Original Message ----- 
  From: Boris Kovalenko 
  To: FreeSWITCH Users Help 
  Sent: Thursday, October 13, 2011 12:38 PM
  Subject: Re: [Freeswitch-users] continue_on_fail


  Hello!

      Yes, in scripts it may be usefull, but I'm useing XML.


    Wiki says: 
      It will return a value suitable for use with the bridge application in the variable ${lcr_auto_route}. It will also populate the variables lcr_route_N (where N is 1 to route count) and lcr_route_count for more flexible use of the lcr information in scripts.
    You can bridge to ${lcr_route_1} and then ${lcr_route_2}, etc. but I don't know how many carriers it will end up returning..


    -Avi



    On Thu, Oct 13, 2011 at 7:07 AM, Madovsky <infos at madovsky.org> wrote:

      no ex sorry.
      try to hash you mod_lcr result in different variables
      and create multiline bridge with continue_on_fail and hangup_cause conidtions between them.

        ----- Original Message ----- 
        From: Boris Kovalenko 
        To: FreeSWITCH Users Help 
        Sent: Thursday, October 13, 2011 12:32 AM
        Subject: Re: [Freeswitch-users] continue_on_fail


        Hello!

            May You give an example? I use mod_lcr 


          try to use bridge witn single line
          and put condition with hangup cause between....

          hope it helps
            ----- Original Message ----- 
            From: Boris Kovalenko 
            To: FreeSWITCH Users Help 
            Sent: Wednesday, October 12, 2011 11:43 PM
            Subject: Re: [Freeswitch-users] continue_on_fail


            Hello!

                No. I even tried continue_on_fail=false and hangup_after_bridge=true

              did you resolve it ?

              after some test I'm facing the same problem
              if for example the callee decides to send a user busy so FS tries with the second bridge and
              makes a second call which can be interpreted as harrassment ;)

                ----- Original Message ----- 
                From: Madovsky 
                To: FreeSWITCH Users Help 
                Sent: Monday, October 10, 2011 11:28 PM
                Subject: Re: [Freeswitch-users] continue_on_fail


                maybe modify continue_on_fail as a leg var on the first bridge
                  ----- Original Message ----- 
                  From: Boris Kovalenko 
                  To: FreeSWITCH Users Help 
                  Sent: Monday, October 10, 2011 11:12 PM
                  Subject: Re: [Freeswitch-users] continue_on_fail


                  Hello!

                      No, the question was how to FAILOVER :) I need to stop processing dialstring if user at carrier1 is busy.


                    the question was about mod_lcr. 
                    it returns something like:


                    sofia/gateway/carrier1/14105780999|sofia/gateway/ carrier2/14105780999|sofia/gateway/ carrier3/14105780999
The question was how to make it when the first returns "user_busy" to not failover and try the other ones.-Avi



                    On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc at freeswitch.org> wrote:

                      Well, now we're talking about something more than just continuing_on_fail. We'll need to look at the actual dialstring to answer any more specific questions. 


                      -MC 



                      On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi at avimarcus.net> wrote:

                        It seems from testing, and I was told on channel:  continue_on_fail is per bridge string, not per leg. So it doesn't actually help with the question.. 
                        Unless I'm mistaken.

                        -Avi



                        On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc at freeswitch.org> wrote:

                          You will need to specify the error conditions on which to continue. See the example here: 
                          http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail


                          -MC


                          P.S. - Check out the "enumeration" column on this page to get an idea of the kinds of failures you may not be aware of: http://wiki.freeswitch.org/wiki/Hangup_causes 



                          On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

                            I've been wondering how to do this. 
                            When I asked in channel, I was basically told this couldn't be done in a single bridge string... I hope that's not actually the case.

                            -Avi



                            On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

                              Hello!

                                  Have read docs but misundertood :( I use mod_lcr which return two
                              or more gateways to the same destination. And when the user is busy with
                              the first gateway the call continues with the second and so on. How may
                              I to not continue gateways if user is busy?

                              --
                              ? ?????????,
                                ????? ?????????
                                ??? "??????"
                                ???. +7 (3435) 230001
                                ???? +7 (3435) 230005



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From asilva at wirelessmundi.com  Thu Oct 13 21:09:33 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Thu, 13 Oct 2011 19:09:33 +0200
Subject: [Freeswitch-users] Loop XML using call_timeout ..
Message-ID: <1318525773.20249.10.camel@marces.madrid.commsmundi.com>


Hi,

Is it possible to loop a bridge until a global call timeout (or other
variable) is reached?

In the wiki is this example is a fixed number of loops...

<extension name="3-way gateway distro"> 

<condition field="destination_number" expression="^(.*)$">
      <action application="set" data="continue_on_fail=true"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge" data="sofia/gateway/${distributor(3gw)}/$1" loop="3"/>
    </condition>
  </extension>

My idea is to call a group sequential, but once it reaches the last member it starts over again.
so when calling a group, once the timeout for each user in the group is reach it hangups the call (the normal behavior),
it jumps again to the begin of the group until a global timeout is reached.

For example:

<extension name="group"> 
<condition field="destination_number" expression="^(.*)$">
      <action application="set" data="continue_on_fail=true"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set" data="call_timeout=60"/>
      <action application="set" data="leg_timeout=10"/>
      <action application="bridge" data=""${group_call(group1 at default+F)}" loop="3"/>
    </condition>
  </extension>

In my group i have two members only.





-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From jeff at jefflenk.com  Thu Oct 13 22:38:57 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 13 Oct 2011 11:38:57 -0700 (PDT)
Subject: [Freeswitch-users] Call/Transaction Does Not Exist
In-Reply-To: <CALH5ZrbxMHUFdBzv_du4t+ji_Hcp9y5qiNs4mvGm86uCsq4wkw@mail.gmail.com>
References: <CALH5ZrbFUBNMrBEwN+uJMO+ee0R=X0=g00rUi6+jPuOT=Y17SQ@mail.gmail.com>
	<CAKbxfG9zcByP3un6RJNRUuw4hJhSd+KZ+tYmLbQd+WmFPwxc8A@mail.gmail.com>
	<CALH5ZrbxMHUFdBzv_du4t+ji_Hcp9y5qiNs4mvGm86uCsq4wkw@mail.gmail.com>
Message-ID: <1318531137389-6890071.post@n2.nabble.com>

You need to make current and reproduce on git head your version is from May.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-Transaction-Does-Not-Exist-tp6882730p6890071.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Oct 14 01:21:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Oct 2011 16:21:04 -0500
Subject: [Freeswitch-users] Call/Transaction Does Not Exist
In-Reply-To: <CALH5ZrbxMHUFdBzv_du4t+ji_Hcp9y5qiNs4mvGm86uCsq4wkw@mail.gmail.com>
References: <CALH5ZrbFUBNMrBEwN+uJMO+ee0R=X0=g00rUi6+jPuOT=Y17SQ@mail.gmail.com>
	<CAKbxfG9zcByP3un6RJNRUuw4hJhSd+KZ+tYmLbQd+WmFPwxc8A@mail.gmail.com>
	<CALH5ZrbxMHUFdBzv_du4t+ji_Hcp9y5qiNs4mvGm86uCsq4wkw@mail.gmail.com>
Message-ID: <CAKbxfG8Xtbb4PJ_4fuvaWtbnBiTf_Vh3XPQAAMnAjaDoYY_ZjA@mail.gmail.com>

4 things to try in order of likelihood to help but you should do all 4
regardless.

1) edit your sofia profile and add <param name="NDLB-force-rport" value="true"/>
2) Update your polycom to latest firmware
3) Update to latest FreeSWITCH
4) make sure your network router does not have any SIP ALG enabled.


On Wed, Oct 12, 2011 at 8:34 PM, Joe Flemmings <joe.jflemmings at gmail.com> wrote:
> Thanks for the help Anthony,
>
> Bellow is the pastebin URL
>
> http://pastebin.freeswitch.org/17518
>
> Joe
>
> On Wed, Oct 12, 2011 at 8:51 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> collect a full trace and put it on pastebin.freeswitch.org
>>
>> console loglevel debug (/log debug on fs_cli)
>> sofia global siptrace on
>>
>>
>>
>>
>> On Tue, Oct 11, 2011 at 4:29 PM, Joe Flemmings <joe.jflemmings at gmail.com>
>> wrote:
>> > When i call from one extension and hangup, the other call does not
>> > hangup
>> > and continues ringing. Please see log bellow.
>> >
>> > Any ideas?
>> >
>> >
>> >
>> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> > ? INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
>> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> > ? <sip:213 at 10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
>> > OPTIONS, INFO, MES
>> > ? SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
>> > en..Supported:
>> > 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards:
>> > ? 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
>> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN IP4
>> > 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
>> > ? 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
>> > telephone-event/8000..
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> > ? SIP/2.0 100 Trying..Via: SIP/2.0/UDP
>> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> > <sip:213 at 10.10.78.36;user=phone>
>> > ? ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1
>> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
>> > 22-51-43 -0500..Content-Length: 0....
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> > ? SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
>> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 6
>> > ? 6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 INVITE..User-Agent:
>> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43
>> > -0500..Accept:
>> > appl
>> > ? ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
>> > UPDATE,
>> > INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
>> > precondition, path, replaces..Allow-Events: talk, hold, prese
>> > ? nce, dialog, line-seize, call-info, sla, include-session-description,
>> > presence.winfo, message-summary, refer..Proxy-Authenticate: Digest
>> > realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac
>> > ? ", algorithm=MD5, qop="auth"..Content-Length: 0....
>> > #
>> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> > ? ACK sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
>> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <si
>> > ? p:213 at 10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 ACK..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
>> > OPTION
>> > ? S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
>> > en..Max-Forwards: 70..Content-Length: 0....
>> > #
>> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> > ? INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
>> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> > ? <sip:213 at 10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL,
>> > OPTIONS, INFO, MES
>> > ? SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
>> > en..Supported:
>> > 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz
>> > ? ation: Digest username="pbxjker_2564", realm="10.10.78.36",
>> > nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth,
>> > cnonce="KnfhF00Oi0GfJCi", nc=00000001,
>> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> > ? response="82a75edb7ba97bfe7dea585684701d0c",
>> > algorithm=MD5..Max-Forwards:
>> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
>> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone
>> > ? ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8
>> > 18
>> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
>> > ? telephone-event/8000..
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> > ? SIP/2.0 100 Trying..Via: SIP/2.0/UDP
>> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> > <sip:213 at 10.10.78.36;user=phone>
>> > ? ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
>> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23
>> > 22-51-43 -0500..Content-Length: 0....
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:52783
>> > ? INVITE sip:pbxjker_213 at 10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP
>> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: 69..From:
>> > "Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81
>> > ? rv8Fp..To: <sip:pbxjker_213 at 10.10.78.24:52783>..Call-ID:
>> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact:
>> > <sip:mod_sofia at 10.10.78.36:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.
>> > ? head-git-1086cba 2011-05-23 22-51-43 -0500..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-Type: application/sdp..Content-Disposition: session..
>> > ? Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID:
>> > "Polycom One"
>> >
>> > <sip:pbxjker_2564 at 10.10.78.36>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
>> > 1318333156 1318333157 IN IP4
>> > ?? 10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio 29124
>> > RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98
>> > bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320
>> > ? 00..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>> > 0-16..a=ptime:20..
>> > #
>> > U 10.10.78.24:55516 -> 10.10.78.36:5060
>> > ? SIP/2.0 100 Trying..Via: SIP/2.0/UDP
>> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
>> > <sip:pbxjker_213 at 99.65.2
>> > ? 17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
>> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
>> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
>> > PolycomSoundPointIP-SPIP_320-UA/3.2
>> > ? .2.0477..Accept-Language: en..Content-Length: 0....
>> > #
>> > U 10.10.78.24:55516 -> 10.10.78.36:5060
>> > ? SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
>> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
>> > <sip:pbxjker_213 at 99.65.
>> > ? 217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
>> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
>> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
>> > PolycomSoundPointIP-SPIP_320-UA/3.
>> > ? 2.2.0477..Allow-Events: talk,hold,conference..Accept-Language:
>> > en..Content-Length: 0....
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> > ? SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
>> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> > <sip:213 at 10.10.78.36;user=phone
>> > ? >;tag=8aQNK2QyUKjXa..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 INVITE..Contact:
>> > <sip:213 at 10.10.78.36:5060;transport=udp>..User-Agent:
>> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2
>> > ? 3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE,
>> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
>> > PUBLISH,
>> > SUBSCRIBE..Supported: timer, precondition, path, replaces..A
>> > ? llow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
>> > include-session-description, presence.winfo, message-summary,
>> > refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker
>> > ? _213>;party=calling;privacy=off;screen=no....
>> > #
>> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> > ? CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
>> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
>> > ? 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
>> > OPTIONS,
>> > INFO, MESSAGE, SUBSCR
>> > ? IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
>> > username="pbxjker_2564", realm="10.10.78.36",
>> > nonce="ab4a9421-fbb9-41fd-943d-462
>> > ? 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
>> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> > response="d47848babd8c396d8da15145503875d0",
>> > algorithm=MD5..Max-Forwards:
>> > 70..Content-Length: 0..
>> > ? ..
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:5060
>> > ? SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
>> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
>> > ? 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
>> > 0....
>> > #
>> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> > ? CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
>> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
>> > ? 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
>> > OPTIONS,
>> > INFO, MESSAGE, SUBSCR
>> > ? IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
>> > username="pbxjker_2564", realm="10.10.78.36",
>> > nonce="ab4a9421-fbb9-41fd-943d-462
>> > ? 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
>> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> > response="d47848babd8c396d8da15145503875d0",
>> > algorithm=MD5..Max-Forwards:
>> > 70..Content-Length: 0..
>> > ? ..
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:5060
>> > ? SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
>> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
>> > ? 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
>> > 0....
>> > #
>> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> > ? CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
>> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
>> > ? 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
>> > OPTIONS,
>> > INFO, MESSAGE, SUBSCR
>> > ? IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest
>> > username="pbxjker_2564", realm="10.10.78.36",
>> > nonce="ab4a9421-fbb9-41fd-943d-462
>> > ? 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
>> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> > response="d47848babd8c396d8da15145503875d0",
>> > algorithm=MD5..Max-Forwards:
>> > 70..Content-Length: 0..
>> > ? ..
>> > #
>> > U 10.10.78.36:5060 -> 10.10.78.24:5060
>> > ? SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
>> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 66.11
>> > ? 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
>> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 CANCEL..Content-Length:
>> > 0....
>> > #
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From bnaylor at sirran.com  Fri Oct 14 01:33:41 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Thu, 13 Oct 2011 22:33:41 +0100
Subject: [Freeswitch-users] TLS problems
In-Reply-To: <CAKDfjgcYgyt_O7VAbAhYZP5qCv-eJtntgJwcXn8N3uvh2WX6mQ@mail.gmail.com>
References: <001e01cc89a1$618b4980$24a1dc80$@sirran.com>	<CAKDfjgcm1ADmsY4Bdj4Kpr7WcvuFrnjCPiP=agADH81n0ge03A@mail.gmail.com>	<005f01cc89b6$d76a1a90$863e4fb0$@sirran.com>	<CAKDfjgdFZah8YWOCAU13EdDjqX6tuAA1isMKRNzx1TZ2hQS4WA@mail.gmail.com>	<006901cc89bf$87b66c20$97234460$@sirran.com>
	<CAKDfjgcYgyt_O7VAbAhYZP5qCv-eJtntgJwcXn8N3uvh2WX6mQ@mail.gmail.com>
Message-ID: <009f01cc89ef$c828ae00$587a0a00$@sirran.com>

Yeah clear as! : )

I will have a go at setting up my own STUN server, although there doesn't
seem to be many options out there!

Do you have any suggestions for possible STUN servers to use?

Kind regards

Ben

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: 13 October 2011 17:04
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TLS problems

Ben,

  If your endpoints support STUN and everything works correctly you won't
need an RTP proxy or anything else (you're using near end NAT traversal).
If you absolutely cannot disable bypass_media mode an RTP proxy is out of
the question.  However - there is a mode of TURN/ICE that effectively is an
RTP proxy...

  Clear as mud, right?

On Thu, Oct 13, 2011 at 11:48 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Many thanks for the info Kristian, it has definetly been a bit of a 
> nightmare understanding how it all works!
>
> Bypass_media is unfortunately unavoidable, so it looks like it's going 
> to be a STUN server. ?You mention SIP latching/RTP proxy, would this 
> replace the need for a STUN server, or work alongside it?
>
> The benefit of a STUN server is that it supports TLS, which is an 
> absolute requirement in our case.
>
> Thanks again for your help.
>
> Ben.

--
Kristian Kielhofner


FreeSWITCH-users mailing list
FreeSWITCH-users at 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 reza at lethalnetworks.com  Fri Oct 14 00:26:17 2011
From: reza at lethalnetworks.com (reza a)
Date: Thu, 13 Oct 2011 13:26:17 -0700 (PDT)
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <0344E3C7596C466F882CCE8292FC796A@e1705>
Message-ID: <112cd66a-f570-4e74-ab44-53b02b87eae5@zcs>

Thanks everyone for the replies, the information was certainly helpful. I was wondering if anyone had any docs, howtos, or anything else on the subject matter to kind of guide along the process of setting something like this up. It would be my first real project with FreeSwitch too, so i'm unsure of any pitfalls i may run into.

----- Original Message -----
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, September 26, 2011 12:08:01 PM
Subject: Re: [Freeswitch-users] prepaid solutions



it works now... 


----- Original Message ----- 
From: Michel Daggelinckx 
To: FreeSWITCH Users Help 
Sent: Monday, September 26, 2011 2:14 PM 
Subject: Re: [Freeswitch-users] prepaid solutions 

digitallinx is behind this 


On Mon, Sep 26, 2011 at 8:10 PM, Muhammad Naseer Bhatti < nbhatti at gmail.com > wrote: 


Check your connectivity. Works fine here. Verified from different locations too. 




On Mon, Sep 26, 2011 at 9:06 PM, Madovsky < infos at madovsky.org > wrote: 
> when I click on 
> http://vbilling.org/about-us/ 
> -> server not found 
> 
> maybe it would be cool to know who are behind this solution... 
> 
> 
> ----- Original Message ----- 
> From: "Muhammad Naseer Bhatti" < nbhatti at gmail.com > 
> To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
> Sent: Monday, September 26, 2011 1:50 PM 
> Subject: Re: [Freeswitch-users] prepaid solutions 
> 
> 
>> Open source solution. http://www.vbilling.org/ 
>> 
>> 
>> On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx 
>> < michel.daggelinckx at gmail.com > wrote: 
>>> mod nibblebill and mod lcr is a good start 
>>> 
>>> On Mon, Sep 26, 2011 at 1:59 AM, reza a < reza at lethalnetworks.com > wrote: 
>>>> 
>>>> Freeswitch users, 
>>>> Does anyone use freeswitch for prepaid applications? Do you have a walk 
>>>> through on setting up an environment like this? I'd like to play with 
>>>> freeswitch a bit, I know there alot of modules out there but I'm 
>>>> wondering 
>>>> what's the best cohesive set of modules/admin tools to use for a prepaid 
>>>> style applications. 
>>>> Thanks a lot 
>>>> 
>>>> 
>>>> FreeSWITCH-users mailing list 
>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>>>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>>> http://www.freeswitch.org 
>>> 
>>> 
>>> 
>>> FreeSWITCH-users mailing list 
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>> http://www.freeswitch.org 
>>> 
>>> 
>> 
>> 
>> FreeSWITCH-users mailing list 
>> FreeSWITCH-users at lists.freeswitch.org 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
>> http://www.freeswitch.org 
> 
> 
> 
> FreeSWITCH-users mailing list 
> FreeSWITCH-users at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
> http://www.freeswitch.org 
> 


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







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


FreeSWITCH-users mailing list
FreeSWITCH-users at 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 joe.jflemmings at gmail.com  Fri Oct 14 01:39:40 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Thu, 13 Oct 2011 14:39:40 -0700
Subject: [Freeswitch-users] Call/Transaction Does Not Exist
In-Reply-To: <CAKbxfG8Xtbb4PJ_4fuvaWtbnBiTf_Vh3XPQAAMnAjaDoYY_ZjA@mail.gmail.com>
References: <CALH5ZrbFUBNMrBEwN+uJMO+ee0R=X0=g00rUi6+jPuOT=Y17SQ@mail.gmail.com>
	<CAKbxfG9zcByP3un6RJNRUuw4hJhSd+KZ+tYmLbQd+WmFPwxc8A@mail.gmail.com>
	<CALH5ZrbxMHUFdBzv_du4t+ji_Hcp9y5qiNs4mvGm86uCsq4wkw@mail.gmail.com>
	<CAKbxfG8Xtbb4PJ_4fuvaWtbnBiTf_Vh3XPQAAMnAjaDoYY_ZjA@mail.gmail.com>
Message-ID: <CALH5Zra3eJFe1KXcL+ObktejUkEG+rL_4zWaNeJ52CQ2QQE1yw@mail.gmail.com>

Actually just updated to the latest Freeswitch and its working.

If i don't write back then consider this to be working.

Thank you

On Thu, Oct 13, 2011 at 2:21 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 4 things to try in order of likelihood to help but you should do all 4
> regardless.
>
> 1) edit your sofia profile and add <param name="NDLB-force-rport"
> value="true"/>
> 2) Update your polycom to latest firmware
> 3) Update to latest FreeSWITCH
> 4) make sure your network router does not have any SIP ALG enabled.
>
>
> On Wed, Oct 12, 2011 at 8:34 PM, Joe Flemmings <joe.jflemmings at gmail.com>
> wrote:
> > Thanks for the help Anthony,
> >
> > Bellow is the pastebin URL
> >
> > http://pastebin.freeswitch.org/17518
> >
> > Joe
> >
> > On Wed, Oct 12, 2011 at 8:51 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> collect a full trace and put it on pastebin.freeswitch.org
> >>
> >> console loglevel debug (/log debug on fs_cli)
> >> sofia global siptrace on
> >>
> >>
> >>
> >>
> >> On Tue, Oct 11, 2011 at 4:29 PM, Joe Flemmings <
> joe.jflemmings at gmail.com>
> >> wrote:
> >> > When i call from one extension and hangup, the other call does not
> >> > hangup
> >> > and continues ringing. Please see log bellow.
> >> >
> >> > Any ideas?
> >> >
> >> >
> >> >
> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >> >   INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
> SIP/2.0/UDP
> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> >> >   <sip:213 at 10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> >> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE,
> CANCEL,
> >> > OPTIONS, INFO, MES
> >> >   SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
> >> > en..Supported:
> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards:
> >> >   70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN IP4
> >> > 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
> >> >   101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
> >> > telephone-event/8000..
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> >> > <sip:213 at 10.10.78.36;user=phone>
> >> >   ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1
> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba
> 2011-05-23
> >> > 22-51-43 -0500..Content-Length: 0....
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >> >   SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 6
> >> >   6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 INVITE..User-Agent:
> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43
> >> > -0500..Accept:
> >> > appl
> >> >   ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
> >> > UPDATE,
> >> > INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
> >> > precondition, path, replaces..Allow-Events: talk, hold, prese
> >> >   nce, dialog, line-seize, call-info, sla,
> include-session-description,
> >> > presence.winfo, message-summary, refer..Proxy-Authenticate: Digest
> >> > realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac
> >> >   ", algorithm=MD5, qop="auth"..Content-Length: 0....
> >> > #
> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >> >   ACK sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <si
> >> >   p:213 at 10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1
> ACK..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> >> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE,
> CANCEL,
> >> > OPTION
> >> >   S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
> REFER..User-Agent:
> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
> >> > en..Max-Forwards: 70..Content-Length: 0....
> >> > #
> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >> >   INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
> SIP/2.0/UDP
> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> >> >   <sip:213 at 10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> >> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE,
> CANCEL,
> >> > OPTIONS, INFO, MES
> >> >   SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
> >> > en..Supported:
> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz
> >> >   ation: Digest username="pbxjker_2564", realm="10.10.78.36",
> >> > nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth,
> >> > cnonce="KnfhF00Oi0GfJCi", nc=00000001,
> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> >> >   response="82a75edb7ba97bfe7dea585684701d0c",
> >> > algorithm=MD5..Max-Forwards:
> >> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone
> >> >   ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0
> 8
> >> > 18
> >> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
> >> >   telephone-event/8000..
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> >> > <sip:213 at 10.10.78.36;user=phone>
> >> >   ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba
> 2011-05-23
> >> > 22-51-43 -0500..Content-Length: 0....
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:52783
> >> >   INVITE sip:pbxjker_213 at 10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP
> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: 69..From:
> >> > "Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81
> >> >   rv8Fp..To: <sip:pbxjker_213 at 10.10.78.24:52783>..Call-ID:
> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact:
> >> > <sip:mod_sofia at 10.10.78.36:5060>..User-Agent:
> FreeSWITCH-mod_sofia/1.0.
> >> >   head-git-1086cba 2011-05-23 22-51-43 -0500..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-Type: application/sdp..Content-Disposition: session..
> >> >   Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID:
> >> > "Polycom One"
> >> >
> >> > <sip:pbxjker_2564 at 10.10.78.36
> >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
> >> > 1318333156 1318333157 IN IP4
> >> >    10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio
> 29124
> >> > RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98
> >> > bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320
> >> >   00..a=fmtp:18 annexb=no..a=rtpmap:101
> telephone-event/8000..a=fmtp:101
> >> > 0-16..a=ptime:20..
> >> > #
> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060
> >> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
> >> > <sip:pbxjker_213 at 99.65.2
> >> >   17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
> >> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
> >> > PolycomSoundPointIP-SPIP_320-UA/3.2
> >> >   .2.0477..Accept-Language: en..Content-Length: 0....
> >> > #
> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060
> >> >   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
> >> > <sip:pbxjker_213 at 99.65.
> >> >   217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
> >> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
> >> > PolycomSoundPointIP-SPIP_320-UA/3.
> >> >   2.2.0477..Allow-Events: talk,hold,conference..Accept-Language:
> >> > en..Content-Length: 0....
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
> >> >   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> >> > <sip:213 at 10.10.78.36;user=phone
> >> >   >;tag=8aQNK2QyUKjXa..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 INVITE..Contact:
> >> > <sip:213 at 10.10.78.36:5060;transport=udp>..User-Agent:
> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2
> >> >   3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE,
> >> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
> >> > PUBLISH,
> >> > SUBSCRIBE..Supported: timer, precondition, path, replaces..A
> >> >   llow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla,
> >> > include-session-description, presence.winfo, message-summary,
> >> > refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker
> >> >   _213>;party=calling;privacy=off;screen=no....
> >> > #
> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
> SIP/2.0/UDP
> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> >> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> >> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
> >> > OPTIONS,
> >> > INFO, MESSAGE, SUBSCR
> >> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization:
> Digest
> >> > username="pbxjker_2564", realm="10.10.78.36",
> >> > nonce="ab4a9421-fbb9-41fd-943d-462
> >> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> >> > response="d47848babd8c396d8da15145503875d0",
> >> > algorithm=MD5..Max-Forwards:
> >> > 70..Content-Length: 0..
> >> >   ..
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060
> >> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> <sip:213 at 66.11
> >> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
> CANCEL..Content-Length:
> >> > 0....
> >> > #
> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
> SIP/2.0/UDP
> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> >> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> >> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
> >> > OPTIONS,
> >> > INFO, MESSAGE, SUBSCR
> >> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization:
> Digest
> >> > username="pbxjker_2564", realm="10.10.78.36",
> >> > nonce="ab4a9421-fbb9-41fd-943d-462
> >> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> >> > response="d47848babd8c396d8da15145503875d0",
> >> > algorithm=MD5..Max-Forwards:
> >> > 70..Content-Length: 0..
> >> >   ..
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060
> >> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> <sip:213 at 66.11
> >> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
> CANCEL..Content-Length:
> >> > 0....
> >> > #
> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
> >> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
> SIP/2.0/UDP
> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
> >> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
> >> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
> >> > OPTIONS,
> >> > INFO, MESSAGE, SUBSCR
> >> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization:
> Digest
> >> > username="pbxjker_2564", realm="10.10.78.36",
> >> > nonce="ab4a9421-fbb9-41fd-943d-462
> >> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
> >> > response="d47848babd8c396d8da15145503875d0",
> >> > algorithm=MD5..Max-Forwards:
> >> > 70..Content-Length: 0..
> >> >   ..
> >> > #
> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060
> >> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
> <sip:213 at 66.11
> >> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
> CANCEL..Content-Length:
> >> > 0....
> >> > #
> >> >
> >> >
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joe.jflemmings at gmail.com  Fri Oct 14 01:44:39 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Thu, 13 Oct 2011 14:44:39 -0700
Subject: [Freeswitch-users] Call/Transaction Does Not Exist - RESOLVED
Message-ID: <CALH5ZrZicFbo7yt5Fo6RB_HEO3x4pSvS+29ng4Jw6ykDiC=k4A@mail.gmail.com>

This has been resolved by upgrading to the latest version

Thank You


On Thu, Oct 13, 2011 at 2:39 PM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:

> Actually just updated to the latest Freeswitch and its working.
>
> If i don't write back then consider this to be working.
>
> Thank you
>
>
> On Thu, Oct 13, 2011 at 2:21 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> 4 things to try in order of likelihood to help but you should do all 4
>> regardless.
>>
>> 1) edit your sofia profile and add <param name="NDLB-force-rport"
>> value="true"/>
>> 2) Update your polycom to latest firmware
>> 3) Update to latest FreeSWITCH
>> 4) make sure your network router does not have any SIP ALG enabled.
>>
>>
>> On Wed, Oct 12, 2011 at 8:34 PM, Joe Flemmings <joe.jflemmings at gmail.com>
>> wrote:
>> > Thanks for the help Anthony,
>> >
>> > Bellow is the pastebin URL
>> >
>> > http://pastebin.freeswitch.org/17518
>> >
>> > Joe
>> >
>> > On Wed, Oct 12, 2011 at 8:51 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> collect a full trace and put it on pastebin.freeswitch.org
>> >>
>> >> console loglevel debug (/log debug on fs_cli)
>> >> sofia global siptrace on
>> >>
>> >>
>> >>
>> >>
>> >> On Tue, Oct 11, 2011 at 4:29 PM, Joe Flemmings <
>> joe.jflemmings at gmail.com>
>> >> wrote:
>> >> > When i call from one extension and hangup, the other call does not
>> >> > hangup
>> >> > and continues ringing. Please see log bellow.
>> >> >
>> >> > Any ideas?
>> >> >
>> >> >
>> >> >
>> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> >> >   INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
>> SIP/2.0/UDP
>> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom
>> One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> >> >   <sip:213 at 10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> >> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE,
>> CANCEL,
>> >> > OPTIONS, INFO, MES
>> >> >   SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
>> >> > en..Supported:
>> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards:
>> >> >   70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
>> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN
>> IP4
>> >> > 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18
>> >> >   101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
>> >> > telephone-event/8000..
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> >> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
>> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom
>> One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> >> > <sip:213 at 10.10.78.36;user=phone>
>> >> >   ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1
>> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba
>> 2011-05-23
>> >> > 22-51-43 -0500..Content-Length: 0....
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> >> >   SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
>> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom
>> One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213 at 6
>> >> >   6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 INVITE..User-Agent:
>> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43
>> >> > -0500..Accept:
>> >> > appl
>> >> >   ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
>> >> > UPDATE,
>> >> > INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
>> >> > precondition, path, replaces..Allow-Events: talk, hold, prese
>> >> >   nce, dialog, line-seize, call-info, sla,
>> include-session-description,
>> >> > presence.winfo, message-summary, refer..Proxy-Authenticate: Digest
>> >> > realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac
>> >> >   ", algorithm=MD5, qop="auth"..Content-Length: 0....
>> >> > #
>> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> >> >   ACK sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP
>> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom
>> One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <si
>> >> >   p:213 at 10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1
>> ACK..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> >> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE,
>> CANCEL,
>> >> > OPTION
>> >> >   S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
>> REFER..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
>> >> > en..Max-Forwards: 70..Content-Length: 0....
>> >> > #
>> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> >> >   INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
>> SIP/2.0/UDP
>> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom
>> One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> >> >   <sip:213 at 10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> >> > <sip:pbxjker_2564 at 10.10.78.24:52732>..Allow: INVITE, ACK, BYE,
>> CANCEL,
>> >> > OPTIONS, INFO, MES
>> >> >   SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language:
>> >> > en..Supported:
>> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz
>> >> >   ation: Digest username="pbxjker_2564", realm="10.10.78.36",
>> >> > nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth,
>> >> > cnonce="KnfhF00Oi0GfJCi", nc=00000001,
>> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> >> >   response="82a75edb7ba97bfe7dea585684701d0c",
>> >> > algorithm=MD5..Max-Forwards:
>> >> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=-
>> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone
>> >> >   ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0
>> 8
>> >> > 18
>> >> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
>> >> >   telephone-event/8000..
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> >> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
>> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom
>> One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> >> > <sip:213 at 10.10.78.36;user=phone>
>> >> >   ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
>> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba
>> 2011-05-23
>> >> > 22-51-43 -0500..Content-Length: 0....
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:52783
>> >> >   INVITE sip:pbxjker_213 at 10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP
>> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards:
>> 69..From:
>> >> > "Polycom One" <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81
>> >> >   rv8Fp..To: <sip:pbxjker_213 at 10.10.78.24:52783>..Call-ID:
>> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact:
>> >> > <sip:mod_sofia at 10.10.78.36:5060>..User-Agent:
>> FreeSWITCH-mod_sofia/1.0.
>> >> >   head-git-1086cba 2011-05-23 22-51-43 -0500..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-Type: application/sdp..Content-Disposition: session..
>> >> >   Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID:
>> >> > "Polycom One"
>> >> >
>> >> > <sip:pbxjker_2564 at 10.10.78.36
>> >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
>> >> > 1318333156 1318333157 IN IP4
>> >> >    10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio
>> 29124
>> >> > RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98
>> >> > bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320
>> >> >   00..a=fmtp:18 annexb=no..a=rtpmap:101
>> telephone-event/8000..a=fmtp:101
>> >> > 0-16..a=ptime:20..
>> >> > #
>> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060
>> >> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
>> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
>> >> > <sip:pbxjker_213 at 99.65.2
>> >> >   17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID:
>> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
>> >> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_320-UA/3.2
>> >> >   .2.0477..Accept-Language: en..Content-Length: 0....
>> >> > #
>> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060
>> >> >   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
>> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two"
>> >> > <sip:pbxjker_213 at 99.65.
>> >> >   217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076
>> INVITE..Call-ID:
>> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact:
>> >> > <sip:pbxjker_213 at 10.10.78.24:52783>..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_320-UA/3.
>> >> >   2.2.0477..Allow-Events: talk,hold,conference..Accept-Language:
>> >> > en..Content-Length: 0....
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732
>> >> >   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
>> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom
>> One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> >> > <sip:213 at 10.10.78.36;user=phone
>> >> >   >;tag=8aQNK2QyUKjXa..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 INVITE..Contact:
>> >> > <sip:213 at 10.10.78.36:5060;transport=udp>..User-Agent:
>> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2
>> >> >   3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE,
>> >> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
>> >> > PUBLISH,
>> >> > SUBSCRIBE..Supported: timer, precondition, path, replaces..A
>> >> >   llow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla,
>> >> > include-session-description, presence.winfo, message-summary,
>> >> > refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker
>> >> >   _213>;party=calling;privacy=off;screen=no....
>> >> > #
>> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> >> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
>> SIP/2.0/UDP
>> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
>> >> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> >> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
>> >> > OPTIONS,
>> >> > INFO, MESSAGE, SUBSCR
>> >> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization:
>> Digest
>> >> > username="pbxjker_2564", realm="10.10.78.36",
>> >> > nonce="ab4a9421-fbb9-41fd-943d-462
>> >> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
>> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> >> > response="d47848babd8c396d8da15145503875d0",
>> >> > algorithm=MD5..Max-Forwards:
>> >> > 70..Content-Length: 0..
>> >> >   ..
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060
>> >> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
>> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> <sip:213 at 66.11
>> >> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
>> CANCEL..Content-Length:
>> >> > 0....
>> >> > #
>> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> >> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
>> SIP/2.0/UDP
>> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
>> >> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> >> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
>> >> > OPTIONS,
>> >> > INFO, MESSAGE, SUBSCR
>> >> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization:
>> Digest
>> >> > username="pbxjker_2564", realm="10.10.78.36",
>> >> > nonce="ab4a9421-fbb9-41fd-943d-462
>> >> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
>> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> >> > response="d47848babd8c396d8da15145503875d0",
>> >> > algorithm=MD5..Max-Forwards:
>> >> > 70..Content-Length: 0..
>> >> >   ..
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060
>> >> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
>> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> <sip:213 at 66.11
>> >> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
>> CANCEL..Content-Length:
>> >> > 0....
>> >> > #
>> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060
>> >> >   CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via:
>> SIP/2.0/UDP
>> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2
>> >> >   13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact:
>> >> > <sip:pbxjker_2564 at 10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL,
>> >> > OPTIONS,
>> >> > INFO, MESSAGE, SUBSCR
>> >> >   IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent:
>> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization:
>> Digest
>> >> > username="pbxjker_2564", realm="10.10.78.36",
>> >> > nonce="ab4a9421-fbb9-41fd-943d-462
>> >> >   89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002,
>> >> > uri="sip:213 at 10.10.78.36:5060;user=phone",
>> >> > response="d47848babd8c396d8da15145503875d0",
>> >> > algorithm=MD5..Max-Forwards:
>> >> > 70..Content-Length: 0..
>> >> >   ..
>> >> > #
>> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060
>> >> >   SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP
>> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One"
>> >> > <sip:pbxjker_2564 at 10.10.78.36>;tag=F5263502-30CB1633..To:
>> <sip:213 at 66.11
>> >> >   3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID:
>> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2
>> CANCEL..Content-Length:
>> >> > 0....
>> >> > #
>> >> >
>> >> >
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From fcintrono at gmail.com  Fri Oct 14 05:36:56 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Thu, 13 Oct 2011 20:36:56 -0500
Subject: [Freeswitch-users] Newbie trouble with default config extension
	9198...
In-Reply-To: <1318473186527-6887414.post@n2.nabble.com>
References: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
	<1318473186527-6887414.post@n2.nabble.com>
Message-ID: <CANtw_ZPgkwH1Cce7dGHNMFRUWiKByRTnF1qsiqQd8UN7Dibtkg@mail.gmail.com>

OK, I am going to open firewall ports for  freeswitch, thank you very much.

On Wed, Oct 12, 2011 at 9:33 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> Trying to use a client and FS on the same box is not a configuration that
> is
> recommended especially for new users. You will have much better success
> using two different IP's(multiple machines). Also make sure that you allow
> exceptions for FS and the client(soft phone) though the windows firewall.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Newbie-trouble-with-default-config-extension-9198-tp6886283p6887414.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cvogel at lyonl.com  Fri Oct 14 06:25:44 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 14 Oct 2011 02:25:44 +0000
Subject: [Freeswitch-users] From param
In-Reply-To: <6DB33DE1-E47A-4C2E-B32D-8D7289C61921@lyonl.com>
References: <6DB33DE1-E47A-4C2E-B32D-8D7289C61921@lyonl.com>
Message-ID: <8E699730-7F56-4158-AD9E-8DE02941EF96@lyonl.com>

I'm still tying to add a param on from header while using a <anti-action application="respond" data="404"/>. is there anyway to do this?


On Oct 8, 2011, at 3:08 PM, Chad Vogel wrote:

Does anyone know of a way to add a param to the from header in a dialplan? I would like to do something like a isup-oli=6  (From: <sip:restricted at 10.100.63.137;OTG=777;isup-oli=6>)

Chad

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 yungwei at resolvity.com  Fri Oct 14 08:22:59 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 14 Oct 2011 00:22:59 -0400
Subject: [Freeswitch-users] Run a js program as soon as someone registers
	with FS
Message-ID: <33095823FD21DF429B481B5163264B7951D4672C2F@VMBX102.ihostexchange.net>

Hi,

I am wondering if it's possible to run a js program as soon as someone registers with a freeswitch.

Thanks.


From SPapineni at enghouse.com  Fri Oct 14 13:08:50 2011
From: SPapineni at enghouse.com (Papineni, Suneel)
Date: Fri, 14 Oct 2011 09:08:50 +0000
Subject: [Freeswitch-users] Invalid Application Hash
Message-ID: <9438D04074E0DE45A49CD76099821272F81279@CORP-MAIL-002.edge.local>

Hi,

Before sending this request, I searched the forum for similar posts and found some solutions for recent versions of Freeswitch. Unfortunately one of my servers is using Freswitch 1.0.5. In this, I am trying to make a call using "originate sofia/external/01183035887 at 135.xxx.xxx.xxx<mailto:sofia/external/01183035887 at 135.xxx.xxx.xxx> 9999" command at console. This is giving an error "Invalid Application hash" and FS is sending CANCEL.

Is there any way to resolve this issue other than upgrading to the latest version of Freeswitch (I am going to upgrade in coming weeks, but I need to test a specific scenario now on existing system).

Thanks & Regards
Suneel


Here is the log for reference...

2011-10-13 16:59:34.341952 [DEBUG] sofia.c:4158 Channel sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> entering state [ready][200]
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3579 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2354 Set Codec sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> PCMU/8000 20 ms 160 samples
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3512 Set 2833 dtmf recv payload to 0
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/external/01183035887 at 135.196.132.200] 192.168.20.240 port 25744 -> 135.196.145.145 port 54206 codec: 0 ms: 20
2011-10-13 16:59:34.341952 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms
2011-10-13 16:59:34.341952 [NOTICE] sofia.c:4738 Channel [sofia/external/01183035887 at 135.196.132.200] has been answered
2011-10-13 16:59:34.341952 [DEBUG] switch_ivr_originate.c:3148 Originate Resulted in Success: [sofia/external/01183035887 at 135.196.132.200]
2011-10-13 16:59:34.341952 [DEBUG] switch_ivr.c:1441 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State Change CS_CONSUME_MEDIA -> CS_ROUTING
2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:638 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.341952 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> to XML[9999 at default]
2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:314 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Running State Change CS_ROUTING
2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:341 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State ROUTING
2011-10-13 16:59:34.341952 [DEBUG] mod_sofia.c:141 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> SOFIA ROUTING
2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:77 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Standard ROUTING
2011-10-13 16:59:34.341952 [INFO] mod_dialplan_xml.c:418 Processing ->9999 in context default
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unloop] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->tod_example] continue=true
Dialplan: day of week[5] =~ 2-6 (PASS)
Dialplan: hour[16] =~ 9-18 (PASS)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action set(open=true)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->holiday_example] continue=true
Dialplan: month[10] =~ 1 (FAIL)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->global-intercept] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [global-intercept] destination_number(9999) =~ /^886$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group-intercept] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group-intercept] destination_number(9999) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->intercept-ext] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [intercept-ext] destination_number(9999) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->redial] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [redial] destination_number(9999) =~ /^870$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->global] continue=true
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Absolute Condition [global]
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->snom-demo-2] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [snom-demo-2] destination_number(9999) =~ /^9001$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->snom-demo-1] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [snom-demo-1] destination_number(9999) =~ /^9000$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [eavesdrop] destination_number(9999) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [eavesdrop] destination_number(9999) =~ /^779$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->call_return] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [call_return] destination_number(9999) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->del-group] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [del-group] destination_number(9999) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->add-group] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [add-group] destination_number(9999) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->call-group-simo] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [call-group-simo] destination_number(9999) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->call-group-order] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [call-group-order] destination_number(9999) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->extension-intercom] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [extension-intercom] destination_number(9999) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->Local_Extension] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [Local_Extension] destination_number(9999) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group_dial_sales] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group_dial_sales] destination_number(9999) =~ /^2000$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group_dial_support] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group_dial_support] destination_number(9999) =~ /^2001$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group_dial_billing] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group_dial_billing] destination_number(9999) =~ /^2002$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->operator] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [operator] destination_number(9999) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->vmain] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [vmain] destination_number(9999) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->sip_uri] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [sip_uri] destination_number(9999) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->nb_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [nb_conferences] destination_number(9999) =~ /^(30\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->wb_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [wb_conferences] destination_number(9999) =~ /^(31\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->uwb_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [uwb_conferences] destination_number(9999) =~ /^(32\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->cdquality_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [cdquality_conferences] destination_number(9999) =~ /^(33\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(9999) =~ /^9(888|1616|3232)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->mad_boss_intercom] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [mad_boss_intercom] destination_number(9999) =~ /^0911$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->mad_boss_intercom] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [mad_boss_intercom] destination_number(9999) =~ /^0912$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->mad_boss] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [mad_boss] destination_number(9999) =~ /^0913$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ivr_demo] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ivr_demo] destination_number(9999) =~ /^5000$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->dynamic_conference] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [dynamic_conference] destination_number(9999) =~ /^5001$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->rtp_multicast_page] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [rtp_multicast_page] destination_number(9999) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [park] destination_number(9999) =~ /^5900$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unpark] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unpark] destination_number(9999) =~ /^5901$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->valet_park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [valet_park] destination_number(9999) =~ /^(6000)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->valet_park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [valet_park] destination_number(9999) =~ /^(60\d[1-9])$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [park] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unpark] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unpark] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [park] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unpark] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unpark] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->wait] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [wait] destination_number(9999) =~ /^wait$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->fax_receive] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [fax_receive] destination_number(9999) =~ /^9978$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->fax_transmit] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [fax_transmit] destination_number(9999) =~ /^9979$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_180] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_180] destination_number(9999) =~ /^9980$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_183_uk_ring] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_183_uk_ring] destination_number(9999) =~ /^9981$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_183_music_ring] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_183_music_ring] destination_number(9999) =~ /^9982$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(9999) =~ /^9983$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_post_answer_music] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_post_answer_music] destination_number(9999) =~ /^9984$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ClueCon] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ClueCon] destination_number(9999) =~ /^9991$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->show_info] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [show_info] destination_number(9999) =~ /^9992$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->video_record] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [video_record] destination_number(9999) =~ /^9993$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->video_playback] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [video_playback] destination_number(9999) =~ /^9994$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->delay_echo] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [delay_echo] destination_number(9999) =~ /^9995$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->echo] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [echo] destination_number(9999) =~ /^9996$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->milliwatt] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [milliwatt] destination_number(9999) =~ /^9997$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->tone_stream] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [tone_stream] destination_number(9999) =~ /^9998$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->zrtp_enrollement] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [zrtp_enrollement] destination_number(9999) =~ /^9787$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->hold_music] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (PASS) [hold_music] destination_number(9999) =~ /^9999$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [hold_music] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> ANTI-Action set(zrtp_secure_media=true)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> ANTI-Action answer()
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> ANTI-Action playback(silence_stream://2000)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> ANTI-Action execute_extension(is_zrtp_secure XML features)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> ANTI-Action playback(local_stream://moh)
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:119 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State Change CS_ROUTING -> CS_EXECUTE
2011-10-13 16:59:34.357577 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:341 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State ROUTING going to sleep
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:314 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Running State Change CS_EXECUTE
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:348 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State EXECUTE
2011-10-13 16:59:34.357577 [DEBUG] mod_sofia.c:227 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> SOFIA EXECUTE
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:157 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Standard EXECUTE
EXECUTE sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> set(open=true)
2011-10-13 16:59:34.357577 [DEBUG] mod_dptools.c:814 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> SET [open]=[true]
2011-10-13 16:59:34.357577 [ERR] switch_core_session.c:1727 Invalid Application hash
2011-10-13 16:59:34.357577 [NOTICE] switch_core_session.c:1728 Hangup sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2011-10-13 16:59:34.357577 [DEBUG] switch_channel.c:2071 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [KILL]
2011-10-13 16:59:34.357577 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:348 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State EXECUTE going to sleep
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:314 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Running State Change CS_HANGUP
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:499 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State HANGUP
2011-10-13 16:59:34.357577 [DEBUG] mod_sofia.c:412 Channel sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> hanging up, cause: DESTINATION_OUT_OF_ORDER
2011-10-13 16:59:34.357577 [DEBUG] mod_sofia.c:455 Sending BYE to sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:46 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:499 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State HANGUP going to sleep
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:333 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State Change CS_HANGUP -> CS_REPORTING
2011-10-13 16:59:34.357577 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:314 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Running State Change CS_REPORTING
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:590 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State REPORTING
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:53 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:590 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State REPORTING going to sleep
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:327 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State Change CS_REPORTING -> CS_DESTROY
2011-10-13 16:59:34.357577 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.357577 [DEBUG] switch_core_session.c:1161 Session 20 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Locked, Waiting on external entities
2011-10-13 16:59:34.357577 [NOTICE] switch_core_session.c:1179 Session 20 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Ended
2011-10-13 16:59:34.357577 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [CS_DESTROY]
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:428 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Running State Change CS_DESTROY
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:439 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State DESTROY
2011-10-13 16:59:34.357577 [DEBUG] mod_sofia.c:339 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> SOFIA DESTROY
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:60 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Standard DESTROY
2011-10-13 16:59:34.357577 [DEBUG] switch_core_state_machine.c:439 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State DESTROY going to sleep
2011-10-13 16:59:34.466952 [DEBUG] switch_core_session.c:856 Send signal sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [BREAK]
2011-10-13 16:59:34.482577 [DEBUG] switch_ivr.c:551 sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> Command Execute redirect(sip:01183035878 at 135.196.132.200)
EXECUTE sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> redirect(sip:01183035878 at 135.196.132.200)
2011-10-13 16:59:34.482577 [DEBUG] mod_sofia.c:1660 Redirecting to "unknown" <sip:01183035878 at 135.196.132.200>
2011-10-13 16:59:34.482577 [DEBUG] switch_core_session.c:638 Send signal sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [BREAK]
2011-10-13 16:59:34.482577 [DEBUG] sofia.c:4158 Channel sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> entering state [terminated][302]
2011-10-13 16:59:34.482577 [NOTICE] switch_core_state_machine.c:185 sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> has executed the last dialplan instruction, hanging up.
2011-10-13 16:59:34.482577 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [CS_EXECUTE] [NORMAL_CLEARING]
2011-10-13 16:59:34.482577 [DEBUG] switch_channel.c:2071 Send signal sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [KILL]
2011-10-13 16:59:34.482577 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [BREAK]
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:348 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State EXECUTE going to sleep
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) Running State Change CS_HANGUP
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State HANGUP
2011-10-13 16:59:34.482577 [DEBUG] mod_sofia.c:406 sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> Overriding SIP cause 480 with 302 from the other leg
2011-10-13 16:59:34.482577 [DEBUG] mod_sofia.c:412 Channel sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> hanging up, cause: NORMAL_CLEARING
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:46 sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> Standard HANGUP, cause: NORMAL_CLEARING
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State HANGUP going to sleep
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:333 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State Change CS_HANGUP -> CS_REPORTING
2011-10-13 16:59:34.482577 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [BREAK]
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) Running State Change CS_REPORTING
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State REPORTING
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:53 sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> Standard REPORTING, cause: NORMAL_CLEARING
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State REPORTING going to sleep
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:327 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State Change CS_REPORTING -> CS_DESTROY
2011-10-13 16:59:34.482577 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [BREAK]
2011-10-13 16:59:34.482577 [DEBUG] switch_core_session.c:1161 Session 23 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) Locked, Waiting on external entities
2011-10-13 16:59:34.482577 [NOTICE] switch_core_session.c:1179 Session 23 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) Ended
2011-10-13 16:59:34.482577 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> [CS_DESTROY]
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:428 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) Running State Change CS_DESTROY
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:439 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State DESTROY
2011-10-13 16:59:34.482577 [DEBUG] mod_sofia.c:339 sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> SOFIA DESTROY
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:60 sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060> Standard DESTROY
2011-10-13 16:59:34.482577 [DEBUG] switch_core_state_machine.c:439 (sofia/external/1183035901 at 135.196.132.200:5060<mailto:sofia/external/1183035901 at 135.196.132.200:5060>) State DESTROY going to sleep
2011-10-13 16:59:34.498202 [NOTICE] switch_channel.c:669 New Channel sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [60e6e03d-75cc-4081-8073-ef8288d472d0]
2011-10-13 16:59:34.498202 [DEBUG] mod_sofia.c:3457 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State Change CS_NEW -> CS_INIT
2011-10-13 16:59:34.498202 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.529452 [DEBUG] switch_core_state_machine.c:314 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Running State Change CS_INIT
2011-10-13 16:59:34.529452 [DEBUG] switch_core_state_machine.c:338 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State INIT
2011-10-13 16:59:34.529452 [DEBUG] mod_sofia.c:84 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> SOFIA INIT
2011-10-13 16:59:34.529452 [DEBUG] mod_sofia.c:118 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State Change CS_INIT -> CS_ROUTING
2011-10-13 16:59:34.529452 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.529452 [DEBUG] switch_core_state_machine.c:338 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State INIT going to sleep
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From steveayre at gmail.com  Fri Oct 14 14:09:06 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Oct 2011 11:09:06 +0100
Subject: [Freeswitch-users] Invalid Application Hash
In-Reply-To: <9438D04074E0DE45A49CD76099821272F81279@CORP-MAIL-002.edge.local>
References: <9438D04074E0DE45A49CD76099821272F81279@CORP-MAIL-002.edge.local>
Message-ID: <C21CEA3D-073A-4B86-ACA7-B115127B7D32@gmail.com>

You need to load mod_hash in modules.conf.xml

Steve on iPhone

On 14 Oct 2011, at 10:08, "Papineni, Suneel" <SPapineni at enghouse.com> wrote:

> Hi,
>  
> Before sending this request, I searched the forum for similar posts and found some solutions for recent versions of Freeswitch. Unfortunately one of my servers is using Freswitch 1.0.5. In this, I am trying to make a call using ?originate sofia/external/01183035887 at 135.xxx.xxx.xxx 9999? command at console. This is giving an error ?Invalid Application hash? and FS is sending CANCEL.
>  
> Is there any way to resolve this issue other than upgrading to the latest version of Freeswitch (I am going to upgrade in coming weeks, but I need to test a specific scenario now on existing system).
>  
> Thanks & Regards
> Suneel
>  
>  
> Here is the log for reference?
>  
> 2011-10-13 16:59:34.341952 [DEBUG] sofia.c:4158 Channel sofia/external/01183035887 at 135.196.132.200 entering state [ready][200]
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3579 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2354 Set Codec sofia/external/01183035887 at 135.196.132.200 PCMU/8000 20 ms 160 samples
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3512 Set 2833 dtmf recv payload to 0
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/external/01183035887 at 135.196.132.200] 192.168.20.240 port 25744 -> 135.196.145.145 port 54206 codec: 0 ms: 20
> 2011-10-13 16:59:34.341952 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms
> 2011-10-13 16:59:34.341952 [NOTICE] sofia.c:4738 Channel [sofia/external/01183035887 at 135.196.132.200] has been answered
> 2011-10-13 16:59:34.341952 [DEBUG] switch_ivr_originate.c:3148 Originate Resulted in Success: [sofia/external/01183035887 at 135.196.132.200]
> 2011-10-13 16:59:34.341952 [DEBUG] switch_ivr.c:1441 (sofia/external/01183035887 at 135.196.132.200) State Change CS_CONSUME_MEDIA -> CS_ROUTING
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200 [BREAK]
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:638 Send signal sofia/external/01183035887 at 135.196.132.200 [BREAK]
> 2011-10-13 16:59:34.341952 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/01183035887 at 135.196.132.200 to XML[9999 at default]
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:314 (sofia/external/01183035887 at 135.196.132.200) Running State Change CS_ROUTING
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:341 (sofia/external/01183035887 at 135.196.132.200) State ROUTING
> 2011-10-13 16:59:34.341952 [DEBUG] mod_sofia.c:141 sofia/external/01183035887 at 135.196.132.200 SOFIA ROUTING
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:77 sofia/external/01183035887 at 135.196.132.200 Standard ROUTING
> 2011-10-13 16:59:34.341952 [INFO] mod_dialplan_xml.c:418 Processing ->9999 in context default
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->unloop] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->tod_example] continue=true
> Dialplan: day of week[5] =~ 2-6 (PASS)
> Dialplan: hour[16] =~ 9-18 (PASS)
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Date/Time Match (PASS) [tod_example] break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action set(open=true)
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->holiday_example] continue=true
> Dialplan: month[10] =~ 1 (FAIL)
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Date/Time Match (FAIL) [holiday_example] break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->global-intercept] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [global-intercept] destination_number(9999) =~ /^886$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->group-intercept] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [group-intercept] destination_number(9999) =~ /^\*8$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->intercept-ext] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [intercept-ext] destination_number(9999) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->redial] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [redial] destination_number(9999) =~ /^870$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->global] continue=true
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Absolute Condition [global]
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action hash(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->snom-demo-2] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [snom-demo-2] destination_number(9999) =~ /^9001$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->snom-demo-1] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [snom-demo-1] destination_number(9999) =~ /^9000$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->eavesdrop] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [eavesdrop] destination_number(9999) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->eavesdrop] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [eavesdrop] destination_number(9999) =~ /^779$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->call_return] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [call_return] destination_number(9999) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->del-group] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [del-group] destination_number(9999) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->add-group] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [add-group] destination_number(9999) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->call-group-simo] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [call-group-simo] destination_number(9999) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->call-group-order] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [call-group-order] destination_number(9999) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->extension-intercom] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [extension-intercom] destination_number(9999) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->Local_Extension] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [Local_Extension] destination_number(9999) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->group_dial_sales] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [group_dial_sales] destination_number(9999) =~ /^2000$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->group_dial_support] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [group_dial_support] destination_number(9999) =~ /^2001$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->group_dial_billing] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [group_dial_billing] destination_number(9999) =~ /^2002$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->operator] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [operator] destination_number(9999) =~ /^(operator|0)$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->vmain] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [vmain] destination_number(9999) =~ /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->sip_uri] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [sip_uri] destination_number(9999) =~ /^sip:(.*)$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->nb_conferences] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [nb_conferences] destination_number(9999) =~ /^(30\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->wb_conferences] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [wb_conferences] destination_number(9999) =~ /^(31\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->uwb_conferences] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [uwb_conferences] destination_number(9999) =~ /^(32\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->cdquality_conferences] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [cdquality_conferences] destination_number(9999) =~ /^(33\d{2})$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->freeswitch_public_conf_via_sip] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(9999) =~ /^9(888|1616|3232)$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->mad_boss_intercom] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [mad_boss_intercom] destination_number(9999) =~ /^0911$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->mad_boss_intercom] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [mad_boss_intercom] destination_number(9999) =~ /^0912$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->mad_boss] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [mad_boss] destination_number(9999) =~ /^0913$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->ivr_demo] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [ivr_demo] destination_number(9999) =~ /^5000$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->dynamic_conference] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [dynamic_conference] destination_number(9999) =~ /^5001$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->rtp_multicast_page] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [rtp_multicast_page] destination_number(9999) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park] destination_number(9999) =~ /^5900$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->unpark] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark] destination_number(9999) =~ /^5901$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->valet_park] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [valet_park] destination_number(9999) =~ /^(6000)$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->valet_park] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [valet_park] destination_number(9999) =~ /^(60\d[1-9])$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->unpark] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->unpark] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->wait] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [wait] destination_number(9999) =~ /^wait$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->fax_receive] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [fax_receive] destination_number(9999) =~ /^9978$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->fax_transmit] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [fax_transmit] destination_number(9999) =~ /^9979$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->ringback_180] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [ringback_180] destination_number(9999) =~ /^9980$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->ringback_183_uk_ring] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [ringback_183_uk_ring] destination_number(9999) =~ /^9981$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->ringback_183_music_ring] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [ringback_183_music_ring] destination_number(9999) =~ /^9982$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->ringback_post_answer_uk_ring] continue=false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(9999) =~ /^9983$/ break=on-false
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->ringback_post_answer_music] continue=false
> Dialplan:
> 
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> FreeSWITCH-users at lists.freeswitch.org
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From SPapineni at enghouse.com  Fri Oct 14 16:40:05 2011
From: SPapineni at enghouse.com (Papineni, Suneel)
Date: Fri, 14 Oct 2011 12:40:05 +0000
Subject: [Freeswitch-users] Invalid Application Hash
In-Reply-To: <C21CEA3D-073A-4B86-ACA7-B115127B7D32@gmail.com>
References: <9438D04074E0DE45A49CD76099821272F81279@CORP-MAIL-002.edge.local>
	<C21CEA3D-073A-4B86-ACA7-B115127B7D32@gmail.com>
Message-ID: <9438D04074E0DE45A49CD76099821272F81324@CORP-MAIL-002.edge.local>

Hi Steve,

I think mod_hash is not present in Freeswitch 1.0.5 ( Ichecked in default modules.conf file). Seems it has mod_limit (which is removed in later releases).

Thanks & Regards
Suneel


From: Steven Ayre [mailto:steveayre at gmail.com]
Sent: 14 October 2011 11:09
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid Application Hash

You need to load mod_hash in modules.conf.xml

Steve on iPhone

On 14 Oct 2011, at 10:08, "Papineni, Suneel" <SPapineni at enghouse.com<mailto:SPapineni at enghouse.com>> wrote:
Hi,

Before sending this request, I searched the forum for similar posts and found some solutions for recent versions of Freeswitch. Unfortunately one of my servers is using Freswitch 1.0.5. In this, I am trying to make a call using ?originate sofia/external/01183035887 at 135.xxx.xxx.xxx<mailto:sofia/external/01183035887 at 135.xxx.xxx.xxx> 9999? command at console. This is giving an error ?Invalid Application hash? and FS is sending CANCEL.

Is there any way to resolve this issue other than upgrading to the latest version of Freeswitch (I am going to upgrade in coming weeks, but I need to test a specific scenario now on existing system).

Thanks & Regards
Suneel


Here is the log for reference?

2011-10-13 16:59:34.341952 [DEBUG] sofia.c:4158 Channel sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> entering state [ready][200]
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3579 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2354 Set Codec sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> PCMU/8000 20 ms 160 samples
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3512 Set 2833 dtmf recv payload to 0
2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>] 192.168.20.240 port 25744 -> 135.196.145.145 port 54206 codec: 0 ms: 20
2011-10-13 16:59:34.341952 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms
2011-10-13 16:59:34.341952 [NOTICE] sofia.c:4738 Channel [sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>] has been answered
2011-10-13 16:59:34.341952 [DEBUG] switch_ivr_originate.c:3148 Originate Resulted in Success: [sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>]
2011-10-13 16:59:34.341952 [DEBUG] switch_ivr.c:1441 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State Change CS_CONSUME_MEDIA -> CS_ROUTING
2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:638 Send signal sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> [BREAK]
2011-10-13 16:59:34.341952 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> to XML[9999 at default]
2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:314 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) Running State Change CS_ROUTING
2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:341 (sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200>) State ROUTING
2011-10-13 16:59:34.341952 [DEBUG] mod_sofia.c:141 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> SOFIA ROUTING
2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:77 sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Standard ROUTING
2011-10-13 16:59:34.341952 [INFO] mod_dialplan_xml.c:418 Processing ->9999 in context default
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unloop] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->tod_example] continue=true
Dialplan: day of week[5] =~ 2-6 (PASS)
Dialplan: hour[16] =~ 9-18 (PASS)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action set(open=true)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->holiday_example] continue=true
Dialplan: month[10] =~ 1 (FAIL)
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->global-intercept] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [global-intercept] destination_number(9999) =~ /^886$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group-intercept] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group-intercept] destination_number(9999) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->intercept-ext] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [intercept-ext] destination_number(9999) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->redial] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [redial] destination_number(9999) =~ /^870$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->global] continue=true
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Absolute Condition [global]
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Action hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->snom-demo-2] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [snom-demo-2] destination_number(9999) =~ /^9001$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->snom-demo-1] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [snom-demo-1] destination_number(9999) =~ /^9000$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [eavesdrop] destination_number(9999) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [eavesdrop] destination_number(9999) =~ /^779$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->call_return] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [call_return] destination_number(9999) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->del-group] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [del-group] destination_number(9999) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->add-group] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [add-group] destination_number(9999) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->call-group-simo] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [call-group-simo] destination_number(9999) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->call-group-order] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [call-group-order] destination_number(9999) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->extension-intercom] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [extension-intercom] destination_number(9999) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->Local_Extension] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [Local_Extension] destination_number(9999) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group_dial_sales] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group_dial_sales] destination_number(9999) =~ /^2000$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group_dial_support] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group_dial_support] destination_number(9999) =~ /^2001$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->group_dial_billing] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [group_dial_billing] destination_number(9999) =~ /^2002$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->operator] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [operator] destination_number(9999) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->vmain] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [vmain] destination_number(9999) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->sip_uri] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [sip_uri] destination_number(9999) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->nb_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [nb_conferences] destination_number(9999) =~ /^(30\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->wb_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [wb_conferences] destination_number(9999) =~ /^(31\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->uwb_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [uwb_conferences] destination_number(9999) =~ /^(32\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->cdquality_conferences] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [cdquality_conferences] destination_number(9999) =~ /^(33\d{2})$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(9999) =~ /^9(888|1616|3232)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->mad_boss_intercom] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [mad_boss_intercom] destination_number(9999) =~ /^0911$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->mad_boss_intercom] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [mad_boss_intercom] destination_number(9999) =~ /^0912$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->mad_boss] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [mad_boss] destination_number(9999) =~ /^0913$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ivr_demo] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ivr_demo] destination_number(9999) =~ /^5000$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->dynamic_conference] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [dynamic_conference] destination_number(9999) =~ /^5001$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->rtp_multicast_page] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [rtp_multicast_page] destination_number(9999) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [park] destination_number(9999) =~ /^5900$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unpark] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unpark] destination_number(9999) =~ /^5901$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->valet_park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [valet_park] destination_number(9999) =~ /^(6000)$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->valet_park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [valet_park] destination_number(9999) =~ /^(60\d[1-9])$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [park] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unpark] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unpark] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->park] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [park] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->unpark] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [unpark] source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->wait] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [wait] destination_number(9999) =~ /^wait$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->fax_receive] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [fax_receive] destination_number(9999) =~ /^9978$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->fax_transmit] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [fax_transmit] destination_number(9999) =~ /^9979$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_180] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_180] destination_number(9999) =~ /^9980$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_183_uk_ring] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_183_uk_ring] destination_number(9999) =~ /^9981$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_183_music_ring] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_183_music_ring] destination_number(9999) =~ /^9982$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(9999) =~ /^9983$/ break=on-false
Dialplan: sofia/external/01183035887 at 135.196.132.200<mailto:sofia/external/01183035887 at 135.196.132.200> parsing [default->ringback_post_answer_music] continue=false
Dialplan:

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 masood24 at gmail.com  Fri Oct 14 17:07:48 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Fri, 14 Oct 2011 16:07:48 +0300
Subject: [Freeswitch-users] Digest Authentication
Message-ID: <CAMd3iJLURDPYn+Oxy_BSsrX1T1ZRSUTT2afk5uFHquirw8NGJQ@mail.gmail.com>

Dear ,

I had successfully installed and configured radiusclient. I do not know that
what next lines should i do to get users registered on my freeswitch using
digest authentication with my radius server,
Like I have my radsius server and billing system. Now what lines should i
add in sofit conf or in dialplans that every time client hits server request
should be sent to radius server and what lines should i add to register them
if i get proper return code and how to do that.

Your Help is needed

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From jeff at jefflenk.com  Fri Oct 14 18:17:00 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 14 Oct 2011 07:17:00 -0700 (PDT)
Subject: [Freeswitch-users] Run a js program as soon as someone
	registers with FS
In-Reply-To: <33095823FD21DF429B481B5163264B7951D4672C2F@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7951D4672C2F@VMBX102.ihostexchange.net>
Message-ID: <1318601820097-6892603.post@n2.nabble.com>

I believe someone gave you an answer the other day in IRC that had something
related to monitoring for the "sofia::register" event. There are many ways
to do this with the simplest way being to create an esl listener and watch
for this event. This of course means that you will detect the condition in
the external esl listener and can do whatever you want then. You also need
to perhaps rethink why you want to do this in the first place(nobody knows
what your intention is here).

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Run-a-js-program-as-soon-as-someone-registers-with-FS-tp6891277p6892603.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From admharris at gmx.com  Fri Oct 14 18:41:58 2011
From: admharris at gmx.com (adam harris)
Date: Fri, 14 Oct 2011 16:41:58 +0200
Subject: [Freeswitch-users] esl functions in c
Message-ID: <20111014144158.199160@gmx.com>

Where can I find out what esl functions are availible in C


From jos at firstcom.dk  Fri Oct 14 18:06:53 2011
From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=)
Date: Fri, 14 Oct 2011 16:06:53 +0200
Subject: [Freeswitch-users] Freeswitch equilavent to Asterisk
	BackgroundDetect
Message-ID: <111509FCE9EBF74BAB2CB38AAC521D9B1313058C49@helle.dk.hq.firstcom.dk>

Hello List.

Ive been trying to find a function equal to what BackgroundDetect does in Asterisk.
Anybody got an idea on how I could implement the functionality in Freeswitch?

I use it in IVR services, to detect when a caller interrupts the currently playing prompt.



Med venlig hilsen/Kind Regards

Jon Leren Sch?pzinsky
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From henrikaagaardsorensen at gmail.com  Fri Oct 14 18:19:45 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Fri, 14 Oct 2011 16:19:45 +0200
Subject: [Freeswitch-users] Kamailio Dispatcher fails in front of FreeSwitch,
	help needed.
In-Reply-To: <CAGH8SeayFuOu5xoOtzhkQknQ31QamRVUAZUdd1Gz+Vtoju_Jcg@mail.gmail.com>
References: <CAGH8SeayFuOu5xoOtzhkQknQ31QamRVUAZUdd1Gz+Vtoju_Jcg@mail.gmail.com>
Message-ID: <CAGH8SeYHcxnZr0kUU0snnXe_pMx4=zCy1oktk7nDw_q+igHazg@mail.gmail.com>

When connecting 2 devices directly to FreeSwitch everything works. When
connecting to Kamailio (as dispatcher), some calls fails with "Timeout" and
some fails with "Too many loops".

This is my kamailio.cfg: http://pastebin.com/KEuZmikh

When connected directly to FreeSwitch where everything works, this is how my
clients are registrered in FreeSwitch: http://pastebin.com/prkjkb7g

When connected to Kamailio where nothing works, this is how my clients are
registrered in FreeSwitch: http://pastebin.com/b5x6NZyv

215.197.145.132 (sip.my-domain.com) is my Kamailio.
pbx.my-domain.com is my FreeSwitch.
96.224.14.164 is my IP at home, where my 2 devices are from.

Can anyone help me out? I'm kinda lost at this point. And I don't know if
the error is on FreeSwitch or Kamailio (or both).
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From rhow at exemail.com.au  Fri Oct 14 10:26:25 2011
From: rhow at exemail.com.au (Ryan How)
Date: Fri, 14 Oct 2011 14:26:25 +0800
Subject: [Freeswitch-users] Change sound someone hears when they ring me
Message-ID: <4E97D611.50204@exemail.com.au>

Hi,

I've set a freeswitch box with bluebox (with a bit of hassle! but it got 
there). I can't find where to start looking regarding changing the sound 
an external person hears when they call me. usually from the standard 
landline when dialing a number I get a ring-ring ring-ring, but when 
someone dials my number (which is a voip DID) it has a long 
riiiiiiiiinnnnngggggg . . . . . riiiiiiiiinngggggg. Is there any way I 
can change that? My asterisk box and other voip phones haven't done it, 
so I figure it must be able to be changed.

Thanks, Ryan




From cmrienzo at gmail.com  Fri Oct 14 19:26:30 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 14 Oct 2011 11:26:30 -0400
Subject: [Freeswitch-users] esl functions in c
In-Reply-To: <20111014144158.199160@gmx.com>
References: <20111014144158.199160@gmx.com>
Message-ID: <CAD-qRkgZ7s36rL-6H6ihsMUA4EV02UBX5zUk4j52Wf=1Mp8J=Q@mail.gmail.com>

C headers are in libs/esl/src/include

Example code is in libs/esl/testclient.c and libs/esl/testserver.c


On Fri, Oct 14, 2011 at 10:41 AM, adam harris <admharris at gmx.com> wrote:

> Where can I find out what esl functions are availible in C
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Oct 14 19:30:43 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Oct 2011 16:30:43 +0100
Subject: [Freeswitch-users] Invalid Application Hash
In-Reply-To: <9438D04074E0DE45A49CD76099821272F81324@CORP-MAIL-002.edge.local>
References: <9438D04074E0DE45A49CD76099821272F81279@CORP-MAIL-002.edge.local>
	<C21CEA3D-073A-4B86-ACA7-B115127B7D32@gmail.com>
	<9438D04074E0DE45A49CD76099821272F81324@CORP-MAIL-002.edge.local>
Message-ID: <CAFiqYuncYY6=DwP91F9ys32UosmM8Fbbfv4Sqo-yStM_e36prg@mail.gmail.com>

Try loading it anyway. If it doesn't load try loading mod_limit. One
of the two will give you the app.

Usage of the app didn't change, just the module providing it so there
should be no difference between switches using the old and new module.

mod_limit will be in newer releases, but as a dummy module that gives
a Warning level message when you load it. If you see that message
you're on a version that should have mod_hash.

-Steve


On 14 October 2011 13:40, Papineni, Suneel <SPapineni at enghouse.com> wrote:
> Hi Steve,
>
>
>
> I think mod_hash is not present in Freeswitch 1.0.5 ( Ichecked in default
> modules.conf file). Seems it has mod_limit (which is removed in later
> releases).
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> From: Steven Ayre [mailto:steveayre at gmail.com]
> Sent: 14 October 2011 11:09
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Invalid Application Hash
>
>
>
> You need to load mod_hash in modules.conf.xml
>
> Steve on iPhone
>
> On 14 Oct 2011, at 10:08, "Papineni, Suneel" <SPapineni at enghouse.com> wrote:
>
> Hi,
>
>
>
> Before sending this request, I searched the forum for similar posts and
> found some solutions for recent versions of Freeswitch. Unfortunately one of
> my servers is using Freswitch 1.0.5. In this, I am trying to make a call
> using ?originate sofia/external/01183035887 at 135.xxx.xxx.xxx 9999? command at
> console. This is giving an error ?Invalid Application hash? and FS is
> sending CANCEL.
>
>
>
> Is there any way to resolve this issue other than upgrading to the latest
> version of Freeswitch (I am going to upgrade in coming weeks, but I need to
> test a specific scenario now on existing system).
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> Here is the log for reference?
>
>
>
> 2011-10-13 16:59:34.341952 [DEBUG] sofia.c:4158 Channel
> sofia/external/01183035887 at 135.196.132.200 entering state [ready][200]
>
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3579 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
>
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/external/01183035887 at 135.196.132.200 PCMU/8000 20 ms 160 samples
>
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3512 Set 2833 dtmf recv
> payload to 0
>
> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2594 AUDIO RTP
> [sofia/external/01183035887 at 135.196.132.200] 192.168.20.240 port 25744 ->
> 135.196.145.145 port 54206 codec: 0 ms: 20
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_rtp.c:1182 Starting timer [soft]
> 160 bytes per 20ms
>
> 2011-10-13 16:59:34.341952 [NOTICE] sofia.c:4738 Channel
> [sofia/external/01183035887 at 135.196.132.200] has been answered
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_ivr_originate.c:3148 Originate
> Resulted in Success: [sofia/external/01183035887 at 135.196.132.200]
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_ivr.c:1441
> (sofia/external/01183035887 at 135.196.132.200) State Change CS_CONSUME_MEDIA
> -> CS_ROUTING
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/01183035887 at 135.196.132.200 [BREAK]
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/01183035887 at 135.196.132.200 [BREAK]
>
> 2011-10-13 16:59:34.341952 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/external/01183035887 at 135.196.132.200 to XML[9999 at default]
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/01183035887 at 135.196.132.200) Running State Change CS_ROUTING
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/01183035887 at 135.196.132.200) State ROUTING
>
> 2011-10-13 16:59:34.341952 [DEBUG] mod_sofia.c:141
> sofia/external/01183035887 at 135.196.132.200 SOFIA ROUTING
>
> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:77
> sofia/external/01183035887 at 135.196.132.200 Standard ROUTING
>
> 2011-10-13 16:59:34.341952 [INFO] mod_dialplan_xml.c:418 Processing ->9999
> in context default
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->unloop] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->tod_example] continue=true
>
> Dialplan: day of week[5] =~ 2-6 (PASS)
>
> Dialplan: hour[16] =~ 9-18 (PASS)
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Date/Time Match (PASS)
> [tod_example] break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action set(open=true)
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->holiday_example] continue=true
>
> Dialplan: month[10] =~ 1 (FAIL)
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Date/Time Match (FAIL)
> [holiday_example] break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->global-intercept] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [global-intercept] destination_number(9999) =~ /^886$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->group-intercept] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [group-intercept] destination_number(9999) =~ /^\*8$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->intercept-ext] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [intercept-ext] destination_number(9999) =~ /^\*\*(\d+)$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->redial] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [redial]
> destination_number(9999) =~ /^870$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->global] continue=true
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [global]
> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Absolute Condition
> [global]
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->snom-demo-2] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [snom-demo-2] destination_number(9999) =~ /^9001$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->snom-demo-1] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [snom-demo-1] destination_number(9999) =~ /^9000$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->eavesdrop] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [eavesdrop] destination_number(9999) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->eavesdrop] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [eavesdrop] destination_number(9999) =~ /^779$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->call_return] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [call_return] destination_number(9999) =~ /^\*69$|^869$|^lcr$/
> break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->del-group] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [del-group] destination_number(9999) =~ /^80(\d{2})$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->add-group] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [add-group] destination_number(9999) =~ /^81(\d{2})$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->call-group-simo] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [call-group-simo] destination_number(9999) =~ /^82(\d{2})$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->call-group-order] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [call-group-order] destination_number(9999) =~ /^83(\d{2})$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->extension-intercom] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [extension-intercom] destination_number(9999) =~ /^8(10[01][0-9])$/
> break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->Local_Extension] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [Local_Extension] destination_number(9999) =~ /^(10[01][0-9])$/
> break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->group_dial_sales] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [group_dial_sales] destination_number(9999) =~ /^2000$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->group_dial_support] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [group_dial_support] destination_number(9999) =~ /^2001$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->group_dial_billing] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [group_dial_billing] destination_number(9999) =~ /^2002$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->operator] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [operator]
> destination_number(9999) =~ /^(operator|0)$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->vmain] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [vmain]
> destination_number(9999) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->sip_uri] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [sip_uri]
> destination_number(9999) =~ /^sip:(.*)$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->nb_conferences] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [nb_conferences] destination_number(9999) =~ /^(30\d{2})$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->wb_conferences] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [wb_conferences] destination_number(9999) =~ /^(31\d{2})$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->uwb_conferences] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [uwb_conferences] destination_number(9999) =~ /^(32\d{2})$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->cdquality_conferences] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [cdquality_conferences] destination_number(9999) =~ /^(33\d{2})$/
> break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(9999) =~
> /^9(888|1616|3232)$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->mad_boss_intercom] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [mad_boss_intercom] destination_number(9999) =~ /^0911$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->mad_boss_intercom] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [mad_boss_intercom] destination_number(9999) =~ /^0912$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->mad_boss] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [mad_boss]
> destination_number(9999) =~ /^0913$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->ivr_demo] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [ivr_demo]
> destination_number(9999) =~ /^5000$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->dynamic_conference] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [dynamic_conference] destination_number(9999) =~ /^5001$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->rtp_multicast_page] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [rtp_multicast_page] destination_number(9999) =~ /^pagegroup$|^7243$/
> break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park]
> continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park]
> destination_number(9999) =~ /^5900$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->unpark] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark]
> destination_number(9999) =~ /^5901$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->valet_park] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [valet_park] destination_number(9999) =~ /^(6000)$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->valet_park] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [valet_park] destination_number(9999) =~ /^(60\d[1-9])$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park]
> continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park]
> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->unpark] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark]
> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park]
> continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park]
> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->unpark] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark]
> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->wait]
> continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [wait]
> destination_number(9999) =~ /^wait$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->fax_receive] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [fax_receive] destination_number(9999) =~ /^9978$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->fax_transmit] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [fax_transmit] destination_number(9999) =~ /^9979$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->ringback_180] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [ringback_180] destination_number(9999) =~ /^9980$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->ringback_183_uk_ring] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [ringback_183_uk_ring] destination_number(9999) =~ /^9981$/ break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->ringback_183_music_ring] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [ringback_183_music_ring] destination_number(9999) =~ /^9982$/
> break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->ringback_post_answer_uk_ring] continue=false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(9999) =~ /^9983$/
> break=on-false
>
> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
> [default->ringback_post_answer_music] continue=false
>
> Dialplan:
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Fri Oct 14 19:34:50 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Oct 2011 11:34:50 -0400
Subject: [Freeswitch-users] conference with pin code and esl
Message-ID: <5F76E2D846A44258837AD996681E5DEB@e1705>

I'm running a conference in esl so in the log

EXECUTE sofia/internal/123456789 at kiki.hehe socket(127.0.0.1:9090 async full)
2011-10-14 11:28:15.534653 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/123456789 at kiki.hehe [BREAK]
2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789 at kiki.hehe Command Execute conference(conf_123456789 at default+2222+flags{moderator,endconf})
EXECUTE sofia/internal/123456789 at kiki.hehe conference(conf_123456789 at default+2222+flags{moderator,endconf})

but if the caller don't type any pin code ther conference is created.

this behaviour exist since I updated to this git

FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500)

Must I change anything ?

Thanks
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From infos at madovsky.org  Fri Oct 14 19:48:14 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Oct 2011 11:48:14 -0400
Subject: [Freeswitch-users] conference with pin code and esl
Message-ID: <8E20DFE641574516B5C6891BA3181F1B@e1705>

mmhmm ok
I didn't see this line in log

2011-10-14 11:28:27.714649 [DEBUG] mod_conference.c:6102 Moderator PIN found!

but I didn't set any pin code in conference.conf.xml



  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 14, 2011 11:34 AM
  Subject: conference with pin code and esl


  I'm running a conference in esl so in the log

  EXECUTE sofia/internal/123456789 at kiki.hehe socket(127.0.0.1:9090 async full)
  2011-10-14 11:28:15.534653 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/123456789 at kiki.hehe [BREAK]
  2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789 at kiki.hehe Command Execute conference(conf_123456789 at default+2222+flags{moderator,endconf})
  EXECUTE sofia/internal/123456789 at kiki.hehe conference(conf_123456789 at default+2222+flags{moderator,endconf})

  but if the caller don't type any pin code ther conference is created.

  this behaviour exist since I updated to this git

  FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500)

  Must I change anything ?

  Thanks
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From SPapineni at enghouse.com  Fri Oct 14 19:56:08 2011
From: SPapineni at enghouse.com (Papineni, Suneel)
Date: Fri, 14 Oct 2011 15:56:08 +0000
Subject: [Freeswitch-users] Error while using origiante command -- "Cannot
 Blind Transfer 1 Legged Calls"
Message-ID: <9438D04074E0DE45A49CD76099821272F8146B@CORP-MAIL-002.edge.local>

Hi,

I have downloaded latest version of Freeswitch (from GIT) and compiled. It went through and created Release version (on a Windows system).

Using X-lite, I am able to register users 1001 and 1003 and able to make a call between two users, but when I try to originate a call to one of extensions from freeswitch console, I am getting an error saying "Cannot Blind Transfer 1 Legged Calls". Following command is used for originating call.
"originate sofia/internal/1003 at 172.22.200.300<mailto:sofia/internal/1003 at 172.22.200.300> 9999"

Could someone help me why it is giving this error and how to resolve this.

Thanks & Regards
Suneel

Here is sofia status and logs

freeswitch at UK-SPapineni-7> sofia status

                     Name          Type                                       Data      State
=================================================================================================
           172.22.200.300         alias                                   internal      ALIASED
                 internal       profile          sip:mod_sofia at 172.22.200.300:5060      RUNNING (0)
            internal-ipv6       profile                   sip:mod_sofia@[::1]:5060      RUNNING (0)
                 external       profile          sip:mod_sofia at 172.22.200.300:5080      RUNNING (0)
    external::example.com       gateway                    sip:joeuser at example.com      NOREG
=================================================================================================
3 profiles 1 alias


Console Log.....
freeswitch at UK-SPapineni-7> originate sofia/internal/1003 at 172.22.200.300 9999
2011-10-14 16:44:24.894231 [NOTICE] switch_channel.c:911 New Channel sofia/internal/1003 at 172.22.200.300 [739a9189-d72b-4092-9c97-04af9d577e40]
2011-10-14 16:44:24.914233 [NOTICE] switch_channel.c:911 New Channel sofia/internal/0000000000 at 172.22.200.300 [56968324-cacc-41e8-834e-224716e666cf]
2011-10-14 16:44:24.914233 [INFO] mod_dialplan_xml.c:336 Processing  <0000000000>->1003 in context public
2011-10-14 16:44:24.914233 [ERR] sofia.c:6591 Cannot Blind Transfer 1 Legged calls
2011-10-14 16:44:25.014243 [NOTICE] switch_core_state_machine.c:226 sofia/internal/0000000000 at 172.22.200.300 has executed the last dialplan instruction, hanging up.

-ERR NO_USER_RESPONSE

2011-10-14 16:44:25.014243 [NOTICE] switch_core_state_machine.c:228 Hangup sofia/internal/0000000000 at 172.22.200.300 [CS_EXECUTE] [NORMAL_CLEARING]
2011-10-14 16:44:25.014243 [NOTICE] sofia.c:5926 Hangup sofia/internal/1003 at 172.22.200.300 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2011-10-14 16:44:25.014243 [NOTICE] switch_core_session.c:1369 Session 27 (sofia/internal/0000000000 at 172.22.200.300) Ended
2011-10-14 16:44:25.014243 [NOTICE] switch_core_session.c:1371 Close Channel sofia/internal/0000000000 at 172.22.200.300 [CS_DESTROY]
2011-10-14 16:44:25.034245 [NOTICE] switch_core_session.c:1369 Session 26 (sofia/internal/1003 at 172.22.200.300) Ended
2011-10-14 16:44:25.034245 [NOTICE] switch_core_session.c:1371 Close Channel sofia/internal/1003 at 172.22.200.300 [CS_DESTROY]
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From infos at madovsky.org  Fri Oct 14 20:37:02 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 Oct 2011 12:37:02 -0400
Subject: [Freeswitch-users] conference with pin code and esl
Message-ID: <B7370501622B417B80C18ED277DBC995@e1705>

I just found

http://jira.freeswitch.org/browse/FS-1565

but closed for incomplete

I wonder why before the pin code was working also for the moderator
but not now.

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 14, 2011 11:48 AM
  Subject: Re: conference with pin code and esl


  mmhmm ok
  I didn't see this line in log

  2011-10-14 11:28:27.714649 [DEBUG] mod_conference.c:6102 Moderator PIN found!

  but I didn't set any pin code in conference.conf.xml



    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Friday, October 14, 2011 11:34 AM
    Subject: conference with pin code and esl


    I'm running a conference in esl so in the log

    EXECUTE sofia/internal/123456789 at kiki.hehe socket(127.0.0.1:9090 async full)
    2011-10-14 11:28:15.534653 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/123456789 at kiki.hehe [BREAK]
    2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789 at kiki.hehe Command Execute conference(conf_123456789 at default+2222+flags{moderator,endconf})
    EXECUTE sofia/internal/123456789 at kiki.hehe conference(conf_123456789 at default+2222+flags{moderator,endconf})

    but if the caller don't type any pin code ther conference is created.

    this behaviour exist since I updated to this git

    FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500)

    Must I change anything ?

    Thanks
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From sirvan.paraste at gmail.com  Fri Oct 14 21:26:19 2011
From: sirvan.paraste at gmail.com (sirvan paraste)
Date: Fri, 14 Oct 2011 20:56:19 +0330
Subject: [Freeswitch-users] There is no voice in Two party call
In-Reply-To: <CAD-qRkjApPOappx4CFaqn1=0qpFbXU-Di=WPCMBK1LKM963_Dw@mail.gmail.com>
References: <20111011091700.16181.qmail@community22.interfree.it>
	<CAAAG0NVaYJbqUCRV76rTkEucWybZ1RzGpRanCqmSnV7E6VP_dA@mail.gmail.com>
	<CAD-qRkjApPOappx4CFaqn1=0qpFbXU-Di=WPCMBK1LKM963_Dw@mail.gmail.com>
Message-ID: <CAAAG0NUA8dta_CUzwwZ-Vpqckmj-LMay3+N0WdK7QfcYijNNSw@mail.gmail.com>

Hi
yes, this problem was related to local_ip_v4, i change it to the freeswitch
server IP, so the problem is solved now
Thanks a million ,

but there is a question too, where local_ip_v4 has been set? it must take FS
server IP automatically ? I just remove the variable $${local_ip_v4} and use
192.168.0.1 instead of.


On Thu, Oct 13, 2011 at 3:24 AM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> Sirvan resolved this issue on irc.  local_ip_v4 was 0.0.0.0, so SDP looked
> like a hold SDP to x-lite.  Setting rtp-ip and sip-ip to 192.168.0.1 fixed
> it.
>
>
>
>
> On Wed, Oct 12, 2011 at 8:15 AM, sirvan paraste <sirvan.paraste at gmail.com>wrote:
>
>> Dear
>>
>> Iused this commands to get log :
>> sodia loglevel all 9
>> sofia global siptrace on
>> sofia tracelevel info
>>
>> now, the resulted log file is attached to this mail. thank you for your
>> regard
>> another thing that is strange is when B call C party and C answering the
>> call this message  be shown on softphone at the start and then there is no
>> way voice : "on hold by another party" ,
>>
>> please help mo to find its solution. I am overwhelmed with it about 3
>> weeks :(((((
>>
>> thank you
>>
>>
>> On Tue, Oct 11, 2011 at 12:47 PM, <chrisbware at interfree.it> wrote:
>>
>>> Send a SIP trace of a call.
>>> Use freeswitch sofia debug (see on wiki) or install ngrep (if you are on
>>> linux) and use the following command
>>>
>>> ngrep port 5060 -Wbyline -q -d any
>>>
>>>
>>> Attach the output on this list.
>>>
>>>
>>>
>>>
>>>
>>> Chrisbware
>>>
>>> -----Messaggio originale-----
>>> Da: sirvan paraste <sirvan.paraste at gmail.com>
>>> Inviato il: 9 Ott 2011 - 22:38
>>> A: freeswitch-users at lists.freeswitch.org
>>>
>>>
>>> Dear all
>>> After regard for your attention to me
>>> I am student how want work on freeswitch. I have read freeSwitch
>>>
>>> documents (introduction, how to install, use default
>>> confuguration,...).
>>> for start of using freeswitch, I make three computer network together
>>> with a hub, after that,I install freeswitch on one of them and X-Lite
>>> on another computers, network checked with sharing file and ping
>>> issue.
>>> After registering of two X-lite hone with No. 1001,1002, they can call
>>>
>>> each other but there is no voice !!! :(
>>> i have checked and did some work but the problem is not solved.
>>>
>>> Can you help me?
>>>
>>> thank you.
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> -------------------------------------------------------------------------------
>>> Valore legale alle tue mail
>>> InterfreePEC - la tua Posta Elettronica Certificata
>>> http://pec.interfree.it
>>>
>>> -------------------------------------------------------------------------------
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Oct 14 21:31:26 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 14 Oct 2011 10:31:26 -0700 (PDT)
Subject: [Freeswitch-users] Error while using origiante command --
 "Cannot Blind Transfer 1 Legged Calls"
In-Reply-To: <9438D04074E0DE45A49CD76099821272F8146B@CORP-MAIL-002.edge.local>
References: <9438D04074E0DE45A49CD76099821272F8146B@CORP-MAIL-002.edge.local>
Message-ID: <1318613486205-6893380.post@n2.nabble.com>

Are you sure you dont have the phone forwarded or something like that?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-while-using-origiante-command-Cannot-Blind-Transfer-1-Legged-Calls-tp6892952p6893380.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From yungwei at resolvity.com  Fri Oct 14 21:48:42 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 14 Oct 2011 13:48:42 -0400
Subject: [Freeswitch-users] Run a js program as soon as someone
 registers with FS
In-Reply-To: <1318601820097-6892603.post@n2.nabble.com>
References: <33095823FD21DF429B481B5163264B7951D4672C2F@VMBX102.ihostexchange.net>
	<1318601820097-6892603.post@n2.nabble.com>
Message-ID: <33095823FD21DF429B481B5163264B7951D4672CFE@VMBX102.ihostexchange.net>

The idea is to automatically sign agents into call center queues as soon as they register with FS instead of having each agent dial 'login' to log into their queues.
Can this be done without relying on an external ESL listener?


From msc at freeswitch.org  Fri Oct 14 22:07:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Oct 2011 11:07:09 -0700
Subject: [Freeswitch-users] Error while using origiante command --
 "Cannot Blind Transfer 1 Legged Calls"
In-Reply-To: <9438D04074E0DE45A49CD76099821272F8146B@CORP-MAIL-002.edge.local>
References: <9438D04074E0DE45A49CD76099821272F8146B@CORP-MAIL-002.edge.local>
Message-ID: <CAKzWOxXocMARRPsfDzQE1SY9H_EydtaxBVeY=GV3Fb0WvAHBGg@mail.gmail.com>

what happens when you try this:
"originate user/1003 9999"

-MC

On Fri, Oct 14, 2011 at 8:56 AM, Papineni, Suneel <SPapineni at enghouse.com>wrote:

>  Hi,****
>
> ** **
>
> I have downloaded latest version of Freeswitch (from GIT) and compiled. It
> went through and created Release version (on a Windows system). ****
>
> ** **
>
> Using X-lite, I am able to register users 1001 and 1003 and able to make a
> call between two users, but when I try to originate a call to one of
> extensions from freeswitch console, I am getting an error saying ?Cannot
> Blind Transfer 1 Legged Calls?. Following command is used for originating
> call.****
>
> ?originate sofia/internal/1003 at 172.22.200.300 9999?****
>
> ** **
>
> Could someone help me why it is giving this error and how to resolve this.
> ****
>
> ** **
>
> Thanks & Regards****
>
> Suneel****
>
> ** **
>
> Here is sofia status and logs****
>
> ** **
>
> freeswitch at UK-SPapineni-7> sofia status****
>
> ** **
>
>                      Name
> Type                                       Data      State****
>
>
> =================================================================================================
> ****
>
>            172.22.200.300         alias
> internal      ALIASED****
>
>                  internal       profile
> sip:mod_sofia at 172.22.200.300:5060      RUNNING (0)****
>
>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060
> RUNNING (0)****
>
>                  external       profile
> sip:mod_sofia at 172.22.200.300:5080      RUNNING (0)****
>
>     external::example.com       gateway
> sip:joeuser at example.com      NOREG****
>
>
> =================================================================================================
> ****
>
> 3 profiles 1 alias****
>
> ** **
>
> ** **
>
> Console Log?..****
>
> freeswitch at UK-SPapineni-7> originate sofia/internal/1003 at 172.22.200.3009999
> ****
>
> 2011-10-14 16:44:24.894231 [NOTICE] switch_channel.c:911 New Channel
> sofia/internal/1003 at 172.22.200.300 [739a9189-d72b-4092-9c97-04af9d577e40]*
> ***
>
> 2011-10-14 16:44:24.914233 [NOTICE] switch_channel.c:911 New Channel
> sofia/internal/0000000000 at 172.22.200.300[56968324-cacc-41e8-834e-224716e666cf]
> ****
>
> 2011-10-14 16:44:24.914233 [INFO] mod_dialplan_xml.c:336 Processing
> <0000000000>->1003 in context public****
>
> 2011-10-14 16:44:24.914233 [ERR] sofia.c:6591 Cannot Blind Transfer 1
> Legged calls****
>
> 2011-10-14 16:44:25.014243 [NOTICE] switch_core_state_machine.c:226
> sofia/internal/0000000000 at 172.22.200.300 has executed the last dialplan
> instruction, hanging up.****
>
> ** **
>
> -ERR NO_USER_RESPONSE****
>
> ** **
>
> 2011-10-14 16:44:25.014243 [NOTICE] switch_core_state_machine.c:228 Hangup
> sofia/internal/0000000000 at 172.22.200.300 [CS_EXECUTE] [NORMAL_CLEARING]***
> *
>
> 2011-10-14 16:44:25.014243 [NOTICE] sofia.c:5926 Hangup
> sofia/internal/1003 at 172.22.200.300 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]**
> **
>
> 2011-10-14 16:44:25.014243 [NOTICE] switch_core_session.c:1369 Session 27
> (sofia/internal/0000000000 at 172.22.200.300) Ended****
>
> 2011-10-14 16:44:25.014243 [NOTICE] switch_core_session.c:1371 Close
> Channel sofia/internal/0000000000 at 172.22.200.300 [CS_DESTROY]****
>
> 2011-10-14 16:44:25.034245 [NOTICE] switch_core_session.c:1369 Session 26
> (sofia/internal/1003 at 172.22.200.300) Ended****
>
> 2011-10-14 16:44:25.034245 [NOTICE] switch_core_session.c:1371 Close
> Channel sofia/internal/1003 at 172.22.200.300 [CS_DESTROY]****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Oct 14 22:28:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Oct 2011 13:28:56 -0500
Subject: [Freeswitch-users] Run a js program as soon as someone
 registers with FS
In-Reply-To: <33095823FD21DF429B481B5163264B7951D4672CFE@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7951D4672C2F@VMBX102.ihostexchange.net>
	<1318601820097-6892603.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7951D4672CFE@VMBX102.ihostexchange.net>
Message-ID: <CAKbxfG87UKg8a=v=o1PYv9uwCHqMZ0ey6_yADf3J9W1hyzbbnQ@mail.gmail.com>

On Fri, Oct 14, 2011 at 12:48 PM, Yungwei Chen <yungwei at resolvity.com> wrote:
> The idea is to automatically sign agents into call center queues as soon as they register with FS instead of having each agent dial 'login' to log into their queues.
> Can this be done without relying on an external ESL listener?
>

Not in any sane way, no.  Plus its trivial to do.

-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Fri Oct 14 22:31:30 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 14 Oct 2011 13:31:30 -0500
Subject: [Freeswitch-users] Change sound someone hears when they ring me
In-Reply-To: <4E97D611.50204@exemail.com.au>
References: <4E97D611.50204@exemail.com.au>
Message-ID: <CAKbxfG8Tg_UxgW9d8XgDiUVgv+DpBHOOOhVWqBxPCoep_cpr0g@mail.gmail.com>

<action application="set" data="ringback=${uk-ring}"/> in your local
extension entry in dialplan/default.xml
somewhere before it calls bridge.

On Fri, Oct 14, 2011 at 1:26 AM, Ryan How <rhow at exemail.com.au> wrote:
> Hi,
>
> I've set a freeswitch box with bluebox (with a bit of hassle! but it got
> there). I can't find where to start looking regarding changing the sound
> an external person hears when they call me. usually from the standard
> landline when dialing a number I get a ring-ring ring-ring, but when
> someone dials my number (which is a voip DID) it has a long
> riiiiiiiiinnnnngggggg . . . . . riiiiiiiiinngggggg. Is there any way I
> can change that? My asterisk box and other voip phones haven't done it,
> so I figure it must be able to be changed.
>
> Thanks, Ryan
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From fcintrono at gmail.com  Fri Oct 14 23:40:37 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Fri, 14 Oct 2011 14:40:37 -0500
Subject: [Freeswitch-users] Newbie problem: IVR at extension 5000 talking
	too fast...
Message-ID: <CANtw_ZMf=_5ZEekiH4i+OOJiwj10BXbufBwSbLP15VyJcAOqMg@mail.gmail.com>

Hi there.
Well I am having some problems with default extension 5000, this is a barely
new freeswitch installation. Actually I haven't changed anything from the
default configuration. This morning I realized that the virtual machine
which I used to install freeswitch had wrong time info, it had UTC time but
my zone is America/Mexico_City

After I adjusted it, I dialed 5000 extension and the IVR's voice was talking
faster than before I changed time info.



I am sure time's difference is the problem, but ?why?

And of course ?How can I solve this problem?

Thanks ind advance for your kind help.

Have a good day.
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From msc at freeswitch.org  Sat Oct 15 00:18:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Oct 2011 13:18:58 -0700
Subject: [Freeswitch-users] Freeswitch equilavent to Asterisk
	BackgroundDetect
In-Reply-To: <111509FCE9EBF74BAB2CB38AAC521D9B1313058C49@helle.dk.hq.firstcom.dk>
References: <111509FCE9EBF74BAB2CB38AAC521D9B1313058C49@helle.dk.hq.firstcom.dk>
Message-ID: <CAKzWOxUrcJMer0XZ_iH1o4ZFqm169RfOgHrAFF5hJRAoc1r2pw@mail.gmail.com>

The closest thing we have is play_and_get_digits. Now, if you're trying to
do answering machine detection there is a commercial module (mod_amd)
available from consulting at freeswitch.org. It is very reasonably priced.

-MC

On Fri, Oct 14, 2011 at 7:06 AM, Jon Sch?pzinsky <jos at firstcom.dk> wrote:

> Hello List.****
>
> ** **
>
> Ive been trying to find a function equal to what BackgroundDetect does in
> Asterisk.****
>
> Anybody got an idea on how I could implement the functionality in
> Freeswitch?****
>
> ** **
>
> I use it in IVR services, to detect when a caller interrupts the currently
> playing prompt.****
>
> ** **
>
> ** **
>
> ** **
>
> Med venlig hilsen/Kind Regards****
>
> ** **
>
> Jon Leren Sch?pzinsky****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From masood24 at gmail.com  Sat Oct 15 00:39:52 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Fri, 14 Oct 2011 23:39:52 +0300
Subject: [Freeswitch-users] Digest Authentication issue
Message-ID: <CAMd3iJJX9887ALe73KYjNd5dWmkG_bdE-L-gGiuE2Bf5vOBMRg@mail.gmail.com>

Dear ,

I had successfully installed and configured radiusclient. I do not know that
what next lines should i do to get users registered on my freeswitch using
digest authentication with my radius server,
Like I have my radsius server and billing system. Now what lines should i
add in sofit conf or in dialplans that every time client hits server request
should be sent to radius server and what lines should i add to register them
if i get proper return code and how to do that.

Your Help is needed
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From Hector.Geraldino at ip-soft.net  Sat Oct 15 00:56:40 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Fri, 14 Oct 2011 16:56:40 -0400
Subject: [Freeswitch-users] Freeswitch equilavent to
	Asterisk	BackgroundDetect
In-Reply-To: <CAKzWOxUrcJMer0XZ_iH1o4ZFqm169RfOgHrAFF5hJRAoc1r2pw@mail.gmail.com>
References: <111509FCE9EBF74BAB2CB38AAC521D9B1313058C49@helle.dk.hq.firstcom.dk>
	<CAKzWOxUrcJMer0XZ_iH1o4ZFqm169RfOgHrAFF5hJRAoc1r2pw@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02209F8427@NY1-EXMB-01.ip-soft.net>

Hi Michael,

I've tried to get a quote request for this module from consulting at freeswitch.org<mailto:consulting at freeswitch.org> a few times with no luck so far. Sorry that I mentioned it on this on this thread, but I wanted to take advantage of the fact that you're talking about it to see if I can get any feedback from the guys at FS.

Thanks,
Hector

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, October 14, 2011 4:19 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch equilavent to Asterisk BackgroundDetect

The closest thing we have is play_and_get_digits. Now, if you're trying to do answering machine detection there is a commercial module (mod_amd) available from consulting at freeswitch.org<mailto:consulting at freeswitch.org>. It is very reasonably priced.

-MC
On Fri, Oct 14, 2011 at 7:06 AM, Jon Sch?pzinsky <jos at firstcom.dk<mailto:jos at firstcom.dk>> wrote:
Hello List.

Ive been trying to find a function equal to what BackgroundDetect does in Asterisk.
Anybody got an idea on how I could implement the functionality in Freeswitch?

I use it in IVR services, to detect when a caller interrupts the currently playing prompt.



Med venlig hilsen/Kind Regards

Jon Leren Sch?pzinsky


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 cesar.bermudez at gmail.com  Sat Oct 15 01:16:02 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Fri, 14 Oct 2011 15:16:02 -0600
Subject: [Freeswitch-users]  Custom rinback tone
Message-ID: <CADrcvV_OLU-UazhsNq=2cS=gvaQ2acvmhCaUyHu-Pd8e=hezUQ@mail.gmail.com>

hi fs users and devs.

mi ringback tone its 450/1200, 0/4900 ( 1.2 on 4.9 off )
how can modify this:
<action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/>
to have the same ringing tone.

Best regards.
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From cmrienzo at gmail.com  Sat Oct 15 01:24:22 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 14 Oct 2011 17:24:22 -0400
Subject: [Freeswitch-users] Custom rinback tone
In-Reply-To: <CADrcvV_OLU-UazhsNq=2cS=gvaQ2acvmhCaUyHu-Pd8e=hezUQ@mail.gmail.com>
References: <CADrcvV_OLU-UazhsNq=2cS=gvaQ2acvmhCaUyHu-Pd8e=hezUQ@mail.gmail.com>
Message-ID: <CAD-qRkjwG8upqkdanHbda=-8rpmkickw0nisTjO_CbvkVdEF4g@mail.gmail.com>

http://wiki.freeswitch.org/wiki/TGML

<action application="set" data="ringback=%(1200, 4900, 450)"/>


On Fri, Oct 14, 2011 at 5:16 PM, Cesar Bermudez <cesar.bermudez at gmail.com>wrote:

> hi fs users and devs.
>
> mi ringback tone its 450/1200, 0/4900 ( 1.2 on 4.9 off )
> how can modify this:
> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/>
> to have the same ringing tone.
>
> Best regards.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cesar.bermudez at gmail.com  Sat Oct 15 01:36:42 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Fri, 14 Oct 2011 15:36:42 -0600
Subject: [Freeswitch-users] Custom rinback tone
In-Reply-To: <CAD-qRkjwG8upqkdanHbda=-8rpmkickw0nisTjO_CbvkVdEF4g@mail.gmail.com>
References: <CADrcvV_OLU-UazhsNq=2cS=gvaQ2acvmhCaUyHu-Pd8e=hezUQ@mail.gmail.com>
	<CAD-qRkjwG8upqkdanHbda=-8rpmkickw0nisTjO_CbvkVdEF4g@mail.gmail.com>
Message-ID: <CADrcvV8rj1epHcWu44ZdWoBy5jgGD0rU9hzZqLjE82L=Dk9c0w@mail.gmail.com>

thanks

On Fri, Oct 14, 2011 at 3:24 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> http://wiki.freeswitch.org/wiki/TGML
>
> <action application="set" data="ringback=%(1200, 4900, 450)"/>
>
>
> On Fri, Oct 14, 2011 at 5:16 PM, Cesar Bermudez <cesar.bermudez at gmail.com>wrote:
>
>> hi fs users and devs.
>>
>> mi ringback tone its 450/1200, 0/4900 ( 1.2 on 4.9 off )
>> how can modify this:
>> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/>
>> to have the same ringing tone.
>>
>> Best regards.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fcintrono at gmail.com  Sat Oct 15 03:08:57 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Fri, 14 Oct 2011 18:08:57 -0500
Subject: [Freeswitch-users] Question about sound files...
Message-ID: <CANtw_ZOkEs1cn3tdg2n6LX2YpTUDRFAxmTPeoV_omNCxndJFXw@mail.gmail.com>

Are there spanish sound files for freeswitch?


I am using version 1.0.6 and  I am realizing that there are only russian and
english sound files.


Thanks in advance for your help.

Have a good day.
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From fcintrono at gmail.com  Sat Oct 15 05:04:11 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Fri, 14 Oct 2011 20:04:11 -0500
Subject: [Freeswitch-users] Question about sound files...
In-Reply-To: <CANtw_ZOkEs1cn3tdg2n6LX2YpTUDRFAxmTPeoV_omNCxndJFXw@mail.gmail.com>
References: <CANtw_ZOkEs1cn3tdg2n6LX2YpTUDRFAxmTPeoV_omNCxndJFXw@mail.gmail.com>
Message-ID: <CANtw_ZNbiOvDLhDQNkzDaHKuD850iyGzEp+DazfByv1brs=TuA@mail.gmail.com>

I have read in this old post that there are an incomplete set of spanish
sounds:

>Diego,
>Our intrepid users have made sound files in these languages. The
>French and German files were mentioned in the mailing list so search
>the archives. We found some missing sounds in the Spanish sound set an
>we are reviewing them. The Spanish sounds are Latin American. We also
>have a bounty for Castilian sounds.
>-MC

>Sent from my iPhone

>On May 8, 2010, at 9:44 AM, Diego Toro <dftoro at yahoo.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:

>>* Hi all,*>>**>>* The lastest announcement the official release of version 1.0.6, it  *>>* says,*>>* Sound files have been created for French, German, and Spanish.*>>**>>* Where Can I get these sound files ?*>>**>>* Thank you*>>**>>* Diego Toro*>>* http://lacarretade.blogspot.com/*>>**>>**>>**>>**>>* _______________________________________________*>>* FreeSWITCH-users mailing list*>>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>*>>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users*>>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- *>>* users*>>* http://www.freeswitch.org*


I installed git and downloaded last repo, but spanish sounds aren't there:

?Where I could get access to this spanish sounds?

Thanks in advance for your help.

Have a good day.
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From rhow at exemail.com.au  Sat Oct 15 05:24:40 2011
From: rhow at exemail.com.au (Ryan How)
Date: Sat, 15 Oct 2011 09:24:40 +0800
Subject: [Freeswitch-users] Change sound someone hears when they ring me
In-Reply-To: <CAKbxfG8Tg_UxgW9d8XgDiUVgv+DpBHOOOhVWqBxPCoep_cpr0g@mail.gmail.com>
References: <4E97D611.50204@exemail.com.au>
	<CAKbxfG8Tg_UxgW9d8XgDiUVgv+DpBHOOOhVWqBxPCoep_cpr0g@mail.gmail.com>
Message-ID: <4E98E0D8.90201@exemail.com.au>

Thanks!, That's great. I can see it from fishing through the config 
files now. ringback is the key :)

On 15/10/2011 2:31 AM, Anthony Minessale wrote:
> <action application="set" data="ringback=${uk-ring}"/>  in your local
> extension entry in dialplan/default.xml
> somewhere before it calls bridge.
>
> On Fri, Oct 14, 2011 at 1:26 AM, Ryan How<rhow at exemail.com.au>  wrote:
>> Hi,
>>
>> I've set a freeswitch box with bluebox (with a bit of hassle! but it got
>> there). I can't find where to start looking regarding changing the sound
>> an external person hears when they call me. usually from the standard
>> landline when dialing a number I get a ring-ring ring-ring, but when
>> someone dials my number (which is a voip DID) it has a long
>> riiiiiiiiinnnnngggggg . . . . . riiiiiiiiinngggggg. Is there any way I
>> can change that? My asterisk box and other voip phones haven't done it,
>> so I figure it must be able to be changed.
>>
>> Thanks, Ryan
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 rhow at exemail.com.au  Sat Oct 15 05:45:41 2011
From: rhow at exemail.com.au (Ryan How)
Date: Sat, 15 Oct 2011 09:45:41 +0800
Subject: [Freeswitch-users] Ring Group with External Number not transferring
	after timeout
Message-ID: <4E98E5C5.2090301@exemail.com.au>

Hi,

I have a ring group with an external number I've created using bluebox. 
Sorry I hope I am posting this in the right forum...

What happens is, it will ring for 30 seconds, then stop ringing, then 
rather than transfer to voicemail, it will start ringing again. If I 
remove the external number from the ring group then it is ok and works 
as expected. Is bluebox setting this up wrong? Or is there something I 
need to change or add? XML is below... On occasion it seems to work, 
maybe if the external number is busy or something?

It is funny because this is exactly the same behaviour I got when 
setting up the same thing with asterisk. Makes me think there is 
something I'm not getting when putting in external numbers.

I'm using the GIT head of freeswitch (and bluebox) as of yesterday.

Thanks,

Ryan


<extension name="main_number_9" continue="true">
<condition field="destination_number" expression="^3000$">
<action application="set" bluebox="settingEndBridge" 
data="hangup_after_bridge=true"/>
<action application="set" bluebox="settingFail" 
data="continue_on_fail=true"/>
<action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
<action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
<action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
<action application="set" bluebox="settingRingXfer" 
data="transfer_ringback=${us-ring}"/>
<action application="export" bluebox="sipCalleeIdName" 
data="sip_callee_id_name=Ring All"/>
<action application="export" bluebox="sipCalleeIdNumber" 
data="sip_callee_id_number=3000"/>
<action application="export" bluebox="settingContinueOnTimeout" 
data="originate_continue_on_timeout=true"/>
<action application="bridge" 
data="[ignore_early_media=true,leg_timeout=30]user/2001@$${location_1}:_:[ignore_early_media=true,leg_timeout=30]user/2002@$${location_1}:_:[ignore_early_media=true,leg_timeout=30]sofia/gateway/trunk_1/0447614933"/>
<action application="answer"/>
<action application="voicemail" data="default voicemail_1 2001"/>
<action application="hangup"/>
</condition>
</extension>



From peter.olsson at visionutveckling.se  Sat Oct 15 09:55:49 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 15 Oct 2011 07:55:49 +0200
Subject: [Freeswitch-users] Newbie problem: IVR at extension 5000
 talking	too fast...
In-Reply-To: <CANtw_ZMf=_5ZEekiH4i+OOJiwj10BXbufBwSbLP15VyJcAOqMg@mail.gmail.com>
References: <CANtw_ZMf=_5ZEekiH4i+OOJiwj10BXbufBwSbLP15VyJcAOqMg@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF148@cooper>

I'm pretty sure that the only reason for this is the virtual machine, it's never safe to run any kind of RTP stream within a virtual machine, beacause of the bad timing.

Run timer_test in the FS console, and get back with the results.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Francisco Javier Cintr?n Olgu?n [fcintrono at gmail.com]
Skickat: den 14 oktober 2011 21:40
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Newbie problem: IVR at extension 5000 talking  too fast...

Hi there.
Well I am having some problems with default extension 5000, this is a barely new freeswitch installation. Actually I haven't changed anything from the default configuration. This morning I realized that the virtual machine which I used to install freeswitch had wrong time info, it had UTC time but my zone is America/Mexico_City

After I adjusted it, I dialed 5000 extension and the IVR's voice was talking faster than before I changed time info.



I am sure time's difference is the problem, but ?why?

And of course ?How can I solve this problem?

Thanks ind advance for your kind help.

Have a good day.




!DSPAM:4e98913632761039510560!


From alec.taylor6 at gmail.com  Sat Oct 15 13:12:33 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sat, 15 Oct 2011 20:12:33 +1100
Subject: [Freeswitch-users] Emulate and script emulation of users calling
 in/receiving calls, transferring calls &etc
Message-ID: <CAO+9iGcWJzafkmTeoSc6ovgN6S+RwhdAoEgt=uEJUrQ=9dcW4g@mail.gmail.com>

Good evening,

If asterisk or freeswitch would be taught in a classroom environment,
is there someway to emulate and script emulation of users calling
in/receiving calls, transferring calls &etc?

The plan is to have each student setup there own Asterisk or
FreeSwitch box, and measure handling efficiency, and communicate
between the servers (by transferring calls from server to server).

Thanks for all suggestions,

Alec Taylor


From anthony.minessale at gmail.com  Sat Oct 15 19:10:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 15 Oct 2011 10:10:21 -0500
Subject: [Freeswitch-users] Invalid Application Hash
In-Reply-To: <CAFiqYuncYY6=DwP91F9ys32UosmM8Fbbfv4Sqo-yStM_e36prg@mail.gmail.com>
References: <9438D04074E0DE45A49CD76099821272F81279@CORP-MAIL-002.edge.local>
	<C21CEA3D-073A-4B86-ACA7-B115127B7D32@gmail.com>
	<9438D04074E0DE45A49CD76099821272F81324@CORP-MAIL-002.edge.local>
	<CAFiqYuncYY6=DwP91F9ys32UosmM8Fbbfv4Sqo-yStM_e36prg@mail.gmail.com>
Message-ID: <CAKbxfG_LN-bLP6Ajv=vK5mKkZNXhd0ucP--x97fVJHe47wyLPw@mail.gmail.com>

or, if you're not really using it, take the line ouf of your dialplan.
 It's just the example config that's using it.


On Fri, Oct 14, 2011 at 10:30 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Try loading it anyway. If it doesn't load try loading mod_limit. One
> of the two will give you the app.
>
> Usage of the app didn't change, just the module providing it so there
> should be no difference between switches using the old and new module.
>
> mod_limit will be in newer releases, but as a dummy module that gives
> a Warning level message when you load it. If you see that message
> you're on a version that should have mod_hash.
>
> -Steve
>
>
> On 14 October 2011 13:40, Papineni, Suneel <SPapineni at enghouse.com> wrote:
>> Hi Steve,
>>
>>
>>
>> I think mod_hash is not present in Freeswitch 1.0.5 ( Ichecked in default
>> modules.conf file). Seems it has mod_limit (which is removed in later
>> releases).
>>
>>
>>
>> Thanks & Regards
>>
>> Suneel
>>
>>
>>
>>
>>
>> From: Steven Ayre [mailto:steveayre at gmail.com]
>> Sent: 14 October 2011 11:09
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Invalid Application Hash
>>
>>
>>
>> You need to load mod_hash in modules.conf.xml
>>
>> Steve on iPhone
>>
>> On 14 Oct 2011, at 10:08, "Papineni, Suneel" <SPapineni at enghouse.com> wrote:
>>
>> Hi,
>>
>>
>>
>> Before sending this request, I searched the forum for similar posts and
>> found some solutions for recent versions of Freeswitch. Unfortunately one of
>> my servers is using Freswitch 1.0.5. In this, I am trying to make a call
>> using ?originate sofia/external/01183035887 at 135.xxx.xxx.xxx 9999? command at
>> console. This is giving an error ?Invalid Application hash? and FS is
>> sending CANCEL.
>>
>>
>>
>> Is there any way to resolve this issue other than upgrading to the latest
>> version of Freeswitch (I am going to upgrade in coming weeks, but I need to
>> test a specific scenario now on existing system).
>>
>>
>>
>> Thanks & Regards
>>
>> Suneel
>>
>>
>>
>>
>>
>> Here is the log for reference?
>>
>>
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] sofia.c:4158 Channel
>> sofia/external/01183035887 at 135.196.132.200 entering state [ready][200]
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3579 Audio Codec Compare
>> [PCMU:0:8000:20]/[PCMU:0:8000:20]
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2354 Set Codec
>> sofia/external/01183035887 at 135.196.132.200 PCMU/8000 20 ms 160 samples
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:3512 Set 2833 dtmf recv
>> payload to 0
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] sofia_glue.c:2594 AUDIO RTP
>> [sofia/external/01183035887 at 135.196.132.200] 192.168.20.240 port 25744 ->
>> 135.196.145.145 port 54206 codec: 0 ms: 20
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_rtp.c:1182 Starting timer [soft]
>> 160 bytes per 20ms
>>
>> 2011-10-13 16:59:34.341952 [NOTICE] sofia.c:4738 Channel
>> [sofia/external/01183035887 at 135.196.132.200] has been answered
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_ivr_originate.c:3148 Originate
>> Resulted in Success: [sofia/external/01183035887 at 135.196.132.200]
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_ivr.c:1441
>> (sofia/external/01183035887 at 135.196.132.200) State Change CS_CONSUME_MEDIA
>> -> CS_ROUTING
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/external/01183035887 at 135.196.132.200 [BREAK]
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_session.c:638 Send signal
>> sofia/external/01183035887 at 135.196.132.200 [BREAK]
>>
>> 2011-10-13 16:59:34.341952 [NOTICE] switch_ivr.c:1447 Transfer
>> sofia/external/01183035887 at 135.196.132.200 to XML[9999 at default]
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/01183035887 at 135.196.132.200) Running State Change CS_ROUTING
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:341
>> (sofia/external/01183035887 at 135.196.132.200) State ROUTING
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] mod_sofia.c:141
>> sofia/external/01183035887 at 135.196.132.200 SOFIA ROUTING
>>
>> 2011-10-13 16:59:34.341952 [DEBUG] switch_core_state_machine.c:77
>> sofia/external/01183035887 at 135.196.132.200 Standard ROUTING
>>
>> 2011-10-13 16:59:34.341952 [INFO] mod_dialplan_xml.c:418 Processing ->9999
>> in context default
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->unloop] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->tod_example] continue=true
>>
>> Dialplan: day of week[5] =~ 2-6 (PASS)
>>
>> Dialplan: hour[16] =~ 9-18 (PASS)
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Date/Time Match (PASS)
>> [tod_example] break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action set(open=true)
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->holiday_example] continue=true
>>
>> Dialplan: month[10] =~ 1 (FAIL)
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Date/Time Match (FAIL)
>> [holiday_example] break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->global-intercept] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [global-intercept] destination_number(9999) =~ /^886$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->group-intercept] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [group-intercept] destination_number(9999) =~ /^\*8$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->intercept-ext] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [intercept-ext] destination_number(9999) =~ /^\*\*(\d+)$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->redial] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [redial]
>> destination_number(9999) =~ /^870$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->global] continue=true
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [global]
>> ${call_debug}(false) =~ /^true$/ break=never
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [global]
>> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
>> break=never
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Absolute Condition
>> [global]
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action
>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action
>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Action
>> hash(insert/${domain_name}-last_dial/global/${uuid})
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->snom-demo-2] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [snom-demo-2] destination_number(9999) =~ /^9001$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->snom-demo-1] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [snom-demo-1] destination_number(9999) =~ /^9000$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->eavesdrop] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [eavesdrop] destination_number(9999) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->eavesdrop] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [eavesdrop] destination_number(9999) =~ /^779$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->call_return] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [call_return] destination_number(9999) =~ /^\*69$|^869$|^lcr$/
>> break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->del-group] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [del-group] destination_number(9999) =~ /^80(\d{2})$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->add-group] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [add-group] destination_number(9999) =~ /^81(\d{2})$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->call-group-simo] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [call-group-simo] destination_number(9999) =~ /^82(\d{2})$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->call-group-order] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [call-group-order] destination_number(9999) =~ /^83(\d{2})$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->extension-intercom] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [extension-intercom] destination_number(9999) =~ /^8(10[01][0-9])$/
>> break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->Local_Extension] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [Local_Extension] destination_number(9999) =~ /^(10[01][0-9])$/
>> break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->group_dial_sales] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [group_dial_sales] destination_number(9999) =~ /^2000$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->group_dial_support] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [group_dial_support] destination_number(9999) =~ /^2001$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->group_dial_billing] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [group_dial_billing] destination_number(9999) =~ /^2002$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->operator] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [operator]
>> destination_number(9999) =~ /^(operator|0)$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->vmain] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [vmain]
>> destination_number(9999) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->sip_uri] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [sip_uri]
>> destination_number(9999) =~ /^sip:(.*)$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->nb_conferences] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [nb_conferences] destination_number(9999) =~ /^(30\d{2})$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->wb_conferences] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [wb_conferences] destination_number(9999) =~ /^(31\d{2})$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->uwb_conferences] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [uwb_conferences] destination_number(9999) =~ /^(32\d{2})$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->cdquality_conferences] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [cdquality_conferences] destination_number(9999) =~ /^(33\d{2})$/
>> break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->freeswitch_public_conf_via_sip] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [freeswitch_public_conf_via_sip] destination_number(9999) =~
>> /^9(888|1616|3232)$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->mad_boss_intercom] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [mad_boss_intercom] destination_number(9999) =~ /^0911$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->mad_boss_intercom] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [mad_boss_intercom] destination_number(9999) =~ /^0912$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->mad_boss] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [mad_boss]
>> destination_number(9999) =~ /^0913$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->ivr_demo] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [ivr_demo]
>> destination_number(9999) =~ /^5000$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->dynamic_conference] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [dynamic_conference] destination_number(9999) =~ /^5001$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->rtp_multicast_page] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [rtp_multicast_page] destination_number(9999) =~ /^pagegroup$|^7243$/
>> break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park]
>> continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park]
>> destination_number(9999) =~ /^5900$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->unpark] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark]
>> destination_number(9999) =~ /^5901$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->valet_park] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [valet_park] destination_number(9999) =~ /^(6000)$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->valet_park] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [valet_park] destination_number(9999) =~ /^(60\d[1-9])$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park]
>> continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park]
>> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->unpark] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark]
>> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->park]
>> continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [park]
>> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->unpark] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [unpark]
>> source(..\..\src\switch_ivr_originate.c) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing [default->wait]
>> continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL) [wait]
>> destination_number(9999) =~ /^wait$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->fax_receive] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [fax_receive] destination_number(9999) =~ /^9978$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->fax_transmit] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [fax_transmit] destination_number(9999) =~ /^9979$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->ringback_180] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [ringback_180] destination_number(9999) =~ /^9980$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->ringback_183_uk_ring] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [ringback_183_uk_ring] destination_number(9999) =~ /^9981$/ break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->ringback_183_music_ring] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [ringback_183_music_ring] destination_number(9999) =~ /^9982$/
>> break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->ringback_post_answer_uk_ring] continue=false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 Regex (FAIL)
>> [ringback_post_answer_uk_ring] destination_number(9999) =~ /^9983$/
>> break=on-false
>>
>> Dialplan: sofia/external/01183035887 at 135.196.132.200 parsing
>> [default->ringback_post_answer_music] continue=false
>>
>> Dialplan:
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From masood24 at gmail.com  Sat Oct 15 19:18:15 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Sat, 15 Oct 2011 18:18:15 +0300
Subject: [Freeswitch-users] Digest authentication
Message-ID: <CAMd3iJJzbnU92mUeYU6YP8M_G2R_zBcQQzn83tN9jaDZ13wP+g@mail.gmail.com>

Dear ,

I had successfully installed and configured radiusclient. I do not know that
what next lines should i do to get users registered on my freeswitch using
digest authentication with my radius server,
Like I have my radsius server and billing system. Now what lines should i
add in sofit conf or in dialplans that every time client hits server request
should be sent to radius server and what lines should i add to register them
if i get proper return code and how to do that.

Your Help is needed

Later on i copied all the attributes same was as mentioned in wiki , but
getting error that unknown parameter h323-conf-id or when i change the
dictionary to dictionary.cisco it says unable to read dictionary.

-- 
*Regards,
Masood Ahmed
*
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From fcintrono at gmail.com  Sat Oct 15 19:24:44 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Sat, 15 Oct 2011 10:24:44 -0500
Subject: [Freeswitch-users] Question about sound files...
In-Reply-To: <CANtw_ZNbiOvDLhDQNkzDaHKuD850iyGzEp+DazfByv1brs=TuA@mail.gmail.com>
References: <CANtw_ZOkEs1cn3tdg2n6LX2YpTUDRFAxmTPeoV_omNCxndJFXw@mail.gmail.com>
	<CANtw_ZNbiOvDLhDQNkzDaHKuD850iyGzEp+DazfByv1brs=TuA@mail.gmail.com>
Message-ID: <CANtw_ZNivr4DSj=WiKxM7T=yrD-mXxdRwK=3Wg9k6Md0B6M6Fg@mail.gmail.com>

 Nevermind :)

I just found site http://www.freeswitch.es

Inside this site I got a package named
http://files.freeswitch.org/freeswitch-sounds-es-mx-maria-44100.tar.gz

I am going to test this voices.




On Fri, Oct 14, 2011 at 8:04 PM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> I have read in this old post that there are an incomplete set of spanish
> sounds:
>
> >Diego,
> >Our intrepid users have made sound files in these languages. The
> >French and German files were mentioned in the mailing list so search
> >the archives. We found some missing sounds in the Spanish sound set an
> >we are reviewing them. The Spanish sounds are Latin American. We also
> >have a bounty for Castilian sounds.
> >-MC
>
> >Sent from my iPhone
>
> >On May 8, 2010, at 9:44 AM, Diego Toro <dftoro at yahoo.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
>
> >>* Hi all,*>>**>>* The lastest announcement the official release of version 1.0.6, it  *>>* says,*>>* Sound files have been created for French, German, and Spanish.*>>**>>* Where Can I get these sound files ?*>>**>>* Thank you*>>**>>* Diego Toro*>>* http://lacarretade.blogspot.com/*>>**>>**>>**>>**>>* _______________________________________________*>>* FreeSWITCH-users mailing list*>>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>*>>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users*>>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- *>>* users*>>* http://www.freeswitch.org*
>
>
> I installed git and downloaded last repo, but spanish sounds aren't there:
>
> ?Where I could get access to this spanish sounds?
>
>
> Thanks in advance for your help.
>
> Have a good day.
>
>
>
>
>
>
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From fcintrono at gmail.com  Sat Oct 15 23:02:13 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Sat, 15 Oct 2011 14:02:13 -0500
Subject: [Freeswitch-users] Newbie problem: IVR at extension 5000
 talking too fast...
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF148@cooper>
References: <CANtw_ZMf=_5ZEekiH4i+OOJiwj10BXbufBwSbLP15VyJcAOqMg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF148@cooper>
Message-ID: <CANtw_ZPt8EMsRCNvjdHv+-WYouF9AZxe8KdcKxKwWxct5zz4VA@mail.gmail.com>

Well, Here I have timer_test output:

http://pastebin.freeswitch.org/17541

Thanks in advance for your help.

On Sat, Oct 15, 2011 at 12:55 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I'm pretty sure that the only reason for this is the virtual machine, it's
> never safe to run any kind of RTP stream within a virtual machine, beacause
> of the bad timing.
>
> Run timer_test in the FS console, and get back with the results.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Francisco Javier
> Cintr?n Olgu?n [fcintrono at gmail.com]
> Skickat: den 14 oktober 2011 21:40
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Newbie problem: IVR at extension 5000 talking  too
> fast...
>
> Hi there.
> Well I am having some problems with default extension 5000, this is a
> barely new freeswitch installation. Actually I haven't changed anything from
> the default configuration. This morning I realized that the virtual machine
> which I used to install freeswitch had wrong time info, it had UTC time but
> my zone is America/Mexico_City
>
> After I adjusted it, I dialed 5000 extension and the IVR's voice was
> talking faster than before I changed time info.
>
>
>
> I am sure time's difference is the problem, but ?why?
>
> And of course ?How can I solve this problem?
>
> Thanks ind advance for your kind help.
>
> Have a good day.
>
>
>
>
> !DSPAM:4e98913632761039510560!
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fcintrono at gmail.com  Sat Oct 15 23:03:21 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Sat, 15 Oct 2011 14:03:21 -0500
Subject: [Freeswitch-users] Newbie problem: IVR at extension 5000
 talking too fast...
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF148@cooper>
References: <CANtw_ZMf=_5ZEekiH4i+OOJiwj10BXbufBwSbLP15VyJcAOqMg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF148@cooper>
Message-ID: <CANtw_ZOrcimdfa06s3u260O+2WwLMy5COe8oG1g30yeu+2Mi0w@mail.gmail.com>

I am going to try to get another machine to make my tests. ?What kind of rig
is needed for a freeswitch home server? I have 5 phones and 2 POT lines.

Thanks in advance for your help.

On Sat, Oct 15, 2011 at 12:55 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I'm pretty sure that the only reason for this is the virtual machine, it's
> never safe to run any kind of RTP stream within a virtual machine, beacause
> of the bad timing.
>
> Run timer_test in the FS console, and get back with the results.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Francisco Javier
> Cintr?n Olgu?n [fcintrono at gmail.com]
> Skickat: den 14 oktober 2011 21:40
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Newbie problem: IVR at extension 5000 talking  too
> fast...
>
> Hi there.
> Well I am having some problems with default extension 5000, this is a
> barely new freeswitch installation. Actually I haven't changed anything from
> the default configuration. This morning I realized that the virtual machine
> which I used to install freeswitch had wrong time info, it had UTC time but
> my zone is America/Mexico_City
>
> After I adjusted it, I dialed 5000 extension and the IVR's voice was
> talking faster than before I changed time info.
>
>
>
> I am sure time's difference is the problem, but ?why?
>
> And of course ?How can I solve this problem?
>
> Thanks ind advance for your kind help.
>
> Have a good day.
>
>
>
>
> !DSPAM:4e98913632761039510560!
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fieldpeak at gmail.com  Sun Oct 16 07:41:32 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 16 Oct 2011 11:41:32 +0800
Subject: [Freeswitch-users] FS -How to identify the A-leg or B-leg inside
	the source code
Message-ID: <CAOTmbMxQt7OHO-RGM_BGOiYdu9c1OC8Nb+U+bGqTM671Qvxw+A@mail.gmail.com>

Hi Friends,

i'm using mod_nibblebill for billing on FS, i bundled event processing
fucntions when status change,
e.g. /* on_routing */ check_billing_before_routing,
and * on_hangup */ process_hangup, as below,
however, i found it excuted the processing function on both A-leg and
B-leg, what i want is that either only A-leg or only B-leg (i can
select by coding), how can i realize it?

P.S. is it possible that i identify the Legs (A or B) inside the
processing function (e.g. check_billing_before_routing,) by intput
parameter *sessionsession?


source codes as below,

switch_state_handler_table_t nibble_state_handler = {
	/* on_init */ NULL,
	/* on_routing */ check_billing_before_routing, /* Need to add a check
here for anything in their account before routing */
	/* on_execute */ sched_billing, 	/* Turn on heartbeat for this
session and do an initial account check */
	/* on_hangup */ process_hangup, 	/* On hangup - most important place
to go bill */
	/* on_exch_media */ NULL,
	/* on_soft_exec */ NULL,
	/* on_consume_med */ NULL,
	/* on_hibernate */ NULL,
	/* on_reset */ NULL,
	/* on_park */ NULL,
	/* on_reporting */ NULL,
	/* on_destroy */ NULL
};


static switch_status_t
check_billing_before_routing(switch_core_session_t *session)
{
	//is it possible to identify the legs (A or B) here?
	...
}
	
	static switch_status_t process_hangup(switch_core_session_t *session)
{
	...
}

BR,
Charles


From rhow at exemail.com.au  Sun Oct 16 09:35:25 2011
From: rhow at exemail.com.au (Ryan How)
Date: Sun, 16 Oct 2011 13:35:25 +0800
Subject: [Freeswitch-users] External number starts ringing again after
	rejecting call until timeout
Message-ID: <4E9A6D1D.1010701@exemail.com.au>

Hi,

I've got an extension set up to redirect to my mobile. It all works 
fine, except if I reject the call on my mobile it starts ringing again. 
It just keeps going until the timeout and then transfers to voicemail. 
How can I make it go to voicemail if I reject the call on my mobile? I 
tried adding fail_on_single_reject=true but it didn't seem to do 
anything. Sorry for the newbie question. I'm slowly learning :). XML is 
below.

Thanks!

Ryan

<extension name="main_number_11" continue="true">
<condition field="destination_number" expression="^3101$">
<action application="set" bluebox="settingEndBridge" 
data="hangup_after_bridge=true"/>
<action application="set" bluebox="settingFail" 
data="continue_on_fail=true"/>
<action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
<action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
<action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
<action application="set" bluebox="settingRingXfer" 
data="transfer_ringback=${us-ring}"/>
<action application="set" data="fail_on_single_reject=true"/>
<action application="export" bluebox="sipCalleeIdName" 
data="sip_callee_id_name=Ryan Work Mobile"/>
<action application="export" bluebox="sipCalleeIdNumber" 
data="sip_callee_id_number=3101"/>
<action application="bridge" 
data="{ignore_early_media=true}sofia/gateway/trunk_1/0447614933"/>
<action application="answer"/>
<action application="voicemail" data="default voicemail_1 2001"/>
<action application="hangup"/>
</condition>
</extension>


From alec.taylor6 at gmail.com  Sun Oct 16 11:44:23 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sun, 16 Oct 2011 18:44:23 +1100
Subject: [Freeswitch-users] [asterisk-users] Emulate and script
 emulation of users calling in/receiving calls, transferring calls &etc
In-Reply-To: <20111015182607.GA12665@tryba.nl>
References: <CAO+9iGcWJzafkmTeoSc6ovgN6S+RwhdAoEgt=uEJUrQ=9dcW4g@mail.gmail.com>
	<20111015182607.GA12665@tryba.nl>
Message-ID: <CAO+9iGfyQx2-V_Wjg5VCv8RhZhsTXqgm7Es408LvP482c4tC6A@mail.gmail.com>

Thanks, looks like: sipp.sourceforge.net supports scripting...

Are there sample University Curricula for teaching VOIP with Asterisk
or FreeSwitch?

On Sun, Oct 16, 2011 at 5:26 AM, Daniel Tryba <daniel at tryba.nl> wrote:
> On Sat, Oct 15, 2011 at 08:12:33PM +1100, Alec Taylor wrote:
>> If asterisk or freeswitch would be taught in a classroom environment,
>> is there someway to emulate and script emulation of users calling
>> in/receiving calls, transferring calls &etc?
>
> The Asterisk Manager Interface (AMI) and callfiles to/from Echo(),
> voicemail or IVRs! But much easier is to have multiple (soft)phones
> available to 1 student.
>
> --
>
> ? Daniel Tryba
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> ? ? ? ? ? ? ? http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> ? http://lists.digium.com/mailman/listinfo/asterisk-users
>


From avi at avimarcus.net  Sun Oct 16 12:17:59 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 16 Oct 2011 10:17:59 +0200
Subject: [Freeswitch-users] FS -How to identify the A-leg or B-leg
 inside the source code
In-Reply-To: <CAOTmbMxQt7OHO-RGM_BGOiYdu9c1OC8Nb+U+bGqTM671Qvxw+A@mail.gmail.com>
References: <CAOTmbMxQt7OHO-RGM_BGOiYdu9c1OC8Nb+U+bGqTM671Qvxw+A@mail.gmail.com>
Message-ID: <CALNZuwkZqyr_7dx29c_bHxTXCT=jYUAF83=0=ZeFic4pzHa5Tg@mail.gmail.com>

Looking at the a_ prefix from mod_xml_cdr.c, it seems like it's this:
>
> is_b = channel && switch_channel_get_originator_caller_profile(channel);


Feel free to check freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c

-Avi


On Sun, Oct 16, 2011 at 5:41 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Friends,
>
> i'm using mod_nibblebill for billing on FS, i bundled event processing
> fucntions when status change,
> e.g. /* on_routing */ check_billing_before_routing,
> and * on_hangup */ process_hangup, as below,
> however, i found it excuted the processing function on both A-leg and
> B-leg, what i want is that either only A-leg or only B-leg (i can
> select by coding), how can i realize it?
>
> P.S. is it possible that i identify the Legs (A or B) inside the
> processing function (e.g. check_billing_before_routing,) by intput
> parameter *sessionsession?
>
>
> source codes as below,
>
> switch_state_handler_table_t nibble_state_handler = {
>        /* on_init */ NULL,
>        /* on_routing */ check_billing_before_routing, /* Need to add a
> check
> here for anything in their account before routing */
>        /* on_execute */ sched_billing,         /* Turn on heartbeat for
> this
> session and do an initial account check */
>        /* on_hangup */ process_hangup,         /* On hangup - most
> important place
> to go bill */
>        /* on_exch_media */ NULL,
>        /* on_soft_exec */ NULL,
>        /* on_consume_med */ NULL,
>        /* on_hibernate */ NULL,
>        /* on_reset */ NULL,
>        /* on_park */ NULL,
>        /* on_reporting */ NULL,
>        /* on_destroy */ NULL
> };
>
>
> static switch_status_t
> check_billing_before_routing(switch_core_session_t *session)
> {
>        //is it possible to identify the legs (A or B) here?
>        ...
> }
>
>        static switch_status_t process_hangup(switch_core_session_t
> *session)
> {
>        ...
> }
>
> BR,
> Charles
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raimund.sacherer at logitravel.com  Mon Oct 17 11:55:15 2011
From: raimund.sacherer at logitravel.com (Raimund Sacherer)
Date: Mon, 17 Oct 2011 09:55:15 +0200 (CEST)
Subject: [Freeswitch-users] Help with choppy audio after
	attended	transfer
In-Reply-To: <CAL0fNgW730m1cKFrCRsA=ACeq=6nOWYWWaubJAeGUaXGk36rZg@mail.gmail.com>
Message-ID: <41b80a3a-4e9e-410f-8c9e-c1eb5cc02f91@Raimund-ThinkPad-X61s>

Are you still gettint this Problem? I ran into similiar issues, but it's intermittent ... 

best 
Ray 

----- Original Message -----

From: "Chris Cureau" <cmcureau at gmail.com> 
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, August 3, 2011 1:28:22 AM 
Subject: Re: [Freeswitch-users] Help with choppy audio after attended transfer 

Latest git doesn't help unfortunately. 

More research is beginning to confirm its a codec issue. When I limit the phone to codes that use 20ms packetization, I hear no incoming audio on outgoing calls. Incoming calls are okay. When using All for codes (and the 30ms packetization) I get the choppy voice when leaving MOH. 




On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale < anthony.minessale at gmail.com > wrote: 


can you do make current and try latest git? 





On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau < cmcureau at gmail.com > wrote: 
> Anthony, 
> 
> Thanks for answering...and sorry for the delay. I've already checked all of 
> the ptime settings I can, and all phones plus freeswitch are set to use 20ms 
> packetization. I've even set "scrooge" in the codec negotiation, but I keep 
> running into this issue. I've updated my post with "sofia global siptrace 
> on". 
> 
> I am assuming that the ptime issue happens around line 2462 
> ( http://pastebin.freeswitch.org/16935 ) 
> 
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare 
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000] 
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU 
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send 
> payload to 101 
> 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP 
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are 
> unchanged for sofia/internal/ sip:1003 at 10.0.1.205:5060 . 
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052 
> sofia/internal/ sip:1003 at 10.0.1.205:5060 Setting audio receive payload in 
> Re-INVITE to 0 
> 
> Could this be an issue with the Aastra'a firmware? Or maybe the MOH is 
> being processed at 30ms instead of 20ms, and the negotiation is not updated 
> somehow? 
> 
> I don't mean to sound ignorant, but I'm really at a loss here...and thanks 
> again for any help! 
> 
> Cheers, 
> Chris 
> 
> On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale 
> < anthony.minessale at gmail.com > wrote: 
>> 
>> probably ptime related thing. 
>> you should have included the sip trace "sofia global siptrace on" 
>> 
>> 
>> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau < cmcureau at gmail.com > wrote: 
>> > I'm having some issues with extremely choppy audio after a call has been 
>> > sent to another extension via an automated transfer. The audio is great 
>> > when the call is answered. Shortly after, the transfer button is 
>> > pressed 
>> > and the incoming call hears music on hold. The music on hold is sent to 
>> > the 
>> > caller sounds fine as does the conversation between extensions. When 
>> > the 
>> > transfer is completed, the caller hears what sounds like someone 
>> > speaking 
>> > through a fan (though slower) but incoming audio sounds fine. 
>> > 
>> > Since it's such a large log, I posted it to the FreeSWITCH pastebin: 
>> > http://pastebin.freeswitch.org/16911 
>> > 
>> > I'm thinking that it has something to do with the transition from MOH to 
>> > the 
>> > internal extension, but I can't figure out what is happening. 
>> > 
>> > Any ideas? 
>> > 
>> > _______________________________________________ 
>> > Join us at ClueCon 2011, Aug 9-11, Chicago 
>> > http://www.cluecon.com 877-7-4ACLUE 
>> > 
>> > FreeSWITCH-users mailing list 
>> > FreeSWITCH-users at lists.freeswitch.org 
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
>> > http://www.freeswitch.org 
>> > 
>> > 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II 
>> 
>> FreeSWITCH http://www.freeswitch.org/ 
>> ClueCon http://www.cluecon.com/ 
>> Twitter: http://twitter.com/FreeSWITCH_wire 
>> 
>> AIM: anthm 
>> MSN:anthony_minessale at hotmail.com 
>> GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
>> IRC: irc.freenode.net #freeswitch 
>> 
>> FreeSWITCH Developer Conference 
>> sip:888 at conference.freeswitch.org 
>> googletalk:conf+888 at conference.freeswitch.org 
>> pstn: +19193869900 
>> 
>> _______________________________________________ 
>> Join us at ClueCon 2011, Aug 9-11, Chicago 
>> http://www.cluecon.com 877-7-4ACLUE 
>> 
>> FreeSWITCH-users mailing list 
>> FreeSWITCH-users at lists.freeswitch.org 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
>> http://www.freeswitch.org 
> 
> 
> _______________________________________________ 
> Join us at ClueCon 2011, Aug 9-11, Chicago 
> http://www.cluecon.com 877-7-4ACLUE 
> 
> FreeSWITCH-users mailing list 
> FreeSWITCH-users at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
> http://www.freeswitch.org 
> 
> 



-- 
Anthony Minessale II 

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

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

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

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

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




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

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



-- 
Logitravel.com
Raimund Sacherer 
Sistemas 
Agencia de Viajes Online 
www.logitravel.com 
Edificio Logitravel, Parcela 3B (Parc Bit) 
Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca 
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From acrow at integrafin.co.uk  Mon Oct 17 18:40:57 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 17 Oct 2011 15:40:57 +0100
Subject: [Freeswitch-users] Hold and BLF - disable flashing or pickup
 for held call?
In-Reply-To: <4E6DD00F.5050303@integrafin.co.uk>
References: <4E652FE9.5010600@integrafin.co.uk>	<EC5966E8-12BE-4C8A-B36E-5E0D0E0120D2@freeswitch.org>
	<4E6DD00F.5050303@integrafin.co.uk>
Message-ID: <4E9C3E79.4050109@integrafin.co.uk>

On 12/09/11 10:25, Alex Crow wrote:
> Brian,
>
> Snom think that this flashing on hold is incorrect - I had this on my 
> snom forum post:
>
> No! I've never seen such behaviour - Neither the LED is supposed to
> blink when the monitored party is on hold, nor it should be possible to
> steal the call by hitting the function key. That indeed is extremely
> confusing and is definitely not intended.
> I don't have much (if at all) experience with Freeswitch but from what I
>   understand, extension monitoring is being configured through dial plans
>   (and?/) or themod_snom module  <http://wiki.freeswitch.org/wiki/Mod_snom>.
> I guess something is not configured properly which results in the problem you are encountering.
> It should not make any difference whether the monitored extension is set
>   on hold or not. Can you post a SIP trace which shows such a scenario?
>
> I replied with:
>
> Extension monitoring is being done with bog-standard subscriptions, not with mod_snom or anything extra in the dialplan.
> The Polycoms flash the button while a monitored line is held, but you can't pick it up.
> I will try to get a trace from the phone, however someone re-used my
> test FS box for a windows desktop while I was away, so I have to build a
>   new one.
>
> Any more ideas as to why this is happening?
>
> Cheers
>
> Alex

Anyone know anything about this?

We are having some problems with confused users in one of our offices.

Thanks,

Alex

This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
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From acrow at integrafin.co.uk  Mon Oct 17 18:44:28 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 17 Oct 2011 15:44:28 +0100
Subject: [Freeswitch-users] Snom 370 Presence Function
In-Reply-To: <CABK0fC7LqFNJT4yMayLeQXhd1O-KzsYn_ppcQiRNXRDx=wiH0g@mail.gmail.com>
References: <CABK0fC7LqFNJT4yMayLeQXhd1O-KzsYn_ppcQiRNXRDx=wiH0g@mail.gmail.com>
Message-ID: <4E9C3F4C.2030304@integrafin.co.uk>

On 09/10/11 15:47, afshin afzali wrote:
> Hi Guys,
>
> I'll appreciate if I know someone has been experienced snom 370
> presence function. As I trace sip messages, it shows that snom does
> not subscribe for itself and just tries to publish its state which
> replies 404 by freeswitch ! I don't know it is the snom or freeswitch
> protocol issue.
>
> BEST,
> -- afshin
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

Did you turn off "peer to peer call completion"? This seems to break BLF.

Cheers

Alex

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From nic at itecity.com  Sun Oct 16 20:15:21 2011
From: nic at itecity.com (Network Information Centre)
Date: Sun, 16 Oct 2011 12:15:21 -0400
Subject: [Freeswitch-users] Freeswitch (Windows) 64-bit - call answering
	issues
Message-ID: <007001cc8c1e$d04879e0$70d96da0$@itecity.com>

Hello,

 

I am rather new to Freeswitch, but have successfully installed and ran the
32-bit windows version.

I have now upgraded to the windows 64-bit, on a larger server (pretty clean
re-install).

 

I can make and receive calls (through multiple gateways).

However, my issue is that the call is never actually answered (but
Freeswitch says it is).

Although, I believe I may actually have two different issues.

   

5555551212 = Freeswitch line/extension

15554445555 = Remote line

   

Calls are being made between a cellular phone and a softphone (connected to
Freeswitch)

   

   

1)      If placing a call, the remote side answers (and Freeswitch also
reports this), but the calling side just keeps ringing.

   

2011-10-16 10:41:44.146140 [NOTICE] switch_channel.c:911 New Channel
sofia/internal/5555551212 at sip.XXXXX.com:5060
[4f422e4e-2513-46a2-8416-cd1a771a231a]

2011-10-16 10:41:44.193015 [INFO] mod_dialplan_xml.c:336 Processing Don
Hunter <5555551212>->15554445555 in context default

2011-10-16 10:41:44.193015 [NOTICE] switch_channel.c:911 New Channel
sofia/external/15554445555 [54540bc5-e9aa-4485-ac6a-88ea43552dde]

2011-10-16 10:41:49.166648 [NOTICE] sofia_glue.c:3868 Pre-Answer
sofia/external/15554445555!

2011-10-16 10:41:49.166648 [INFO] switch_ivr_originate.c:3198 Sending early
media

2011-10-16 10:41:49.166648 [NOTICE] mod_sofia.c:2464 Pre-Answer
sofia/internal/5555551212 at sip.XXXXX.com:5060!

2011-10-16 10:41:51.018210 [NOTICE] sofia.c:5780 Channel
[sofia/external/15554445555] has been answered

2011-10-16 10:41:51.018210 [NOTICE] switch_ivr_bridge.c:417 Channel
[sofia/internal/5555551212 at sip.XXXXX.com:5060] has been answered

2011-10-16 10:41:59.188132 [NOTICE] sofia.c:560 Hangup
sofia/external/15554445555 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]

2011-10-16 10:41:59.209617 [NOTICE] switch_ivr_bridge.c:1355 Hangup
sofia/internal/5555551212 at sip.XXXXX.com:5060 [CS_EXECUTE] [NORMAL_CLEARING]

2011-10-16 10:41:59.209617 [NOTICE] switch_core_session.c:1367 Session 6
(sofia/external/15554445555) Ended

2011-10-16 10:41:59.209617 [NOTICE] switch_core_session.c:1369 Close Channel
sofia/external/15554445555 [CS_DESTROY]

2011-10-16 10:41:59.495750 [NOTICE] switch_core_session.c:1367 Session 5
(sofia/internal/5555551212 at sip.XXXXX.com:5060) Ended

2011-10-16 10:41:59.495750 [NOTICE] switch_core_session.c:1369 Close Channel
sofia/internal/5555551212 at sip.XXXXX.com:5060 [CS_DESTROY]

   

As you can see by the [2011-10-16 10:41:51.018210] lines, the call is
reporting as being answered, but is not actually picked up.

   

   

2)      When receiving a call, the calling side rings and eventually goes to
voicemail. But the called side also rings, but even when answered (phone set
shows answered) both sides continue to ring.

   

2011-10-16 10:58:52.552476 [NOTICE] switch_channel.c:911 New Channel
sofia/internal/15554445555 at 46.19.209.14
[ff76c548-816f-4313-8b15-dbca896e1264]

2011-10-16 10:58:52.589585 [INFO] mod_dialplan_xml.c:336 Processing
4167869038 <15554445555>->5555551212 in context public

2011-10-16 10:58:52.589585 [NOTICE] switch_ivr.c:1699 Transfer
sofia/internal/15554445555 at 46.19.209.14 to XML[5555551212 at default]

2011-10-16 10:58:52.589585 [INFO] mod_dialplan_xml.c:336 Processing
4167869038 <15554445555>->5555551212 in context default

2011-10-16 10:58:52.589585 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *1
execute_extension::dx XMLfeatures

2011-10-16 10:58:52.589585 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *2
record_session::C:\Programecordings/archive/2011/Oct/16/ff76c548-816f-4313-8
b15-dbca896e1264.wav

2011-10-16 10:58:52.589585 [INFO] switch_ivr_async.c:3107 Bound B-Leg: *3
execute_extension::cf XMLfeatures

2011-10-16 10:58:52.589585 [NOTICE] switch_channel.c:911 New Channel
sofia/internal/sip:5555551212 at 69.196.144.178:33677
[36da2d38-0b99-47ae-9f1b-5fa7785159ee]

2011-10-16 10:58:52.741929 [NOTICE] sofia.c:5235 Ring-Ready
sofia/internal/sip:5555551212 at 69.196.144.178:33677!

2011-10-16 10:58:52.741929 [INFO] switch_ivr_originate.c:1113 Sending early
media

2011-10-16 10:58:52.741929 [NOTICE] mod_sofia.c:2464 Pre-Answer
sofia/internal/15554445555 at 46.19.209.14!

2011-10-16 10:59:22.008531 [NOTICE] switch_ivr_originate.c:3165 Hangup
sofia/internal/sip:5555551212 at 69.196.144.178:33677 [CS_CONSUME_MEDIA]
[NO_ANSWER]

2011-10-16 10:59:22.008531 [NOTICE] switch_ivr_originate.c:2457 Cannot
create outgoing channel of type [user] cause: [NO_ANSWER]

2011-10-16 10:59:22.008531 [INFO] mod_dptools.c:2810 Originate Failed.
Cause: NO_ANSWER

2011-10-16 10:59:22.008531 [NOTICE] mod_dptools.c:1043 Channel
[sofia/internal/15554445555 at 46.19.209.14] has been answered

2011-10-16 10:59:22.008531 [NOTICE] switch_core_session.c:1367 Session 10
(sofia/internal/sip:5555551212 at 69.196.144.178:33677) Ended

2011-10-16 10:59:22.008531 [NOTICE] switch_core_session.c:1369 Close Channel
sofia/internal/sip:5555551212 at 69.196.144.178:33677 [CS_DESTROY]

2011-10-16 10:59:32.526109 [NOTICE] sofia.c:560 Hangup
sofia/internal/15554445555 at 46.19.209.14 [CS_EXECUTE] [NORMAL_CLEARING]

2011-10-16 10:59:32.867906 [NOTICE] switch_core_session.c:1367 Session 9
(sofia/internal/15554445555 at 46.19.209.14) Ended

2011-10-16 10:59:32.867906 [NOTICE] switch_core_session.c:1369 Close Channel
sofia/internal/15554445555 at 46.19.209.14 [CS_DESTROY]

   

   

I have compared my configs against the version that works, but cannot see
any differences.  I know I am missing something small and stupid.  

   

Any assistance would be appreciated (even a pointer as to where to look)...
as I am not sure where this is actually handled and in which XML(s).

  

 

 

 

Regards,

 

Don Hunter

e-City Solutions Inc.

 

email:                          don.hunter at 44ecity.com

CA/US Toll-Free:     +1-855-IT-ECITY, ext 200

UK National:             +44 870-820-0286

iNum (inum.net):   +883 5100-0827-0154

 

ITeCity.com  |  FoundU.com  |  StoreLocatorService.com  |  go-Postal.ca  |
getZiped.com | FraudCloud.com

WebPageAlerts.com | TurboBrochures.com | DomainStruck.com | EZ-Verify.com |
VOIPMuch.com

 

 

 

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From fieldpeak at gmail.com  Mon Oct 17 18:50:53 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 17 Oct 2011 22:50:53 +0800
Subject: [Freeswitch-users] FS -How to identify the A-leg or B-leg
 inside the source code
In-Reply-To: <CALNZuwkZqyr_7dx29c_bHxTXCT=jYUAF83=0=ZeFic4pzHa5Tg@mail.gmail.com>
References: <CAOTmbMxQt7OHO-RGM_BGOiYdu9c1OC8Nb+U+bGqTM671Qvxw+A@mail.gmail.com>
	<CALNZuwkZqyr_7dx29c_bHxTXCT=jYUAF83=0=ZeFic4pzHa5Tg@mail.gmail.com>
Message-ID: <CAOTmbMzdQ+ZxwHru=bAJXFiJyvt=TyKQA0VF4PZ28ZpvNeR5Pg@mail.gmail.com>

Hi Avi,

Thanks your valuable info, and i will try later, Thanks.

BR,
Charles

2011/10/16 Avi Marcus <avi at avimarcus.net>

> Looking at the a_ prefix from mod_xml_cdr.c, it seems like it's this:
>>
>> is_b = channel && switch_channel_get_originator_caller_profile(channel);
>
>
> Feel free to check freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c
>
> -Avi
>
>
> On Sun, Oct 16, 2011 at 5:41 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Friends,
>>
>> i'm using mod_nibblebill for billing on FS, i bundled event processing
>> fucntions when status change,
>> e.g. /* on_routing */ check_billing_before_routing,
>> and * on_hangup */ process_hangup, as below,
>> however, i found it excuted the processing function on both A-leg and
>> B-leg, what i want is that either only A-leg or only B-leg (i can
>> select by coding), how can i realize it?
>>
>> P.S. is it possible that i identify the Legs (A or B) inside the
>> processing function (e.g. check_billing_before_routing,) by intput
>> parameter *sessionsession?
>>
>>
>> source codes as below,
>>
>> switch_state_handler_table_t nibble_state_handler = {
>>        /* on_init */ NULL,
>>        /* on_routing */ check_billing_before_routing, /* Need to add a
>> check
>> here for anything in their account before routing */
>>        /* on_execute */ sched_billing,         /* Turn on heartbeat for
>> this
>> session and do an initial account check */
>>        /* on_hangup */ process_hangup,         /* On hangup - most
>> important place
>> to go bill */
>>        /* on_exch_media */ NULL,
>>        /* on_soft_exec */ NULL,
>>        /* on_consume_med */ NULL,
>>        /* on_hibernate */ NULL,
>>        /* on_reset */ NULL,
>>        /* on_park */ NULL,
>>        /* on_reporting */ NULL,
>>        /* on_destroy */ NULL
>> };
>>
>>
>> static switch_status_t
>> check_billing_before_routing(switch_core_session_t *session)
>> {
>>        //is it possible to identify the legs (A or B) here?
>>        ...
>> }
>>
>>        static switch_status_t process_hangup(switch_core_session_t
>> *session)
>> {
>>        ...
>> }
>>
>> BR,
>> Charles
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Charles
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From wampir98 at gmail.com  Mon Oct 17 18:59:26 2011
From: wampir98 at gmail.com (Jacek)
Date: Mon, 17 Oct 2011 16:59:26 +0200
Subject: [Freeswitch-users] How do I  debugging  Client Auth?
Message-ID: <4E9C42CE.9000502@gmail.com>

Hi

I have FreeSWITCH on Debian Squeeze.
I have 2 questions:

1: How to configure FreeSWITCH, any list handling user checked the sql
database (mysql) instead of the xml files in the folder conf / directory.
Or, how to use external script for authentication (eg, use PAM, Radius,
or otherwise authorized client)?

2: How to debug (verbose) a problem with user authentication (the client
auth)?
for example, the program (Twinkle) reports the error:

+++ 17-10-2011 16:23:18.998049 INFO SIP ::process_sip_msg
Received from: udp:83.230.27.214:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 83.230.27.214:5063;rport=5063;branch=z9hG4bKclobopii
From: "user" <sip:user at domain.tld>;tag=nmehl
To: "user" <sip:user at domain.tld>;tag=vQ0ma0yHt7UmH
Call-ID: wiqutrgrresdddt at domain.tld
CSeq: 381 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fdb4031 2011-10-14
14-13-16 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0

and no trace of FreeSWITCH logs logon failure, or other cause of the
problem.
Console says only:

soa_set_params(static::0x818deb0, ...) called
tport_tsend(0x8172090) tpn = UDP/83.230.27.214:5063
tport_resolve addrinfo = 83.230.27.214:5063
tport_by_addrinfo(0x8172090): not found by name UDP/83.230.27.214:5063
tport_vsend returned 510
nta: sent 403 Forbidden for REGISTER (651)
nta_leg_destroy((nil))

Where UDP/83.230.27.214:5063 - client addres.

My Freeswitch comes from subversioon:
version: Freeswitch (1.0.7)
config.log:
.....
uname -m = i686
uname -r = 2.6.39-bpo.2-686-pae
uname -s = Linux
uname -v = #1 SMP Thu Aug 4 11:02:22 UTC 2011
....

Cheers
;)

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From jeff at jefflenk.com  Mon Oct 17 20:21:00 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 17 Oct 2011 09:21:00 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch (Windows) 64-bit - call answering
	issues
In-Reply-To: <007001cc8c1e$d04879e0$70d96da0$@itecity.com>
References: <007001cc8c1e$d04879e0$70d96da0$@itecity.com>
Message-ID: <1318868460385-6901234.post@n2.nabble.com>

Did you copy your config files from the other machine? Did you modify the
default configuration to set local ip addresses to deal with NAT perhaps?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Windows-64-bit-call-answering-issues-tp6900876p6901234.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Mon Oct 17 22:14:00 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 17 Oct 2011 11:14:00 -0700
Subject: [Freeswitch-users] mod_blacklist functionality change input
Message-ID: <CAMWiMy8=c5DeC98WW45HhjBXVzf1LjhKbQdCbJ__q8Mp=F2rvg@mail.gmail.com>

Hey all,

Currently I am looking for community input on how mod_blacklist should
be changed. Right now I find that when the blacklist dump api command
is executed in the console, it only dumps the numbers it has on the
hash table. That may not be much of a concern for most users using
this, but what if you have to remove a few numbers from the blacklist?
Currently from what I can see on mod_blacklist is it only dumps the
numbers contained in the hash table and it won't make any attempts to
clear out the old numbers that are already stored in the blacklist
file.

My proposed change would be to make blacklist dump to flush out the
contents first and THEN write out the contents on the hash table to
that file as specified on the xml config, this change has been already
pushed out on my github repo as commented out code as I have not even
bothered to make any attempts to test this functionality change...
yet... I might test this when I get home tonight from school, but that
depends if I do have time on hand to even test this out.

Regards,

curriegrad2004


From nic at itecity.com  Mon Oct 17 23:30:21 2011
From: nic at itecity.com (Network Information Centre)
Date: Mon, 17 Oct 2011 19:30:21 +0000
Subject: [Freeswitch-users] Freeswitch (Windows) 64-bit - call
	answering	issues
Message-ID: <1341827762-1318879879-cardhu_decombobulator_blackberry.rim.net-470636866-@b13.c13.bise6.blackberry>

Oddly, testing this from another external sip (android) phone on my switch to another external phone - WORKS.

I suspect this may be either a) an issue with my softphone client - although it worked prior)

Or b) something on my internal network is not nating or routing traffic to/from my softphone (as my switch is now externally hosted).

Either way, It appears to be working outside my test network.

 
------Original Message------
From: Jeff Lenk
Sender: freeswitch-users-bounces at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
ReplyTo: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch (Windows) 64-bit - call answering	issues
Sent: Oct 17, 2011 12:21 PM

Did you copy your config files from the other machine? Did you modify the
default configuration to set local ip addresses to deal with NAT perhaps?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Windows-64-bit-call-answering-issues-tp6900876p6901234.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


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

Regards,
Network Informatiom Centre
e-City Solutions Inc.
nic at itecity.com

From msc at freeswitch.org  Tue Oct 18 00:12:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Oct 2011 13:12:07 -0700
Subject: [Freeswitch-users] Loop XML using call_timeout ..
In-Reply-To: <1318525773.20249.10.camel@marces.madrid.commsmundi.com>
References: <1318525773.20249.10.camel@marces.madrid.commsmundi.com>
Message-ID: <CAKzWOxULmqimmCvD34f065vus6a8LE=k-cA1ixfAJ3aDhUUyWw@mail.gmail.com>

Did you figure this out yet? Not sure exactly what you're trying to do here,
but you could use the sched_hangup API and schedule a hangup or
sched_transfer to schedule a transfer to a specific dialplan extension that
does your cleanup work.

<action application="set" data="api_res=${sched_transfer +60 ${uuid}
handle_timeout XML handlers}"/>

Then something like this for the dp

<context name="handlers">
  <extension name="handle loop timeouts">
    <condition field="destination_number" expression="^handle_timeout$">
      <action application="log" data="INFO OB call timed out..."/>
      <action application="foo" data="bar"/> <!-- do something useful -->
      <action application="hangup" data="pick_a_hangup_cause_if_you_wish"/>
    </condition>
  </extensions>
</context>

-MC

On Thu, Oct 13, 2011 at 10:09 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
>
> Hi,
>
> Is it possible to loop a bridge until a global call timeout (or other
> variable) is reached?
>
> In the wiki is this example is a fixed number of loops...
>
> <extension name="3-way gateway distro">
>
> <condition field="destination_number" expression="^(.*)$">
>       <action application="set" data="continue_on_fail=true"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>       <action application="bridge" data="sofia/gateway/${distributor(3gw)}/$1" loop="3"/>
>     </condition>
>   </extension>
>
> My idea is to call a group sequential, but once it reaches the last member it starts over again.
> so when calling a group, once the timeout for each user in the group is reach it hangups the call (the normal behavior),
> it jumps again to the begin of the group until a global timeout is reached.
>
> For example:
>
> <extension name="group">
> <condition field="destination_number" expression="^(.*)$">
>       <action application="set" data="continue_on_fail=true"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>       <action application="set" data="call_timeout=60"/>
>       <action application="set" data="leg_timeout=10"/>
>       <action application="bridge" data=""${group_call(group1 at default+F)}" loop="3"/>
>     </condition>
>   </extension>
>
> In my group i have two members only.
>
>
>
>
>
>   --
>
> Un cordial saludo / Best regards,
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.com
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 18 00:17:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Oct 2011 13:17:09 -0700
Subject: [Freeswitch-users] Newbie trouble with default config extension
	9198...
In-Reply-To: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
References: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
Message-ID: <CAKzWOxXdV2N_utOYxBV-BFTJdD1qNxZQGVDg69qOdM7oENZxmQ@mail.gmail.com>

Sorry for the late reply... You really should not be using 1.0.6 - it is
quite old. Get on the latest git HEAD. Be sure to mv or cp the
/usr/local/freeswitch directory before reinstalling. The latest version of
FS has an updated default config set and 9198 should be working just fine.

-MC

P.S. - You can look in /usr/local/freeswitch/conf/dialplan/default.xml for
"9198" and see if it's actually there. I suspect that if 9198 is missing
then 9998 will be there, which is a sure sign that your install is *REALLY*
old and is in need of a serious update.

On Wed, Oct 12, 2011 at 11:50 AM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Hi there. :)
> After spending hours trying to register Xlite 4.1 in freeswitch 1.0.6 both
> installed in my Windows 7 pro rig I gave up.
>
> Now I installed Freeswtich 1.0.6 in a centos 5.7 x86 virtual machine with
> the help of freeswitch book.
>
> Xlite is registering fine but now the problem is this:
>
> When  I dial 9198 extension Xlite tells me "Failed to establish call" and
> This log messages are written in freeswitch console:
>
> freeswitch at pbx> 2011-10-12 09:01:17.469489 [NOTICE] switch_channel.c:669
> New Channel sofia/internal/1000 at 10.10.100.35[e3f7697a-74ff-46a6-a3ac-c5b95f58d103]
> 2011-10-12 09:01:17.474982 [INFO] mod_dialplan_xml.c:418 Processing
> 1000->9198 in context default
> 2011-10-12 09:01:17.480554 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/internal/1000 at 10.10.100.35 to enum[9198 at default]
> 2011-10-12 09:01:17.497178 [INFO] switch_core_state_machine.c:142 No Route,
> Aborting
> 2011-10-12 09:01:17.497178 [NOTICE] switch_core_state_machine.c:143 Hangup
> sofia/internal/1000 at 10.10.100.35 [CS_ROUTING] [NO_ROUTE_DESTINATION]
> 2011-10-12 09:01:17.502715 [NOTICE] switch_core_session.c:1182 Session 6
> (sofia/internal/1000 at 10.10.100.35) Ended
> 2011-10-12 09:01:17.504038 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1000 at 10.10.100.35 [CS_DESTROY]
>
>
> It looks that 9198 extension is not defined.
> This is the default configuration I haven't change anything.
>
> If it helps extension 5000 is working.
>
> What am I doing wrong?
>
> Thanks in advance for your  help and have a good day.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 18 00:25:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Oct 2011 13:25:12 -0700
Subject: [Freeswitch-users] conference with pin code and esl
In-Reply-To: <8E20DFE641574516B5C6891BA3181F1B@e1705>
References: <8E20DFE641574516B5C6891BA3181F1B@e1705>
Message-ID: <CAKzWOxUO7vNZezeWhdBn5QmAtgzOH4v8bN-b1kb2oeQRHJ_Rug@mail.gmail.com>

Madovsky,

I think the conf params you sent in created the moderator PIN:

2011-10-14 11:28:15.534653 [*DEBUG*] *switch_ivr*.*c*:577 *sofia*/*internal*
/123456789@*kiki*.*hehe* *Command* *Execute**conference*(*conf_123456789*@*
default*+2222+*flags*{*moderator*,*endconf*})

You say +2222 which sets the PIN and you set +flags{moderator} which means
this caller is the moderator if he/she auth's with the PIN you specified...

-MC

On Fri, Oct 14, 2011 at 8:48 AM, Madovsky <infos at madovsky.org> wrote:

> **
> mmhmm ok
> I didn't see this line in log
>
> 2011-10-14 11:28:27.714649 [DEBUG] mod_conference.c:6102 Moderator PIN
> found!
> but I didn't set any pin code in conference.conf.xml
>
>
>
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, October 14, 2011 11:34 AM
> *Subject:* conference with pin code and esl
>
> I'm running a conference in esl so in the log
>
> *
>
> EXECUTE
> * *sofia <sofia/internal/123456789 at kiki.hehe>*/*internal*/123456789@*kiki*
> .*hehe* *socket*(127.0.0.1:9090 *async* *full*)
> 2011-10-14 11:28:15.534653 [*DEBUG*] *switch_core_session*.*c*:1009 *Send*
> *signal* *sofia <sofia/internal/123456789 at kiki.hehe>*/*internal*/123456789
> @*kiki*.*hehe* [*BREAK*]
> 2011-10-14 11:28:15.534653 [*DEBUG*] *switch_ivr*.*c*:577 *sofia<sofia/internal/123456789 at kiki.hehe>
> */*internal*/123456789@*kiki*.*hehe* *Command* *Execute* *conference*(*
> conf_123456789 <conf_123456789 at default+2222+flags%7Bmoderator,endconf>*@*
> default*+2222+*flags*{*moderator*,*endconf*})
> *EXECUTE* *sofia*/*internal*/123456789@*kiki*.*hehe* *conference*(*
> conf_123456789 <conf_123456789 at default+2222+flags%7Bmoderator,endconf>*@*
> default*+2222+*flags*{*moderator*,*endconf*})
>
> but if the caller don't type any pin code ther conference is created.
>
> this behaviour exist since I updated to this git
>
> FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500)
>
> Must I change anything ?
>
> Thanks
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Oct 18 00:37:43 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Oct 2011 16:37:43 -0400
Subject: [Freeswitch-users] conference with pin code and esl
References: <8E20DFE641574516B5C6891BA3181F1B@e1705>
	<CAKzWOxUO7vNZezeWhdBn5QmAtgzOH4v8bN-b1kb2oeQRHJ_Rug@mail.gmail.com>
Message-ID: <FD6ED18F454249B497CE21B79AA6A44D@e1705>

Hi Michael,

I use also 2222 pin code with flags "waste" but
the result is the same. If the caller send DTMF so hte internal play and get digits work and the voice
says invalid pin number or pin ok, but if the caller (waste or moderator) doesn't type any DTMF so
the caller enters in the conference anyway rather than block the leg and say "invalid pin number"
I'm not sure I'm explaining well...

Thanks

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, October 17, 2011 4:25 PM
  Subject: Re: [Freeswitch-users] conference with pin code and esl


  Madovsky,


  I think the conf params you sent in created the moderator PIN:


  2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789 at kiki.hehe Command Executeconference(conf_123456789 at default+2222+flags{moderator,endconf})

  You say +2222 which sets the PIN and you set +flags{moderator} which means this caller is the moderator if he/she auth's with the PIN you specified...


  -MC


  On Fri, Oct 14, 2011 at 8:48 AM, Madovsky <infos at madovsky.org> wrote:

    mmhmm ok
    I didn't see this line in log

    2011-10-14 11:28:27.714649 [DEBUG] mod_conference.c:6102 Moderator PIN found!

    but I didn't set any pin code in conference.conf.xml



      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, October 14, 2011 11:34 AM
      Subject: conference with pin code and esl


      I'm running a conference in esl so in the log

      EXECUTE

      sofia/internal/123456789 at kiki.hehe socket(127.0.0.1:9090 async full)
      2011-10-14 11:28:15.534653 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/123456789 at kiki.hehe [BREAK]
      2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789 at kiki.hehe Command Execute conference(conf_123456789 at default+2222+flags{moderator,endconf})
      EXECUTE sofia/internal/123456789 at kiki.hehe conference(conf_123456789 at default+2222+flags{moderator,endconf}) 
      but if the caller don't type any pin code ther conference is created.

      this behaviour exist since I updated to this git

      FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500)

      Must I change anything ?

      Thanks



    FreeSWITCH-users mailing list
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From fcintrono at gmail.com  Tue Oct 18 01:55:28 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Mon, 17 Oct 2011 16:55:28 -0500
Subject: [Freeswitch-users] Newbie trouble with default config extension
	9198...
In-Reply-To: <CAKzWOxXdV2N_utOYxBV-BFTJdD1qNxZQGVDg69qOdM7oENZxmQ@mail.gmail.com>
References: <CANtw_ZPMntFrFwUgN_oJuJ5YxsB0YACpZHjw_ZN-b8yUmU3xUg@mail.gmail.com>
	<CAKzWOxXdV2N_utOYxBV-BFTJdD1qNxZQGVDg69qOdM7oENZxmQ@mail.gmail.com>
Message-ID: <CANtw_ZPYGmedQ0xTnPNUJhbfTDs2oG+P2szfn1CswPkjAZNdmQ@mail.gmail.com>

OK, let me upgrade the version and try again.

Thank you very much for your help.

On Mon, Oct 17, 2011 at 3:17 PM, Michael Collins <msc at freeswitch.org> wrote:

> Sorry for the late reply... You really should not be using 1.0.6 - it is
> quite old. Get on the latest git HEAD. Be sure to mv or cp the
> /usr/local/freeswitch directory before reinstalling. The latest version of
> FS has an updated default config set and 9198 should be working just fine.
>
> -MC
>
> P.S. - You can look in /usr/local/freeswitch/conf/dialplan/default.xml for
> "9198" and see if it's actually there. I suspect that if 9198 is missing
> then 9998 will be there, which is a sure sign that your install is *REALLY*
> old and is in need of a serious update.
>
> On Wed, Oct 12, 2011 at 11:50 AM, Francisco Javier Cintr?n Olgu?n <
> fcintrono at gmail.com> wrote:
>
>> Hi there. :)
>> After spending hours trying to register Xlite 4.1 in freeswitch 1.0.6 both
>> installed in my Windows 7 pro rig I gave up.
>>
>> Now I installed Freeswtich 1.0.6 in a centos 5.7 x86 virtual machine with
>> the help of freeswitch book.
>>
>> Xlite is registering fine but now the problem is this:
>>
>> When  I dial 9198 extension Xlite tells me "Failed to establish call" and
>> This log messages are written in freeswitch console:
>>
>> freeswitch at pbx> 2011-10-12 09:01:17.469489 [NOTICE] switch_channel.c:669
>> New Channel sofia/internal/1000 at 10.10.100.35[e3f7697a-74ff-46a6-a3ac-c5b95f58d103]
>> 2011-10-12 09:01:17.474982 [INFO] mod_dialplan_xml.c:418 Processing
>> 1000->9198 in context default
>> 2011-10-12 09:01:17.480554 [NOTICE] switch_ivr.c:1447 Transfer
>> sofia/internal/1000 at 10.10.100.35 to enum[9198 at default]
>> 2011-10-12 09:01:17.497178 [INFO] switch_core_state_machine.c:142 No
>> Route, Aborting
>> 2011-10-12 09:01:17.497178 [NOTICE] switch_core_state_machine.c:143 Hangup
>> sofia/internal/1000 at 10.10.100.35 [CS_ROUTING] [NO_ROUTE_DESTINATION]
>> 2011-10-12 09:01:17.502715 [NOTICE] switch_core_session.c:1182 Session 6
>> (sofia/internal/1000 at 10.10.100.35) Ended
>> 2011-10-12 09:01:17.504038 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/internal/1000 at 10.10.100.35 [CS_DESTROY]
>>
>>
>> It looks that 9198 extension is not defined.
>> This is the default configuration I haven't change anything.
>>
>> If it helps extension 5000 is working.
>>
>> What am I doing wrong?
>>
>> Thanks in advance for your  help and have a good day.
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 18 02:24:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Oct 2011 15:24:33 -0700
Subject: [Freeswitch-users] conference with pin code and esl
In-Reply-To: <FD6ED18F454249B497CE21B79AA6A44D@e1705>
References: <8E20DFE641574516B5C6891BA3181F1B@e1705>
	<CAKzWOxUO7vNZezeWhdBn5QmAtgzOH4v8bN-b1kb2oeQRHJ_Rug@mail.gmail.com>
	<FD6ED18F454249B497CE21B79AA6A44D@e1705>
Message-ID: <CAKzWOxVq+-d+N_aZ_98nkR74JG_26ECp+wZ8RbFyU1LhVb17OA@mail.gmail.com>

Well, I surely did not understand a word you said... If someone can
translate that for me I would appreciate it.

-MC

On Mon, Oct 17, 2011 at 1:37 PM, Madovsky <infos at madovsky.org> wrote:

> **
> Hi Michael,
>
> I use also 2222 pin code with flags "waste" but
> the result is the same. If the caller send DTMF so hte internal play and
> get digits work and the voice
> says invalid pin number or pin ok, but if the caller (waste or moderator)
> doesn't type any DTMF so
> the caller enters in the conference anyway rather than block the leg and
> say "invalid pin number"
> I'm not sure I'm explaining well...
>
> Thanks
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, October 17, 2011 4:25 PM
> *Subject:* Re: [Freeswitch-users] conference with pin code and esl
>
> Madovsky,
>
> I think the conf params you sent in created the moderator PIN:
>
> 2011-10-14 11:28:15.534653 [*DEBUG*] *switch_ivr*.*c*:577 *sofia<sofia/internal/123456789 at kiki.hehe>
> */*internal*/123456789@*kiki*.*hehe* *Command* *Execute**conference*(*
> conf_123456789 <conf_123456789 at default+2222+flags%7Bmoderator,endconf>*@*
> default*+2222+*flags*{*moderator*,*endconf*})
>
> You say +2222 which sets the PIN and you set +flags{moderator} which means
> this caller is the moderator if he/she auth's with the PIN you specified...
>
> -MC
>
> On Fri, Oct 14, 2011 at 8:48 AM, Madovsky <infos at madovsky.org> wrote:
>
>> **
>> mmhmm ok
>> I didn't see this line in log
>>
>> 2011-10-14 11:28:27.714649 [DEBUG] mod_conference.c:6102 Moderator PIN
>> found!
>> but I didn't set any pin code in conference.conf.xml
>>
>>
>>
>>
>> ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Friday, October 14, 2011 11:34 AM
>> *Subject:* conference with pin code and esl
>>
>> I'm running a conference in esl so in the log
>>
>> *
>>
>> EXECUTE
>> **sofia <sofia/internal/123456789 at kiki.hehe>*/*internal*/123456789@*kiki*
>> .*hehe* *socket*(127.0.0.1:9090 *async* *full*)
>> 2011-10-14 11:28:15.534653 [*DEBUG*] *switch_core_session*.*c*:1009 *Send
>> * *signal* *sofia <sofia/internal/123456789 at kiki.hehe>*/*internal*/
>> 123456789@*kiki*.*hehe* [*BREAK*]
>> 2011-10-14 11:28:15.534653 [*DEBUG*] *switch_ivr*.*c*:577 *sofia<sofia/internal/123456789 at kiki.hehe>
>> */*internal*/123456789@*kiki*.*hehe* *Command* *Execute* *conference*(*
>> conf_123456789 <conf_123456789 at default+2222+flags%7Bmoderator,endconf>*@*
>> default*+2222+*flags*{*moderator*,*endconf*})
>> *EXECUTE* *sofia*/*internal*/123456789@*kiki*.*hehe* *conference*(*
>> conf_123456789 <conf_123456789 at default+2222+flags%7Bmoderator,endconf>*@*
>> default*+2222+*flags*{*moderator*,*endconf*})
>>
>> but if the caller don't type any pin code ther conference is created.
>>
>> this behaviour exist since I updated to this git
>>
>> FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500)
>>
>> Must I change anything ?
>>
>> Thanks
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From msc at freeswitch.org  Tue Oct 18 02:28:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Oct 2011 15:28:32 -0700
Subject: [Freeswitch-users] Ring Group with External Number not
 transferring after timeout
In-Reply-To: <4E98E5C5.2090301@exemail.com.au>
References: <4E98E5C5.2090301@exemail.com.au>
Message-ID: <CAKzWOxWzc_16o0cwf-HB5C2mW6X1AKaXZ+2HspcFMRZ3FKuMUQ@mail.gmail.com>

This is definitely a question for Bluebox and/or 2600hz forum/mailing list.
-MC

On Fri, Oct 14, 2011 at 6:45 PM, Ryan How <rhow at exemail.com.au> wrote:

> Hi,
>
> I have a ring group with an external number I've created using bluebox.
> Sorry I hope I am posting this in the right forum...
>
> What happens is, it will ring for 30 seconds, then stop ringing, then
> rather than transfer to voicemail, it will start ringing again. If I
> remove the external number from the ring group then it is ok and works
> as expected. Is bluebox setting this up wrong? Or is there something I
> need to change or add? XML is below... On occasion it seems to work,
> maybe if the external number is busy or something?
>
> It is funny because this is exactly the same behaviour I got when
> setting up the same thing with asterisk. Makes me think there is
> something I'm not getting when putting in external numbers.
>
> I'm using the GIT head of freeswitch (and bluebox) as of yesterday.
>
> Thanks,
>
> Ryan
>
>
> <extension name="main_number_9" continue="true">
> <condition field="destination_number" expression="^3000$">
> <action application="set" bluebox="settingEndBridge"
> data="hangup_after_bridge=true"/>
> <action application="set" bluebox="settingFail"
> data="continue_on_fail=true"/>
> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
> <action application="set" bluebox="settingRing"
> data="ringback=${us-ring}"/>
> <action application="set" bluebox="settingRingXfer"
> data="transfer_ringback=${us-ring}"/>
> <action application="export" bluebox="sipCalleeIdName"
> data="sip_callee_id_name=Ring All"/>
> <action application="export" bluebox="sipCalleeIdNumber"
> data="sip_callee_id_number=3000"/>
> <action application="export" bluebox="settingContinueOnTimeout"
> data="originate_continue_on_timeout=true"/>
> <action application="bridge"
> data="[ignore_early_media=true,leg_timeout=30]user/2001@
> $${location_1}:_:[ignore_early_media=true,leg_timeout=30]user/2002@
> $${location_1}:_:[ignore_early_media=true,leg_timeout=30]sofia/gateway/trunk_1/0447614933"/>
> <action application="answer"/>
> <action application="voicemail" data="default voicemail_1 2001"/>
> <action application="hangup"/>
> </condition>
> </extension>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 18 02:31:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Oct 2011 15:31:21 -0700
Subject: [Freeswitch-users] External number starts ringing again after
 rejecting call until timeout
In-Reply-To: <4E9A6D1D.1010701@exemail.com.au>
References: <4E9A6D1D.1010701@exemail.com.au>
Message-ID: <CAKzWOxWdgddhpMai1vpbUYA4onvjcVMSH=Tt5p9w7DexXuD1Sw@mail.gmail.com>

Is this really a bluebox system? If so you may want to check in on the
2600hz forums/irc/mailing list for help. The configuration scheme used by bb
is very different from the generic FS configs that we're all used to using.

-MC

On Sat, Oct 15, 2011 at 10:35 PM, Ryan How <rhow at exemail.com.au> wrote:

> Hi,
>
> I've got an extension set up to redirect to my mobile. It all works
> fine, except if I reject the call on my mobile it starts ringing again.
> It just keeps going until the timeout and then transfers to voicemail.
> How can I make it go to voicemail if I reject the call on my mobile? I
> tried adding fail_on_single_reject=true but it didn't seem to do
> anything. Sorry for the newbie question. I'm slowly learning :). XML is
> below.
>
> Thanks!
>
> Ryan
>
> <extension name="main_number_11" continue="true">
> <condition field="destination_number" expression="^3101$">
> <action application="set" bluebox="settingEndBridge"
> data="hangup_after_bridge=true"/>
> <action application="set" bluebox="settingFail"
> data="continue_on_fail=true"/>
> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
> <action application="set" bluebox="settingRing"
> data="ringback=${us-ring}"/>
> <action application="set" bluebox="settingRingXfer"
> data="transfer_ringback=${us-ring}"/>
> <action application="set" data="fail_on_single_reject=true"/>
> <action application="export" bluebox="sipCalleeIdName"
> data="sip_callee_id_name=Ryan Work Mobile"/>
> <action application="export" bluebox="sipCalleeIdNumber"
> data="sip_callee_id_number=3101"/>
> <action application="bridge"
> data="{ignore_early_media=true}sofia/gateway/trunk_1/0447614933"/>
> <action application="answer"/>
> <action application="voicemail" data="default voicemail_1 2001"/>
> <action application="hangup"/>
> </condition>
> </extension>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rhow at exemail.com.au  Tue Oct 18 02:57:02 2011
From: rhow at exemail.com.au (Ryan How)
Date: Tue, 18 Oct 2011 06:57:02 +0800
Subject: [Freeswitch-users] External number starts ringing again after
	rejecting call until timeout
In-Reply-To: <CAKzWOxWdgddhpMai1vpbUYA4onvjcVMSH=Tt5p9w7DexXuD1Sw@mail.gmail.com>
References: <4E9A6D1D.1010701@exemail.com.au>
	<CAKzWOxWdgddhpMai1vpbUYA4onvjcVMSH=Tt5p9w7DexXuD1Sw@mail.gmail.com>
Message-ID: <4BE32CA6-8F64-471E-B160-143F9FB4A8E4@exemail.com.au>

Hi, thanks I'll check over there. It mustn't be a freeswitch thing or I'm sure someone would have heard of it before. Thanks! Ryan



On 18/10/2011, at 6:31 AM, Michael Collins <msc at freeswitch.org> wrote:

> Is this really a bluebox system? If so you may want to check in on the 2600hz forums/irc/mailing list for help. The configuration scheme used by bb is very different from the generic FS configs that we're all used to using.
> 
> -MC
> 
> On Sat, Oct 15, 2011 at 10:35 PM, Ryan How <rhow at exemail.com.au> wrote:
> Hi,
> 
> I've got an extension set up to redirect to my mobile. It all works
> fine, except if I reject the call on my mobile it starts ringing again.
> It just keeps going until the timeout and then transfers to voicemail.
> How can I make it go to voicemail if I reject the call on my mobile? I
> tried adding fail_on_single_reject=true but it didn't seem to do
> anything. Sorry for the newbie question. I'm slowly learning :). XML is
> below.
> 
> Thanks!
> 
> Ryan
> 
> <extension name="main_number_11" continue="true">
> <condition field="destination_number" expression="^3101$">
> <action application="set" bluebox="settingEndBridge"
> data="hangup_after_bridge=true"/>
> <action application="set" bluebox="settingFail"
> data="continue_on_fail=true"/>
> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
> <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
> <action application="set" bluebox="settingRingXfer"
> data="transfer_ringback=${us-ring}"/>
> <action application="set" data="fail_on_single_reject=true"/>
> <action application="export" bluebox="sipCalleeIdName"
> data="sip_callee_id_name=Ryan Work Mobile"/>
> <action application="export" bluebox="sipCalleeIdNumber"
> data="sip_callee_id_number=3101"/>
> <action application="bridge"
> data="{ignore_early_media=true}sofia/gateway/trunk_1/0447614933"/>
> <action application="answer"/>
> <action application="voicemail" data="default voicemail_1 2001"/>
> <action application="hangup"/>
> </condition>
> </extension>
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 18 04:02:18 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 18 Oct 2011 02:02:18 +0200
Subject: [Freeswitch-users] X-Headers on BYE- How to save?
Message-ID: <CALNZuw=QXyxmMWQ=5eNK7yK=wBe7gHKJKCnaFKunMqSayv1W8A@mail.gmail.com>

I have a carrier supplying the LRN in the BYE / OK to my BYE.
It SEEMS that when the other party hangs up first, I get a sip_h_X-LRN in my
xml_cdrs, but not when my FS hangs up first.
Presumably this is because their bye happens before the reporting begins,
but my bye starts reporting before the OK to the bye is received.

So, how would I be able to save access this header? A hangup hook of some
sort.. well, that won't have access either. Any ideas?
Thanks,
-Avi
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From infos at madovsky.org  Tue Oct 18 05:59:44 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 Oct 2011 21:59:44 -0400
Subject: [Freeswitch-users] conference with pin code and esl
References: <8E20DFE641574516B5C6891BA3181F1B@e1705><CAKzWOxUO7vNZezeWhdBn5QmAtgzOH4v8bN-b1kb2oeQRHJ_Rug@mail.gmail.com><FD6ED18F454249B497CE21B79AA6A44D@e1705>
	<CAKzWOxVq+-d+N_aZ_98nkR74JG_26ECp+wZ8RbFyU1LhVb17OA@mail.gmail.com>
Message-ID: <FE619B1E9F9144D48432D48BFF3FA30B@e1705>

ok deosn't matter.
thanks anyway

Franck
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, October 17, 2011 6:24 PM
  Subject: Re: [Freeswitch-users] conference with pin code and esl


  Well, I surely did not understand a word you said... If someone can translate that for me I would appreciate it.


  -MC


  On Mon, Oct 17, 2011 at 1:37 PM, Madovsky <infos at madovsky.org> wrote:

    Hi Michael,

    I use also 2222 pin code with flags "waste" but
    the result is the same. If the caller send DTMF so hte internal play and get digits work and the voice
    says invalid pin number or pin ok, but if the caller (waste or moderator) doesn't type any DTMF so
    the caller enters in the conference anyway rather than block the leg and say "invalid pin number"
    I'm not sure I'm explaining well...

    Thanks

      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Monday, October 17, 2011 4:25 PM
      Subject: Re: [Freeswitch-users] conference with pin code and esl


      Madovsky, 


      I think the conf params you sent in created the moderator PIN:


      2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789 at kiki.hehe Command Executeconference(conf_123456789 at default+2222+flags{moderator,endconf})

      You say +2222 which sets the PIN and you set +flags{moderator} which means this caller is the moderator if he/she auth's with the PIN you specified...


      -MC


      On Fri, Oct 14, 2011 at 8:48 AM, Madovsky <infos at madovsky.org> wrote:

        mmhmm ok
        I didn't see this line in log

        2011-10-14 11:28:27.714649 [DEBUG] mod_conference.c:6102 Moderator PIN found!

        but I didn't set any pin code in conference.conf.xml



          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Friday, October 14, 2011 11:34 AM
          Subject: conference with pin code and esl


          I'm running a conference in esl so in the log

          EXECUTE

          sofia/internal/123456789 at kiki.hehe socket(127.0.0.1:9090 async full)
          2011-10-14 11:28:15.534653 [DEBUG] switch_core_session.c:1009 Send signal sofia/internal/123456789 at kiki.hehe [BREAK]
          2011-10-14 11:28:15.534653 [DEBUG] switch_ivr.c:577 sofia/internal/123456789 at kiki.hehe Command Execute conference(conf_123456789 at default+2222+flags{moderator,endconf})
          EXECUTE sofia/internal/123456789 at kiki.hehe conference(conf_123456789 at default+2222+flags{moderator,endconf}) 
          but if the caller don't type any pin code ther conference is created.

          this behaviour exist since I updated to this git

          FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500)

          Must I change anything ?

          Thanks



        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From asilva at wirelessmundi.com  Tue Oct 18 13:29:02 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 18 Oct 2011 11:29:02 +0200
Subject: [Freeswitch-users] Loop XML using call_timeout ..
In-Reply-To: <CAKzWOxULmqimmCvD34f065vus6a8LE=k-cA1ixfAJ3aDhUUyWw@mail.gmail.com>
References: <1318525773.20249.10.camel@marces.madrid.commsmundi.com>
	<CAKzWOxULmqimmCvD34f065vus6a8LE=k-cA1ixfAJ3aDhUUyWw@mail.gmail.com>
Message-ID: <1318930142.21847.25.camel@marces.madrid.commsmundi.com>

Yes, a decide to do a lua script for this task.

What I'm doing is a group call that loops all the members until a global
timeout is reached.

Is the code:

call_timeout = session:getVariable("call_timeout") or 60
timestamp_a = os.time()
-- while the call is not answered
while session:answered()==false do
-- dial the group
session:setVariable("call_timeout", call_timeout)
session:execute("bridge","\${group_call(two at default+F)}")

-- bridge state
if (session:ready() == false) then break end
if (session:hangupState()) then break end

-- check global time, and remove the used amount of time already used in
the call
timestamp_b = os.time()
sec = timestamp_b - timestamp_a
timestamp_a = timestamp_b
call_timeout = call_timeout - sec
if (call_timeout <= 0) then
session:setVariable("originate_disposition", "ALLOTTED_TIMEOUT")
debug.info("GROUP: CALL TIMEOUT!")
break
end
end





On Mon, 2011-10-17 at 13:12 -0700, Michael Collins wrote:

> Did you figure this out yet? Not sure exactly what you're trying to do
> here, but you could use the sched_hangup API and schedule a hangup or
> sched_transfer to schedule a transfer to a specific dialplan extension
> that does your cleanup work.
> 
> 
> <action application="set" data="api_res=${sched_transfer +60 ${uuid}
> handle_timeout XML handlers}"/>
> 
> 
> Then something like this for the dp
> 
> 
> <context name="handlers">
>   <extension name="handle loop timeouts">
> 
>     <condition field="destination_number" expression="^handle_timeout
> $">
>       <action application="log" data="INFO OB call timed out..."/>
>       <action application="foo" data="bar"/> <!-- do something useful
> -->
>       <action application="hangup"
> data="pick_a_hangup_cause_if_you_wish"/>
>     </condition>
>   </extensions>
> </context>
> 
> 
> -MC
> 
> On Thu, Oct 13, 2011 at 10:09 AM, Antonio <asilva at wirelessmundi.com>
> wrote:
> 
>         
>         Hi,
>         
>         Is it possible to loop a bridge until a global call timeout
>         (or other variable) is reached?
>         
>         In the wiki is this example is a fixed number of loops...
>         
>         <extension name="3-way gateway distro"> 
>         
>         <condition field="destination_number" expression="^(.*)$">
>               <action application="set" data="continue_on_fail=true"/>
>               <action application="set" data="hangup_after_bridge=true"/>
>               <action application="bridge" data="sofia/gateway/${distributor(3gw)}/$1" loop="3"/>
>             </condition>
>           </extension>
>         
>         My idea is to call a group sequential, but once it reaches the last member it starts over again.
>         so when calling a group, once the timeout for each user in the group is reach it hangups the call (the normal behavior),
>         it jumps again to the begin of the group until a global timeout is reached.
>         
>         For example:
>         
>         <extension name="group"> 
>         <condition field="destination_number" expression="^(.*)$">
>               <action application="set" data="continue_on_fail=true"/>
>               <action application="set" data="hangup_after_bridge=true"/>
>               <action application="set" data="call_timeout=60"/>
>               <action application="set" data="leg_timeout=10"/>
>               <action application="bridge" data=""${group_call(group1 at default+F)}" loop="3"/>
>             </condition>
>           </extension>
>         
>         In my group i have two members only.
>         
>         
>         
>         
>         
>         -- 
>         
>         Un cordial saludo / Best regards, 
>         
>          _________________________
>         
>         Ant?nio Silva
>         
>         E-mail:asilva at wirelessmundi.com
>         
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bnaylor at sirran.com  Tue Oct 18 15:37:33 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Tue, 18 Oct 2011 12:37:33 +0100
Subject: [Freeswitch-users] Lock down default config
Message-ID: <008e01cc8d8a$542f6fe0$fc8e4fa0$@sirran.com>

Hi All

 

I am going live with a Freeswitch server in a couple of weeks, and it will
be using a public IP.  I am a bit concerned that I may not have tightened
security on the box enough to protect it (as best I can) from attack.

 

I have had a look online for something similar to this but a freeswitch
version, but can't find anything -
http://blogs.digium.com/2009/03/28/sip-security/

 

Can anyone suggest a list of things I should check before I go live with
this server?  I have already changed the default password, removed the
default user accounts/passwords and removed the external SIP account.

 

Any ideas  are greatly appreciated, also any links to websites that I may
not have found yet.

 

Thanks for your help!

 

Kind regards

 

Ben

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From avi at avimarcus.net  Tue Oct 18 15:45:33 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 18 Oct 2011 13:45:33 +0200
Subject: [Freeswitch-users] Lock down default config
In-Reply-To: <008e01cc8d8a$542f6fe0$fc8e4fa0$@sirran.com>
References: <008e01cc8d8a$542f6fe0$fc8e4fa0$@sirran.com>
Message-ID: <CALNZuwkH_rDvdXOMLiBrivf+0iHULyP-6-MgJk=nmY2qo3yu_A@mail.gmail.com>

Here's a start:
-fail2ban on DDOS and excessive attempts to register with the wrong
password. See the wiki page on fail2ban.
-ufw or similar to manage iptables, or iptables (or it's bsd equivalent) to
drop everything but these ports: http://wiki.freeswitch.org/wiki/Firewall
-ssh - if  you use ssh, use keys (with a passphrase?) rather than just plain
passwords. I rarely get sip-attacks, but my logs show several SSH brute
force attempts each day.
-Do you have any other users or services running on the server? if they have
access to config files, or just to using resources that FS needs, it can
compromise the security or just the quality of the call.

-Avi


On Tue, Oct 18, 2011 at 1:37 PM, Ben Naylor <bnaylor at sirran.com> wrote:

> Hi All****
>
> ** **
>
> I am going live with a Freeswitch server in a couple of weeks, and it will
> be using a public IP.  I am a bit concerned that I may not have tightened
> security on the box enough to protect it (as best I can) from attack.****
>
> ** **
>
> I have had a look online for something similar to this but a freeswitch
> version, but can?t find anything -
> http://blogs.digium.com/2009/03/28/sip-security/****
>
> ** **
>
> Can anyone suggest a list of things I should check before I go live with
> this server?  I have already changed the default password, removed the
> default user accounts/passwords and removed the external SIP account.****
>
> ** **
>
> Any ideas  are greatly appreciated, also any links to websites that I may
> not have found yet.****
>
> ** **
>
> Thanks for your help!****
>
> ** **
>
> Kind regards****
>
> ** **
>
> Ben****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 18 18:09:56 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 18 Oct 2011 07:09:56 -0700
Subject: [Freeswitch-users] Lock down default config
In-Reply-To: <CALNZuwkH_rDvdXOMLiBrivf+0iHULyP-6-MgJk=nmY2qo3yu_A@mail.gmail.com>
References: <008e01cc8d8a$542f6fe0$fc8e4fa0$@sirran.com>
	<CALNZuwkH_rDvdXOMLiBrivf+0iHULyP-6-MgJk=nmY2qo3yu_A@mail.gmail.com>
Message-ID: <CAMWiMy8BJ2SrdTBgn=W-Em07yDwrXF5LvpmU1J1-M=0KFDgcUQ@mail.gmail.com>

Remove those default users! At least that's what FusionPBX does (I'm
talking about users from 1000-1019)

On Tue, Oct 18, 2011 at 4:45 AM, Avi Marcus <avi at avimarcus.net> wrote:
> Here's a start:
> -fail2ban on DDOS and excessive attempts to register with the wrong
> password. See the wiki page on fail2ban.
> -ufw or similar to manage iptables, or iptables (or it's bsd equivalent) to
> drop everything but these ports:?http://wiki.freeswitch.org/wiki/Firewall
> -ssh - if ?you use ssh, use keys (with a passphrase?) rather than just plain
> passwords. I rarely get sip-attacks, but my logs show several SSH brute
> force attempts each day.
> -Do you have any other users or services running on the server? if they have
> access to config files, or just to using resources that FS needs, it can
> compromise the security or just the quality of the call.
> -Avi
>
> On Tue, Oct 18, 2011 at 1:37 PM, Ben Naylor <bnaylor at sirran.com> wrote:
>>
>> Hi All
>>
>>
>>
>> I am going live with a Freeswitch server in a couple of weeks, and it will
>> be using a public IP.? I am a bit concerned that I may not have tightened
>> security on the box enough to protect it (as best I can) from attack.
>>
>>
>>
>> I have had a look online for something similar to this but a freeswitch
>> version, but can?t find anything -
>> http://blogs.digium.com/2009/03/28/sip-security/
>>
>>
>>
>> Can anyone suggest a list of things I should check before I go live with
>> this server?? I have already changed the default password, removed the
>> default user accounts/passwords and removed the external SIP account.
>>
>>
>>
>> Any ideas? are greatly appreciated, also any links to websites that I may
>> not have found yet.
>>
>>
>>
>> Thanks for your help!
>>
>>
>>
>> Kind regards
>>
>>
>>
>> Ben
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bnaylor at sirran.com  Tue Oct 18 18:27:24 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Tue, 18 Oct 2011 15:27:24 +0100
Subject: [Freeswitch-users] Lock down default config
In-Reply-To: <CALNZuwkH_rDvdXOMLiBrivf+0iHULyP-6-MgJk=nmY2qo3yu_A@mail.gmail.com>
References: <008e01cc8d8a$542f6fe0$fc8e4fa0$@sirran.com>
	<CALNZuwkH_rDvdXOMLiBrivf+0iHULyP-6-MgJk=nmY2qo3yu_A@mail.gmail.com>
Message-ID: <00db01cc8da2$0e8ad660$2ba08320$@sirran.com>

Hi Avi

 

Thanks for the info, I will certainly look into using fail2ban for failed
SIP attempts.

For the other items I should be covered ok.  Iptables is already restricted
to what I will be using on the server, and I am using software called
'denyhosts' which blocks hosts based on failed SSH login attempts.

 

No other services run on the server which could compromise the system, so
hopefully I should be fairly safe!

 

Cheers

 

Ben

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: 18 October 2011 12:46
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lock down default config

 

Here's a start:

-fail2ban on DDOS and excessive attempts to register with the wrong
password. See the wiki page on fail2ban.

-ufw or similar to manage iptables, or iptables (or it's bsd equivalent) to
drop everything but these ports: http://wiki.freeswitch.org/wiki/Firewall

-ssh - if  you use ssh, use keys (with a passphrase?) rather than just plain
passwords. I rarely get sip-attacks, but my logs show several SSH brute
force attempts each day.

-Do you have any other users or services running on the server? if they have
access to config files, or just to using resources that FS needs, it can
compromise the security or just the quality of the call.




-Avi

 

On Tue, Oct 18, 2011 at 1:37 PM, Ben Naylor <bnaylor at sirran.com> wrote:

Hi All

 

I am going live with a Freeswitch server in a couple of weeks, and it will
be using a public IP.  I am a bit concerned that I may not have tightened
security on the box enough to protect it (as best I can) from attack.

 

I have had a look online for something similar to this but a freeswitch
version, but can't find anything -
http://blogs.digium.com/2009/03/28/sip-security/

 

Can anyone suggest a list of things I should check before I go live with
this server?  I have already changed the default password, removed the
default user accounts/passwords and removed the external SIP account.

 

Any ideas  are greatly appreciated, also any links to websites that I may
not have found yet.

 

Thanks for your help!

 

Kind regards

 

Ben



FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Tue Oct 18 19:00:22 2011
From: vipkilla at gmail.com (vip killa)
Date: Tue, 18 Oct 2011 11:00:22 -0400
Subject: [Freeswitch-users] Kamailio Dispatcher fails in front of
 FreeSwitch, help needed.
In-Reply-To: <CAGH8SeYHcxnZr0kUU0snnXe_pMx4=zCy1oktk7nDw_q+igHazg@mail.gmail.com>
References: <CAGH8SeayFuOu5xoOtzhkQknQ31QamRVUAZUdd1Gz+Vtoju_Jcg@mail.gmail.com>
	<CAGH8SeYHcxnZr0kUU0snnXe_pMx4=zCy1oktk7nDw_q+igHazg@mail.gmail.com>
Message-ID: <CAC-LwPObK0jMecq_Qq-YF0SsQXEJnE6ELEsnc9nD=M+8-BDTnA@mail.gmail.com>

can you pb the /usr/local/etc/kamailio/dispatcher.list file?

2011/10/14 Henrik Aagaard S?rensen <henrikaagaardsorensen at gmail.com>

> When connecting 2 devices directly to FreeSwitch everything works. When
> connecting to Kamailio (as dispatcher), some calls fails with "Timeout" and
> some fails with "Too many loops".
>
> This is my kamailio.cfg: http://pastebin.com/KEuZmikh
>
> When connected directly to FreeSwitch where everything works, this is how
> my clients are registrered in FreeSwitch: http://pastebin.com/prkjkb7g
>
> When connected to Kamailio where nothing works, this is how my clients are
> registrered in FreeSwitch: http://pastebin.com/b5x6NZyv
>
> 215.197.145.132 (sip.my-domain.com) is my Kamailio.
> pbx.my-domain.com is my FreeSwitch.
> 96.224.14.164 is my IP at home, where my 2 devices are from.
>
> Can anyone help me out? I'm kinda lost at this point. And I don't know if
> the error is on FreeSwitch or Kamailio (or both).
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cinthia721 at gmail.com  Tue Oct 18 04:37:39 2011
From: cinthia721 at gmail.com (Cindy Leung)
Date: Mon, 17 Oct 2011 20:37:39 -0400
Subject: [Freeswitch-users] Mismatched video port number in bridge after
	answer
Message-ID: <027A7B90-8145-4FCC-83EA-E56C681DFDDD@gmail.com>

Hello all,

I'm have 1.0.head (git-e463321 2011-10-05 19-13-21 -0300) installed and running.

I am able to get video both ways if I send the call straight to bridge, like this.
  <extension name="name1">
    <condition field="destination_number" expression="^2015">
        <action application="set" data="bypass_media=true"/>
        <action application="bridge" data="sofia/${use_profile}/2001:2001 at dst.proxy.com"/>
    </condition>
  </extension>

But I'm not getting video in either direction (audio is fine) if I want to get digits from the caller (i.e, a number that is going to be the final destination number.)
  <extension name="name2">
    <condition field="destination_number" expression="^(2014)$">
        <action application="answer"/>
        <action application="sleep" data="1500"/>
        <action application="spandsp_start_dtmf" />
        <action application="read" data="4 5 ivr/Welcome.wav conf_id 10000 #"/>
        <action application="spandsp_stop_dtmf"/>
        <action application="set" data="bypass_media=true"/>
        <action application="bridge" data="sofia/${use_profile}/${conf_id}@dst.proxy.com"/>
    </condition>
  </extension>


Here's the tshark capture.  I'm sorry if this is a repeat.  I tried to search the mailing list but I couldn't find anything.
Any suggestion is welcome.  TIA!!

1  <caller's IP> -> <freeswitch IP> SIP/SDP Request: INVITE sip:2014 at my.freeswitch.com, with session description
	c= contains caller's IP
	m=audio 26646 RTP/AVP 0 8 18 101
	m=video 30114 RTP/AVP 123 124

2  <freeswitch IP> -> <caller's IP> SIP Status: 100 Trying

3  <freeswitch IP> -> <caller's IP> SIP/SDP Status: 200 OK, with session description
4  <freeswitch IP> -> <caller's IP> SIP/SDP Status: 200 OK, with session description
	c= contains FS's IP
	m=audio 21098 RTP/AVP 0 101
	m=video 28620 RTP/AVP 123

5  <caller's IP> -> <freeswitch IP> SIP Request: ACK sip:2014@<freeswitch IP>:5060;transport=tcp
6  <caller's IP> -> <freeswitch IP> SIP Request: ACK sip:2014@<freeswitch IP>:5060;transport=tcp



7  <freeswitch IP> -> <dst proxy ip> SIP/SDP Request: INVITE sip:2001 at dst.proxy.com, with session description
	c= contains FS's IP
	m=audio 21402 RTP/AVP 0 8 101 13
	m=video 22854 RTP/AVP 98

8  <dst proxy ip> -> <freeswitch IP> SIP Status: 100 Giving a try
9  <dst proxy ip> -> <freeswitch IP> SIP Status: 180 Ringing
10 <dst proxy ip> -> <freeswitch IP> SIP/SDP Status: 200 OK, with session description
	c= contains dst proxy's IP
	m=audio 29462 RTP/AVP 0 8 101
	m=video 26506 RTP/AVP 98

11 <freeswitch IP> -> <dst proxy ip> SIP Request: ACK sip:10.10.0.12:5060



12 <freeswitch IP> -> <dst proxy ip> SIP/SDP Request: INVITE sip:10.10.0.12:5060, in-dialog, with session description
	c= contains the caller's IP
	m=audio 26646 RTP/AVP 0 8 101
	m=video 22854 RTP/AVP 98

13 <dst proxy ip> -> <freeswitch IP> SIP Status: 100 Giving a try
14 <dst proxy ip> -> <freeswitch IP> SIP/SDP Status: 200 OK, with session description
	c= contains dst proxy's IP
	m=audio 29462 RTP/AVP 0 8 101
	m=video 26506 RTP/AVP 98

15 <freeswitch IP> -> <dst proxy ip> SIP Request: ACK sip:<some ip>:5060



16 <freeswitch IP> -> <caller's IP> SIP/SDP Request: INVITE sip:2060@<some ip>:40731;transport=TCP, in-dialog, with session description
	c= contains dst proxy's IP
	m=audio 29462 RTP/AVP 0 8 101
	m=video 28620 RTP/AVP 123

17 <caller's IP> -> <freeswitch IP> SIP Status: 100 Giving a try
18 <caller's IP> -> <freeswitch IP> SIP/SDP Status: 200 OK, with session description
	c= contains the caller's IP
	m=audio 26646 RTP/AVP 0 8 101
	m=video 30114 RTP/AVP 123

19 <freeswitch IP> -> <caller's IP> SIP Request: ACK sip:2060@<some ip>:40731;transport=TCP



20 <caller's IP> -> <freeswitch IP> SIP Request: BYE sip:2014@<freeswitch IP>:5060;transport=tcp
21 <freeswitch IP> -> <caller's IP> SIP Status: 200 OK
22 <freeswitch IP> -> <dst proxy ip> SIP Request: BYE sip:<some ip>:5060
23 <dst proxy ip> -> <freeswitch IP> SIP Status: 200 OK




Cindy

From tomp at tomp.co.uk  Tue Oct 18 14:34:01 2011
From: tomp at tomp.co.uk (Tom Parrott)
Date: Tue, 18 Oct 2011 11:34:01 +0100
Subject: [Freeswitch-users] audo sync issues with record_session to mp3
Message-ID: <36bc584d980ce80fe6a6f6e7d7383db9.squirrel@my.tomp.co.uk>

Hi,

We are running a cluster of 3 SIP servers all running Freeswitch.

Each call is recorded using the record_session application to record to
mp3 files using mod_shout.

Longer calls, after about 10 minutes start to introduce sync issues
between the A-leg and the B-leg.

We are running record_session on the A-leg, and it seems to get ahead of
the B-leg.

For example the caller on the A-leg will be heard to answer a question
whilst the person on the B-leg is asking it.

I am aware that recording to .mp3 can use additional resources, but these
are quad core servers with 4GB of ram and they are only doing around 10
concurrent calls each.

Additionally I have seen others experiencing these issues with Wav files too:
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060107.html

I am using the git head version from Tue 11 Oct 2011.

I have also tried turning off enable_file_write_buffering on the A-leg
however this makes no difference either.

Does anyone have any suggestions?

Thanks



From manish.khilwani at clariontechnologies.co.in  Tue Oct 18 14:54:53 2011
From: manish.khilwani at clariontechnologies.co.in (Manish K - Clarion, India)
Date: Tue, 18 Oct 2011 16:24:53 +0530
Subject: [Freeswitch-users] custom events
Message-ID: <7DBF30F35AA24C8C8E0DDB323D831791@ManishKhilwani>

Hello Ron, 

I'm trying to register SIP via sofia::register event. 

can you help me?? 

below script showing error(Content-Type: command/reply Reply-Text: -ERR no keywords supplied 
) in telnet 

Event-Subclass: sofia::register 
Event-Name: CUSTOM 
Core-UUID: 5263c34d-8ac9-46dc-8984-64c9571316a4 
FreeSWITCH-Hostname: localhost.localdomain 
FreeSWITCH-IPv4: 192.168.1.167 
FreeSWITCH-IPv6: ::1 
Event-Date-Local: 2011-10-18 15:00:00 
Event-Date-GMT: Tue,18 October 2011 03:00:00 GMT 
Event-Date-Timestamp: 1245190546126571 
Event-Calling-File: sofia_reg.c 
Event-Calling-Function: sofia_reg_handle_register 
Event-Calling-Line-Number: 1113 
profile-name: internal 
from-user: 420 
from-host: 192.168.1.167 
presence-hosts: 192.168.1.167 
contact: sip:420 at 192.168.1.118:5060;fs_nat=yes;fs_path=sip:420 at 192.168.1.118:5060 
call-id: 002D61B2-5F3A-DD11-BF4B-00132019B750 at 192.168.1.118 
rpid: unknown 
statusd: Registered(UDP-NAT) 
expires: 900 
to-user: 420 
to-host: 192.168.1.167 
network-ip: 192.168.1.118 
network-port: 5060 
username: Rana 
realm: 192.168.1.167 
user-agent: vtiger 

Please help 

      Thanks 
      Manish Khilwani 
      Software Engineer

      manish.khilwani at clariontechnologies.co.in  
      Clarion Technologies
      SEI CMMI Level 3

      803, GNFC Info Tower, 
      Sarkhej-Gandhinagar Highway,
      Bodakdev,
      Ahmedabad - 380054, 
      Gujarat, India. 
      Mobile : +91 9099923079 
      www.clariontechnologies.co.in 
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From Giuseppe.Grillo at italtel.it  Tue Oct 18 15:58:20 2011
From: Giuseppe.Grillo at italtel.it (Grillo Giuseppe)
Date: Tue, 18 Oct 2011 13:58:20 +0200
Subject: [Freeswitch-users] Sip register with contact reg-id and
	+sip.instance
Message-ID: <47F792F8B9380249AD789203F4C0CF10049C4F43@BESTWO.corp.dom>

Hi all,

 

I have the need that the "Contact" in SIP register is formatted as
follows including reg-id and +sip.instance.

How do I configure freeswitch to send the "Contact" so ?

 

REGISTER sips:10.10.2.110 SIP/2.0

From: sips:1000 at 10.10.2.110;tag=-5159386d4396-8_F192.168.178.29

To: sips:1000 at 10.10.2.110

Call-ID: 1_909-5f0974b1386d4ca0_R at 192.168.178.29

CSeq: 1 REGISTER

Via: SIP/2.0/TLS
192.168.178.29:5061;branch=z9hG4bK1_90a4615b657386d4ca8_R

Content-Length: 0

Max-Forwards: 70

Contact:
<sips:1000 at 192.168.178.29:5061;avaya-sc-enabled;transport=tls>;q=1;expir
es=900;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00040de
c6f47>"

Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE

User-Agent: Avaya one-X Deskphone

Supported: replaces

 

Thanks for your help!

 

Kind regards

 

Giuseppe

 


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From fcintrono at gmail.com  Wed Oct 19 00:32:46 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Tue, 18 Oct 2011 15:32:46 -0500
Subject: [Freeswitch-users] Make freeswitch use spanish sounds...
Message-ID: <CANtw_ZOVVHN=xD-yHafyrYaMx5+snSfMr3RjKxUS43DAY57V-w@mail.gmail.com>

I downloaded this file which is a set of spanish sounds for freeswitch:

http://files.freeswitch.org/freeswitch-sounds-es-mx-maria-44100.tar.gz

As you see It only has sounds for the 44100 frecuency.
To generate files for the 8000, 16000 and 32000 frecuencies,  I untarred the
file inside freeswitch/sounds directory.

After I untarred the file, there was a directory called es/mx/maria which
has all this subdirectories:

ascii
currency
ivr
phonetic-ascii
voicemail
conference
digits
misc
time
zrtp

For each of these subdirectories I made this:

cd "one of the directories inside sounds/es/mx/maria "
mkdir 8000
mkdir 16000
mkdir 16000

cd 44100
for a in *.wav;
do
     /usr/bin/ffmpeg -i "$a" -ar 8000 -ac 1 -ab 128 "../8000/"$a;
     /usr/bin/ffmpeg -i "$a" -ar 16000 "../16000/"$a;
     /usr/bin/ffmpeg -i "$a" -ar 32000 "../32000/"$a;
done

Now I have a complete set of spanish sounds for the 8000, 16000, 32000 y
44100 frecuencies

?What would I need to configure in freeswitch to use these spanish sounds
instead of the english ones?

Thanks in advance for your kind help.
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From msc at freeswitch.org  Wed Oct 19 00:57:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Oct 2011 13:57:22 -0700
Subject: [Freeswitch-users] Make freeswitch use spanish sounds...
In-Reply-To: <CANtw_ZOVVHN=xD-yHafyrYaMx5+snSfMr3RjKxUS43DAY57V-w@mail.gmail.com>
References: <CANtw_ZOVVHN=xD-yHafyrYaMx5+snSfMr3RjKxUS43DAY57V-w@mail.gmail.com>
Message-ID: <CAKzWOxVzutgu=mPz7j1fEU1kQ_KELmZwq9-BG7uen84j0Bn28A@mail.gmail.com>

Well, be sure to build mod_say_es. From your FS source tree:

make mod_say_es-install

Unfortunately it looks like the installer does not create a conf/lang/es/
directory for you, so you will need to create one yourself. Copy the
conf/lang/en/* directory structure and files into conf/lang/es/*

After that go into conf/lang/es and rename en.xml to es.xml. Open that file
and look for this line:
<language name="en" say-module="en"
sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="cepstral"
tts-voice="callie">

Change it to:
<language name="es" say-module="es"
sound-prefix="$${sounds_dir}/es/mx/maria" tts-engine="cepstral"
tts-voice="callie">

Be sure to put your maria sounds in the proper directory structure, just
like the sounds/en/us/callie files are.

You will also need to add a line to freeswitch.xml like the other langs:
<X-PRE-PROCESS cmd="include" data="lang/es/*.xml"/>

If you want to load mod_say_es automatically then add it to
conf/autoload_configs/modules.conf, otherwise restart FreeSWITCH and do
"load mod_say_es" and hopefully nothing blows up! :)

Also, you may want to set the default language on calls that you know are
Spanish. Check out this wiki page for assistance. (
http://wiki.freeswitch.org/wiki/Channel_Variables#default_language)

-MC

On Tue, Oct 18, 2011 at 1:32 PM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> I downloaded this file which is a set of spanish sounds for freeswitch:
>
> http://files.freeswitch.org/freeswitch-sounds-es-mx-maria-44100.tar.gz
>
> As you see It only has sounds for the 44100 frecuency.
> To generate files for the 8000, 16000 and 32000 frecuencies,  I untarred
> the file inside freeswitch/sounds directory.
>
> After I untarred the file, there was a directory called es/mx/maria which
> has all this subdirectories:
>
> ascii
> currency
> ivr
> phonetic-ascii
> voicemail
>  conference
> digits
> misc
> time
> zrtp
>
> For each of these subdirectories I made this:
>
> cd "one of the directories inside sounds/es/mx/maria "
> mkdir 8000
> mkdir 16000
> mkdir 16000
>
> cd 44100
> for a in *.wav;
> do
>      /usr/bin/ffmpeg -i "$a" -ar 8000 -ac 1 -ab 128 "../8000/"$a;
>      /usr/bin/ffmpeg -i "$a" -ar 16000 "../16000/"$a;
>      /usr/bin/ffmpeg -i "$a" -ar 32000 "../32000/"$a;
> done
>
> Now I have a complete set of spanish sounds for the 8000, 16000, 32000 y
> 44100 frecuencies
>
> ?What would I need to configure in freeswitch to use these spanish sounds
> instead of the english ones?
>
> Thanks in advance for your kind help.
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From henrikaagaardsorensen at gmail.com  Wed Oct 19 01:01:18 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Tue, 18 Oct 2011 23:01:18 +0200
Subject: [Freeswitch-users] Kamailio Dispatcher fails in front of
 FreeSwitch, help needed.
In-Reply-To: <CAGH8Seah2D_LKVgwcq1yzRPTzkzHUwaD7MQ69OsC32SNdW2mSQ@mail.gmail.com>
References: <CAGH8SeayFuOu5xoOtzhkQknQ31QamRVUAZUdd1Gz+Vtoju_Jcg@mail.gmail.com>
	<CAGH8SeYHcxnZr0kUU0snnXe_pMx4=zCy1oktk7nDw_q+igHazg@mail.gmail.com>
	<CAC-LwPObK0jMecq_Qq-YF0SsQXEJnE6ELEsnc9nD=M+8-BDTnA@mail.gmail.com>
	<CAGH8Seah2D_LKVgwcq1yzRPTzkzHUwaD7MQ69OsC32SNdW2mSQ@mail.gmail.com>
Message-ID: <CAGH8SeZ0fQE2kp94zxBLAKW=fJZ9aVbe23w8URkR6XL=9s14Og@mail.gmail.com>

My dispatcher.list only contains:

1 pbx.my-domain.com:5060
On Oct 18, 2011 11:04 AM, "vip killa" <vipkilla at gmail.com> wrote:
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From larclap at yahoo.com  Wed Oct 19 03:00:50 2011
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 18 Oct 2011 16:00:50 -0700
Subject: [Freeswitch-users] Follow me - 2nd number rings but silence on both
	ends
Message-ID: <00e301cc8de9$c7cdee40$5769cac0$@yahoo.com>

http://pastebin.com/9PR0yCf8

I setup a follow-me scenario in the dialplan above. It does ring through to
the second number, but most times there is silence on both ends. If it
matters, the second device is a Blackberry.

Any ideas on what might correct this?

Thanks, Lars



From admharris at gmx.com  Wed Oct 19 03:06:47 2011
From: admharris at gmx.com (adam harris)
Date: Wed, 19 Oct 2011 01:06:47 +0200
Subject: [Freeswitch-users] esl_event_get_header error
Message-ID: <20111018230647.199160@gmx.com>


I am trying to check if event received was CHANNEL_HANUP 


I am doing somthing like this

 const char *last_event = esl_event_get_header(call_info->global_handle->last_event,"Event-Name");

 if (strcmp(last_event,"CHANNEL_HANGUP")) {
 exit(0);
 }

However an error occurs

src/esl_event.c:244: esl_event_get_header_ptr: Assertion `event' failed.


From eman at chabotel.com  Wed Oct 19 05:11:53 2011
From: eman at chabotel.com (eman)
Date: Tue, 18 Oct 2011 21:11:53 -0400
Subject: [Freeswitch-users] Ring Group with External Number not
 transferring after timeout
In-Reply-To: <CAKzWOxWzc_16o0cwf-HB5C2mW6X1AKaXZ+2HspcFMRZ3FKuMUQ@mail.gmail.com>
References: <4E98E5C5.2090301@exemail.com.au>
	<CAKzWOxWzc_16o0cwf-HB5C2mW6X1AKaXZ+2HspcFMRZ3FKuMUQ@mail.gmail.com>
Message-ID: <CAEzwxGSf8W3trp3UjFEkJt7zDVooGxCS6nSyAuPgd1qukrgHnA@mail.gmail.com>

I have run into the same problem.  I had a ring group with multiple
extensions separated by a ',' and multiple registrations enabled so I
switched it to ':_:' as suggested by the Ring Group Wiki page and now it
never goes to voicemail.

Any help would be appreciated
Eric
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From gcd at i.ph  Wed Oct 19 06:32:08 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 19 Oct 2011 10:32:08 +0800
Subject: [Freeswitch-users] GSMopen builds
Message-ID: <CAHKqMb49rfLYmdM+19M0K21uVpzmpSE2RpHALf+ivo_9KRkj6Q@mail.gmail.com>

hello,

referring to the Wiki, there 4 different builds for voice and SMS support -
w/ (or w/o) gsmlib, alsa vs portaudio, etc. i want to test using (A) a
mobile phone + USB cable + audio cable and later on, (B) w/ an embedded GSM
gateway e.g. Mobigater.  will one type of build work in both A and B setups?
is there any important item to watch?

tks,
nandy
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From rhow at exemail.com.au  Wed Oct 19 08:03:00 2011
From: rhow at exemail.com.au (Ryan How)
Date: Wed, 19 Oct 2011 12:03:00 +0800
Subject: [Freeswitch-users] Ring Group with External Number not
 transferring after timeout
In-Reply-To: <CAEzwxGSf8W3trp3UjFEkJt7zDVooGxCS6nSyAuPgd1qukrgHnA@mail.gmail.com>
References: <4E98E5C5.2090301@exemail.com.au>
	<CAKzWOxWzc_16o0cwf-HB5C2mW6X1AKaXZ+2HspcFMRZ3FKuMUQ@mail.gmail.com>
	<CAEzwxGSf8W3trp3UjFEkJt7zDVooGxCS6nSyAuPgd1qukrgHnA@mail.gmail.com>
Message-ID: <4E9E4BF4.60302@exemail.com.au>

Hi,

I played around with contexts and it seems to be working properly now. 
So I don't really actually know what I did, contexts still confuse me... 
The mobile still keeps ringing over and over even if I reject the call, 
but at least after 30 seconds it goes to voicemail...  Now to just get 
it so it doesn't ring the mobile over and over when I reject it. At 
least you can just click a mobile onto silent :)

Ryan


On 19/10/2011 9:11 AM, eman wrote:
> I have run into the same problem.  I had a ring group with multiple 
> extensions separated by a ',' and multiple registrations enabled so I 
> switched it to ':_:' as suggested by the Ring Group Wiki page and now 
> it never goes to voicemail.
>
> Any help would be appreciated
> Eric
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From trtr at 2ride.com  Wed Oct 19 08:46:11 2011
From: trtr at 2ride.com (T.R. Missner)
Date: Wed, 19 Oct 2011 00:46:11 -0400
Subject: [Freeswitch-users] Kamailio Dispatcher fails in front of
 FreeSwitch, help needed.
In-Reply-To: <CAGH8SeZ0fQE2kp94zxBLAKW=fJZ9aVbe23w8URkR6XL=9s14Og@mail.gmail.com>
References: <CAGH8SeayFuOu5xoOtzhkQknQ31QamRVUAZUdd1Gz+Vtoju_Jcg@mail.gmail.com>
	<CAGH8SeYHcxnZr0kUU0snnXe_pMx4=zCy1oktk7nDw_q+igHazg@mail.gmail.com>
	<CAC-LwPObK0jMecq_Qq-YF0SsQXEJnE6ELEsnc9nD=M+8-BDTnA@mail.gmail.com>
	<CAGH8Seah2D_LKVgwcq1yzRPTzkzHUwaD7MQ69OsC32SNdW2mSQ@mail.gmail.com>
	<CAGH8SeZ0fQE2kp94zxBLAKW=fJZ9aVbe23w8URkR6XL=9s14Og@mail.gmail.com>
Message-ID: <3DBA3588465242FDA9E10DE298702B67@2ride.com>

 try sip:pbx.my-domain.com:5060 (http://pbx.my-domain.com:5060)  

--  
T.R. Missner
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

On Tuesday, October 18, 2011 at 5:01 PM, Henrik Aagaard S?rensen wrote:

> My dispatcher.list only contains:
> 1 pbx.my-domain.com:5060 (http://pbx.my-domain.com:5060)
> On Oct 18, 2011 11:04 AM, "vip killa" <vipkilla at gmail.com (mailto:vipkilla at gmail.com)> wrote:
>  
> 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 william at xofap.com  Wed Oct 19 13:25:09 2011
From: william at xofap.com (William Alianto)
Date: Wed, 19 Oct 2011 16:25:09 +0700
Subject: [Freeswitch-users] mod_freetdm incoming dialplan
Message-ID: <4E9E9775.2050006@xofap.com>

Hi,

Has anybody got a sample of dialplan for incoming from freetdm?

I've been working to get the incoming from my A400P card working, but 
still fail to do so.

Following is my current incoming dialplan :

<extension name="analog" >
<condition field="context" expression="public"/>
<condition field="destination_number" expression="^(\d+)$"/>
<condition field="chan_name" expression="FreeTDM/1/[1-4]">
<action application="set" data="domain_name=$${domain}"/>
<action application="set" data="transfer_ringback=$${us-ring}"/>
<action application="answer"/>
<action application="transfer" data="800 XML default"/>
</condition>
</extension>



From luixsansan at hotmail.com  Wed Oct 19 14:37:40 2011
From: luixsansan at hotmail.com (luixsansan at hotmail.com)
Date: Wed, 19 Oct 2011 10:37:40 +0000
Subject: [Freeswitch-users] Hi freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP2003B9E927D9781F49D66D9C7EA0@phx.gbl>

hello freeswitch-users at lists.freeswitch.org are you doing good? this just jumped out at me http://t.co/2vmP0XHP


From covici at ccs.covici.com  Wed Oct 19 17:16:51 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 19 Oct 2011 09:16:51 -0400
Subject: [Freeswitch-users] mod_freetdm incoming dialplan
In-Reply-To: <4E9E9775.2050006@xofap.com>
References: <4E9E9775.2050006@xofap.com>
Message-ID: <10432.1319030211@ccs.covici.com>

I would just use the freetdm condition and see if that works.

William Alianto <william at xofap.com> wrote:

> Hi,
> 
> Has anybody got a sample of dialplan for incoming from freetdm?
> 
> I've been working to get the incoming from my A400P card working, but 
> still fail to do so.
> 
> Following is my current incoming dialplan :
> 
> <extension name="analog" >
> <condition field="context" expression="public"/>
> <condition field="destination_number" expression="^(\d+)$"/>
> <condition field="chan_name" expression="FreeTDM/1/[1-4]">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="set" data="transfer_ringback=$${us-ring}"/>
> <action application="answer"/>
> <action application="transfer" data="800 XML default"/>
> </condition>
> </extension>
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 henrikaagaardsorensen at gmail.com  Wed Oct 19 17:50:50 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Wed, 19 Oct 2011 15:50:50 +0200
Subject: [Freeswitch-users] Kamailio Dispatcher fails in front of
 FreeSwitch, help needed.
In-Reply-To: <3DBA3588465242FDA9E10DE298702B67@2ride.com>
References: <CAGH8SeayFuOu5xoOtzhkQknQ31QamRVUAZUdd1Gz+Vtoju_Jcg@mail.gmail.com>
	<CAGH8SeYHcxnZr0kUU0snnXe_pMx4=zCy1oktk7nDw_q+igHazg@mail.gmail.com>
	<CAC-LwPObK0jMecq_Qq-YF0SsQXEJnE6ELEsnc9nD=M+8-BDTnA@mail.gmail.com>
	<CAGH8Seah2D_LKVgwcq1yzRPTzkzHUwaD7MQ69OsC32SNdW2mSQ@mail.gmail.com>
	<CAGH8SeZ0fQE2kp94zxBLAKW=fJZ9aVbe23w8URkR6XL=9s14Og@mail.gmail.com>
	<3DBA3588465242FDA9E10DE298702B67@2ride.com>
Message-ID: <CAGH8SeaXzuF2jgMwbGuQfu9L2qyYDPZ+woWRLm4dqYkQ+AjCYw@mail.gmail.com>

So sorry, it is actually sip:pbx.my-domain.com:5060 which is in my
dispatcher.list.

On Wed, Oct 19, 2011 at 6:46 AM, T.R. Missner <trtr at 2ride.com> wrote:

>  try sip:pbx.my-domain.com:5060
>
> --
> T.R. Missner
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Tuesday, October 18, 2011 at 5:01 PM, Henrik Aagaard S?rensen wrote:
>
> My dispatcher.list only contains:
>
> 1 pbx.my-domain.com:5060
> On Oct 18, 2011 11:04 AM, "vip killa" <vipkilla at gmail.com> wrote:
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hexade at hotmail.com  Wed Oct 19 11:06:27 2011
From: hexade at hotmail.com (Adelia C.)
Date: Wed, 19 Oct 2011 03:06:27 -0400
Subject: [Freeswitch-users] Running FreeSwitch as Windows Service
Message-ID: <BLU166-W196CD7DE2FB3227327E813DAEA0@phx.gbl>


Hello,

I am building FreeSwitch through VS 2010 (not VC++ Express) and I generate FreeSwitchConsole.exe as a console application. No FreeSwitch.exe is generated.

FreeSwitchConsole runs fine as an application but I need to run the SW as a service.

I installed Freeswitch as a local service that calls FreeSwitchConsole.exe, and this is the output of "sc qc FreeSWITCH":

C:\FreeSWITCH>sc qc FreeSWITCH
[SC] QueryServiceConfig SUCCESS

SERVICE_NAME: FreeSWITCH
        TYPE               : 10  WIN32_OWN_PROCESS
        START_TYPE         : 2   AUTO_START
        ERROR_CONTROL      : 1   NORMAL
        BINARY_PATH_NAME   : C:\FreeSWITCH\FreeSwitchConsole.exe
        LOAD_ORDER_GROUP   :
        TAG                : 0
        DISPLAY_NAME       : FreeSWITCH
        DEPENDENCIES       :
        SERVICE_START_NAME : LocalSystem


Trying to start FreeSwitch as a service results in  Error 1053 Timeout - Event Viewer is : Timeout (30000 milliseconds) waiting for the FreeSWITCH service to connect.

Or: 
C:\FreeSWITCH>net start freeswitch
The service is not responding to the control function.

Is it because I am trying to start a console application as a service? 

Do I need to have a FreeSwitch.exe and if so, how do I build it? Do I need to create a Windows Service wrapper project?

I tried using winserv (supposed to start console apps as service), with same result:
C:\FreeSWITCH>..\tools\winserv\winserv start FreeSwitch
ERROR 1053: The service did not respond to the start or control request in a timely fashion.

What am I missing?

Thank you.
A.C.
 		 	   		  
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From dmatveev at swdrom.com  Wed Oct 19 11:44:17 2011
From: dmatveev at swdrom.com (Dmitry Matveev)
Date: Wed, 19 Oct 2011 11:44:17 +0400
Subject: [Freeswitch-users] Proxying video stream, recording audo stream
Message-ID: <CAKA2esk_DCVqhA4mB7Pt5h6umZZC541XR7eGp_HMoNj=Wxd+=g@mail.gmail.com>

Hello,

I have a VoIP client software with a video codec unsupported by FreeSWITCH.
To make video calls possible, I am currently using 'bypass_media' setting.

But I would like to record all audio calls with FreeSWITCH. Is it possible
to configure FreeSWITCH to
1. Proxy and process (record) audio RTP streams, and
2. Proxy (but not process) video RTP streams?

Thanks in advance,
Dmitry
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From rsaavedra at ecogizmos.com  Wed Oct 19 18:13:40 2011
From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com)
Date: Wed, 19 Oct 2011 09:13:40 -0500
Subject: [Freeswitch-users] mod_freetdm incoming dialplan
In-Reply-To: <10432.1319030211@ccs.covici.com>
References: <4E9E9775.2050006@xofap.com> <10432.1319030211@ccs.covici.com>
Message-ID: <ccb4fc8b76331c4855f1a5e82d048bfe.squirrel@emailmg.globat.com>

Hi,

I'm working with the Sangoma's U100.

I found on Internet that you must use:

<condition field=?source? expression=?mod_freetdm?/>

My experience is that with the U100 you can put context="public" but the
calls always have the context in default, so I put all the instrucctions
in the file 00_inbound_did.xml in:
/usr/local/freeswitch/conf/dialplan/default and everything now is working.

Best regards,

Ricardo Saavedra

> I would just use the freetdm condition and see if that works.
>
> William Alianto <william at xofap.com> wrote:
>
>> Hi,
>>
>> Has anybody got a sample of dialplan for incoming from freetdm?
>>
>> I've been working to get the incoming from my A400P card working, but
>> still fail to do so.
>>
>> Following is my current incoming dialplan :
>>
>> <extension name="analog" >
>> <condition field="context" expression="public"/>
>> <condition field="destination_number" expression="^(\d+)$"/>
>> <condition field="chan_name" expression="FreeTDM/1/[1-4]">
>> <action application="set" data="domain_name=$${domain}"/>
>> <action application="set" data="transfer_ringback=$${us-ring}"/>
>> <action application="answer"/>
>> <action application="transfer" data="800 XML default"/>
>> </condition>
>> </extension>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Oct 19 20:02:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Oct 2011 09:02:02 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxVddXvVRMUFDD3iL45T4zD=-sgfsfP+zRFK-N=9ArT8DQ@mail.gmail.com>

Hello all,

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

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

-Michael
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From infos at madovsky.org  Wed Oct 19 20:41:06 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 19 Oct 2011 12:41:06 -0400
Subject: [Freeswitch-users] Proxying video stream, recording audo stream
References: <CAKA2esk_DCVqhA4mB7Pt5h6umZZC541XR7eGp_HMoNj=Wxd+=g@mail.gmail.com>
Message-ID: <664069B053F248D69F0EBFDEC080265F@e1705>

I think if you bypass_media so the audio/video flow
doesn't go through FS. so the answer is no.
but maybe I'm wrong..
  ----- Original Message ----- 
  From: Dmitry Matveev 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, October 19, 2011 3:44 AM
  Subject: [Freeswitch-users] Proxying video stream, recording audo stream


  Hello,


  I have a VoIP client software with a video codec unsupported by FreeSWITCH.
  To make video calls possible, I am currently using 'bypass_media' setting.


  But I would like to record all audio calls with FreeSWITCH. Is it possible to configure FreeSWITCH to
  1. Proxy and process (record) audio RTP streams, and
  2. Proxy (but not process) video RTP streams?


  Thanks in advance,
  Dmitry




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



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From peter.olsson at visionutveckling.se  Wed Oct 19 21:26:07 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 19 Oct 2011 19:26:07 +0200
Subject: [Freeswitch-users] Running FreeSwitch as Windows Service
Message-ID: <C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>

Remove the service again, and the execute FreeSwitchConsole.exe -install. And try again

/Peter

----- Reply message -----
Fr?n: "Adelia C." <hexade at hotmail.com>
Datum: ons, okt 19, 2011 18:49
Rubrik: [Freeswitch-users] Running FreeSwitch as Windows Service
Till: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>

Hello,

I am building FreeSwitch through VS 2010 (not VC++ Express) and I generate FreeSwitchConsole.exe as a console application. No FreeSwitch.exe is generated.

FreeSwitchConsole runs fine as an application but I need to run the SW as a service.

I installed Freeswitch as a local service that calls FreeSwitchConsole.exe, and this is the output of "sc qc FreeSWITCH":

C:\FreeSWITCH>sc qc FreeSWITCH
[SC] QueryServiceConfig SUCCESS

SERVICE_NAME: FreeSWITCH
        TYPE               : 10  WIN32_OWN_PROCESS
        START_TYPE         : 2   AUTO_START
        ERROR_CONTROL      : 1   NORMAL
        BINARY_PATH_NAME   : C:\FreeSWITCH\FreeSwitchConsole.exe
        LOAD_ORDER_GROUP   :
        TAG                : 0
        DISPLAY_NAME       : FreeSWITCH
        DEPENDENCIES       :
        SERVICE_START_NAME : LocalSystem


Trying to start FreeSwitch as a service results in  Error 1053 Timeout - Event Viewer is : Timeout (30000 milliseconds) waiting for the FreeSWITCH service to connect.

Or:
C:\FreeSWITCH>net start freeswitch
The service is not responding to the control function.

Is it because I am trying to start a console application as a service?

Do I need to have a FreeSwitch.exe and if so, how do I build it? Do I need to create a Windows Service wrapper project?

I tried using winserv (supposed to start console apps as service), with same result:
C:\FreeSWITCH>..\tools\winserv\winserv start FreeSwitch
ERROR 1053: The service did not respond to the start or control request in a timely fashion.

What am I missing?

Thank you.
A.C.
!DSPAM:4e9efdb732761275568181!


From sherifomran2000 at yahoo.com  Wed Oct 19 17:47:07 2011
From: sherifomran2000 at yahoo.com (sherif omran)
Date: Wed, 19 Oct 2011 13:47:07 +0000 (UTC)
Subject: [Freeswitch-users] Different SAS with ZRTP
References: <64710029-617A-4648-8BF7-68249B3BD8F0@gmail.com>
Message-ID: <loom.20111019T154437-357@post.gmane.org>




Banhi Dutra <banhi.dutra at ...> writes:

> 
> Hi all.
> 
> Could anyone help me with ZRTP configuration ? For me all works fine, except
for SAS exchange, which are
> different. I'm using default configuration because it is a test environment. 
> 
> Where I could find configuration or documentation about "Trusted MITM" ?
> 
> Thank you.
> 
> Banhi
> 


Hi,

I have the same issue, FreeSwitch with Zrtp library from zphone but zphone does
not show secure channel button to press on.

Any help.

thanks
Sherif



From fcintrono at gmail.com  Wed Oct 19 22:45:21 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Wed, 19 Oct 2011 13:45:21 -0500
Subject: [Freeswitch-users] Make freeswitch use spanish sounds...
In-Reply-To: <CAKzWOxVzutgu=mPz7j1fEU1kQ_KELmZwq9-BG7uen84j0Bn28A@mail.gmail.com>
References: <CANtw_ZOVVHN=xD-yHafyrYaMx5+snSfMr3RjKxUS43DAY57V-w@mail.gmail.com>
	<CAKzWOxVzutgu=mPz7j1fEU1kQ_KELmZwq9-BG7uen84j0Bn28A@mail.gmail.com>
Message-ID: <CANtw_ZPZN4enVMAyD1XPzj9vWEVD-8aiBL+LtXB09CyGdkKALg@mail.gmail.com>

This morning I followed all your directions.
I only needed to add this two lines to vars.xml:

 <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/es/mx/maria"/>
 <X-PRE-PROCESS cmd="set" data="default_language=es"/>

And It worked perfectly.

Thank you very much.

On Tue, Oct 18, 2011 at 3:57 PM, Michael Collins <msc at freeswitch.org> wrote:

> Well, be sure to build mod_say_es. From your FS source tree:
>
> make mod_say_es-install
>
> Unfortunately it looks like the installer does not create a conf/lang/es/
> directory for you, so you will need to create one yourself. Copy the
> conf/lang/en/* directory structure and files into conf/lang/es/*
>
> After that go into conf/lang/es and rename en.xml to es.xml. Open that file
> and look for this line:
> <language name="en" say-module="en"
> sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="cepstral"
> tts-voice="callie">
>
> Change it to:
> <language name="es" say-module="es"
> sound-prefix="$${sounds_dir}/es/mx/maria" tts-engine="cepstral"
> tts-voice="callie">
>
> Be sure to put your maria sounds in the proper directory structure, just
> like the sounds/en/us/callie files are.
>
> You will also need to add a line to freeswitch.xml like the other langs:
> <X-PRE-PROCESS cmd="include" data="lang/es/*.xml"/>
>
> If you want to load mod_say_es automatically then add it to
> conf/autoload_configs/modules.conf, otherwise restart FreeSWITCH and do
> "load mod_say_es" and hopefully nothing blows up! :)
>
> Also, you may want to set the default language on calls that you know are
> Spanish. Check out this wiki page for assistance. (
> http://wiki.freeswitch.org/wiki/Channel_Variables#default_language)
>
> -MC
>
> On Tue, Oct 18, 2011 at 1:32 PM, Francisco Javier Cintr?n Olgu?n <
> fcintrono at gmail.com> wrote:
>
>> I downloaded this file which is a set of spanish sounds for freeswitch:
>>
>> http://files.freeswitch.org/freeswitch-sounds-es-mx-maria-44100.tar.gz
>>
>> As you see It only has sounds for the 44100 frecuency.
>> To generate files for the 8000, 16000 and 32000 frecuencies,  I untarred
>> the file inside freeswitch/sounds directory.
>>
>> After I untarred the file, there was a directory called es/mx/maria which
>> has all this subdirectories:
>>
>> ascii
>> currency
>> ivr
>> phonetic-ascii
>> voicemail
>>  conference
>> digits
>> misc
>> time
>> zrtp
>>
>> For each of these subdirectories I made this:
>>
>> cd "one of the directories inside sounds/es/mx/maria "
>> mkdir 8000
>> mkdir 16000
>> mkdir 16000
>>
>> cd 44100
>> for a in *.wav;
>> do
>>      /usr/bin/ffmpeg -i "$a" -ar 8000 -ac 1 -ab 128 "../8000/"$a;
>>      /usr/bin/ffmpeg -i "$a" -ar 16000 "../16000/"$a;
>>      /usr/bin/ffmpeg -i "$a" -ar 32000 "../32000/"$a;
>> done
>>
>> Now I have a complete set of spanish sounds for the 8000, 16000, 32000 y
>> 44100 frecuencies
>>
>> ?What would I need to configure in freeswitch to use these spanish sounds
>> instead of the english ones?
>>
>> Thanks in advance for your kind help.
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Oct 19 22:49:16 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 19 Oct 2011 14:49:16 -0400
Subject: [Freeswitch-users] transfer_ringback from a conference
Message-ID: <B55CB35F9A1746669F128729A3E9B728@e1705>

After read this thread of 2009

http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8469

Brian said that it was not possible to use transfer_ringback.

but with last git is it possible now ?

Thanks
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From msc at freeswitch.org  Thu Oct 20 01:45:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Oct 2011 14:45:38 -0700
Subject: [Freeswitch-users] Make freeswitch use spanish sounds...
In-Reply-To: <CANtw_ZPZN4enVMAyD1XPzj9vWEVD-8aiBL+LtXB09CyGdkKALg@mail.gmail.com>
References: <CANtw_ZOVVHN=xD-yHafyrYaMx5+snSfMr3RjKxUS43DAY57V-w@mail.gmail.com>
	<CAKzWOxVzutgu=mPz7j1fEU1kQ_KELmZwq9-BG7uen84j0Bn28A@mail.gmail.com>
	<CANtw_ZPZN4enVMAyD1XPzj9vWEVD-8aiBL+LtXB09CyGdkKALg@mail.gmail.com>
Message-ID: <CAKzWOxVSoY14bf5VHFb_B78iVtnp3D3mLx7NAptQAqFpA-=eZg@mail.gmail.com>

?Buen trabajo!

Thanks for letting us know that it's working. I have an item on my to-do
list to get the Spanish files updated but it's WAY down on the priority
list...

-MC

On Wed, Oct 19, 2011 at 11:45 AM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> This morning I followed all your directions.
> I only needed to add this two lines to vars.xml:
>
>  <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/es/mx/maria"/>
>  <X-PRE-PROCESS cmd="set" data="default_language=es"/>
>
> And It worked perfectly.
>
> Thank you very much.
>
>
> On Tue, Oct 18, 2011 at 3:57 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Well, be sure to build mod_say_es. From your FS source tree:
>>
>> make mod_say_es-install
>>
>> Unfortunately it looks like the installer does not create a conf/lang/es/
>> directory for you, so you will need to create one yourself. Copy the
>> conf/lang/en/* directory structure and files into conf/lang/es/*
>>
>> After that go into conf/lang/es and rename en.xml to es.xml. Open that
>> file and look for this line:
>> <language name="en" say-module="en"
>> sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="cepstral"
>> tts-voice="callie">
>>
>> Change it to:
>> <language name="es" say-module="es"
>> sound-prefix="$${sounds_dir}/es/mx/maria" tts-engine="cepstral"
>> tts-voice="callie">
>>
>> Be sure to put your maria sounds in the proper directory structure, just
>> like the sounds/en/us/callie files are.
>>
>> You will also need to add a line to freeswitch.xml like the other langs:
>> <X-PRE-PROCESS cmd="include" data="lang/es/*.xml"/>
>>
>> If you want to load mod_say_es automatically then add it to
>> conf/autoload_configs/modules.conf, otherwise restart FreeSWITCH and do
>> "load mod_say_es" and hopefully nothing blows up! :)
>>
>> Also, you may want to set the default language on calls that you know are
>> Spanish. Check out this wiki page for assistance. (
>> http://wiki.freeswitch.org/wiki/Channel_Variables#default_language)
>>
>> -MC
>>
>> On Tue, Oct 18, 2011 at 1:32 PM, Francisco Javier Cintr?n Olgu?n <
>> fcintrono at gmail.com> wrote:
>>
>>> I downloaded this file which is a set of spanish sounds for freeswitch:
>>>
>>> http://files.freeswitch.org/freeswitch-sounds-es-mx-maria-44100.tar.gz
>>>
>>> As you see It only has sounds for the 44100 frecuency.
>>> To generate files for the 8000, 16000 and 32000 frecuencies,  I untarred
>>> the file inside freeswitch/sounds directory.
>>>
>>> After I untarred the file, there was a directory called es/mx/maria which
>>> has all this subdirectories:
>>>
>>> ascii
>>> currency
>>> ivr
>>> phonetic-ascii
>>> voicemail
>>>  conference
>>> digits
>>> misc
>>> time
>>> zrtp
>>>
>>> For each of these subdirectories I made this:
>>>
>>> cd "one of the directories inside sounds/es/mx/maria "
>>> mkdir 8000
>>> mkdir 16000
>>> mkdir 16000
>>>
>>> cd 44100
>>> for a in *.wav;
>>> do
>>>      /usr/bin/ffmpeg -i "$a" -ar 8000 -ac 1 -ab 128 "../8000/"$a;
>>>      /usr/bin/ffmpeg -i "$a" -ar 16000 "../16000/"$a;
>>>      /usr/bin/ffmpeg -i "$a" -ar 32000 "../32000/"$a;
>>> done
>>>
>>> Now I have a complete set of spanish sounds for the 8000, 16000, 32000 y
>>> 44100 frecuencies
>>>
>>> ?What would I need to configure in freeswitch to use these spanish sounds
>>> instead of the english ones?
>>>
>>> Thanks in advance for your kind help.
>>>
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mhammond at gcicom.net  Thu Oct 20 00:07:42 2011
From: mhammond at gcicom.net (Mike Hammond)
Date: Wed, 19 Oct 2011 21:07:42 +0100
Subject: [Freeswitch-users] BLF not working
Message-ID: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>

Hello all,
?
I'm new to FS and I'm having a problem getting BLF working, initially I'm trying with a Cisco SPA504G.
?
I've configured a test switch directly on a public IP using the latest download (about a week ago) with a fusionpbx front end gui, running on an ubuntu 10.04 LTS server.
?
I've set it up for multi tenant and have had some domain names pointed at my public IP, I can register and use handsets and gateways without an issue on the different tenants offsite from behind nat'd routers.
?
Next I wanted to try BLF, so I've configured a spare key on the Cisco handset (extn 204) with fnc=blf+sd+cp;sub=202@$PROXY;ext=202@$PROXY to monitor extension 202 which is within the same tenant domain. But the key is glowing orange so obviously not happy
?
If set sofia to debug presence and turn on the sip trace I see the following messaging
?
http://pastebin.freeswitch.org/17557
?
Can anyone help me with this one as I'm struggling to know what to do next other than strip it all back out and start again with clean a single domain setup.
?
Many thanks,
?


?
Mike Hammond

??
GCI Com
Cedar Court Office Park
Denby Dale Road
Calder Grove
Wakefield
WF4 3QZ
?
Office:???0844 443 3537
Fax:??????0844?443 3540
http://www.gcicom.net/


This email has been swept by Webroot for viruses. Any files transmitted with it are confidential and intended solely for the email recipient. If you are not the intended recipient please delete this email immediately. Be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error please notify the system administrator. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company. Finally, the recipient should check this email and any attachments for the presence of viruses.


GCI Com incorporates the following Group Companies:
GCI Telecom Group Limited Reg. No. 5396496, Edge Telecommunications Ltd Reg. No. 5748740, Edge Telecom Ltd Reg. No. 3101247, IP Infrastructures Ltd Reg. No. 4657026, Invomo Ltd Reg. No. 6267056, NetServices UK Ltd Reg. No. 7118768, WAN Services Ltd Reg. No. 4082862. All Registered in England and Wales, Registered Office: Global House, 2 Crofton Close, Lincoln, LN3 4NT

From randy.andrade at gmail.com  Thu Oct 20 02:42:21 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Wed, 19 Oct 2011 18:42:21 -0400
Subject: [Freeswitch-users] BLF not working
In-Reply-To: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>
References: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>
Message-ID: <-5270605825443932569@unknownmsgid>

I'm curious what version of firmware you're running on your 504's. I
work with them professionally and have found that certain functions
work better / don't work depending on your version of firmware.
Personally I've found 7.4.8a to be the best stability / feature
balance. 7.4.9a was just released and though it fixes a few bugs,
especiallly related to BLF's, against our servers a lot of more basic
functionality, such as blind transfers and others which were related
to the 500S sidecar, but we weren't testing against FreeSwitch, and we
were testing SPA509G's. Perhaps you'll have better luck with it? Or
better luck not using it, if you are? Point I'm trying to make is,
Cisco has been known to break some functionality in virtually every
recent release of the firmware, so try a couple different revisions,
because your code looks correct. I use it on my Linksys SPA942's and
Cisco SPA508's, which both have working BLF's against my FreeSwitch
box, at home. The SPA508's are running FW v7.4.8a with the BLF's
working fine.

Randy

Sent from my iPhone

On Oct 19, 2011, at 5:46 PM, Mike Hammond <mhammond at gcicom.net> wrote:

> Hello all,
>
> I'm new to FS and I'm having a problem getting BLF working, initially I'm trying with a Cisco SPA504G.
>
> I've configured a test switch directly on a public IP using the latest download (about a week ago) with a fusionpbx front end gui, running on an ubuntu 10.04 LTS server.
>
> I've set it up for multi tenant and have had some domain names pointed at my public IP, I can register and use handsets and gateways without an issue on the different tenants offsite from behind nat'd routers.
>
> Next I wanted to try BLF, so I've configured a spare key on the Cisco handset (extn 204) with fnc=blf+sd+cp;sub=202@$PROXY;ext=202@$PROXY to monitor extension 202 which is within the same tenant domain. But the key is glowing orange so obviously not happy
>
> If set sofia to debug presence and turn on the sip trace I see the following messaging
>
> http://pastebin.freeswitch.org/17557
>
> Can anyone help me with this one as I'm struggling to know what to do next other than strip it all back out and start again with clean a single domain setup.
>
> Many thanks,
>
>
>
>
> Mike Hammond
>
>
> GCI Com
> Cedar Court Office Park
> Denby Dale Road
> Calder Grove
> Wakefield
> WF4 3QZ
>
> Office:   0844 443 3537
> Fax:      0844 443 3540
> http://www.gcicom.net/
>
>
> This email has been swept by Webroot for viruses. Any files transmitted with it are confidential and intended solely for the email recipient. If you are not the intended recipient please delete this email immediately. Be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error please notify the system administrator. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company. Finally, the recipient should check this email and any attachments for the presence of viruses.
>
>
> GCI Com incorporates the following Group Companies:
> GCI Telecom Group Limited Reg. No. 5396496, Edge Telecommunications Ltd Reg. No. 5748740, Edge Telecom Ltd Reg. No. 3101247, IP Infrastructures Ltd Reg. No. 4657026, Invomo Ltd Reg. No. 6267056, NetServices UK Ltd Reg. No. 7118768, WAN Services Ltd Reg. No. 4082862. All Registered in England and Wales, Registered Office: Global House, 2 Crofton Close, Lincoln, LN3 4NT
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jmesquita at freeswitch.org  Thu Oct 20 02:52:24 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 19 Oct 2011 19:52:24 -0300
Subject: [Freeswitch-users] esl_event_get_header error
In-Reply-To: <20111018230647.199160@gmx.com>
References: <20111018230647.199160@gmx.com>
Message-ID: <CABH8QX0WZGo1k97-pPg4wS8P79xeS7Pt3BFboaJsTKMAdWM9_g@mail.gmail.com>

What is the call_info object? Best way to get help is to provide a simple
working (or not working in this case) example of your code. Then I can
compile/test and figure it out for ya.

It seems like last_event is NULL in this case...

Regards,
Jo?o Mesquita



On Tue, Oct 18, 2011 at 8:06 PM, adam harris <admharris at gmx.com> wrote:

>
> I am trying to check if event received was CHANNEL_HANUP
>
>
> I am doing somthing like this
>
>  const char *last_event =
> esl_event_get_header(call_info->global_handle->last_event,"Event-Name");
>
>  if (strcmp(last_event,"CHANNEL_HANGUP")) {
>  exit(0);
>  }
>
> However an error occurs
>
> src/esl_event.c:244: esl_event_get_header_ptr: Assertion `event' failed.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ddgiants at gmail.com  Thu Oct 20 03:01:09 2011
From: ddgiants at gmail.com (Darren DeLitizia)
Date: Wed, 19 Oct 2011 19:01:09 -0400
Subject: [Freeswitch-users] Freeswitch handling 300 multiple choice redirect
Message-ID: <CAKvB_N7Bf33hFjFwJipfX+JuZjycUf88vC4JkYTRkx_HBmGO4w@mail.gmail.com>

All,
I am having a hell of a time getting Freeswitch to handle multiple choice
300 redirects. I've followed directions on -->
http://wiki.freeswitch.org/wiki/SBC_FreeSWITCH_Configuration_Example_2 but
it does not work. It never gets past the first contact. I've done Googled my
arse off and still nothing. Can anyone help? Currently my config is exactly
what you see on that link above. I am using F.S. as topology hiding entrance
point and Opensips as LCR redirect.

The call flow is

Customer --> F.S. --> Vendors from redirect
                         |
                         |
                    Opensips (returns 300 multiple)


Thanks
DD
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From william at xofap.com  Thu Oct 20 08:58:59 2011
From: william at xofap.com (William Alianto)
Date: Thu, 20 Oct 2011 11:58:59 +0700
Subject: [Freeswitch-users] mod_freetdm incoming dialplan
In-Reply-To: <ccb4fc8b76331c4855f1a5e82d048bfe.squirrel@emailmg.globat.com>
References: <4E9E9775.2050006@xofap.com> <10432.1319030211@ccs.covici.com>
	<ccb4fc8b76331c4855f1a5e82d048bfe.squirrel@emailmg.globat.com>
Message-ID: <4E9FAA93.8050704@xofap.com>

Ok, I have put my dialplan on default, and change it to :

<extension name="analog" >
<condition field="source" expression="mod_freetdm" >
<action application="set" data="domain_name=$${domain}" />
<action application="set" data="transfer_ringback=$${us-ring}" />
<action application="answer"
<action application="transfer" data="800 XML default" />
</condition>
</extension>

The FS finally able to pass the dialplan, but it still not ringing on my 
phone. Any idea how?

Regards,

William

On 2:59, rsaavedra at ecogizmos.com wrote:
> Hi,
>
> I'm working with the Sangoma's U100.
>
> I found on Internet that you must use:
>
> <condition field=?source? expression=?mod_freetdm?/>
>
> My experience is that with the U100 you can put context="public" but the
> calls always have the context in default, so I put all the instrucctions
> in the file 00_inbound_did.xml in:
> /usr/local/freeswitch/conf/dialplan/default and everything now is working.
>
> Best regards,
>
> Ricardo Saavedra
>
>> I would just use the freetdm condition and see if that works.
>>
>> William Alianto<william at xofap.com>  wrote:
>>
>>> Hi,
>>>
>>> Has anybody got a sample of dialplan for incoming from freetdm?
>>>
>>> I've been working to get the incoming from my A400P card working, but
>>> still fail to do so.
>>>
>>> Following is my current incoming dialplan :
>>>
>>> <extension name="analog">
>>> <condition field="context" expression="public"/>
>>> <condition field="destination_number" expression="^(\d+)$"/>
>>> <condition field="chan_name" expression="FreeTDM/1/[1-4]">
>>> <action application="set" data="domain_name=$${domain}"/>
>>> <action application="set" data="transfer_ringback=$${us-ring}"/>
>>> <action application="answer"/>
>>> <action application="transfer" data="800 XML default"/>
>>> </condition>
>>> </extension>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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,

William
CV.XOFAP.COM
Engineering Dept.



From gabe at gundy.org  Thu Oct 20 10:45:07 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Oct 2011 00:45:07 -0600
Subject: [Freeswitch-users] Digest Authentication issue
In-Reply-To: <CAMd3iJJX9887ALe73KYjNd5dWmkG_bdE-L-gGiuE2Bf5vOBMRg@mail.gmail.com>
References: <CAMd3iJJX9887ALe73KYjNd5dWmkG_bdE-L-gGiuE2Bf5vOBMRg@mail.gmail.com>
Message-ID: <CAMwZ-tL3nVzFUU=syNTogwMw=sP_e3Rds93vnBySyPxWLheKOw@mail.gmail.com>

On Fri, Oct 14, 2011 at 2:39 PM, Masood Ahmed <masood24 at gmail.com> wrote:
> I had successfully installed and configured radiusclient. I do not know that
> what next lines should i do to get users registered on my freeswitch using
> digest authentication with my radius server,
> Like I have my radsius server and billing system. Now what lines should i
> add in sofit conf or in dialplans that every time client hits server request
> should be sent to radius server and what lines should i add to register them
> if i get proper return code and how to do that.

Have you looked here?

http://wiki.freeswitch.org/wiki/Mod_xml_radius


Best,
Gabe


From gabe at gundy.org  Thu Oct 20 10:50:42 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Oct 2011 00:50:42 -0600
Subject: [Freeswitch-users] custom events
In-Reply-To: <7DBF30F35AA24C8C8E0DDB323D831791@ManishKhilwani>
References: <7DBF30F35AA24C8C8E0DDB323D831791@ManishKhilwani>
Message-ID: <CAMwZ-t+ppzztM2ekQUS2Taq=hWooZO5jBBVwgeP-Wd2UQcArhQ@mail.gmail.com>

On Tue, Oct 18, 2011 at 4:54 AM, Manish K - Clarion, India <
manish.khilwani at clariontechnologies.co.in> wrote:

> I'm trying to register SIP via sofia::register event.
>
> can you help me??
>

You can't register via an event. The event lets you know that someone has
registered. Instead, you register by supplying the proper info in the
directory section of the XML config.

Hope that helps (well, and that I even understood what you were asking).

Gabe
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From gabe at gundy.org  Thu Oct 20 10:52:17 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Oct 2011 00:52:17 -0600
Subject: [Freeswitch-users] transfer_ringback from a conference
In-Reply-To: <B55CB35F9A1746669F128729A3E9B728@e1705>
References: <B55CB35F9A1746669F128729A3E9B728@e1705>
Message-ID: <CAMwZ-tKZCZXxxgYQXD+3iyVE3b2u12j=B_reJZ_XuBR1YWctvA@mail.gmail.com>

On Wed, Oct 19, 2011 at 12:49 PM, Madovsky <infos at madovsky.org> wrote:
> but with last git is it possible now ?

The best way to find out would be to give it a try!  Do let us know
what you find and update the docs :)

Good luck,
Gabe


From gabe at gundy.org  Thu Oct 20 11:07:23 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Oct 2011 01:07:23 -0600
Subject: [Freeswitch-users] audo sync issues with record_session to mp3
In-Reply-To: <36bc584d980ce80fe6a6f6e7d7383db9.squirrel@my.tomp.co.uk>
References: <36bc584d980ce80fe6a6f6e7d7383db9.squirrel@my.tomp.co.uk>
Message-ID: <CAMwZ-t+=eu1_ocEV5WKgx-r5exuHcWi-g0hNGESBVB9BUEudCw@mail.gmail.com>

On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott <tomp at tomp.co.uk> wrote:
> Longer calls, after about 10 minutes start to introduce sync issues
> between the A-leg and the B-leg.
>
> We are running record_session on the A-leg, and it seems to get ahead of
> the B-leg.
>
> For example the caller on the A-leg will be heard to answer a question
> whilst the person on the B-leg is asking it.

What's on the other end of each leg? That might help us figure this out.

Gabe


From grsingh750 at gmail.com  Thu Oct 20 12:17:26 2011
From: grsingh750 at gmail.com (guru singh)
Date: Thu, 20 Oct 2011 13:47:26 +0530
Subject: [Freeswitch-users] mod_freetdm incoming dialplan
In-Reply-To: <4E9FAA93.8050704@xofap.com>
References: <4E9E9775.2050006@xofap.com> <10432.1319030211@ccs.covici.com>
	<ccb4fc8b76331c4855f1a5e82d048bfe.squirrel@emailmg.globat.com>
	<4E9FAA93.8050704@xofap.com>
Message-ID: <CAFSQhGXDbMtHv-F3GpxKuLsnX3isXX7PzzNtv=KKyRwgtZQS0w@mail.gmail.com>

Hi William,

What phone do  you expect it to ring?
Do you have an extension in your default dialplan for destination number 800?
You should pastebin your freeswitch logs, so we can see whats going on.

Regards,
guru

On Thu, Oct 20, 2011 at 10:28 AM, William Alianto <william at xofap.com> wrote:
> Ok, I have put my dialplan on default, and change it to :
>
> <extension name="analog" >
> <condition field="source" expression="mod_freetdm" >
> <action application="set" data="domain_name=$${domain}" />
> <action application="set" data="transfer_ringback=$${us-ring}" />
> <action application="answer"
> <action application="transfer" data="800 XML default" />
> </condition>
> </extension>
>
> The FS finally able to pass the dialplan, but it still not ringing on my
> phone. Any idea how?
>
> Regards,
>
> William
>
> On 2:59, rsaavedra at ecogizmos.com wrote:
>> Hi,
>>
>> I'm working with the Sangoma's U100.
>>
>> I found on Internet that you must use:
>>
>> <condition field=?source? expression=?mod_freetdm?/>
>>
>> My experience is that with the U100 you can put context="public" but the
>> calls always have the context in default, so I put all the instrucctions
>> in the file 00_inbound_did.xml in:
>> /usr/local/freeswitch/conf/dialplan/default and everything now is working.
>>
>> Best regards,
>>
>> Ricardo Saavedra
>>
>>> I would just use the freetdm condition and see if that works.
>>>
>>> William Alianto<william at xofap.com> ?wrote:
>>>
>>>> Hi,
>>>>
>>>> Has anybody got a sample of dialplan for incoming from freetdm?
>>>>
>>>> I've been working to get the incoming from my A400P card working, but
>>>> still fail to do so.
>>>>
>>>> Following is my current incoming dialplan :
>>>>
>>>> <extension name="analog">
>>>> <condition field="context" expression="public"/>
>>>> <condition field="destination_number" expression="^(\d+)$"/>
>>>> <condition field="chan_name" expression="FreeTDM/1/[1-4]">
>>>> <action application="set" data="domain_name=$${domain}"/>
>>>> <action application="set" data="transfer_ringback=$${us-ring}"/>
>>>> <action application="answer"/>
>>>> <action application="transfer" data="800 XML default"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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,
>
> William
> CV.XOFAP.COM
> Engineering Dept.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 faisal.rehman22 at hotmail.com  Thu Oct 20 13:21:58 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Thu, 20 Oct 2011 14:21:58 +0500
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
Message-ID: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>


Hi Everyone,

First of all I would like to say that I wasn't supposed to ask this question here in this forum that belongs to Free-switch, but I think I can find any help related to Asterisk ASR & ACD monitoring.
So I am in search of a reliable open source tool for the real time monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or suggestions will be highly appreciated.


Regards,

Faisal Rehman
 		 	   		  
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From eagle.antonio at gmail.com  Thu Oct 20 14:01:28 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 20 Oct 2011 10:01:28 +0000
Subject: [Freeswitch-users] Fs Core Dumps On Mod Python
Message-ID: <CAF+3j-sLP6WG=pf4FodF6n_h2gcf0h9doEAYGoB7hgWUFiSL3g@mail.gmail.com>

Just to give all you guys a heads up and since the devs are still sleeping
:) and you guys maybe tempted to upgrade.

On GIT HEAD as of now FreeSWITCH Version 1.0.head (git-957361c 2011-10-19
16-50-46 -0500)

FS Will core dump if mod_python is enabled on modules.conf.xml so try it
before deploying :) , i just tested on one machine but just to be sure ...


Related info :

http://jira.freeswitch.org/browse/FS-3622
http://pastebin.freeswitch.org/17561


P.S Sorry to double post :)
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From netcentrica at gmail.com  Thu Oct 20 14:50:16 2011
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Thu, 20 Oct 2011 12:50:16 +0200
Subject: [Freeswitch-users] Bridge problem - (randomly) leg A looses events
 and communication with leg B
Message-ID: <CAJWEF2avKuKrCuAMnKQ3-RaMWaEQJzkyqyPGEvY=Do8+nrEScg@mail.gmail.com>

Hi All,

My scenario:
number dialed by user test_user
dial plan configured to send outbound call to event socket listening on
127.0.0.1:11111
outbound call is intercepted, bridge application is executed to connect leg
A (test_user) with dialed number via my SIP provider (leg B)

The problem is, that (completely randomly - sometimes it's OK, sometimes
not) user's session is desynchronized with events from leg B.
Leg B dials number and connects via SIP provider to the destination number
(that's always OK)
But (randomly) leg A doesn't get ring back, no connect event, still in
"calling" state until 30 second timeout (in the same time leg B is connected
OK) but there is some problem with passing events from leg B to leg A

Bridge params:
{ignore_early_media=true,bypass_media=false,origination_caller_id_name=004822333222111,origination_caller_id_number=004822333222111,effective_caller_id_number=004822333222111,effective_caller_id_name=004822333222111,execute_on_answer=sched_hangup
+5700 alloted_timeout}sofia/gateway/supergateway/0000048222019475

I've tried to change socket mode to sync but with no success

I use following Java code to execute bridge application:

SendMsg msg = new SendMsg();
msg.addCallCommand("execute");
msg.addExecuteAppName("bridge");
msg.addExecuteAppArg(dialstring);
msg.addEventLock();

Full freeswitch console log link: http://pastebin.com/eLCmMXRB

I will greatly appreciate your help solving this problem
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From govoiper at gmail.com  Thu Oct 20 15:00:08 2011
From: govoiper at gmail.com (Sammy Govind)
Date: Thu, 20 Oct 2011 16:00:08 +0500
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
Message-ID: <CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>

I hope you've already heard about it, not an opensource one but VQManager is
the best thing so far I've came across.

On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman
<faisal.rehman22 at hotmail.com>wrote:

>  *Hi Everyone,*
>
>
> First of all I would like to say that I wasn't supposed to ask this
> question here in this forum that belongs to Free-switch, but I think I can
> find any help related to Asterisk ASR & ACD monitoring.
>
> So I am in search of a reliable open source tool for the real time
> monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
> suggestions will be highly appreciated.
>
>
>
> Regards,
>
> *Faisal Rehman*
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From faisal.rehman22 at hotmail.com  Thu Oct 20 15:14:31 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Thu, 20 Oct 2011 16:14:31 +0500
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>,
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
Message-ID: <BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>


Hi Sam,
Will that also generate an email if the ASR & ACD is low from the desired requirements & also it is a little expensive :) so any further idea?

Regards, 

Faisal Rehman

Date: Thu, 20 Oct 2011 16:00:08 +0500
From: govoiper at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring

I hope you've already heard about it, not an opensource one but VQManager is the best thing so far I've came across.

On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:






Hi Everyone,

First of all I would like to say that I wasn't supposed to ask this question here in this forum that belongs to Free-switch, but I think I can find any help related to Asterisk ASR & ACD monitoring.

So I am in search of a reliable open source tool for the real time monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or suggestions will be highly appreciated.



Regards,

Faisal Rehman
 		 	   		  



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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

http://www.freeswitch.org





FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From nbhatti at gmail.com  Thu Oct 20 15:21:07 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Thu, 20 Oct 2011 14:21:07 +0300
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
Message-ID: <CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>

Hello Faisal,
We have developed scripts which may do the purpose. Let me know if you
are interested. I would to get more details about your system as well.

Naseer

On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
<faisal.rehman22 at hotmail.com> wrote:
> Hi Sam,
> Will that also generate an email if the ASR & ACD is low from the desired
> requirements & also it is a little expensive :) so any further idea?
>
> Regards,
>
> Faisal Rehman
>
> ________________________________
> Date: Thu, 20 Oct 2011 16:00:08 +0500
> From: govoiper at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>
> I hope you've already heard about it, not an opensource one but VQManager is
> the best thing so far I've came across.
>
> On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <faisal.rehman22 at hotmail.com>
> wrote:
>
> Hi Everyone,
>
> First of all I would like to say that I wasn't supposed to ask this question
> here in this forum that belongs to Free-switch, but I think I can find any
> help related to Asterisk ASR & ACD monitoring.
> So I am in search of a reliable open source tool for the real time
> monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
> suggestions will be highly appreciated.
>
>
> Regards,
>
> Faisal Rehman
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 govoiper at gmail.com  Thu Oct 20 15:34:36 2011
From: govoiper at gmail.com (Sammy Govind)
Date: Thu, 20 Oct 2011 16:34:36 +0500
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
Message-ID: <CAJUJwtjb1PYntmtgCDmbPmYTg_d1p0Yk_qExAXVvipvD-AeQhA@mail.gmail.com>

VQManager can do everything, trigger alarm and alerts on low/high usages
ASR/ACD etc. Also monitoring QoS.
Self written scripts, using MySQL CDRs table is definitely a good choice if
you are good at it.

On Thu, Oct 20, 2011 at 4:21 PM, Muhammad Naseer Bhatti
<nbhatti at gmail.com>wrote:

> Hello Faisal,
> We have developed scripts which may do the purpose. Let me know if you
> are interested. I would to get more details about your system as well.
>
> Naseer
>
> On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
> <faisal.rehman22 at hotmail.com> wrote:
> > Hi Sam,
> > Will that also generate an email if the ASR & ACD is low from the desired
> > requirements & also it is a little expensive :) so any further idea?
> >
> > Regards,
> >
> > Faisal Rehman
> >
> > ________________________________
> > Date: Thu, 20 Oct 2011 16:00:08 +0500
> > From: govoiper at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> >
> > I hope you've already heard about it, not an opensource one but VQManager
> is
> > the best thing so far I've came across.
> >
> > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <
> faisal.rehman22 at hotmail.com>
> > wrote:
> >
> > Hi Everyone,
> >
> > First of all I would like to say that I wasn't supposed to ask this
> question
> > here in this forum that belongs to Free-switch, but I think I can find
> any
> > help related to Asterisk ASR & ACD monitoring.
> > So I am in search of a reliable open source tool for the real time
> > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
> > suggestions will be highly appreciated.
> >
> >
> > Regards,
> >
> > Faisal Rehman
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mhammond at gcicom.net  Thu Oct 20 15:56:59 2011
From: mhammond at gcicom.net (Mike Hammond)
Date: Thu, 20 Oct 2011 12:56:59 +0100
Subject: [Freeswitch-users] BLF not working
References: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>
	<-5270605825443932569@unknownmsgid>
Message-ID: <1C748D48EFD36B4AA0B934E8B4E2998003615A0B@ipi-cc-srv04.ipinfrastructures.com>

Hi Randy, 

funnily enough I've just upgraded the phone this morning to 7.4.9.a
after digging around on Cisco's website last night and I have now got
some functionality on the 504... yay !

I'm now working through some tests to see what works and what doesn't
with Fusionpbx/FS.

I've started by removing all of the default * feature codes from the
handset itself and set just **# against the pickup code in the attendant
screen and I can now intercept a ringing call by hitting the flashing
blf key which is cool.

I get grn solid for idle.

Red solid when busy on a call.

Red flashing when ringing.
 
DND doesn't seem to affect the BLF though which is a shame as that would
be nice.

I assume that's as good as it gets as unplugging one of the monitored
extensions and waiting for the thing to drop it's registration doesn't
seem to make any difference ? although I assume there will be some sort
of subscription timer involved that FS hasn't reached yet ? given you've
obviously been through the mill with cisco's before, do you have any
recommendations for timer settings on the handsets for things like NAT
keep alives and subscriptions etc ?

Thanks, Mike 




-----Original Message-----
From: Randy Andrade [mailto:randy.andrade at gmail.com] 
Sent: 19 October 2011 23:42
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] BLF not working

I'm curious what version of firmware you're running on your 504's. I
work with them professionally and have found that certain functions
work better / don't work depending on your version of firmware.
Personally I've found 7.4.8a to be the best stability / feature
balance. 7.4.9a was just released and though it fixes a few bugs,
especiallly related to BLF's, against our servers a lot of more basic
functionality, such as blind transfers and others which were related
to the 500S sidecar, but we weren't testing against FreeSwitch, and we
were testing SPA509G's. Perhaps you'll have better luck with it? Or
better luck not using it, if you are? Point I'm trying to make is,
Cisco has been known to break some functionality in virtually every
recent release of the firmware, so try a couple different revisions,
because your code looks correct. I use it on my Linksys SPA942's and
Cisco SPA508's, which both have working BLF's against my FreeSwitch
box, at home. The SPA508's are running FW v7.4.8a with the BLF's
working fine.

Randy

Sent from my iPhone

On Oct 19, 2011, at 5:46 PM, Mike Hammond <mhammond at gcicom.net> wrote:

> Hello all,
>
> I'm new to FS and I'm having a problem getting BLF working, initially
I'm trying with a Cisco SPA504G.
>
> I've configured a test switch directly on a public IP using the latest
download (about a week ago) with a fusionpbx front end gui, running on
an ubuntu 10.04 LTS server.
>
> I've set it up for multi tenant and have had some domain names pointed
at my public IP, I can register and use handsets and gateways without an
issue on the different tenants offsite from behind nat'd routers.
>
> Next I wanted to try BLF, so I've configured a spare key on the Cisco
handset (extn 204) with fnc=blf+sd+cp;sub=202@$PROXY;ext=202@$PROXY to
monitor extension 202 which is within the same tenant domain. But the
key is glowing orange so obviously not happy
>
> If set sofia to debug presence and turn on the sip trace I see the
following messaging
>
> http://pastebin.freeswitch.org/17557
>
> Can anyone help me with this one as I'm struggling to know what to do
next other than strip it all back out and start again with clean a
single domain setup.
>
> Many thanks,
>
>
>
>
> Mike Hammond
>
>
> GCI Com
> Cedar Court Office Park
> Denby Dale Road
> Calder Grove
> Wakefield
> WF4 3QZ
>
> Office:   0844 443 3537
> Fax:      0844 443 3540
> http://www.gcicom.net/
>
>
> This email has been swept by Webroot for viruses. Any files
transmitted with it are confidential and intended solely for the email
recipient. If you are not the intended recipient please delete this
email immediately. Be aware that any disclosure, copying, distribution
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Please note that any views or opinions presented in this email are
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>
>
> GCI Com incorporates the following Group Companies:
> GCI Telecom Group Limited Reg. No. 5396496, Edge Telecommunications
Ltd Reg. No. 5748740, Edge Telecom Ltd Reg. No. 3101247, IP
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All Registered in England and Wales, Registered Office: Global House, 2
Crofton Close, Lincoln, LN3 4NT
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



 
Mike Hammond

  
GCI Com
Cedar Court Office Park
Denby Dale Road
Calder Grove
Wakefield
WF4 3QZ
 
Office:   0844 443 3537
Fax:      0844 443 3540
http://www.gcicom.net/


From faisal.rehman22 at hotmail.com  Thu Oct 20 16:01:20 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Thu, 20 Oct 2011 17:01:20 +0500
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>,
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>,
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>,
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
Message-ID: <BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>


Hi Naseer,
I am going to monitor the ASR and ACD from the Master.csv file of asterisk (/var/log/asterisk/cdr-csv/Master.csv) & actually the basic requirement is that the script or tool should read the whole file and when ASR, ACD is low from the desired requirement, then we should get an email alert or any notification.

Regards,

Faisal Rehman

> From: nbhatti at gmail.com
> Date: Thu, 20 Oct 2011 14:21:07 +0300
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> 
> Hello Faisal,
> We have developed scripts which may do the purpose. Let me know if you
> are interested. I would to get more details about your system as well.
> 
> Naseer
> 
> On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
> <faisal.rehman22 at hotmail.com> wrote:
> > Hi Sam,
> > Will that also generate an email if the ASR & ACD is low from the desired
> > requirements & also it is a little expensive :) so any further idea?
> >
> > Regards,
> >
> > Faisal Rehman
> >
> > ________________________________
> > Date: Thu, 20 Oct 2011 16:00:08 +0500
> > From: govoiper at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> >
> > I hope you've already heard about it, not an opensource one but VQManager is
> > the best thing so far I've came across.
> >
> > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <faisal.rehman22 at hotmail.com>
> > wrote:
> >
> > Hi Everyone,
> >
> > First of all I would like to say that I wasn't supposed to ask this question
> > here in this forum that belongs to Free-switch, but I think I can find any
> > help related to Asterisk ASR & ACD monitoring.
> > So I am in search of a reliable open source tool for the real time
> > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
> > suggestions will be highly appreciated.
> >
> >
> > Regards,
> >
> > Faisal Rehman
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From ljjimenez at gmail.com  Thu Oct 20 16:33:19 2011
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Thu, 20 Oct 2011 12:33:19 +0000
Subject: [Freeswitch-users] Make freeswitch use spanish sounds...
In-Reply-To: <CAKzWOxVSoY14bf5VHFb_B78iVtnp3D3mLx7NAptQAqFpA-=eZg@mail.gmail.com>
References: <CANtw_ZOVVHN=xD-yHafyrYaMx5+snSfMr3RjKxUS43DAY57V-w@mail.gmail.com>
	<CAKzWOxVzutgu=mPz7j1fEU1kQ_KELmZwq9-BG7uen84j0Bn28A@mail.gmail.com>
	<CANtw_ZPZN4enVMAyD1XPzj9vWEVD-8aiBL+LtXB09CyGdkKALg@mail.gmail.com>
	<CAKzWOxVSoY14bf5VHFb_B78iVtnp3D3mLx7NAptQAqFpA-=eZg@mail.gmail.com>
Message-ID: <1552494118-1319114000-cardhu_decombobulator_blackberry.rim.net-1593421769-@b11.c27.bise6.blackberry>

Hola Michael, i am using spanish prompts, but mod_say_es is incomplete and i have been doing changes to the code, later i will pastebin those changes so you can check and approve.

Luis Jimenez

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Wed, 19 Oct 2011 14:45:38 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Make freeswitch use spanish sounds...


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




From jmesquita at freeswitch.org  Thu Oct 20 16:34:19 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Oct 2011 09:34:19 -0300
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
	<BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>
Message-ID: <CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>

I am amazed that no one asked that yet, but ... Why don't you use FreeSWITCH
instead? It is a bit strange that you ask Asterisk questions here, don't
won't help you on their list? Funny, I've heard that before.... :-)

Regards,
Jo?o Mesquita



On Thu, Oct 20, 2011 at 9:01 AM, Faisal Rehman
<faisal.rehman22 at hotmail.com>wrote:

>  *Hi Naseer,*
>
> I am going to monitor the ASR and ACD from the Master.csv file of asterisk
> (/var/log/asterisk/cdr-csv/Master.csv) & actually the basic requirement is
> that the script or tool should read the whole file and when ASR, ACD is low
> from the desired requirement, then we should get an email alert or any
> notification.
>
>
> Regards,
>
> *
> *
> *Faisal Rehman*
>
> > From: nbhatti at gmail.com
> > Date: Thu, 20 Oct 2011 14:21:07 +0300
>
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> >
> > Hello Faisal,
> > We have developed scripts which may do the purpose. Let me know if you
> > are interested. I would to get more details about your system as well.
> >
> > Naseer
> >
> > On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
> > <faisal.rehman22 at hotmail.com> wrote:
> > > Hi Sam,
> > > Will that also generate an email if the ASR & ACD is low from the
> desired
> > > requirements & also it is a little expensive :) so any further idea?
> > >
> > > Regards,
> > >
> > > Faisal Rehman
> > >
> > > ________________________________
> > > Date: Thu, 20 Oct 2011 16:00:08 +0500
> > > From: govoiper at gmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> > >
> > > I hope you've already heard about it, not an opensource one but
> VQManager is
> > > the best thing so far I've came across.
> > >
> > > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <
> faisal.rehman22 at hotmail.com>
> > > wrote:
> > >
> > > Hi Everyone,
> > >
> > > First of all I would like to say that I wasn't supposed to ask this
> question
> > > here in this forum that belongs to Free-switch, but I think I can find
> any
> > > help related to Asterisk ASR & ACD monitoring.
> > > So I am in search of a reliable open source tool for the real time
> > > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
> > > suggestions will be highly appreciated.
> > >
> > >
> > > Regards,
> > >
> > > Faisal Rehman
> > >
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From faisal.rehman22 at hotmail.com  Thu Oct 20 16:40:57 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Thu, 20 Oct 2011 17:40:57 +0500
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>,
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>,
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>,
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>,
	<BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>,
	<CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>
Message-ID: <BAY167-W112A88856EB58230359FD87ECEB0@phx.gbl>


Hi,
These are all open source softwares and linked with each other that is why I am getting help from everywhere openly :) By the way I am also using FS but my requirement is a bit different indeed. So any ideas about my demand?

Regards,
Faisal Rehman

Date: Thu, 20 Oct 2011 09:34:19 -0300
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring

I am amazed that no one asked that yet, but ... Why don't you use FreeSWITCH instead? It is a bit strange that you ask Asterisk questions here, don't won't help you on their list? Funny, I've heard that before.... :-)

Regards,Jo?o Mesquita




On Thu, Oct 20, 2011 at 9:01 AM, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:






Hi Naseer,
I am going to monitor the ASR and ACD from the Master.csv file of asterisk (/var/log/asterisk/cdr-csv/Master.csv) & actually the basic requirement is that the script or tool should read the whole file and when ASR, ACD is low from the desired requirement, then we should get an email alert or any notification.


Regards,

Faisal Rehman

> From: nbhatti at gmail.com
> Date: Thu, 20 Oct 2011 14:21:07 +0300

> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring

> 
> Hello Faisal,
> We have developed scripts which may do the purpose. Let me know if you
> are interested. I would to get more details about your system as well.
> 
> Naseer
> 

> On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
> <faisal.rehman22 at hotmail.com> wrote:
> > Hi Sam,
> > Will that also generate an email if the ASR & ACD is low from the desired

> > requirements & also it is a little expensive :) so any further idea?
> >
> > Regards,
> >
> > Faisal Rehman
> >
> > ________________________________
> > Date: Thu, 20 Oct 2011 16:00:08 +0500

> > From: govoiper at gmail.com
> > To: freeswitch-users at lists.freeswitch.org

> > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> >
> > I hope you've already heard about it, not an opensource one but VQManager is
> > the best thing so far I've came across.

> >
> > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <faisal.rehman22 at hotmail.com>
> > wrote:
> >
> > Hi Everyone,

> >
> > First of all I would like to say that I wasn't supposed to ask this question
> > here in this forum that belongs to Free-switch, but I think I can find any
> > help related to Asterisk ASR & ACD monitoring.

> > So I am in search of a reliable open source tool for the real time
> > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
> > suggestions will be highly appreciated.

> >
> >
> > Regards,
> >
> > Faisal Rehman
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org

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

> > http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org

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

> > http://www.freeswitch.org
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org

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

> > http://www.freeswitch.org
> >
> >
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org

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

> http://www.freeswitch.org
 		 	   		  



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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

http://www.freeswitch.org





FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From dgarcia at anew.com.ve  Thu Oct 20 16:46:25 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Thu, 20 Oct 2011 08:16:25 -0430
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
	<BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>
	<CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>
Message-ID: <4EA01821.20503@anew.com.ve>

Good point...

I will ask then...

For freeswicth users,,,
which tool do you use to monitor freeswicth (ASR, ACD)?
DO you develop your own monitoring scripts/tools?
Would like share your experience?

On 10/20/2011 8:04 AM, Jo?o Mesquita wrote:
> I am amazed that no one asked that yet, but ... Why don't you use 
> FreeSWITCH instead? It is a bit strange that you ask Asterisk 
> questions here, don't won't help you on their list? Funny, I've heard 
> that before.... :-)
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Thu, Oct 20, 2011 at 9:01 AM, Faisal Rehman 
> <faisal.rehman22 at hotmail.com <mailto:faisal.rehman22 at hotmail.com>> wrote:
>
>     *Hi Naseer,*
>
>     I am going to monitor the ASR and ACD from the Master.csv file of
>     asterisk (/var/log/asterisk/cdr-csv/Master.csv) & actually the
>     basic requirement is that the script or tool should read the whole
>     file and when ASR, ACD is low from the desired requirement, then
>     we should get an email alert or any notification.
>
>
>     Regards,
>
>     *
>     *
>     *Faisal Rehman*
>
>     > From: nbhatti at gmail.com <mailto:nbhatti at gmail.com>
>     > Date: Thu, 20 Oct 2011 14:21:07 +0300
>
>     > To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>     >
>     > Hello Faisal,
>     > We have developed scripts which may do the purpose. Let me know
>     if you
>     > are interested. I would to get more details about your system as
>     well.
>     >
>     > Naseer
>     >
>     > On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
>     > <faisal.rehman22 at hotmail.com
>     <mailto:faisal.rehman22 at hotmail.com>> wrote:
>     > > Hi Sam,
>     > > Will that also generate an email if the ASR & ACD is low from
>     the desired
>     > > requirements & also it is a little expensive :) so any further
>     idea?
>     > >
>     > > Regards,
>     > >
>     > > Faisal Rehman
>     > >
>     > > ________________________________
>     > > Date: Thu, 20 Oct 2011 16:00:08 +0500
>     > > From: govoiper at gmail.com <mailto:govoiper at gmail.com>
>     > > To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     > > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>     > >
>     > > I hope you've already heard about it, not an opensource one
>     but VQManager is
>     > > the best thing so far I've came across.
>     > >
>     > > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman
>     <faisal.rehman22 at hotmail.com <mailto:faisal.rehman22 at hotmail.com>>
>     > > wrote:
>     > >
>     > > Hi Everyone,
>     > >
>     > > First of all I would like to say that I wasn't supposed to ask
>     this question
>     > > here in this forum that belongs to Free-switch, but I think I
>     can find any
>     > > help related to Asterisk ASR & ACD monitoring.
>     > > So I am in search of a reliable open source tool for the real time
>     > > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any
>     help or
>     > > suggestions will be highly appreciated.
>     > >
>     > >
>     > > Regards,
>     > >
>     > > Faisal Rehman
>     > >
>     > >
>     > > FreeSWITCH-users mailing list
>     > > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > > http://www.freeswitch.org
>     > >
>     > >
>     > >
>     > > FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > > http://www.freeswitch.org
>     > >
>     > > FreeSWITCH-users mailing list
>     > > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > > http://www.freeswitch.org
>     > >
>     > >
>     >
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.1831 / Virus Database: 2092/4563 - Release Date: 10/20/11
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From govoiper at gmail.com  Thu Oct 20 16:51:58 2011
From: govoiper at gmail.com (Sammy Govind)
Date: Thu, 20 Oct 2011 17:51:58 +0500
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <BAY167-W112A88856EB58230359FD87ECEB0@phx.gbl>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
	<BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>
	<CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>
	<BAY167-W112A88856EB58230359FD87ECEB0@phx.gbl>
Message-ID: <CAJUJwtjnMfxsauSPp+4uqXxVd+ogNzWLbonS8xkbyBFidYK=bQ@mail.gmail.com>

This thing came to my mind since Faisal said "These are all open source
softwares and linked with each other" - Relating to the CDR CSV manipulation
here, do both Asterisk and FreeSWITCH cdrs correspond to each other? (both
have same dispositions and other fields)

if there be any way to make both (*,FS) dump CDR data into just one common
backend! , this would definitely require to have almost equivalent and same
fields of the CDR structure !


On Thu, Oct 20, 2011 at 5:40 PM, Faisal Rehman
<faisal.rehman22 at hotmail.com>wrote:

>  *Hi,*
>
> These are all open source softwares and linked with each other that is why
> I am getting help from everywhere openly :) By the way I am also using FS
> but my requirement is a bit different indeed. So any ideas about my demand?
>
>
> Regards,
>
> *Faisal Rehman*
>
> ------------------------------
> Date: Thu, 20 Oct 2011 09:34:19 -0300
> From: jmesquita at freeswitch.org
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>
> I am amazed that no one asked that yet, but ... Why don't you use
> FreeSWITCH instead? It is a bit strange that you ask Asterisk questions
> here, don't won't help you on their list? Funny, I've heard that before....
> :-)
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Thu, Oct 20, 2011 at 9:01 AM, Faisal Rehman <
> faisal.rehman22 at hotmail.com> wrote:
>
>  *Hi Naseer,*
>
> I am going to monitor the ASR and ACD from the Master.csv file of asterisk
> (/var/log/asterisk/cdr-csv/Master.csv) & actually the basic requirement is
> that the script or tool should read the whole file and when ASR, ACD is low
> from the desired requirement, then we should get an email alert or any
> notification.
>
>
> Regards,
>
> *
> *
> *Faisal Rehman*
>
> > From: nbhatti at gmail.com
> > Date: Thu, 20 Oct 2011 14:21:07 +0300
>
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> >
> > Hello Faisal,
> > We have developed scripts which may do the purpose. Let me know if you
> > are interested. I would to get more details about your system as well.
> >
> > Naseer
> >
> > On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
> > <faisal.rehman22 at hotmail.com> wrote:
> > > Hi Sam,
> > > Will that also generate an email if the ASR & ACD is low from the
> desired
> > > requirements & also it is a little expensive :) so any further idea?
> > >
> > > Regards,
> > >
> > > Faisal Rehman
> > >
> > > ________________________________
> > > Date: Thu, 20 Oct 2011 16:00:08 +0500
> > > From: govoiper at gmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
> > >
> > > I hope you've already heard about it, not an opensource one but
> VQManager is
> > > the best thing so far I've came across.
> > >
> > > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <
> faisal.rehman22 at hotmail.com>
> > > wrote:
> > >
> > > Hi Everyone,
> > >
> > > First of all I would like to say that I wasn't supposed to ask this
> question
> > > here in this forum that belongs to Free-switch, but I think I can find
> any
> > > help related to Asterisk ASR & ACD monitoring.
> > > So I am in search of a reliable open source tool for the real time
> > > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
> > > suggestions will be highly appreciated.
> > >
> > >
> > > Regards,
> > >
> > > Faisal Rehman
> > >
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nbhatti at gmail.com  Thu Oct 20 16:55:52 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Thu, 20 Oct 2011 15:55:52 +0300
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <CAJUJwtjnMfxsauSPp+4uqXxVd+ogNzWLbonS8xkbyBFidYK=bQ@mail.gmail.com>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
	<BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>
	<CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>
	<BAY167-W112A88856EB58230359FD87ECEB0@phx.gbl>
	<CAJUJwtjnMfxsauSPp+4uqXxVd+ogNzWLbonS8xkbyBFidYK=bQ@mail.gmail.com>
Message-ID: <CACak_ywgqKpQkk0HT+2nQE3e88ZdPfupeV5m9xoLr1EZpA_+VQ@mail.gmail.com>

Try this one, http://www.cdr-stats.org/


On Thu, Oct 20, 2011 at 3:51 PM, Sammy Govind <govoiper at gmail.com> wrote:
> This thing came to my mind since Faisal said "These are all open source
> softwares and linked with each other" - Relating to the CDR CSV manipulation
> here, do both Asterisk and FreeSWITCH cdrs correspond to each other? (both
> have same dispositions and other fields)
> if there be any way to make both (*,FS) dump CDR data into just one common
> backend! , this would definitely require to have almost equivalent and same
> fields of the CDR structure !
>
> On Thu, Oct 20, 2011 at 5:40 PM, Faisal Rehman <faisal.rehman22 at hotmail.com>
> wrote:
>>
>> Hi,
>> These are all open source softwares and linked with each other that is why
>> I am getting help from everywhere openly :) By the way I am also using FS
>> but my requirement is a bit different indeed. So any ideas about my demand?
>>
>> Regards,
>> Faisal Rehman
>>
>> ________________________________
>> Date: Thu, 20 Oct 2011 09:34:19 -0300
>> From: jmesquita at freeswitch.org
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>>
>> I am amazed that no one asked that yet, but ... Why don't you use
>> FreeSWITCH instead? It is a bit strange that you ask Asterisk questions
>> here, don't won't help you on their list? Funny, I've heard that before....
>> :-)
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Thu, Oct 20, 2011 at 9:01 AM, Faisal Rehman
>> <faisal.rehman22 at hotmail.com> wrote:
>>
>> Hi Naseer,
>> I am going to monitor the ASR and ACD from the Master.csv file of asterisk
>> (/var/log/asterisk/cdr-csv/Master.csv) & actually the basic requirement is
>> that the script or tool should read the whole file and when ASR, ACD is low
>> from the desired requirement, then we should get an email alert or any
>> notification.
>>
>> Regards,
>>
>> Faisal Rehman
>>
>> > From: nbhatti at gmail.com
>> > Date: Thu, 20 Oct 2011 14:21:07 +0300
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>> >
>> > Hello Faisal,
>> > We have developed scripts which may do the purpose. Let me know if you
>> > are interested. I would to get more details about your system as well.
>> >
>> > Naseer
>> >
>> > On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
>> > <faisal.rehman22 at hotmail.com> wrote:
>> > > Hi Sam,
>> > > Will that also generate an email if the ASR & ACD is low from the
>> > > desired
>> > > requirements & also it is a little expensive :) so any further idea?
>> > >
>> > > Regards,
>> > >
>> > > Faisal Rehman
>> > >
>> > > ________________________________
>> > > Date: Thu, 20 Oct 2011 16:00:08 +0500
>> > > From: govoiper at gmail.com
>> > > To: freeswitch-users at lists.freeswitch.org
>> > > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>> > >
>> > > I hope you've already heard about it, not an opensource one but
>> > > VQManager is
>> > > the best thing so far I've came across.
>> > >
>> > > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman
>> > > <faisal.rehman22 at hotmail.com>
>> > > wrote:
>> > >
>> > > Hi Everyone,
>> > >
>> > > First of all I would like to say that I wasn't supposed to ask this
>> > > question
>> > > here in this forum that belongs to Free-switch, but I think I can find
>> > > any
>> > > help related to Asterisk ASR & ACD monitoring.
>> > > So I am in search of a reliable open source tool for the real time
>> > > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
>> > > suggestions will be highly appreciated.
>> > >
>> > >
>> > > Regards,
>> > >
>> > > Faisal Rehman
>> > >
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> > >
>> > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 eagle.antonio at gmail.com  Thu Oct 20 17:06:04 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 20 Oct 2011 13:06:04 +0000
Subject: [Freeswitch-users] ASR & ACD Analysis & Monitoring
In-Reply-To: <4EA01821.20503@anew.com.ve>
References: <BAY167-W1156E4CB52853464A53A162ECEB0@phx.gbl>
	<CAJUJwth2bWyPWjfYuCi8HhbJf8JBov31i6SnFaK8e6OycapQuQ@mail.gmail.com>
	<BAY167-W50E80D71BFE194F9DA7591ECEB0@phx.gbl>
	<CACak_ywoU0c5cr5A_KmM=SZSdyh2HR0veY-Kf6a5KbHKCRPYEA@mail.gmail.com>
	<BAY167-W738F008350B3B469F33B4FECEB0@phx.gbl>
	<CABH8QX3LsDb565OaOnaHeyOwaDbx0QqXkJZYMKR-x3_GbGropQ@mail.gmail.com>
	<4EA01821.20503@anew.com.ve>
Message-ID: <CAF+3j-ujVz1qhnXiHvfE+ZoTwWw7SkTNjeaHvZYQuBedFrorSw@mail.gmail.com>

Hello :)

Well i can only assume this varies on your needs ( data  per client ,per
carrier, per endpoint ) by i assume a Daemon with Event Socket or even
better http://wiki.freeswitch.org/wiki/Mod_event_zmq

+

Poll of Workers ( Using ZMQ) would be able to generate the data in realtime
( that usually lacks in 90 % of the software out there) , even if you are
trowing thousands of calls at FS.

If you are reading CSV i think Multiprocessing will be able to handle the
load , at least in my opinion i can't see the difficulty in deploying this ,
in the CSV case will only be the DISK IO that may hold you back .. and in
ZMQ the network IO.

Dunno if this was what you asked for :)

P.S Regarding different back ends , this is easily bypassed with a simple
function to "normalize your data"  into a neutral format that is transversal
to all systems and that integrates into your APP :)

I do this to "compile" different provider routes into our LCR DB :)

Regards
A/T



2011/10/20 Saugort Dario Garcia Tovar <dgarcia at anew.com.ve>

>  Good point...
>
> I will ask then...
>
> For freeswicth users,,,
> which tool do you use to monitor freeswicth (ASR, ACD)?
> DO you develop your own monitoring scripts/tools?
> Would like share your experience?
>
> On 10/20/2011 8:04 AM, Jo?o Mesquita wrote:
>
> I am amazed that no one asked that yet, but ... Why don't you use
> FreeSWITCH instead? It is a bit strange that you ask Asterisk questions
> here, don't won't help you on their list? Funny, I've heard that before....
> :-)
>
>  Regards,
> Jo?o Mesquita
>
>
>
> On Thu, Oct 20, 2011 at 9:01 AM, Faisal Rehman <
> faisal.rehman22 at hotmail.com> wrote:
>
>>  *Hi Naseer,*
>>
>>  I am going to monitor the ASR and ACD from the Master.csv file of
>> asterisk (/var/log/asterisk/cdr-csv/Master.csv) & actually the basic
>> requirement is that the script or tool should read the whole file and when
>> ASR, ACD is low from the desired requirement, then we should get an email
>> alert or any notification.
>>
>>
>>  Regards,
>>
>>  *
>> *
>> *Faisal Rehman*
>>
>> > From: nbhatti at gmail.com
>> > Date: Thu, 20 Oct 2011 14:21:07 +0300
>>
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>> >
>> > Hello Faisal,
>> > We have developed scripts which may do the purpose. Let me know if you
>> > are interested. I would to get more details about your system as well.
>> >
>> > Naseer
>> >
>> > On Thu, Oct 20, 2011 at 2:14 PM, Faisal Rehman
>> > <faisal.rehman22 at hotmail.com> wrote:
>> > > Hi Sam,
>> > > Will that also generate an email if the ASR & ACD is low from the
>> desired
>> > > requirements & also it is a little expensive :) so any further idea?
>> > >
>> > > Regards,
>> > >
>> > > Faisal Rehman
>> > >
>> > > ________________________________
>> > > Date: Thu, 20 Oct 2011 16:00:08 +0500
>> > > From: govoiper at gmail.com
>> > > To: freeswitch-users at lists.freeswitch.org
>> > > Subject: Re: [Freeswitch-users] ASR & ACD Analysis & Monitoring
>> > >
>> > > I hope you've already heard about it, not an opensource one but
>> VQManager is
>> > > the best thing so far I've came across.
>> > >
>> > > On Thu, Oct 20, 2011 at 2:21 PM, Faisal Rehman <
>> faisal.rehman22 at hotmail.com>
>> > > wrote:
>> > >
>> > > Hi Everyone,
>> > >
>> > > First of all I would like to say that I wasn't supposed to ask this
>> question
>> > > here in this forum that belongs to Free-switch, but I think I can find
>> any
>> > > help related to Asterisk ASR & ACD monitoring.
>> > > So I am in search of a reliable open source tool for the real time
>> > > monitoring of ASR & ACD (Using Asterisk's Master.csv), so any help or
>> > > suggestions will be highly appreciated.
>> > >
>> > >
>> > > Regards,
>> > >
>> > > Faisal Rehman
>> > >
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> > >
>> > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.1831 / Virus Database: 2092/4563 - Release Date: 10/20/11
>
>
>
> --
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From randy.andrade at gmail.com  Thu Oct 20 18:16:22 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Thu, 20 Oct 2011 10:16:22 -0400
Subject: [Freeswitch-users] BLF not working
In-Reply-To: <1C748D48EFD36B4AA0B934E8B4E2998003615A0B@ipi-cc-srv04.ipinfrastructures.com>
References: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>
	<-5270605825443932569@unknownmsgid>
	<1C748D48EFD36B4AA0B934E8B4E2998003615A0B@ipi-cc-srv04.ipinfrastructures.com>
Message-ID: <CAJZcYrfMSoWGkDp9jsZc1PJmVxvNfLXFbCie742cd7eOTbQiow@mail.gmail.com>

Mike,

I'd have to play with the various timer settings against my FS box at home,
to say for sure. It sounds like 7.4.9a may be a solid firmware after all.
Besides yourself having good luck with it, some co-workers have just
returned from a conference where they spoke with other organizations using
the same back-end that we are who have had fantastic success with it; it
seems our servers may be out of date on some patches that are more critical
than they seemed. We'll be updating our servers over the next couple weeks
and bringing them up to date with their various software loads, so hopefully
that will show that the issues were not in the Cisco firmware after all. I
would be very happy to move to 7.4.9a, as it addresses several outstanding
issues with the 7.4.8a firmware that we're currently running as standard.

I'll let you know how I make out with the BLF's & timers etc after I've had
a chance to play with them more after work, tonight.

Randy

On Thu, Oct 20, 2011 at 7:56 AM, Mike Hammond <mhammond at gcicom.net> wrote:

> Hi Randy,
>
> funnily enough I've just upgraded the phone this morning to 7.4.9.a
> after digging around on Cisco's website last night and I have now got
> some functionality on the 504... yay !
>
> I'm now working through some tests to see what works and what doesn't
> with Fusionpbx/FS.
>
> I've started by removing all of the default * feature codes from the
> handset itself and set just **# against the pickup code in the attendant
> screen and I can now intercept a ringing call by hitting the flashing
> blf key which is cool.
>
> I get grn solid for idle.
>
> Red solid when busy on a call.
>
> Red flashing when ringing.
>
> DND doesn't seem to affect the BLF though which is a shame as that would
> be nice.
>
> I assume that's as good as it gets as unplugging one of the monitored
> extensions and waiting for the thing to drop it's registration doesn't
> seem to make any difference ? although I assume there will be some sort
> of subscription timer involved that FS hasn't reached yet ? given you've
> obviously been through the mill with cisco's before, do you have any
> recommendations for timer settings on the handsets for things like NAT
> keep alives and subscriptions etc ?
>
> Thanks, Mike
>
>
>
>
> -----Original Message-----
> From: Randy Andrade [mailto:randy.andrade at gmail.com]
> Sent: 19 October 2011 23:42
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] BLF not working
>
> I'm curious what version of firmware you're running on your 504's. I
> work with them professionally and have found that certain functions
> work better / don't work depending on your version of firmware.
> Personally I've found 7.4.8a to be the best stability / feature
> balance. 7.4.9a was just released and though it fixes a few bugs,
> especiallly related to BLF's, against our servers a lot of more basic
> functionality, such as blind transfers and others which were related
> to the 500S sidecar, but we weren't testing against FreeSwitch, and we
> were testing SPA509G's. Perhaps you'll have better luck with it? Or
> better luck not using it, if you are? Point I'm trying to make is,
> Cisco has been known to break some functionality in virtually every
> recent release of the firmware, so try a couple different revisions,
> because your code looks correct. I use it on my Linksys SPA942's and
> Cisco SPA508's, which both have working BLF's against my FreeSwitch
> box, at home. The SPA508's are running FW v7.4.8a with the BLF's
> working fine.
>
> Randy
>
> Sent from my iPhone
>
> On Oct 19, 2011, at 5:46 PM, Mike Hammond <mhammond at gcicom.net> wrote:
>
> > Hello all,
> >
> > I'm new to FS and I'm having a problem getting BLF working, initially
> I'm trying with a Cisco SPA504G.
> >
> > I've configured a test switch directly on a public IP using the latest
> download (about a week ago) with a fusionpbx front end gui, running on
> an ubuntu 10.04 LTS server.
> >
> > I've set it up for multi tenant and have had some domain names pointed
> at my public IP, I can register and use handsets and gateways without an
> issue on the different tenants offsite from behind nat'd routers.
> >
> > Next I wanted to try BLF, so I've configured a spare key on the Cisco
> handset (extn 204) with fnc=blf+sd+cp;sub=202@$PROXY;ext=202@$PROXY to
> monitor extension 202 which is within the same tenant domain. But the
> key is glowing orange so obviously not happy
> >
> > If set sofia to debug presence and turn on the sip trace I see the
> following messaging
> >
> > http://pastebin.freeswitch.org/17557
> >
> > Can anyone help me with this one as I'm struggling to know what to do
> next other than strip it all back out and start again with clean a
> single domain setup.
> >
> > Many thanks,
> >
> >
> >
> >
> > Mike Hammond
> >
> >
> > GCI Com
> > Cedar Court Office Park
> > Denby Dale Road
> > Calder Grove
> > Wakefield
> > WF4 3QZ
> >
> > Office:   0844 443 3537
> > Fax:      0844 443 3540
> > http://www.gcicom.net/
> >
> >
> > This email has been swept by Webroot for viruses. Any files
> transmitted with it are confidential and intended solely for the email
> recipient. If you are not the intended recipient please delete this
> email immediately. Be aware that any disclosure, copying, distribution
> or use of the contents of this information is prohibited. If you have
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> Please note that any views or opinions presented in this email are
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> company. Finally, the recipient should check this email and any
> attachments for the presence of viruses.
> >
> >
> > GCI Com incorporates the following Group Companies:
> > GCI Telecom Group Limited Reg. No. 5396496, Edge Telecommunications
> Ltd Reg. No. 5748740, Edge Telecom Ltd Reg. No. 3101247, IP
> Infrastructures Ltd Reg. No. 4657026, Invomo Ltd Reg. No. 6267056,
> NetServices UK Ltd Reg. No. 7118768, WAN Services Ltd Reg. No. 4082862.
> All Registered in England and Wales, Registered Office: Global House, 2
> Crofton Close, Lincoln, LN3 4NT
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
> Mike Hammond
>
>
> GCI Com
> Cedar Court Office Park
> Denby Dale Road
> Calder Grove
> Wakefield
> WF4 3QZ
>
> Office:   0844 443 3537
> Fax:      0844 443 3540
> http://www.gcicom.net/
>
>
> FreeSWITCH-users mailing 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 jeff at jefflenk.com  Thu Oct 20 19:09:34 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 20 Oct 2011 08:09:34 -0700 (PDT)
Subject: [Freeswitch-users] Fs Core Dumps On Mod Python
In-Reply-To: <CAF+3j-sLP6WG=pf4FodF6n_h2gcf0h9doEAYGoB7hgWUFiSL3g@mail.gmail.com>
References: <CAF+3j-sLP6WG=pf4FodF6n_h2gcf0h9doEAYGoB7hgWUFiSL3g@mail.gmail.com>
Message-ID: <1319123374635-6913252.post@n2.nabble.com>

This is resolved in latest Git

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Fs-Core-Dumps-On-Mod-Python-tp6912325p6913252.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From basit.engg at gmail.com  Thu Oct 20 19:20:34 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Thu, 20 Oct 2011 20:20:34 +0500
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
Message-ID: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>

Hi!

I am trying to compile freeswitch-1.0.6. while make following error occur.

Any suggestion why its happening? Solution....


creating libpcreposix.la
(cd .libs && rm -f libpcreposix.la && ln -s ../libpcreposix.la
libpcreposix.la)
if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
-I.     -g -O2 -MT pcrecpp.lo -MD -MP -MF ".deps/pcrecpp.Tpo" -c -o
pcrecpp.lo pcrecpp.cc; \
        then mv -f ".deps/pcrecpp.Tpo" ".deps/pcrecpp.Plo"; else rm -f
".deps/pcrecpp.Tpo"; exit 1; fi
 g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcrecpp.lo -MD -MP -MF
.deps/pcrecpp.Tpo -c pcrecpp.cc  -fPIC -DPIC -o pcrecpp.o
if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
-I.     -g -O2 -MT pcre_scanner.lo -MD -MP -MF ".deps/pcre_scanner.Tpo" -c
-o pcre_scanner.l
o pcre_scanner.cc; \
        then mv -f ".deps/pcre_scanner.Tpo" ".deps/pcre_scanner.Plo"; else
rm -f ".deps/pcre_scanner.Tpo"; exit 1; fi
 g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_scanner.lo -MD -MP -MF
.deps/pcre_scanner.Tpo -c pcre_scanner.cc  -fPIC -DPIC -o pcre_scanner.o
if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
-I.     -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF
".deps/pcre_stringpiece.Tpo" -c -o pcre_s
tringpiece.lo pcre_stringpiece.cc; \
        then mv -f ".deps/pcre_stringpiece.Tpo"
".deps/pcre_stringpiece.Plo"; else rm -f ".deps/pcre_stringpiece.Tpo"; exit
1; fi
 g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF
.deps/pcre_stringpiece.Tpo -c pcre_stringpiece.cc  -fPIC -DPIC -o
pcre_stringpiece.o
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o
libpcrecpp.la-rpath /usr/local/freeswitch/lib -version-info 0:0:0
   pcrecpp.lo pcre_scanner.lo pcre_stringpiece.lo libpcre.la
rm -fr  .libs/libpcrecpp.a .libs/libpcrecpp.la .libs/libpcrecpp.lai
ar cru .libs/libpcrecpp.a  pcrecpp.o pcre_scanner.o pcre_stringpiece.o
ranlib .libs/libpcrecpp.a
creating libpcrecpp.la
(cd .libs && rm -f libpcrecpp.la && ln -s ../libpcrecpp.la libpcrecpp.la)
if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcretest.o -MD -MP -MF
".deps/pcretest.Tpo" -c -o pcretest.o pcretest.c; \
        then mv -f ".deps/pcretest.Tpo" ".deps/pcretest.Po"; else rm -f
".deps/pcretest.Tpo"; exit 1; fi
pcretest.c: In function ?extend_inputline?:
pcretest.c:253:7: warning: format not a string literal and no format
arguments [-Wformat-security]
/bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcretest
 pcretest.o libpcreposix.la
gcc -g -O2 -o pcretest pcretest.o  ./.libs/libpcreposix.a
/usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcregrep.o -MD -MP -MF
".deps/pcregrep.Tpo" -c -o pcregrep.o pcregrep.c; \
        then mv -f ".deps/pcregrep.Tpo" ".deps/pcregrep.Po"; else rm -f
".deps/pcregrep.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcregrep
 pcregrep.o libpcreposix.la
gcc -g -O2 -o pcregrep pcregrep.o  ./.libs/libpcreposix.a
/usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcrecpp_unittest.o -MD -MP
-MF ".deps/pcrecpp_unittest.Tpo" -c -o pcrecpp_unittest.o
pcrecpp_unittest.cc; \
        then mv -f ".deps/pcrecpp_unittest.Tpo" ".deps/pcrecpp_unittest.Po";
else rm -f ".deps/pcrecpp_unittest.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o pcrecpp_unittest
 pcrecpp_unittest.o libpcrecpp.la
g++ -g -O2 -o pcrecpp_unittest pcrecpp_unittest.o  ./.libs/libpcrecpp.a
/usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcre_scanner_unittest.o
-MD -MP -MF ".deps/pcre_scanner_unittest.Tpo" -c -o pcre_scanner_unittest.o
pcre_scanner_unittest.cc; \
        then mv -f ".deps/pcre_scanner_unittest.Tpo"
".deps/pcre_scanner_unittest.Po"; else rm -f
".deps/pcre_scanner_unittest.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o
pcre_scanner_unittest  pcre_scanner_unittest.o libpcrecpp.la
g++ -g -O2 -o pcre_scanner_unittest pcre_scanner_unittest.o
 ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT
pcre_stringpiece_unittest.o -MD -MP -MF
".deps/pcre_stringpiece_unittest.Tpo" -c -o pcre_stringpiece_unittest.o
pcre_stringpiece_unittest.cc; \
        then mv -f ".deps/pcre_stringpiece_unittest.Tpo"
".deps/pcre_stringpiece_unittest.Po"; else rm -f
".deps/pcre_stringpiece_unittest.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o
pcre_stringpiece_unittest  pcre_stringpiece_unittest.o libpcrecpp.la
g++ -g -O2 -o pcre_stringpiece_unittest pcre_stringpiece_unittest.o
 ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
make[2]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'
make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'
make[1]: Entering directory `/usr/src/freeswitch-1.0.6/libs/srtp'
make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/srtp'
gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
-DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
-DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null |
grep -v "^#" > src/include/switch_swigable_cpp.h
make: *** [src/include/switch_swigable_cpp.h] Error 1
root at ubuntufs:/usr/src/freeswitch-1.0.6#
root at ubuntufs:/usr/src/freeswitch-1.0.6#

Last action

gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
-DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
-DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null |
grep -v "^#" > src/include/switch_swigable_cpp.h

is returning empty src/include/switch_swigable_cpp.h file



-- 
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
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From basit.engg at gmail.com  Thu Oct 20 19:24:59 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Thu, 20 Oct 2011 20:24:59 +0500
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
Message-ID: <CAGQG5MaSkt7B6wJc_22Sr00TapszhAszo451nkVhUpsbx6W0Nw@mail.gmail.com>

just to add

root at ubuntufs:~# uname -a
Linux ubuntufs 3.0.0-12-generic-pae #20-Ubuntu SMP Fri Oct 7 16:37:17 UTC
2011 i686 i686 i386 GNU/Linux


On Thu, Oct 20, 2011 at 8:20 PM, Abdul Basit <basit.engg at gmail.com> wrote:

> Hi!
>
> I am trying to compile freeswitch-1.0.6. while make following error occur.
>
> Any suggestion why its happening? Solution....
>
>
> creating libpcreposix.la
> (cd .libs && rm -f libpcreposix.la && ln -s ../libpcreposix.la
> libpcreposix.la)
> if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
> -I.     -g -O2 -MT pcrecpp.lo -MD -MP -MF ".deps/pcrecpp.Tpo" -c -o
> pcrecpp.lo pcrecpp.cc; \
>         then mv -f ".deps/pcrecpp.Tpo" ".deps/pcrecpp.Plo"; else rm -f
> ".deps/pcrecpp.Tpo"; exit 1; fi
>  g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcrecpp.lo -MD -MP -MF
> .deps/pcrecpp.Tpo -c pcrecpp.cc  -fPIC -DPIC -o pcrecpp.o
> if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
> -I.     -g -O2 -MT pcre_scanner.lo -MD -MP -MF ".deps/pcre_scanner.Tpo" -c
> -o pcre_scanner.l
> o pcre_scanner.cc; \
>         then mv -f ".deps/pcre_scanner.Tpo" ".deps/pcre_scanner.Plo"; else
> rm -f ".deps/pcre_scanner.Tpo"; exit 1; fi
>  g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_scanner.lo -MD -MP -MF
> .deps/pcre_scanner.Tpo -c pcre_scanner.cc  -fPIC -DPIC -o pcre_scanner.o
> if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
> -I.     -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF
> ".deps/pcre_stringpiece.Tpo" -c -o pcre_s
> tringpiece.lo pcre_stringpiece.cc; \
>         then mv -f ".deps/pcre_stringpiece.Tpo"
> ".deps/pcre_stringpiece.Plo"; else rm -f ".deps/pcre_stringpiece.Tpo"; exit
> 1; fi
>  g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF
> .deps/pcre_stringpiece.Tpo -c pcre_stringpiece.cc  -fPIC -DPIC -o
> pcre_stringpiece.o
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o libpcrecpp.la-rpath /usr/local/freeswitch/lib -version-info 0:0:0
>    pcrecpp.lo pcre_scanner.lo pcre_stringpiece.lo libpcre.la
> rm -fr  .libs/libpcrecpp.a .libs/libpcrecpp.la .libs/libpcrecpp.lai
> ar cru .libs/libpcrecpp.a  pcrecpp.o pcre_scanner.o pcre_stringpiece.o
> ranlib .libs/libpcrecpp.a
> creating libpcrecpp.la
> (cd .libs && rm -f libpcrecpp.la && ln -s ../libpcrecpp.la libpcrecpp.la)
> if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcretest.o -MD -MP -MF
> ".deps/pcretest.Tpo" -c -o pcretest.o pcretest.c; \
>         then mv -f ".deps/pcretest.Tpo" ".deps/pcretest.Po"; else rm -f
> ".deps/pcretest.Tpo"; exit 1; fi
> pcretest.c: In function ?extend_inputline?:
> pcretest.c:253:7: warning: format not a string literal and no format
> arguments [-Wformat-security]
> /bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcretest
>  pcretest.o libpcreposix.la
> gcc -g -O2 -o pcretest pcretest.o  ./.libs/libpcreposix.a
> /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
> if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcregrep.o -MD -MP -MF
> ".deps/pcregrep.Tpo" -c -o pcregrep.o pcregrep.c; \
>         then mv -f ".deps/pcregrep.Tpo" ".deps/pcregrep.Po"; else rm -f
> ".deps/pcregrep.Tpo"; exit 1; fi
> /bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcregrep
>  pcregrep.o libpcreposix.la
> gcc -g -O2 -o pcregrep pcregrep.o  ./.libs/libpcreposix.a
> /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
> if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcrecpp_unittest.o -MD
> -MP -MF ".deps/pcrecpp_unittest.Tpo" -c -o pcrecpp_unittest.o
> pcrecpp_unittest.cc; \
>         then mv -f ".deps/pcrecpp_unittest.Tpo"
> ".deps/pcrecpp_unittest.Po"; else rm -f ".deps/pcrecpp_unittest.Tpo"; exit
> 1; fi
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o pcrecpp_unittest
>  pcrecpp_unittest.o libpcrecpp.la
> g++ -g -O2 -o pcrecpp_unittest pcrecpp_unittest.o  ./.libs/libpcrecpp.a
> /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
> if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcre_scanner_unittest.o
> -MD -MP -MF ".deps/pcre_scanner_unittest.Tpo" -c -o pcre_scanner_unittest.o
> pcre_scanner_unittest.cc; \
>         then mv -f ".deps/pcre_scanner_unittest.Tpo"
> ".deps/pcre_scanner_unittest.Po"; else rm -f
> ".deps/pcre_scanner_unittest.Tpo"; exit 1; fi
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o
> pcre_scanner_unittest  pcre_scanner_unittest.o libpcrecpp.la
> g++ -g -O2 -o pcre_scanner_unittest pcre_scanner_unittest.o
>  ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
> if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT
> pcre_stringpiece_unittest.o -MD -MP -MF
> ".deps/pcre_stringpiece_unittest.Tpo" -c -o pcre_stringpiece_unittest.o
> pcre_stringpiece_unittest.cc; \
>         then mv -f ".deps/pcre_stringpiece_unittest.Tpo"
> ".deps/pcre_stringpiece_unittest.Po"; else rm -f
> ".deps/pcre_stringpiece_unittest.Tpo"; exit 1; fi
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o
> pcre_stringpiece_unittest  pcre_stringpiece_unittest.o libpcrecpp.la
> g++ -g -O2 -o pcre_stringpiece_unittest pcre_stringpiece_unittest.o
>  ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
> make[2]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'
> make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'
> make[1]: Entering directory `/usr/src/freeswitch-1.0.6/libs/srtp'
> make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/srtp'
> gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
> -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
> -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null |
> grep -v "^#" > src/include/switch_swigable_cpp.h
> make: *** [src/include/switch_swigable_cpp.h] Error 1
> root at ubuntufs:/usr/src/freeswitch-1.0.6#
> root at ubuntufs:/usr/src/freeswitch-1.0.6#
>
> Last action
>
> gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
> -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
> -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null |
> grep -v "^#" > src/include/switch_swigable_cpp.h
>
> is returning empty src/include/switch_swigable_cpp.h file
>
>
>
> --
> Regards,
>
> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
>



-- 
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
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From peter.olsson at visionutveckling.se  Thu Oct 20 19:30:56 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 20 Oct 2011 17:30:56 +0200
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>

My suggestion is to use the current git HEAD. 1.0.6 is really old by now...

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Abdul Basit
Skickat: den 20 oktober 2011 17:21
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] error while compiling switch_swigable_cpp.h

Hi!

I am trying to compile freeswitch-1.0.6. while make following error occur.

Any suggestion why its happening? Solution....


creating libpcreposix.la<http://libpcreposix.la>
(cd .libs && rm -f libpcreposix.la<http://libpcreposix.la> && ln -s ../libpcreposix.la<http://libpcreposix.la> libpcreposix.la<http://libpcreposix.la>)
if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcrecpp.lo -MD -MP -MF ".deps/pcrecpp.Tpo" -c -o pcrecpp.lo pcrecpp.cc; \
        then mv -f ".deps/pcrecpp.Tpo" ".deps/pcrecpp.Plo"; else rm -f ".deps/pcrecpp.Tpo"; exit 1; fi
 g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcrecpp.lo -MD -MP -MF .deps/pcrecpp.Tpo -c pcrecpp.cc  -fPIC -DPIC -o pcrecpp.o
if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcre_scanner.lo -MD -MP -MF ".deps/pcre_scanner.Tpo" -c -o pcre_scanner.l
o pcre_scanner.cc; \
        then mv -f ".deps/pcre_scanner.Tpo" ".deps/pcre_scanner.Plo"; else rm -f ".deps/pcre_scanner.Tpo"; exit 1; fi
 g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_scanner.lo -MD -MP -MF .deps/pcre_scanner.Tpo -c pcre_scanner.cc  -fPIC -DPIC -o pcre_scanner.o
if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF ".deps/pcre_stringpiece.Tpo" -c -o pcre_s
tringpiece.lo pcre_stringpiece.cc; \
        then mv -f ".deps/pcre_stringpiece.Tpo" ".deps/pcre_stringpiece.Plo"; else rm -f ".deps/pcre_stringpiece.Tpo"; exit 1; fi
 g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF .deps/pcre_stringpiece.Tpo -c pcre_stringpiece.cc  -fPIC -DPIC -o pcre_stringpiece.o
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o libpcrecpp.la<http://libpcrecpp.la> -rpath /usr/local/freeswitch/lib -version-info 0:0:0                            pcrecpp.lo pcre_scanner.lo pcre_stringpiece.lo libpcre.la<http://libpcre.la>
rm -fr  .libs/libpcrecpp.a .libs/libpcrecpp.la<http://libpcrecpp.la> .libs/libpcrecpp.lai
ar cru .libs/libpcrecpp.a  pcrecpp.o pcre_scanner.o pcre_stringpiece.o
ranlib .libs/libpcrecpp.a
creating libpcrecpp.la<http://libpcrecpp.la>
(cd .libs && rm -f libpcrecpp.la<http://libpcrecpp.la> && ln -s ../libpcrecpp.la<http://libpcrecpp.la> libpcrecpp.la<http://libpcrecpp.la>)
if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcretest.o -MD -MP -MF ".deps/pcretest.Tpo" -c -o pcretest.o pcretest.c; \
        then mv -f ".deps/pcretest.Tpo" ".deps/pcretest.Po"; else rm -f ".deps/pcretest.Tpo"; exit 1; fi
pcretest.c: In function ?extend_inputline?:
pcretest.c:253:7: warning: format not a string literal and no format arguments [-Wformat-security]
/bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcretest  pcretest.o libpcreposix.la<http://libpcreposix.la>
gcc -g -O2 -o pcretest pcretest.o  ./.libs/libpcreposix.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcregrep.o -MD -MP -MF ".deps/pcregrep.Tpo" -c -o pcregrep.o pcregrep.c; \
        then mv -f ".deps/pcregrep.Tpo" ".deps/pcregrep.Po"; else rm -f ".deps/pcregrep.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcregrep  pcregrep.o libpcreposix.la<http://libpcreposix.la>
gcc -g -O2 -o pcregrep pcregrep.o  ./.libs/libpcreposix.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcrecpp_unittest.o -MD -MP -MF ".deps/pcrecpp_unittest.Tpo" -c -o pcrecpp_unittest.o pcrecpp_unittest.cc; \
        then mv -f ".deps/pcrecpp_unittest.Tpo" ".deps/pcrecpp_unittest.Po"; else rm -f ".deps/pcrecpp_unittest.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o pcrecpp_unittest  pcrecpp_unittest.o libpcrecpp.la<http://libpcrecpp.la>
g++ -g -O2 -o pcrecpp_unittest pcrecpp_unittest.o  ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcre_scanner_unittest.o -MD -MP -MF ".deps/pcre_scanner_unittest.Tpo" -c -o pcre_scanner_unittest.o pcre_scanner_unittest.cc; \
        then mv -f ".deps/pcre_scanner_unittest.Tpo" ".deps/pcre_scanner_unittest.Po"; else rm -f ".deps/pcre_scanner_unittest.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o pcre_scanner_unittest  pcre_scanner_unittest.o libpcrecpp.la<http://libpcrecpp.la>
g++ -g -O2 -o pcre_scanner_unittest pcre_scanner_unittest.o  ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcre_stringpiece_unittest.o -MD -MP -MF ".deps/pcre_stringpiece_unittest.Tpo" -c -o pcre_stringpiece_unittest.o pcre_stringpiece_unittest.cc; \
        then mv -f ".deps/pcre_stringpiece_unittest.Tpo" ".deps/pcre_stringpiece_unittest.Po"; else rm -f ".deps/pcre_stringpiece_unittest.Tpo"; exit 1; fi
/bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o pcre_stringpiece_unittest  pcre_stringpiece_unittest.o libpcrecpp.la<http://libpcrecpp.la>
g++ -g -O2 -o pcre_stringpiece_unittest pcre_stringpiece_unittest.o  ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a
make[2]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'
make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'
make[1]: Entering directory `/usr/src/freeswitch-1.0.6/libs/srtp'
make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/srtp'
gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h
make: *** [src/include/switch_swigable_cpp.h] Error 1
root at ubuntufs:/usr/src/freeswitch-1.0.6#
root at ubuntufs:/usr/src/freeswitch-1.0.6#

Last action

gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h

is returning empty src/include/switch_swigable_cpp.h file



--
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
!DSPAM:4ea03c8532762022373163!
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From basit.engg at gmail.com  Thu Oct 20 19:41:55 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Thu, 20 Oct 2011 20:41:55 +0500
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>
Message-ID: <CAGQG5Ma1MrSYVWvRcRAWjby9DP2m5WRQ4=6=UNESbNsnyh1aSQ@mail.gmail.com>

yes. but 1.0.6 was quite good as i heard. This might be some simple issue.

Also i noticed

root at ubuntufs:/usr/src/freeswitch-1.0.6# gcc -E
/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS=
-DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR=
-DSWITCH_DECLARE_NONSTD\(x\)=x
# 1 "/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h"
# 1 "<built-in>"
# 1 "<command-line>"
# 1 "/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h"
/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h:11:20: fatal error:
switch.h: No such file or directory
compilation terminated.

both files are present at
/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
/usr/src/freeswitch-1.0.6/src/include/switch.h

any idea?


On Thu, Oct 20, 2011 at 8:30 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> My suggestion is to use the current git HEAD. 1.0.6 is really old by now...
> ****
>
> ** **
>
> /Peter****
>
> ** **
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Abdul Basit
> *Skickat:* den 20 oktober 2011 17:21
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] error while compiling switch_swigable_cpp.h****
>
> ** **
>
> Hi!****
>
> ** **
>
> I am trying to compile freeswitch-1.0.6. while make following error occur.
> ****
>
> ** **
>
> Any suggestion why its happening? Solution.... ****
>
> ** **
>
> ** **
>
> creating libpcreposix.la****
>
> (cd .libs && rm -f libpcreposix.la && ln -s ../libpcreposix.la
> libpcreposix.la)****
>
> if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
> -I.     -g -O2 -MT pcrecpp.lo -MD -MP -MF ".deps/pcrecpp.Tpo" -c -o
> pcrecpp.lo pcrecpp.cc; \****
>
>         then mv -f ".deps/pcrecpp.Tpo" ".deps/pcrecpp.Plo"; else rm -f
> ".deps/pcrecpp.Tpo"; exit 1; fi****
>
>  g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcrecpp.lo -MD -MP -MF
> .deps/pcrecpp.Tpo -c pcrecpp.cc  -fPIC -DPIC -o pcrecpp.o****
>
> if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
> -I.     -g -O2 -MT pcre_scanner.lo -MD -MP -MF ".deps/pcre_scanner.Tpo" -c
> -o pcre_scanner.l****
>
> o pcre_scanner.cc; \****
>
>         then mv -f ".deps/pcre_scanner.Tpo" ".deps/pcre_scanner.Plo"; else
> rm -f ".deps/pcre_scanner.Tpo"; exit 1; fi****
>
>  g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_scanner.lo -MD -MP -MF
> .deps/pcre_scanner.Tpo -c pcre_scanner.cc  -fPIC -DPIC -o pcre_scanner.o**
> **
>
> if /bin/bash ./libtool --tag=CXX --mode=compile g++ -DHAVE_CONFIG_H -I. -I.
> -I.     -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF
> ".deps/pcre_stringpiece.Tpo" -c -o pcre_s****
>
> tringpiece.lo pcre_stringpiece.cc; \****
>
>         then mv -f ".deps/pcre_stringpiece.Tpo"
> ".deps/pcre_stringpiece.Plo"; else rm -f ".deps/pcre_stringpiece.Tpo"; exit
> 1; fi****
>
>  g++ -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT pcre_stringpiece.lo -MD -MP -MF
> .deps/pcre_stringpiece.Tpo -c pcre_stringpiece.cc  -fPIC -DPIC -o
> pcre_stringpiece.o****
>
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o libpcrecpp.la-rpath /usr/local/freeswitch/lib -version-info 0:0:0
>    pcrecpp.lo pcre_scanner.lo pcre_stringpiece.lo libpcre.la****
>
> rm -fr  .libs/libpcrecpp.a .libs/libpcrecpp.la .libs/libpcrecpp.lai****
>
> ar cru .libs/libpcrecpp.a  pcrecpp.o pcre_scanner.o pcre_stringpiece.o****
>
> ranlib .libs/libpcrecpp.a****
>
> creating libpcrecpp.la****
>
> (cd .libs && rm -f libpcrecpp.la && ln -s ../libpcrecpp.la libpcrecpp.la)*
> ***
>
> if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcretest.o -MD -MP -MF
> ".deps/pcretest.Tpo" -c -o pcretest.o pcretest.c; \****
>
>         then mv -f ".deps/pcretest.Tpo" ".deps/pcretest.Po"; else rm -f
> ".deps/pcretest.Tpo"; exit 1; fi****
>
> pcretest.c: In function ?extend_inputline?:****
>
> pcretest.c:253:7: warning: format not a string literal and no format
> arguments [-Wformat-security]****
>
> /bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcretest
>  pcretest.o libpcreposix.la****
>
> gcc -g -O2 -o pcretest pcretest.o  ./.libs/libpcreposix.a
> /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a****
>
> if gcc -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcregrep.o -MD -MP -MF
> ".deps/pcregrep.Tpo" -c -o pcregrep.o pcregrep.c; \****
>
>         then mv -f ".deps/pcregrep.Tpo" ".deps/pcregrep.Po"; else rm -f
> ".deps/pcregrep.Tpo"; exit 1; fi****
>
> /bin/bash ./libtool --tag=CC --mode=link gcc  -g -O2   -o pcregrep
>  pcregrep.o libpcreposix.la****
>
> gcc -g -O2 -o pcregrep pcregrep.o  ./.libs/libpcreposix.a
> /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a****
>
> if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcrecpp_unittest.o -MD
> -MP -MF ".deps/pcrecpp_unittest.Tpo" -c -o pcrecpp_unittest.o
> pcrecpp_unittest.cc; \****
>
>         then mv -f ".deps/pcrecpp_unittest.Tpo"
> ".deps/pcrecpp_unittest.Po"; else rm -f ".deps/pcrecpp_unittest.Tpo"; exit
> 1; fi****
>
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o pcrecpp_unittest
>  pcrecpp_unittest.o libpcrecpp.la****
>
> g++ -g -O2 -o pcrecpp_unittest pcrecpp_unittest.o  ./.libs/libpcrecpp.a
> /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a****
>
> if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT pcre_scanner_unittest.o
> -MD -MP -MF ".deps/pcre_scanner_unittest.Tpo" -c -o pcre_scanner_unittest.o
> pcre_scanner_unittest.cc; \****
>
>         then mv -f ".deps/pcre_scanner_unittest.Tpo"
> ".deps/pcre_scanner_unittest.Po"; else rm -f
> ".deps/pcre_scanner_unittest.Tpo"; exit 1; fi****
>
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o
> pcre_scanner_unittest  pcre_scanner_unittest.o libpcrecpp.la****
>
> g++ -g -O2 -o pcre_scanner_unittest pcre_scanner_unittest.o
>  ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a*
> ***
>
> if g++ -DHAVE_CONFIG_H -I. -I. -I.     -g -O2 -MT
> pcre_stringpiece_unittest.o -MD -MP -MF
> ".deps/pcre_stringpiece_unittest.Tpo" -c -o pcre_stringpiece_unittest.o
> pcre_stringpiece_unittest.cc; \****
>
>         then mv -f ".deps/pcre_stringpiece_unittest.Tpo"
> ".deps/pcre_stringpiece_unittest.Po"; else rm -f
> ".deps/pcre_stringpiece_unittest.Tpo"; exit 1; fi****
>
> /bin/bash ./libtool --tag=CXX --mode=link g++  -g -O2   -o
> pcre_stringpiece_unittest  pcre_stringpiece_unittest.o libpcrecpp.la****
>
> g++ -g -O2 -o pcre_stringpiece_unittest pcre_stringpiece_unittest.o
>  ./.libs/libpcrecpp.a /usr/src/freeswitch-1.0.6/libs/pcre/.libs/libpcre.a*
> ***
>
> make[2]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'****
>
> make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/pcre'****
>
> make[1]: Entering directory `/usr/src/freeswitch-1.0.6/libs/srtp'****
>
> make[1]: Leaving directory `/usr/src/freeswitch-1.0.6/libs/srtp'****
>
> gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
> -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
> -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null |
> grep -v "^#" > src/include/switch_swigable_cpp.h****
>
> make: *** [src/include/switch_swigable_cpp.h] Error 1****
>
> root at ubuntufs:/usr/src/freeswitch-1.0.6#****
>
> root at ubuntufs:/usr/src/freeswitch-1.0.6#****
>
> ** **
>
> Last action ****
>
> ** **
>
> gcc -E /usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
> -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
> -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null |
> grep -v "^#" > src/include/switch_swigable_cpp.h****
>
> ** **
>
> is returning empty src/include/switch_swigable_cpp.h file****
>
> ** **
>
> ** **
>
> ** **
>
> -- ****
>
> Regards,****
>
>
> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
> !DSPAM:4ea03c8532762022373163! ****
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 | +92 32 1416 4196 | +92 30 0841 1445
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From infos at madovsky.org  Thu Oct 20 19:58:35 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 20 Oct 2011 11:58:35 -0400
Subject: [Freeswitch-users] transfer_ringback from a conference
References: <B55CB35F9A1746669F128729A3E9B728@e1705>
	<CAMwZ-tKZCZXxxgYQXD+3iyVE3b2u12j=B_reJZ_XuBR1YWctvA@mail.gmail.com>
Message-ID: <E6B1E5E61AE44BD6A1A80E8554CD27BE@e1705>

Hi Gabe,

no it doesn't work. I had to create a fake ringback on my phone app.

Thanks

----- Original Message ----- 
From: "Gabriel Gunderson" <gabe at gundy.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, October 20, 2011 2:52 AM
Subject: Re: [Freeswitch-users] transfer_ringback from a conference


> On Wed, Oct 19, 2011 at 12:49 PM, Madovsky <infos at madovsky.org> wrote:
>> but with last git is it possible now ?
> 
> The best way to find out would be to give it a try!  Do let us know
> what you find and update the docs :)
> 
> Good luck,
> Gabe
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cesar.bermudez at gmail.com  Thu Oct 20 21:52:26 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Thu, 20 Oct 2011 11:52:26 -0600
Subject: [Freeswitch-users]  Desk Sip Phone with PSTN port, anyone?
Message-ID: <CADrcvV-69k7XxO5pcpPj3DN+7awRySzpmaqeR4R6o3eRA833ag@mail.gmail.com>

Hi Fs'users

Anyone know a desktop phone that can handle
multiple sip acc, and have pstn port?

Best regards.
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From devel at omninet.eu  Fri Oct 21 01:27:54 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Fri, 21 Oct 2011 00:27:54 +0300
Subject: [Freeswitch-users] LUA transfer result:0
Message-ID: <2C9B38E79A44400D8F7509B1010A00C4@omni1.local>

Hi,

 

I have tried two different ways of transfer in LUA:

session:transfer(number,"XML","default")

 

And 

 

Session:execute("transfer","12345 XML default")

 

And I can't make it working.

 

I get "switch_cpp.cpp:797 transfer result: 0"

 

The same transfer from within the dialplan works fine

 

The call is not answered before the transfer.

 

Does anybody have an idea?

 

Thanks

Anestis

 

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From luixsansan at hotmail.com  Fri Oct 21 04:10:15 2011
From: luixsansan at hotmail.com (luixsansan at hotmail.com)
Date: Fri, 21 Oct 2011 00:10:15 +0000
Subject: [Freeswitch-users] hi freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP30257FE9123DB424145C81FC7E80@phx.gbl>

hi freeswitch-users at lists.freeswitch.org we havent talked in awhile. i noticed this the other day http://t.co/FUwv44v9


From vallimamod.abdullah at imtelecom.fr  Thu Oct 20 11:19:17 2011
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Thu, 20 Oct 2011 09:19:17 +0200
Subject: [Freeswitch-users] BLF not working
In-Reply-To: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>
References: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>
Message-ID: <21EE2CE7-8F37-4450-B776-98F77120C52B@imtelecom.fr>

Hi,

From the logs you posted, I don't see any subscribe request coming from the phone. 
The phone should send the subscribe request right after registering and the LED goes green when it receives a notify reply.
Try to make a sip capture of the phone outgoing packets to check if there is any subscribe request sent.

Regards,
- vma
.



On Oct 19, 2011, at 10:07 PM, Mike Hammond wrote:

> Hello all,
>  
> I'm new to FS and I'm having a problem getting BLF working, initially I'm trying with a Cisco SPA504G.
>  
> I've configured a test switch directly on a public IP using the latest download (about a week ago) with a fusionpbx front end gui, running on an ubuntu 10.04 LTS server.
>  
> I've set it up for multi tenant and have had some domain names pointed at my public IP, I can register and use handsets and gateways without an issue on the different tenants offsite from behind nat'd routers.
>  
> Next I wanted to try BLF, so I've configured a spare key on the Cisco handset (extn 204) with fnc=blf+sd+cp;sub=202@$PROXY;ext=202@$PROXY to monitor extension 202 which is within the same tenant domain. But the key is glowing orange so obviously not happy
>  
> If set sofia to debug presence and turn on the sip trace I see the following messaging
>  
> http://pastebin.freeswitch.org/17557
>  
> Can anyone help me with this one as I'm struggling to know what to do next other than strip it all back out and start again with clean a single domain setup.
>  
> Many thanks,
>  
> 
> 
>  
> Mike Hammond
> 
>   
> GCI Com
> Cedar Court Office Park
> Denby Dale Road
> Calder Grove
> Wakefield
> WF4 3QZ
>  
> Office:   0844 443 3537
> Fax:      0844 443 3540
> http://www.gcicom.net/
> 
> 
> This email has been swept by Webroot for viruses. Any files transmitted with it are confidential and intended solely for the email recipient. If you are not the intended recipient please delete this email immediately. Be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error please notify the system administrator. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company. Finally, the recipient should check this email and any attachments for the presence of viruses.
> 
> 
> GCI Com incorporates the following Group Companies:
> GCI Telecom Group Limited Reg. No. 5396496, Edge Telecommunications Ltd Reg. No. 5748740, Edge Telecom Ltd Reg. No. 3101247, IP Infrastructures Ltd Reg. No. 4657026, Invomo Ltd Reg. No. 6267056, NetServices UK Ltd Reg. No. 7118768, WAN Services Ltd Reg. No. 4082862. All Registered in England and Wales, Registered Office: Global House, 2 Crofton Close, Lincoln, LN3 4NT
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sherif.omran at gmx.de  Thu Oct 20 13:42:36 2011
From: sherif.omran at gmx.de (Sherif Omran)
Date: Thu, 20 Oct 2011 11:42:36 +0200
Subject: [Freeswitch-users] ip_queue help
Message-ID: <DFB2AEC2-8D25-4DC8-AC11-C157EF471026@gmx.de>

Hello guys,

I installed freeswitch with zfone for testing purposes on a centos machine. After installing zfone, i find that ip_queue is not installed.

Could any body help me to continue?

/sbin/modprobe ip_queue
FATAL: Could not load /lib/modules/2.6.18-028stab092.1/modules.dep: No such file or directory

thank you

Sherif

From ayobami at programmer.net  Thu Oct 20 14:57:26 2011
From: ayobami at programmer.net (ayobami)
Date: Thu, 20 Oct 2011 03:57:26 -0700 (PDT)
Subject: [Freeswitch-users] Problem sending E-mail in Freeswitch on windows
Message-ID: <1319108246414-6912478.post@n2.nabble.com>

Please people help me out, I want to send email in a lua script of mine in
freeswitch the code is below

freeswitch.email("ayobami at programmer.net",
                 "ayobami.adewole at gmail.com",
                 "subject: testing fs mail function",
                 "checking to see if this guy works")
and I have downloaded the msmtp.exe and placed in c:\windows\system32 folder
as directed
i have equally changed the configuration in switch.conf.xml
but each time i keep running the script,
I keep getting the same error message, which is
sendmail is not recognised as an internal or external command

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-sending-E-mail-in-Freeswitch-on-windows-tp6912478p6912478.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sherif.omran at gmx.de  Thu Oct 20 16:29:36 2011
From: sherif.omran at gmx.de (Sherif Omran)
Date: Thu, 20 Oct 2011 14:29:36 +0200
Subject: [Freeswitch-users] zrtp
Message-ID: <70474E48-9807-45A1-8F29-DFFC4E1DB8AE@gmx.de>

Hi all,

I am trying to install zrtp zphone but it says that module ip_queue is missing. Any help is appreciated

thanks

regards,

Sherif



From hexade at hotmail.com  Fri Oct 21 05:43:16 2011
From: hexade at hotmail.com (Adelia C.)
Date: Thu, 20 Oct 2011 21:43:16 -0400
Subject: [Freeswitch-users] Running FreeSwitch as Windows Service
In-Reply-To: <C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>
References: <C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>
Message-ID: <BLU166-W13D5865416F0C2616ED8B9DAE80@phx.gbl>


Thank you, this worked.

> From: peter.olsson at visionutveckling.se
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 19 Oct 2011 19:26:07 +0200
> Subject: Re: [Freeswitch-users] Running FreeSwitch as Windows Service
> 
> Remove the service again, and the execute FreeSwitchConsole.exe -install. And try again
> 
> /Peter
> 
> ----- Reply message -----
> Fr?n: "Adelia C." <hexade at hotmail.com>
> Datum: ons, okt 19, 2011 18:49
> Rubrik: [Freeswitch-users] Running FreeSwitch as Windows Service
> Till: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
> 
> Hello,
> 
> I am building FreeSwitch through VS 2010 (not VC++ Express) and I generate FreeSwitchConsole.exe as a console application. No FreeSwitch.exe is generated.
> 
> FreeSwitchConsole runs fine as an application but I need to run the SW as a service.
> 
> I installed Freeswitch as a local service that calls FreeSwitchConsole.exe, and this is the output of "sc qc FreeSWITCH":
> 
> C:\FreeSWITCH>sc qc FreeSWITCH
> [SC] QueryServiceConfig SUCCESS
> 
> SERVICE_NAME: FreeSWITCH
>         TYPE               : 10  WIN32_OWN_PROCESS
>         START_TYPE         : 2   AUTO_START
>         ERROR_CONTROL      : 1   NORMAL
>         BINARY_PATH_NAME   : C:\FreeSWITCH\FreeSwitchConsole.exe
>         LOAD_ORDER_GROUP   :
>         TAG                : 0
>         DISPLAY_NAME       : FreeSWITCH
>         DEPENDENCIES       :
>         SERVICE_START_NAME : LocalSystem
> 
> 
> Trying to start FreeSwitch as a service results in  Error 1053 Timeout - Event Viewer is : Timeout (30000 milliseconds) waiting for the FreeSWITCH service to connect.
> 
> Or:
> C:\FreeSWITCH>net start freeswitch
> The service is not responding to the control function.
> 
> Is it because I am trying to start a console application as a service?
> 
> Do I need to have a FreeSwitch.exe and if so, how do I build it? Do I need to create a Windows Service wrapper project?
> 
> I tried using winserv (supposed to start console apps as service), with same result:
> C:\FreeSWITCH>..\tools\winserv\winserv start FreeSwitch
> ERROR 1053: The service did not respond to the start or control request in a timely fashion.
> 
> What am I missing?
> 
> Thank you.
> A.C.
> !DSPAM:4e9efdb732761275568181!
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From hexade at hotmail.com  Fri Oct 21 06:14:28 2011
From: hexade at hotmail.com (Adelia C.)
Date: Thu, 20 Oct 2011 22:14:28 -0400
Subject: [Freeswitch-users] Problems loading managed dll
In-Reply-To: <C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>
References: <C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>
Message-ID: <BLU166-W8E596DF597007FB70EFACDAE80@phx.gbl>


Out of my 3 VMs (Windows Server 2003), one fails to load my managed dll.

Bad server:
[ALERT] switch_cpp.cpp:1197 Exception loading C:\FreeSWITCH\mod\managed\MyManagedApp.dll: System.InvalidCastException: Unable to cast transparent proxy to type 'FreeSWITCH.PluginManager'.
   at FreeSWITCH.Loader.loadFile(String fileName)

Good servers:
[NOTICE] switch_cpp.cpp:1197 Loaded App MyManagedApp.Handle, aliases 'MyManagedApp.Handle, Handle', into domain MyManagedApp.dll_1.

All seems equal, I am running the same code, under the 
same "C:\FreeSWITCH" location on all servers. No host specific hard codings.

Please help.

Thank you.
A.C.
 		 	   		  
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From hexade at hotmail.com  Fri Oct 21 08:23:44 2011
From: hexade at hotmail.com (Adelia C.)
Date: Fri, 21 Oct 2011 00:23:44 -0400
Subject: [Freeswitch-users] Problems loading managed dll
In-Reply-To: <C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>
References: <C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>
Message-ID: <BLU166-W3307B50B6C22750FA03C3ADAE80@phx.gbl>


Out of my 3 VMs (Windows Server 2003), one fails to load my managed dll.

Bad server:
[ALERT] switch_cpp.cpp:1197 Exception loading C:\FreeSWITCH\mod\managed\MyManagedApp.dll: System.InvalidCastException: Unable to cast transparent proxy to type 'FreeSWITCH.PluginManager'.
   at FreeSWITCH.Loader.loadFile(String fileName)

Good servers:
[NOTICE] switch_cpp.cpp:1197 Loaded App MyManagedApp.Handle, aliases 'MyManagedApp.Handle, Handle', into domain MyManagedApp.dll_1.

All seems equal, I am running the same code, under the 
same "C:\FreeSWITCH" location on all servers. No host specific hard codings.

Please help.

Thank you.
A.C.
 		 	   		  
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From william at xofap.com  Fri Oct 21 09:22:54 2011
From: william at xofap.com (William Alianto)
Date: Fri, 21 Oct 2011 12:22:54 +0700
Subject: [Freeswitch-users] mod_freetdm incoming dialplan
In-Reply-To: <CAFSQhGXDbMtHv-F3GpxKuLsnX3isXX7PzzNtv=KKyRwgtZQS0w@mail.gmail.com>
References: <4E9E9775.2050006@xofap.com> <10432.1319030211@ccs.covici.com>
	<ccb4fc8b76331c4855f1a5e82d048bfe.squirrel@emailmg.globat.com>
	<4E9FAA93.8050704@xofap.com>
	<CAFSQhGXDbMtHv-F3GpxKuLsnX3isXX7PzzNtv=KKyRwgtZQS0w@mail.gmail.com>
Message-ID: <4EA101AE.3090906@xofap.com>

Hi Guru Singh,

extension 800 is actually a hunt group for my reception phones.
And here is the log you requested : http://pastebin.freeswitch.com/17567

Regards,

William

On 2:59, guru singh wrote:
> Hi William,
>
> What phone do  you expect it to ring?
> Do you have an extension in your default dialplan for destination number 800?
> You should pastebin your freeswitch logs, so we can see whats going on.
>
> Regards,
> guru
>
> On Thu, Oct 20, 2011 at 10:28 AM, William Alianto<william at xofap.com>  wrote:
>> Ok, I have put my dialplan on default, and change it to :
>>
>> <extension name=nalog">
>> <condition field=ource" expression="mod_freetdm">
>> <action application=et" data="domain_name=$${domain}" />
>> <action application=et" data="transfer_ringback=$${us-ring}" />
>> <action application=nswer"
>> <action application=ransfer" data="800 XML default" />
>> </condition>
>> </extension>
>>
>> The FS finally able to pass the dialplan, but it still not ringing on my
>> phone. Any idea how?
>>
>> Regards,
>>
>> William
>>
>> On 2:59, rsaavedra at ecogizmos.com wrote:
>>> Hi,
>>>
>>> I'm working with the Sangoma's U100.
>>>
>>> I found on Internet that you must use:
>>>
>>> <condition field=ource? expression=?mod_freetdm?/>
>>>
>>> My experience is that with the U100 you can put context=ublic" but the
>>> calls always have the context in default, so I put all the instrucctions
>>> in the file 00_inbound_did.xml in:
>>> /usr/local/freeswitch/conf/dialplan/default and everything now is working.
>>>
>>> Best regards,
>>>
>>> Ricardo Saavedra
>>>
>>>> I would just use the freetdm condition and see if that works.
>>>>
>>>> William Alianto<william at xofap.com>   wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> Has anybody got a sample of dialplan for incoming from freetdm?
>>>>>
>>>>> I've been working to get the incoming from my A400P card working, but
>>>>> still fail to do so.
>>>>>
>>>>> Following is my current incoming dialplan :
>>>>>
>>>>> <extension name=nalog">
>>>>> <condition field=ontext" expression="public"/>
>>>>> <condition field=estination_number" expression="^(\d+)$"/>
>>>>> <condition field=han_name" expression="FreeTDM/1/[1-4]">
>>>>> <action application=et" data="domain_name=$${domain}"/>
>>>>> <action application=et" data="transfer_ringback=$${us-ring}"/>
>>>>> <action application=nswer"/>
>>>>> <action application=ransfer" data="800 XML default"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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,
>>
>> William
>> CV.XOFAP.COM
>> Engineering Dept.
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 at xofap.com  Fri Oct 21 09:28:21 2011
From: william at xofap.com (William Alianto)
Date: Fri, 21 Oct 2011 12:28:21 +0700
Subject: [Freeswitch-users] Desk Sip Phone with PSTN port, anyone?
In-Reply-To: <CADrcvV-69k7XxO5pcpPj3DN+7awRySzpmaqeR4R6o3eRA833ag@mail.gmail.com>
References: <CADrcvV-69k7XxO5pcpPj3DN+7awRySzpmaqeR4R6o3eRA833ag@mail.gmail.com>
Message-ID: <4EA102F5.8030305@xofap.com>

Hi Cesar,

There is one I'm using right now. It's Planet VIP-254NT.
http://www.planet.com.tw/en/product/product_ov.php?id=34064

Check if it is the one you need

Regards,

William

On 2:59, Cesar Bermudez wrote:
> Hi Fs'users
>
> Anyone know a desktop phone that can handle
> multiple sip acc, and have pstn port?
>
> Best regards.




From hynek.cihlar at gmail.com  Fri Oct 21 10:19:35 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 21 Oct 2011 08:19:35 +0200
Subject: [Freeswitch-users] Lua directory lookup and multihoming
Message-ID: <CAJh8XdWB0vmidsNfOj5i0opR=cn+jFOoYi0G4h9o1TmFcCnqgQ@mail.gmail.com>

I have enabled Lua directory lookup based on the example at
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh#Lua_Script_2. It works
ok, however I didn't figure out, how to properly group and isolate
registered users to allow multihoming.

IMHO the best isolation is through different contexts. The context names are
saved also in the DB together with the individual users. Hence the context
dialplans cannot be specified on the file system as xml files. The dialplans
are fairly static, they don't vary with each context.

Any idea how to make Freeswitch create those dialplans dynamically?

Regards,
Hynek
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From hynek.cihlar at gmail.com  Fri Oct 21 10:22:05 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 21 Oct 2011 08:22:05 +0200
Subject: [Freeswitch-users] Host and domain in the dial string
Message-ID: <CAJh8XdVs8DB6x7cU-qd743k8R9_J_Aif9-hgyxPXOTRKLLEe2g@mail.gmail.com>

Is there a way to originate or bridge calls specifying destination host AND
destination domain in the SIP dial string? I think I've spot it somewhere in
the wiki, but I cannot find again.

Hynek
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From william at xofap.com  Fri Oct 21 14:43:07 2011
From: william at xofap.com (William Alianto)
Date: Fri, 21 Oct 2011 17:43:07 +0700
Subject: [Freeswitch-users] mod_freetdm incoming dialplan
In-Reply-To: <4EA101AE.3090906@xofap.com>
References: <4E9E9775.2050006@xofap.com> <10432.1319030211@ccs.covici.com>
	<ccb4fc8b76331c4855f1a5e82d048bfe.squirrel@emailmg.globat.com>
	<4E9FAA93.8050704@xofap.com>
	<CAFSQhGXDbMtHv-F3GpxKuLsnX3isXX7PzzNtv=KKyRwgtZQS0w@mail.gmail.com>
	<4EA101AE.3090906@xofap.com>
Message-ID: <4EA14CBB.7050103@xofap.com>

Hi Everyone,

I've successfully done the incoming from freetdm. This is my dialplan :

transfer to Single Extension :

<extension name="analog" >
<condition field="source" expression="mod_freetdm" >
<action application="set" data="context=inbound" />
<action application="set" data="domain_name=$${domain}" />
<action application="set" data="transfer_ringback=$${us-ring}" />
<action application="bridge" data="user/201@$${domain}" />
</condition>
</extension>

transfer to Hunt Group :

<extension name="analog" >
<condition field="source" expression="mod_freetdm" >
<action application="set" data="context=inbound" />
<action application="set" data="domain_name=$${domain}" />
<action application="set" data="transfer_ringback=$${us-ring}" />
<action application="lua" data="v_huntgroup_192.168.1.3_800.lua" />
</condition>
</extension>

the lua script is generated for huntgroup creation using fusionpbx.

If somebody has better dialplan, please tell me.

Regards,

William

On 2:59, William Alianto wrote:
> Hi Guru Singh,
>
> extension 800 is actually a hunt group for my reception phones.
> And here is the log you requested : http://pastebin.freeswitch.com/17567
>
> Regards,
>
> William
>
> On 2:59, guru singh wrote:
>> Hi William,
>>
>> What phone do you expect it to ring?
>> Do you have an extension in your default dialplan for destination 
>> number 800?
>> You should pastebin your freeswitch logs, so we can see whats going on.
>>
>> Regards,
>> guru
>>
>> On Thu, Oct 20, 2011 at 10:28 AM, William Alianto<william at xofap.com> 
>> wrote:
>>> Ok, I have put my dialplan on default, and change it to :
>>>
>>> <extension name=nalog">
>>> <condition field=ource" expression="mod_freetdm">
>>> <action application=et" data="domain_name=$${domain}" />
>>> <action application=et" data="transfer_ringback=$${us-ring}" />
>>> <action application=nswer"
>>> <action application=ransfer" data="800 XML default" />
>>> </condition>
>>> </extension>
>>>
>>> The FS finally able to pass the dialplan, but it still not ringing 
>>> on my
>>> phone. Any idea how?
>>>
>>> Regards,
>>>
>>> William
>>>
>>> On 2:59, rsaavedra at ecogizmos.com wrote:
>>>> Hi,
>>>>
>>>> I'm working with the Sangoma's U100.
>>>>
>>>> I found on Internet that you must use:
>>>>
>>>> <condition field=ource? expression=?mod_freetdm?/>
>>>>
>>>> My experience is that with the U100 you can put context=ublic" but the
>>>> calls always have the context in default, so I put all the 
>>>> instrucctions
>>>> in the file 00_inbound_did.xml in:
>>>> /usr/local/freeswitch/conf/dialplan/default and everything now is 
>>>> working.
>>>>
>>>> Best regards,
>>>>
>>>> Ricardo Saavedra
>>>>
>>>>> I would just use the freetdm condition and see if that works.
>>>>>
>>>>> William Alianto<william at xofap.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Has anybody got a sample of dialplan for incoming from freetdm?
>>>>>>
>>>>>> I've been working to get the incoming from my A400P card working, 
>>>>>> but
>>>>>> still fail to do so.
>>>>>>
>>>>>> Following is my current incoming dialplan :
>>>>>>
>>>>>> <extension name=nalog">
>>>>>> <condition field=ontext" expression="public"/>
>>>>>> <condition field=estination_number" expression="^(\d+)$"/>
>>>>>> <condition field=han_name" expression="FreeTDM/1/[1-4]">
>>>>>> <action application=et" data="domain_name=$${domain}"/>
>>>>>> <action application=et" data="transfer_ringback=$${us-ring}"/>
>>>>>> <action application=nswer"/>
>>>>>> <action application=ransfer" data="800 XML default"/>
>>>>>> </condition>
>>>>>> </extension>
>>>>>>
>>>>>>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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,
>>>
>>> William
>>> CV.XOFAP.COM
>>> Engineering Dept.
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 mhammond at gcicom.net  Fri Oct 21 14:53:48 2011
From: mhammond at gcicom.net (Mike Hammond)
Date: Fri, 21 Oct 2011 11:53:48 +0100
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 64, Issue 90
References: <mailman.7528.1319174613.29749.freeswitch-users@lists.freeswitch.org>
Message-ID: <1C748D48EFD36B4AA0B934E8B4E2998003615E42@ipi-cc-srv04.ipinfrastructures.com>

Hi, 

The issue has been resolved with a firmware upgrade on the Cisco
handset's and I've also now got it working well on a Yealink T26p. 

Thanks for the replies folks.

Mike
 
-----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: 21 October 2011 06:24
To: freeswitch-users at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 64, Issue 90

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..."

 
Mike Hammond

  
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From hynek.cihlar at gmail.com  Fri Oct 21 17:09:13 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 21 Oct 2011 15:09:13 +0200
Subject: [Freeswitch-users] Transparent external call
Message-ID: <CAJh8XdXYkTMNevfHibR++hrju+c2V3PnZptgXXPiaey1sBtPCQ@mail.gmail.com>

  <context name="default">
     <extension name="ext">
      <condition field="destination_number" expression="^(\d+)$">
          <action application="bridge"
data="sofia/internal/$1%${domain_name}"/>

           *HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE DST
NUMBER IS NOT AN EXTENSION OF A REGED USER)?*
          <action application="bridge" data="sofia/external/$1@${host}"/>
      </condition>
    </extension>



Hynek
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From erik.dekkers at wvds.nl  Fri Oct 21 17:17:55 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Fri, 21 Oct 2011 15:17:55 +0200
Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on internal
	calls
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433C5@FS1.wvds.local>

Hey list,

Recently (at least since two days) freeswitch is sending UPDATE messages with  P-Asserted-Identity: "Outbound Call" <sip:NUMBER at DOMAIN>  after a call is established on the callers phone.
It looks like it?s not updating the callee?s name with this UPDATE message since the callers name is displayed correctly.

This is the dialplan I?m using for internal call routing:

    <extension name="Local_Extension_SIP">
      <condition field="destination_number" expression="^([1-4]\d{2})$">
            <action application="bridge" data="user/$1"/>
      </condition>
    </extension>

Full debug + siptrace: http://pastebin.freeswitch.org/17568

If there?s anything I can test please let me know

Regards,

Erik Dekkers (wvds-nl)

[cid:image78e1b4.JPG at fcaf16f0.48a24e65]
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________________________________
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From Giuseppe.Grillo at italtel.it  Fri Oct 21 17:27:43 2011
From: Giuseppe.Grillo at italtel.it (Grillo Giuseppe)
Date: Fri, 21 Oct 2011 15:27:43 +0200
Subject: [Freeswitch-users] R: Sip register with contact reg-id
	and+sip.instance
In-Reply-To: <47F792F8B9380249AD789203F4C0CF10049C4F43@BESTWO.corp.dom>
References: <47F792F8B9380249AD789203F4C0CF10049C4F43@BESTWO.corp.dom>
Message-ID: <47F792F8B9380249AD789203F4C0CF1004A3F9FD@BESTWO.corp.dom>

Someone can help me ?

 

Kind regards

 

Giuseppe

 

 

 

Da: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Per conto di
Grillo Giuseppe
Inviato: 18 October, 2011 1:58 PM
A: freeswitch-users at lists.freeswitch.org
Oggetto: [Freeswitch-users] Sip register with contact reg-id
and+sip.instance

 

Hi all,

 

I have the need that the "Contact" in SIP register is formatted as
follows including reg-id and +sip.instance.

How do I configure freeswitch to send the "Contact" so ?

 

REGISTER sips:10.10.2.110 SIP/2.0

From: sips:1000 at 10.10.2.110;tag=-5159386d4396-8_F192.168.178.29

To: sips:1000 at 10.10.2.110

Call-ID: 1_909-5f0974b1386d4ca0_R at 192.168.178.29

CSeq: 1 REGISTER

Via: SIP/2.0/TLS
192.168.178.29:5061;branch=z9hG4bK1_90a4615b657386d4ca8_R

Content-Length: 0

Max-Forwards: 70

Contact:
<sips:1000 at 192.168.178.29:5061;avaya-sc-enabled;transport=tls>;q=1;expir
es=900;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00040de
c6f47>"

Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE

User-Agent: Avaya one-X Deskphone

Supported: replaces

 

Thanks for your help!

 

Kind regards

 

Giuseppe

 

 

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From jeff at jefflenk.com  Fri Oct 21 17:30:45 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 21 Oct 2011 06:30:45 -0700 (PDT)
Subject: [Freeswitch-users] Problems loading managed dll
In-Reply-To: <BLU166-W3307B50B6C22750FA03C3ADAE80@phx.gbl>
References: <BLU166-W196CD7DE2FB3227327E813DAEA0@phx.gbl>
	<C456E42C-A680-40FE-AC72-0FD6686AE217@visionutveckling.se>
	<BLU166-W3307B50B6C22750FA03C3ADAE80@phx.gbl>
Message-ID: <1319203845658-6916919.post@n2.nabble.com>

There is not enough information given here to determine what may be wrong. Do
you you have dependencies missing? Do all systems have the same Framework
loaded?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Running-FreeSwitch-as-Windows-Service-tp6909473p6916919.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From freeswitch-list at puzzled.xs4all.nl  Fri Oct 21 17:48:50 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 21 Oct 2011 15:48:50 +0200
Subject: [Freeswitch-users] R: Sip register with contact
	reg-id	and+sip.instance
In-Reply-To: <47F792F8B9380249AD789203F4C0CF1004A3F9FD@BESTWO.corp.dom>
References: <47F792F8B9380249AD789203F4C0CF10049C4F43@BESTWO.corp.dom>
	<47F792F8B9380249AD789203F4C0CF1004A3F9FD@BESTWO.corp.dom>
Message-ID: <4EA17842.8020300@puzzled.xs4all.nl>

On 10/21/2011 03:27 PM, Grillo Giuseppe wrote:
> Someonecanhelp me ?

I am not sure if that is possible unless you code it in the sofia-sip 
stack. Maybe someone more knowledgeable will reply.

Can't you just add some headers with that information? See:
http://wiki.freeswitch.org/wiki/Sofia#Channel_Variables

Regards,
Patrick


From Giuseppe.Grillo at italtel.it  Fri Oct 21 18:00:21 2011
From: Giuseppe.Grillo at italtel.it (Grillo Giuseppe)
Date: Fri, 21 Oct 2011 16:00:21 +0200
Subject: [Freeswitch-users] R: R: Sip register with
	contactreg-id	and+sip.instance
In-Reply-To: <4EA17842.8020300@puzzled.xs4all.nl>
References: <47F792F8B9380249AD789203F4C0CF10049C4F43@BESTWO.corp.dom><47F792F8B9380249AD789203F4C0CF1004A3F9FD@BESTWO.corp.dom>
	<4EA17842.8020300@puzzled.xs4all.nl>
Message-ID: <47F792F8B9380249AD789203F4C0CF1004A3FA08@BESTWO.corp.dom>

I saw this in many trace, and I found this comment in nua_register.c line 326

*
 * Likewise, nua may add transport parameters (such as "transport=tcp" or
 * "maddr") to the @Contact URI. It can add addtional header parameters, like
 * "+sip.instance" or "reg-id", too.
 *

Thanks,  Giuseppe


Giuseppe Grillo
MPS-RTA-PA-AP
Phone: +39-0918615523
Collaboration : MPS-RTA

Please do not print this email unless it is   absolutely necessary. 
Spread environmental awareness.

 


-----Messaggio originale-----
Da: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Per conto di Patrick Lists
Inviato: 21 October, 2011 3:49 PM
A: FreeSWITCH Users Help
Oggetto: Re: [Freeswitch-users] R: Sip register with contactreg-id and+sip.instance

On 10/21/2011 03:27 PM, Grillo Giuseppe wrote:
> Someonecanhelp me ?

I am not sure if that is possible unless you code it in the sofia-sip 
stack. Maybe someone more knowledgeable will reply.

Can't you just add some headers with that information? See:
http://wiki.freeswitch.org/wiki/Sofia#Channel_Variables

Regards,
Patrick


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


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-----------------------------------------------------------------------------------------------------
La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. 

This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. 
-----------------------------------------------------------------------------------------------------



From infos at madovsky.org  Fri Oct 21 20:40:30 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Oct 2011 12:40:30 -0400
Subject: [Freeswitch-users] Transparent external call
References: <CAJh8XdXYkTMNevfHibR++hrju+c2V3PnZptgXXPiaey1sBtPCQ@mail.gmail.com>
Message-ID: <7123B286F65146CDA22C92F0651457AE@e1705>

take a look at hangup_after_bridge and continue_on_fail wiki
and make condition with hangup_cause
  ----- Original Message ----- 
  From: Hynek Cihlar 
  To: FreeSWITCH Users Help 
  Sent: Friday, October 21, 2011 9:09 AM
  Subject: [Freeswitch-users] Transparent external call


    <context name="default">
       <extension name="ext">
        <condition field="destination_number" expression="^(\d+)$">
            <action application="bridge" data="sofia/internal/$1%${domain_name}"/>
            
             HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?
            <action application="bridge" data="sofia/external/$1@${host}"/>
        </condition>
      </extension>





  Hynek




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



  FreeSWITCH-users mailing 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 hynek.cihlar at gmail.com  Fri Oct 21 20:56:26 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 21 Oct 2011 18:56:26 +0200
Subject: [Freeswitch-users] Transparent external call
In-Reply-To: <7123B286F65146CDA22C92F0651457AE@e1705>
References: <CAJh8XdXYkTMNevfHibR++hrju+c2V3PnZptgXXPiaey1sBtPCQ@mail.gmail.com>
	<7123B286F65146CDA22C92F0651457AE@e1705>
Message-ID: <CAJh8XdXRNwPR1evLbM3tdzS9CyOcoau9bED-tWPenuYUqJHvyQ@mail.gmail.com>

My apology. Forgot to mention that I need the hangup_after_bridge to be
false. There is additional processing on the legs after hangup.


  <context name="default">
     <extension name="ext">
      <condition field="destination_number" expression="^(\d+)$">
          *NO HANGUP IS ESSENTIAL*
          <action application="set" data="hangup_after_bridge=false"/>
          <action application="bridge"
data="sofia/internal/$1%${domain_name}"/>

           *HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE DST
NUMBER IS NOT AN EXTENSION OF A REGED USER)?*
          <action application="bridge" data="sofia/external/$1@${host}"/>
      </condition>
    </extension>


Hynek



On Fri, Oct 21, 2011 at 6:40 PM, Madovsky <infos at madovsky.org> wrote:

> **
> take a look at hangup_after_bridge and continue_on_fail wiki
> and make condition with hangup_cause
>
> ----- Original Message -----
> *From:* Hynek Cihlar <hynek.cihlar at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, October 21, 2011 9:09 AM
> *Subject:* [Freeswitch-users] Transparent external call
>
>    <context name="default">
>      <extension name="ext">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="bridge"
> data="sofia/internal/$1%${domain_name}"/>
>
>            *HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE
> DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?*
>           <action application="bridge" data="sofia/external/$1@${host}"/>
>       </condition>
>     </extension>
>
>
>
> Hynek
>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Oct 21 20:59:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Oct 2011 11:59:17 -0500
Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on
 internal calls
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433C5@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433C5@FS1.wvds.local>
Message-ID: <CAKbxfG_JR3wD+UyNziiXvuLZYR29RckcUq_10Mk9TPdESAAArg@mail.gmail.com>

if you don't know who you are calling that is the display update you will
get.
if you do, set it in the bridge with origination_callee_id_name

{origination_callee_id_name='Fred Smith'}user/1234


On Fri, Oct 21, 2011 at 8:17 AM, Erik Dekkers <erik.dekkers at wvds.nl> wrote:

>  Hey list,****
>
> ** **
>
> Recently (at least since two days) freeswitch is sending UPDATE messages
> with  P-Asserted-Identity: "Outbound Call" <sip:NUMBER at DOMAIN>  after a
> call is established on the callers phone.****
>
> It looks like it?s not updating the callee?s name with this UPDATE message
> since the callers name is displayed correctly.****
>
> ** **
>
> This is the dialplan I?m using for internal call routing:****
>
> ** **
>
>     <extension name="Local_Extension_SIP">****
>
>       <condition field="destination_number" expression="^([1-4]\d{2})$">**
> **
>
>             <action application="bridge" data="user/$1"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> Full debug + siptrace: http://pastebin.freeswitch.org/17568****
>
> ** **
>
> If there?s anything I can test please let me know****
>
> ** **
>
> Regards,****
>
> ** **
>
> Erik Dekkers (wvds-nl)****
>
>
>   ABC Westland 555 Tel: +31 174 245 727 P.O. Box 90 Fax: +31 174 248 739 2685
> ZH Poeldijk Mob: +31 624 423 009 www.wvds.nl erik.dekkers at wvds.nl
> ------------------------------
> Bezoek ons op de Horti Fair / Visit us at the Horti Fair
> 1-4 november 2011 in Amsterdam, stand 11.0302.
> <https://form.n200.com/?s=1dvsast0dsqul&c=2quptgqmwol3s&ac=NTWO000229FIK>
> ------------------------------
> Wilk van der Sande is gecertificeerd Philips Led Horti Partner.
>
> DISCLAIMER
> Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka
> B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van
> toepassing
> All our quotations, all orders placed and all contracts concluded with
> Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the
> AVAG-Conditions.
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From infos at madovsky.org  Fri Oct 21 21:14:42 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Oct 2011 13:14:42 -0400
Subject: [Freeswitch-users] Transparent external call
References: <CAJh8XdXYkTMNevfHibR++hrju+c2V3PnZptgXXPiaey1sBtPCQ@mail.gmail.com><7123B286F65146CDA22C92F0651457AE@e1705>
	<CAJh8XdXRNwPR1evLbM3tdzS9CyOcoau9bED-tWPenuYUqJHvyQ@mail.gmail.com>
Message-ID: <F4AA541819174F8F8EE6CCFC7A671BCD@e1705>

so use hangup_cause in condition
  ----- Original Message ----- 
  From: Hynek Cihlar 
  To: FreeSWITCH Users Help 
  Sent: Friday, October 21, 2011 12:56 PM
  Subject: Re: [Freeswitch-users] Transparent external call


  My apology. Forgot to mention that I need the hangup_after_bridge to be false. There is additional processing on the legs after hangup.




    <context name="default">
       <extension name="ext">
        <condition field="destination_number" expression="^(\d+)$">
            NO HANGUP IS ESSENTIAL
            <action application="set" data="hangup_after_bridge=false"/>
            <action application="bridge" data="sofia/internal/$1%${domain_name}"/>
            
             HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?
            <action application="bridge" data="sofia/external/$1@${host}"/>
        </condition>
      </extension>



  Hynek




  On Fri, Oct 21, 2011 at 6:40 PM, Madovsky <infos at madovsky.org> wrote:

    take a look at hangup_after_bridge and continue_on_fail wiki
    and make condition with hangup_cause
      ----- Original Message ----- 
      From: Hynek Cihlar 
      To: FreeSWITCH Users Help 
      Sent: Friday, October 21, 2011 9:09 AM
      Subject: [Freeswitch-users] Transparent external call


        <context name="default">
           <extension name="ext">
            <condition field="destination_number" expression="^(\d+)$">
                <action application="bridge" data="sofia/internal/$1%${domain_name}"/>
                
                 HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?
                <action application="bridge" data="sofia/external/$1@${host}"/>
            </condition>
          </extension>





      Hynek




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From infos at madovsky.org  Fri Oct 21 21:18:39 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 Oct 2011 13:18:39 -0400
Subject: [Freeswitch-users] Transparent external call
References: <CAJh8XdXYkTMNevfHibR++hrju+c2V3PnZptgXXPiaey1sBtPCQ@mail.gmail.com><7123B286F65146CDA22C92F0651457AE@e1705><CAJh8XdXRNwPR1evLbM3tdzS9CyOcoau9bED-tWPenuYUqJHvyQ@mail.gmail.com>
	<F4AA541819174F8F8EE6CCFC7A671BCD@e1705>
Message-ID: <4986D0F9BF254398A54BA50DAD9D2918@e1705>

sorry you have to hangup the bridge before to try another bridge..
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Friday, October 21, 2011 1:14 PM
  Subject: Re: [Freeswitch-users] Transparent external call


  so use hangup_cause in condition
    ----- Original Message ----- 
    From: Hynek Cihlar 
    To: FreeSWITCH Users Help 
    Sent: Friday, October 21, 2011 12:56 PM
    Subject: Re: [Freeswitch-users] Transparent external call


    My apology. Forgot to mention that I need the hangup_after_bridge to be false. There is additional processing on the legs after hangup.




      <context name="default">
         <extension name="ext">
          <condition field="destination_number" expression="^(\d+)$">
              NO HANGUP IS ESSENTIAL
              <action application="set" data="hangup_after_bridge=false"/>
              <action application="bridge" data="sofia/internal/$1%${domain_name}"/>
              
               HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?
              <action application="bridge" data="sofia/external/$1@${host}"/>
          </condition>
        </extension>



    Hynek




    On Fri, Oct 21, 2011 at 6:40 PM, Madovsky <infos at madovsky.org> wrote:

      take a look at hangup_after_bridge and continue_on_fail wiki
      and make condition with hangup_cause
        ----- Original Message ----- 
        From: Hynek Cihlar 
        To: FreeSWITCH Users Help 
        Sent: Friday, October 21, 2011 9:09 AM
        Subject: [Freeswitch-users] Transparent external call


          <context name="default">
             <extension name="ext">
              <condition field="destination_number" expression="^(\d+)$">
                  <action application="bridge" data="sofia/internal/$1%${domain_name}"/>
                  
                   HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?
                  <action application="bridge" data="sofia/external/$1@${host}"/>
              </condition>
            </extension>





        Hynek




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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From hynek.cihlar at gmail.com  Fri Oct 21 21:19:36 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 21 Oct 2011 19:19:36 +0200
Subject: [Freeswitch-users] Transparent external call
In-Reply-To: <F4AA541819174F8F8EE6CCFC7A671BCD@e1705>
References: <CAJh8XdXYkTMNevfHibR++hrju+c2V3PnZptgXXPiaey1sBtPCQ@mail.gmail.com>
	<7123B286F65146CDA22C92F0651457AE@e1705>
	<CAJh8XdXRNwPR1evLbM3tdzS9CyOcoau9bED-tWPenuYUqJHvyQ@mail.gmail.com>
	<F4AA541819174F8F8EE6CCFC7A671BCD@e1705>
Message-ID: <CAJh8XdU_XfOYub+RaMqJZWunaZp4Wf89jFjH58GGkEkU=Kvyqg@mail.gmail.com>

Clever, thanks!

Hynek



On Fri, Oct 21, 2011 at 7:14 PM, Madovsky <infos at madovsky.org> wrote:

> **
> so use hangup_cause in condition
>
> ----- Original Message -----
> *From:* Hynek Cihlar <hynek.cihlar at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, October 21, 2011 12:56 PM
> *Subject:* Re: [Freeswitch-users] Transparent external call
>
> My apology. Forgot to mention that I need the hangup_after_bridge to be
> false. There is additional processing on the legs after hangup.
>
>
>    <context name="default">
>      <extension name="ext">
>       <condition field="destination_number" expression="^(\d+)$">
>           *NO HANGUP IS ESSENTIAL*
>           <action application="set" data="hangup_after_bridge=false"/>
>           <action application="bridge"
> data="sofia/internal/$1%${domain_name}"/>
>
>            *HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE
> DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?*
>           <action application="bridge" data="sofia/external/$1@${host}"/>
>       </condition>
>     </extension>
>
>
> Hynek
>
>
>
> On Fri, Oct 21, 2011 at 6:40 PM, Madovsky <infos at madovsky.org> wrote:
>
>> **
>> take a look at hangup_after_bridge and continue_on_fail wiki
>> and make condition with hangup_cause
>>
>>   ----- Original Message -----
>> *From:* Hynek Cihlar <hynek.cihlar at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Friday, October 21, 2011 9:09 AM
>> *Subject:* [Freeswitch-users] Transparent external call
>>
>>    <context name="default">
>>      <extension name="ext">
>>       <condition field="destination_number" expression="^(\d+)$">
>>           <action application="bridge"
>> data="sofia/internal/$1%${domain_name}"/>
>>
>>            *HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE
>> DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?*
>>           <action application="bridge" data="sofia/external/$1@${host}"/>
>>       </condition>
>>     </extension>
>>
>>
>>
>> Hynek
>>
>>  ------------------------------
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hynek.cihlar at gmail.com  Fri Oct 21 21:23:08 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 21 Oct 2011 19:23:08 +0200
Subject: [Freeswitch-users] Transparent external call
In-Reply-To: <4986D0F9BF254398A54BA50DAD9D2918@e1705>
References: <CAJh8XdXYkTMNevfHibR++hrju+c2V3PnZptgXXPiaey1sBtPCQ@mail.gmail.com>
	<7123B286F65146CDA22C92F0651457AE@e1705>
	<CAJh8XdXRNwPR1evLbM3tdzS9CyOcoau9bED-tWPenuYUqJHvyQ@mail.gmail.com>
	<F4AA541819174F8F8EE6CCFC7A671BCD@e1705>
	<4986D0F9BF254398A54BA50DAD9D2918@e1705>
Message-ID: <CAJh8XdUD+9tAMACF4DMyR28xoRBtM3KwFfaoSpaBe34FT+gSiQ@mail.gmail.com>

Right... or multiple destinations in the bridge with 'or' semantics, if
bridge supports that. The think is, as you got it correctly, that there may
be only one bridge for the incoming session.

Hynek



On Fri, Oct 21, 2011 at 7:18 PM, Madovsky <infos at madovsky.org> wrote:

> **
> sorry you have to hangup the bridge before to try another bridge..
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, October 21, 2011 1:14 PM
> *Subject:* Re: [Freeswitch-users] Transparent external call
>
> so use hangup_cause in condition
>
> ----- Original Message -----
> *From:* Hynek Cihlar <hynek.cihlar at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, October 21, 2011 12:56 PM
> *Subject:* Re: [Freeswitch-users] Transparent external call
>
> My apology. Forgot to mention that I need the hangup_after_bridge to be
> false. There is additional processing on the legs after hangup.
>
>
>    <context name="default">
>      <extension name="ext">
>       <condition field="destination_number" expression="^(\d+)$">
>           *NO HANGUP IS ESSENTIAL*
>           <action application="set" data="hangup_after_bridge=false"/>
>           <action application="bridge"
> data="sofia/internal/$1%${domain_name}"/>
>
>            *HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE
> DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?*
>           <action application="bridge" data="sofia/external/$1@${host}"/>
>       </condition>
>     </extension>
>
>
> Hynek
>
>
>
> On Fri, Oct 21, 2011 at 6:40 PM, Madovsky <infos at madovsky.org> wrote:
>
>> **
>> take a look at hangup_after_bridge and continue_on_fail wiki
>> and make condition with hangup_cause
>>
>>   ----- Original Message -----
>> *From:* Hynek Cihlar <hynek.cihlar at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Friday, October 21, 2011 9:09 AM
>> *Subject:* [Freeswitch-users] Transparent external call
>>
>>    <context name="default">
>>      <extension name="ext">
>>       <condition field="destination_number" expression="^(\d+)$">
>>           <action application="bridge"
>> data="sofia/internal/$1%${domain_name}"/>
>>
>>            *HOW TO RUN THIS ACTION, ONLY IF THE ABOVE BRIDGE FAILS (THE
>> DST NUMBER IS NOT AN EXTENSION OF A REGED USER)?*
>>           <action application="bridge" data="sofia/external/$1@${host}"/>
>>       </condition>
>>     </extension>
>>
>>
>>
>> Hynek
>>
>>  ------------------------------
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cesar.bermudez at gmail.com  Sat Oct 22 01:56:38 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Fri, 21 Oct 2011 15:56:38 -0600
Subject: [Freeswitch-users]  cdr reconciliation
Message-ID: <CADrcvV_jdeyszW=B+vp2qEo4=R2kd7v1UqA1OxSDno=o2QjbYw@mail.gmail.com>

Hi guys,
what you do whan have a cdr dispute?
what mechanic do, how you compare
what tools use?

Best regards.
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From rhuddleston at gmail.com  Sat Oct 22 02:15:44 2011
From: rhuddleston at gmail.com (Robert-IPhone)
Date: Fri, 21 Oct 2011 18:15:44 -0400
Subject: [Freeswitch-users] cdr reconciliation
In-Reply-To: <CADrcvV_jdeyszW=B+vp2qEo4=R2kd7v1UqA1OxSDno=o2QjbYw@mail.gmail.com>
References: <CADrcvV_jdeyszW=B+vp2qEo4=R2kd7v1UqA1OxSDno=o2QjbYw@mail.gmail.com>
Message-ID: <C5FEA9F6-2055-424F-8501-721A4AD1F38D@gmail.com>

SQL SELECT * FROM

Sent from my iPhone 4S

On Oct 21, 2011, at 5:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com> wrote:

> Hi guys, 
> what you do whan have a cdr dispute?
> what mechanic do, how you compare
> what tools use?
> 
> Best regards.
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Oct 22 02:27:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Oct 2011 15:27:48 -0700
Subject: [Freeswitch-users] Bridge problem - (randomly) leg A looses
 events and communication with leg B
In-Reply-To: <CAJWEF2avKuKrCuAMnKQ3-RaMWaEQJzkyqyPGEvY=Do8+nrEScg@mail.gmail.com>
References: <CAJWEF2avKuKrCuAMnKQ3-RaMWaEQJzkyqyPGEvY=Do8+nrEScg@mail.gmail.com>
Message-ID: <CAKzWOxWV4YmA9WxgR5Vza-K+71Cyu0ysHbBtFohagD8KEGAR9Q@mail.gmail.com>

I would get this with a siptrace included:
sofia global siptrace on

Also, use pastebin.freeswitch.org and set syntax highlighting to "FreeSWITCH
Log"

-MC

On Thu, Oct 20, 2011 at 3:50 AM, Mateusz Bartczak <netcentrica at gmail.com>wrote:

> Hi All,
>
> My scenario:
> number dialed by user test_user
> dial plan configured to send outbound call to event socket listening on
> 127.0.0.1:11111
> outbound call is intercepted, bridge application is executed to connect leg
> A (test_user) with dialed number via my SIP provider (leg B)
>
> The problem is, that (completely randomly - sometimes it's OK, sometimes
> not) user's session is desynchronized with events from leg B.
> Leg B dials number and connects via SIP provider to the destination number
> (that's always OK)
> But (randomly) leg A doesn't get ring back, no connect event, still in
> "calling" state until 30 second timeout (in the same time leg B is connected
> OK) but there is some problem with passing events from leg B to leg A
>
> Bridge params:
> {ignore_early_media=true,bypass_media=false,origination_caller_id_name=004822333222111,origination_caller_id_number=004822333222111,effective_caller_id_number=004822333222111,effective_caller_id_name=004822333222111,execute_on_answer=sched_hangup
> +5700 alloted_timeout}sofia/gateway/supergateway/0000048222019475
>
> I've tried to change socket mode to sync but with no success
>
> I use following Java code to execute bridge application:
>
> SendMsg msg = new SendMsg();
> msg.addCallCommand("execute");
> msg.addExecuteAppName("bridge");
> msg.addExecuteAppArg(dialstring);
> msg.addEventLock();
>
> Full freeswitch console log link: http://pastebin.com/eLCmMXRB
>
> I will greatly appreciate your help solving this problem
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Oct 22 02:29:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Oct 2011 15:29:51 -0700
Subject: [Freeswitch-users] Make freeswitch use spanish sounds...
In-Reply-To: <1552494118-1319114000-cardhu_decombobulator_blackberry.rim.net-1593421769-@b11.c27.bise6.blackberry>
References: <CANtw_ZOVVHN=xD-yHafyrYaMx5+snSfMr3RjKxUS43DAY57V-w@mail.gmail.com>
	<CAKzWOxVzutgu=mPz7j1fEU1kQ_KELmZwq9-BG7uen84j0Bn28A@mail.gmail.com>
	<CANtw_ZPZN4enVMAyD1XPzj9vWEVD-8aiBL+LtXB09CyGdkKALg@mail.gmail.com>
	<CAKzWOxVSoY14bf5VHFb_B78iVtnp3D3mLx7NAptQAqFpA-=eZg@mail.gmail.com>
	<1552494118-1319114000-cardhu_decombobulator_blackberry.rim.net-1593421769-@b11.c27.bise6.blackberry>
Message-ID: <CAKzWOxWgxm-hsHpBQdHCns7uaD8PZtO7j=-GN7zHyKrnj8HkNg@mail.gmail.com>

?Excelente! Let me know when you are done and I'll test it as best I can. I
only know a few dozen words in Spanish but I can at least test the basic
stuff.

-MC

On Thu, Oct 20, 2011 at 5:33 AM, Luis Jimenez <ljjimenez at gmail.com> wrote:

> Hola Michael, i am using spanish prompts, but mod_say_es is incomplete and
> i have been doing changes to the code, later i will pastebin those changes
> so you can check and approve.
>
> Luis Jimenez
>
> -----Original Message-----
> From: Michael Collins <msc at freeswitch.org>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Wed, 19 Oct 2011 14:45:38
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Make freeswitch use spanish sounds...
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sat Oct 22 02:33:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Oct 2011 15:33:07 -0700
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <CAGQG5Ma1MrSYVWvRcRAWjby9DP2m5WRQ4=6=UNESbNsnyh1aSQ@mail.gmail.com>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>
	<CAGQG5Ma1MrSYVWvRcRAWjby9DP2m5WRQ4=6=UNESbNsnyh1aSQ@mail.gmail.com>
Message-ID: <CAKzWOxV+Ob=u3j=bYSar0942QhhfcU=Ty9JsjhSJ+-3+vHWoMQ@mail.gmail.com>

On Thu, Oct 20, 2011 at 8:41 AM, Abdul Basit <basit.engg at gmail.com> wrote:

> yes. but 1.0.6 was quite good as i heard. This might be some simple issue.


Whoever told you that was either high or living in the past. :) The latest
git HEAD is way better than 1.0.6

-MC
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From msc at freeswitch.org  Sat Oct 22 02:35:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Oct 2011 15:35:03 -0700
Subject: [Freeswitch-users] LUA transfer result:0
In-Reply-To: <2C9B38E79A44400D8F7509B1010A00C4@omni1.local>
References: <2C9B38E79A44400D8F7509B1010A00C4@omni1.local>
Message-ID: <CAKzWOxWrPegf6X4C0FBSmowsBxRoOvx2J40ZSLXmNMoHe+QgTQ@mail.gmail.com>

Don't forget that in a Lua dp script the script must "end" before the
transfer takes place. Make sure that there isn't a logic loop or something
in your code preventing it from ending. If you pastebin your script we'll
take a look.
-MC

On Thu, Oct 20, 2011 at 2:27 PM, Anestis Mavro <devel at omninet.eu> wrote:

>  Hi,****
>
> ** **
>
> I have tried two different ways of transfer in LUA:****
>
> session:transfer(number,?XML?,?default?)****
>
> ** **
>
> And ****
>
> ** **
>
> Session:execute(?transfer?,?12345 XML default?)****
>
> ** **
>
> And I can?t make it working.****
>
> ** **
>
> I get ?switch_cpp.cpp:797 transfer result: 0?****
>
> ** **
>
> The same transfer from within the dialplan works fine****
>
> ** **
>
> The call is not answered before the transfer.****
>
> ** **
>
> Does anybody have an idea?****
>
> ** **
>
> Thanks****
>
> Anestis****
>
> ** **
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 5054 (20100423) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Oct 22 02:40:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Oct 2011 15:40:24 -0700
Subject: [Freeswitch-users] cdr reconciliation
In-Reply-To: <CADrcvV_jdeyszW=B+vp2qEo4=R2kd7v1UqA1OxSDno=o2QjbYw@mail.gmail.com>
References: <CADrcvV_jdeyszW=B+vp2qEo4=R2kd7v1UqA1OxSDno=o2QjbYw@mail.gmail.com>
Message-ID: <CAKzWOxVsmMySmpuf5OYH0Zc1gpNUuo3RTr6wFZw0TrSg0s=SuQ@mail.gmail.com>

A dispute between which two parties?
-MC

On Fri, Oct 21, 2011 at 2:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com>wrote:

> Hi guys,
> what you do whan have a cdr dispute?
> what mechanic do, how you compare
> what tools use?
>
> Best regards.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cesar.bermudez at gmail.com  Sat Oct 22 03:04:48 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Fri, 21 Oct 2011 17:04:48 -0600
Subject: [Freeswitch-users] cdr reconciliation
In-Reply-To: <CAKzWOxVsmMySmpuf5OYH0Zc1gpNUuo3RTr6wFZw0TrSg0s=SuQ@mail.gmail.com>
References: <CADrcvV_jdeyszW=B+vp2qEo4=R2kd7v1UqA1OxSDno=o2QjbYw@mail.gmail.com>
	<CAKzWOxVsmMySmpuf5OYH0Zc1gpNUuo3RTr6wFZw0TrSg0s=SuQ@mail.gmail.com>
Message-ID: <CADrcvV8FGV4s1O6VRM1mFkYr7pf06FZyS6hv_8QZJ6w_Ju8VXA@mail.gmail.com>

 A dispute between me and one provider.
They give me they CDR and they have more calls than my cdr
how you guys do that job?

On Fri, Oct 21, 2011 at 4:40 PM, Michael Collins <msc at freeswitch.org> wrote:

> A dispute between which two parties?
> -MC
>
> On Fri, Oct 21, 2011 at 2:56 PM, Cesar Bermudez <cesar.bermudez at gmail.com>wrote:
>
>> Hi guys,
>> what you do whan have a cdr dispute?
>> what mechanic do, how you compare
>> what tools use?
>>
>> Best regards.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rhow at exemail.com.au  Sat Oct 22 03:53:17 2011
From: rhow at exemail.com.au (Ryan How)
Date: Sat, 22 Oct 2011 07:53:17 +0800
Subject: [Freeswitch-users] Problem sending E-mail in Freeswitch on
	windows
In-Reply-To: <1319108246414-6912478.post@n2.nabble.com>
References: <1319108246414-6912478.post@n2.nabble.com>
Message-ID: <4EA205ED.70202@exemail.com.au>

This might be a dumb question, but have you tried restarting freeswitch 
after changing the xml file ?

On 20/10/2011 6:57 PM, ayobami wrote:
> Please people help me out, I want to send email in a lua script of mine in
> freeswitch the code is below
>
> freeswitch.email("ayobami at programmer.net",
>                   "ayobami.adewole at gmail.com",
>                   "subject: testing fs mail function",
>                   "checking to see if this guy works")
> and I have downloaded the msmtp.exe and placed in c:\windows\system32 folder
> as directed
> i have equally changed the configuration in switch.conf.xml
> but each time i keep running the script,
> I keep getting the same error message, which is
> sendmail is not recognised as an internal or external command
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-sending-E-mail-in-Freeswitch-on-windows-tp6912478p6912478.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Sat Oct 22 09:58:46 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Oct 2011 01:58:46 -0400
Subject: [Freeswitch-users] bind_digits_action and conference
Message-ID: <A26AECCFF2E54BFC8FAE06225FA70D0B@e1705>

Hi all,

when bind_digit_action dtmf keys are fired I noticed
that the whole conference audio is muted.
is there any option to the audio of conference while bind_digits_action is fired ?

Thanks

Franck
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From infos at madovsky.org  Sat Oct 22 10:32:40 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Oct 2011 02:32:40 -0400
Subject: [Freeswitch-users] bind_digits_action and conference
References: <A26AECCFF2E54BFC8FAE06225FA70D0B@e1705>
Message-ID: <4B55697080274280AB5AB4F9B4248B1F@e1705>

I answer to myself,
I forgot the fact that if bind_digit_action execute an extension so the leg is moded out from the conference when it fires the keys.

thanks

Franck
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, October 22, 2011 1:58 AM
  Subject: [Freeswitch-users] bind_digits_action and conference


  Hi all,

  when bind_digit_action dtmf keys are fired I noticed
  that the whole conference audio is muted.
  is there any option to the audio of conference while bind_digits_action is fired ?

  Thanks

  Franck


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



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.villasmil.work at gmail.com  Sat Oct 22 19:15:05 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 22 Oct 2011 17:15:05 +0200
Subject: [Freeswitch-users] Reloading sofia+distributor
Message-ID: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>

Hello,

I understand that if I make a change in a gateway, like IP address I need to
first kill the gw because fs doesn't updated the config until all sessions
are finished.
If the gateway has no traffic, that's fine. But the idea would be for the
change to take effect starting with the new incoming calls and allow the old
ones to end normally.

Also, when you kill a gw and then reload it, and incoming call between the
kill and the reload will NOT be handled... this is just not right for a
production environment..

Is there any other way of doing this WITHOUT actually:

- not handling new incoming calls?
- having to kill the gw?

Thanks!

David
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From jack at livecall.com  Sat Oct 22 19:27:52 2011
From: jack at livecall.com (Jack)
Date: Sat, 22 Oct 2011 08:27:52 -0700
Subject: [Freeswitch-users] Problem sending E-mail in Freeswitch on
	windows
In-Reply-To: <4EA205ED.70202@exemail.com.au>
References: <1319108246414-6912478.post@n2.nabble.com>
	<4EA205ED.70202@exemail.com.au>
Message-ID: <4EA2E0F8.9000502@livecall.com>

I have not used freeswitch.email, and I have not used msmtp.exe  but I 
have used sendmail in web applications and VB scripting.

When you get the error "sendmail is not recognised as an internal or 
external command" it means that windows is not finding sendmail.exe.
If you are using msmtp.exe instead of sendmail.exe you need to call 
msmtp.exe instead of sendmail or get sendmail and use it.
Downloadsendmail.zip <http://www.petri.co.il/software/sendmail.zip>(6kb)
I read where someone just changed the filename from msmtp.exe to 
sendmail.exe and that worked for them.

Jack



On 10/21/2011 4:53 PM, Ryan How wrote:
> This might be a dumb question, but have you tried restarting freeswitch
> after changing the xml file ?
>
> On 20/10/2011 6:57 PM, ayobami wrote:
>> Please people help me out, I want to send email in a lua script of mine in
>> freeswitch the code is below
>>
>> freeswitch.email("ayobami at programmer.net",
>>                    "ayobami.adewole at gmail.com",
>>                    "subject: testing fs mail function",
>>                    "checking to see if this guy works")
>> and I have downloaded the msmtp.exe and placed in c:\windows\system32 folder
>> as directed
>> i have equally changed the configuration in switch.conf.xml
>> but each time i keep running the script,
>> I keep getting the same error message, which is
>> sendmail is not recognised as an internal or external command
>>
>> --
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-sending-E-mail-in-Freeswitch-on-windows-tp6912478p6912478.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gcd at i.ph  Sat Oct 22 19:58:33 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 22 Oct 2011 23:58:33 +0800
Subject: [Freeswitch-users] gsmopen compile error
Message-ID: <CAHKqMb7Ff8d5AnWY+qZz9Oan9v82PWrJS+1DFpwx2vzn2W0hkA@mail.gmail.com>

hello guys,

i encounter a compilation error after i execute "make install" command w/
the ff messages:

Compiling
/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp...
Compiling
/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp ...
/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp: In
function 'switch_status_t
mod_gsmopen_load(switch_loadable_module_interface_t**,
switch_memory_pool_t*)':
/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp:2158:
error: invalid conversion from 'switch_status_t (*)(const char*, const
char*, const char*, const char*, const char*, const char*, const char*)' to
'switch_status_t (*)(switch_event_t*)'
make[1]: *** [mod_gsmopen.lo] Error 1
make: *** [install] Error 1

i've got the latest build, by doing "make current":
FreeSWITCH Version 1.0.head (git-4ceca18 2011-10-21 20-38-41 -0500)

pls help me point out the problem.

tks,
-nandy
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From hynek.cihlar at gmail.com  Sat Oct 22 20:03:45 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Sat, 22 Oct 2011 18:03:45 +0200
Subject: [Freeswitch-users] Reloading sofia+distributor
In-Reply-To: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
Message-ID: <CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com>

I was also having issues with changing gateways. A solution I came up with
was to have globally unique names and changing a gateway was actually
destroying the old one and recreating a new one under new name. The downside
is that DB lookup must be performed before new call is initiated.

Hynek



On Sat, Oct 22, 2011 at 5:15 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> I understand that if I make a change in a gateway, like IP address I need
> to first kill the gw because fs doesn't updated the config until all
> sessions are finished.
> If the gateway has no traffic, that's fine. But the idea would be for the
> change to take effect starting with the new incoming calls and allow the old
> ones to end normally.
>
> Also, when you kill a gw and then reload it, and incoming call between the
> kill and the reload will NOT be handled... this is just not right for a
> production environment..
>
> Is there any other way of doing this WITHOUT actually:
>
> - not handling new incoming calls?
> - having to kill the gw?
>
> Thanks!
>
> David
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble_01394 at slickdeals.endjunk.com  Sat Oct 22 20:08:05 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sat, 22 Oct 2011 09:08:05 -0700 (PDT)
Subject: [Freeswitch-users] gsmopen compile error
In-Reply-To: <CAHKqMb7Ff8d5AnWY+qZz9Oan9v82PWrJS+1DFpwx2vzn2W0hkA@mail.gmail.com>
References: <CAHKqMb7Ff8d5AnWY+qZz9Oan9v82PWrJS+1DFpwx2vzn2W0hkA@mail.gmail.com>
Message-ID: <1319299685372-6920361.post@n2.nabble.com>

I just updated to git 4ceca1890d10787be6c86446599b95f63d60b014 and cross
compiled mod_gsmopen without a problem.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
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From david.villasmil.work at gmail.com  Sat Oct 22 20:10:13 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 22 Oct 2011 18:10:13 +0200
Subject: [Freeswitch-users] Reloading sofia+distributor
In-Reply-To: <CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com>
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
	<CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com>
Message-ID: <CAFGRPVpBEgV82zz7o7WFvVtWHg=t+cFJQEmBqMEZ3H3wQULuxw@mail.gmail.com>

Hello,

Thanks for your answer...

But then that's the same as killing it and reloading it, no?

David

On Sat, Oct 22, 2011 at 6:03 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> I was also having issues with changing gateways. A solution I came up with
> was to have globally unique names and changing a gateway was actually
> destroying the old one and recreating a new one under new name. The downside
> is that DB lookup must be performed before new call is initiated.
>
> Hynek
>
>
>
> On Sat, Oct 22, 2011 at 5:15 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>>
>> I understand that if I make a change in a gateway, like IP address I need
>> to first kill the gw because fs doesn't updated the config until all
>> sessions are finished.
>> If the gateway has no traffic, that's fine. But the idea would be for the
>> change to take effect starting with the new incoming calls and allow the old
>> ones to end normally.
>>
>> Also, when you kill a gw and then reload it, and incoming call between the
>> kill and the reload will NOT be handled... this is just not right for a
>> production environment..
>>
>> Is there any other way of doing this WITHOUT actually:
>>
>> - not handling new incoming calls?
>> - having to kill the gw?
>>
>> Thanks!
>>
>> David
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nbhatti at gmail.com  Sat Oct 22 20:16:29 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sat, 22 Oct 2011 19:16:29 +0300
Subject: [Freeswitch-users] Reloading sofia+distributor
In-Reply-To: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
Message-ID: <CACak_yzLjJckA7iKb-NfX6wpmguqMgqTLvT5NUBd5Dt+ge_1wg@mail.gmail.com>

Can't we make a new temporary gateway, route the calls to the new
gateway, let the calls finish the existing old GW, re-configure the
old gateway and then re-route all the calls back to the old one,
remote the temporary one. Offcourse this would require a lot of
changes but looks like a better option?

-B

On Sat, Oct 22, 2011 at 6:15 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Hello,
> I understand that if I make a change in a gateway, like IP address I need to
> first kill the gw because fs doesn't updated the config until all sessions
> are finished.
> If the gateway has no traffic, that's fine. But the idea would be for the
> change to take effect starting with the new incoming calls and allow the old
> ones to end normally.
> Also, when you kill a gw and then reload it, and incoming call between the
> kill and the reload will NOT be handled... this is just not right for a
> production environment..
> Is there any other way of doing this WITHOUT actually:
> - not handling new incoming calls?
> - having to kill the gw?
> Thanks!
> David
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Sat Oct 22 20:17:32 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Oct 2011 12:17:32 -0400
Subject: [Freeswitch-users] Reloading sofia+distributor
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
Message-ID: <AAD4FEC3368A4CFB8735360C2DDF6BC4@e1705>

I think you should ready more carefully the wiki doc on this subject (reload)

Regards
  ----- Original Message ----- 
  From: David Villasmil 
  To: FreeSWITCH Users Help 
  Sent: Saturday, October 22, 2011 11:15 AM
  Subject: [Freeswitch-users] Reloading sofia+distributor


  Hello,


  I understand that if I make a change in a gateway, like IP address I need to first kill the gw because fs doesn't updated the config until all sessions are finished.
  If the gateway has no traffic, that's fine. But the idea would be for the change to take effect starting with the new incoming calls and allow the old ones to end normally.


  Also, when you kill a gw and then reload it, and incoming call between the kill and the reload will NOT be handled... this is just not right for a production environment..


  Is there any other way of doing this WITHOUT actually:


  - not handling new incoming calls?
  - having to kill the gw?


  Thanks!


  David


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



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Oct 22 20:20:14 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Oct 2011 12:20:14 -0400
Subject: [Freeswitch-users] Reloading sofia+distributor
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
Message-ID: <BAB4FB5D4C294553A32EBDA6E0DA3D6F@e1705>

example: to reload idstributor
reload mod_distributor
  ----- Original Message ----- 
  From: David Villasmil 
  To: FreeSWITCH Users Help 
  Sent: Saturday, October 22, 2011 11:15 AM
  Subject: [Freeswitch-users] Reloading sofia+distributor


  Hello,


  I understand that if I make a change in a gateway, like IP address I need to first kill the gw because fs doesn't updated the config until all sessions are finished.
  If the gateway has no traffic, that's fine. But the idea would be for the change to take effect starting with the new incoming calls and allow the old ones to end normally.


  Also, when you kill a gw and then reload it, and incoming call between the kill and the reload will NOT be handled... this is just not right for a production environment..


  Is there any other way of doing this WITHOUT actually:


  - not handling new incoming calls?
  - having to kill the gw?


  Thanks!


  David


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



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Oct 22 20:23:50 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Oct 2011 12:23:50 -0400
Subject: [Freeswitch-users] Reloading sofia+distributor
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com><CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com>
	<CAFGRPVpBEgV82zz7o7WFvVtWHg=t+cFJQEmBqMEZ3H3wQULuxw@mail.gmail.com>
Message-ID: <503994D4609E453E891E2E9F2E5EB307@e1705>

and if you reload the profile that includes these gateways ?
  ----- Original Message ----- 
  From: David Villasmil 
  To: FreeSWITCH Users Help 
  Sent: Saturday, October 22, 2011 12:10 PM
  Subject: Re: [Freeswitch-users] Reloading sofia+distributor


  Hello,


  Thanks for your answer... 


  But then that's the same as killing it and reloading it, no?


  David


  On Sat, Oct 22, 2011 at 6:03 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

    I was also having issues with changing gateways. A solution I came up with was to have globally unique names and changing a gateway was actually destroying the old one and recreating a new one under new name. The downside is that DB lookup must be performed before new call is initiated.

    Hynek




    On Sat, Oct 22, 2011 at 5:15 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:

      Hello,


      I understand that if I make a change in a gateway, like IP address I need to first kill the gw because fs doesn't updated the config until all sessions are finished.
      If the gateway has no traffic, that's fine. But the idea would be for the change to take effect starting with the new incoming calls and allow the old ones to end normally.


      Also, when you kill a gw and then reload it, and incoming call between the kill and the reload will NOT be handled... this is just not right for a production environment..


      Is there any other way of doing this WITHOUT actually:


      - not handling new incoming calls?
      - having to kill the gw?


      Thanks!


      David



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





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






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



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From hynek.cihlar at gmail.com  Sat Oct 22 20:26:53 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Sat, 22 Oct 2011 18:26:53 +0200
Subject: [Freeswitch-users] Reloading sofia+distributor
In-Reply-To: <CAFGRPVpBEgV82zz7o7WFvVtWHg=t+cFJQEmBqMEZ3H3wQULuxw@mail.gmail.com>
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
	<CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com>
	<CAFGRPVpBEgV82zz7o7WFvVtWHg=t+cFJQEmBqMEZ3H3wQULuxw@mail.gmail.com>
Message-ID: <CAJh8XdWbdLa0Zk05wNDcx4FbHgSJsF5Z1aWf4zda=ZGWLAbYRw@mail.gmail.com>

No. Killing the gateway isn't instantaneous. There is some delay before the
gateway disappears completely. Why it works like that or whether it can be
tuned I have no idea. For my case, the unique names work just fine.

Hynek



On Sat, Oct 22, 2011 at 6:10 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> Thanks for your answer...
>
> But then that's the same as killing it and reloading it, no?
>
> David
>
>
> On Sat, Oct 22, 2011 at 6:03 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> I was also having issues with changing gateways. A solution I came up with
>> was to have globally unique names and changing a gateway was actually
>> destroying the old one and recreating a new one under new name. The downside
>> is that DB lookup must be performed before new call is initiated.
>>
>> Hynek
>>
>>
>>
>> On Sat, Oct 22, 2011 at 5:15 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I understand that if I make a change in a gateway, like IP address I need
>>> to first kill the gw because fs doesn't updated the config until all
>>> sessions are finished.
>>> If the gateway has no traffic, that's fine. But the idea would be for the
>>> change to take effect starting with the new incoming calls and allow the old
>>> ones to end normally.
>>>
>>> Also, when you kill a gw and then reload it, and incoming call between
>>> the kill and the reload will NOT be handled... this is just not right for a
>>> production environment..
>>>
>>> Is there any other way of doing this WITHOUT actually:
>>>
>>> - not handling new incoming calls?
>>> - having to kill the gw?
>>>
>>> Thanks!
>>>
>>> David
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Oct 22 20:40:57 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 22 Oct 2011 18:40:57 +0200
Subject: [Freeswitch-users] Reloading sofia+distributor
In-Reply-To: <503994D4609E453E891E2E9F2E5EB307@e1705>
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com>
	<CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com>
	<CAFGRPVpBEgV82zz7o7WFvVtWHg=t+cFJQEmBqMEZ3H3wQULuxw@mail.gmail.com>
	<503994D4609E453E891E2E9F2E5EB307@e1705>
Message-ID: <CAFGRPVpxOfSM8gVzyStmcibwto3rm-Puw2ecMA+BTBtQwDgwaA@mail.gmail.com>

Reloading sofia doesn't changes the IP address, for one.
Relaoding mod_distributor has nothing to do with the GW configuration, only
the distribution if the calls amongst the gateways.

you *must* kill it, then reload it.

Of course, you can always create a tmp gw, and then rename it, but that
sounds horrible to me, not elegant at all.


David

On Sat, Oct 22, 2011 at 6:23 PM, Madovsky <infos at madovsky.org> wrote:

> **
> and if you reload the profile that includes these gateways ?
>
> ----- Original Message -----
> *From:* David Villasmil <david.villasmil.work at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, October 22, 2011 12:10 PM
> *Subject:* Re: [Freeswitch-users] Reloading sofia+distributor
>
> Hello,
>
> Thanks for your answer...
>
> But then that's the same as killing it and reloading it, no?
>
> David
>
> On Sat, Oct 22, 2011 at 6:03 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> I was also having issues with changing gateways. A solution I came up with
>> was to have globally unique names and changing a gateway was actually
>> destroying the old one and recreating a new one under new name. The downside
>> is that DB lookup must be performed before new call is initiated.
>>
>> Hynek
>>
>>
>>
>>   On Sat, Oct 22, 2011 at 5:15 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> I understand that if I make a change in a gateway, like IP address I need
>>> to first kill the gw because fs doesn't updated the config until all
>>> sessions are finished.
>>> If the gateway has no traffic, that's fine. But the idea would be for the
>>> change to take effect starting with the new incoming calls and allow the old
>>> ones to end normally.
>>>
>>> Also, when you kill a gw and then reload it, and incoming call between
>>> the kill and the reload will NOT be handled... this is just not right for a
>>> production environment..
>>>
>>> Is there any other way of doing this WITHOUT actually:
>>>
>>> - not handling new incoming calls?
>>> - having to kill the gw?
>>>
>>> Thanks!
>>>
>>> David
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From masood24 at gmail.com  Sat Oct 22 20:52:45 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Sat, 22 Oct 2011 19:52:45 +0300
Subject: [Freeswitch-users] Digest Authentication issue
Message-ID: <CAMd3iJLP5a8J+zB6xpJ=+B+0DsJtiO_t=_kG=CJfgyQTZ47jTw@mail.gmail.com>

Dear Gabriel,
What I am trying to do here is that there should be no entry in Directory
like there are 1000 and other entries. I only want that every request for
register or authentication should be processed by radius client. like
everytime hits come radius client should send request to server and if
returns code is +ve or 0 then it should register otherwise not.

I have seen the topic link that you sent but still i am not clear by that.

Thanks in advanced for your help.



>
>
> ---------- Forwarded message ----------
> From: Gabriel Gunderson <gabe at gundy.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 20 Oct 2011 00:45:07 -0600
> Subject: Re: [Freeswitch-users] Digest Authentication issue
> On Fri, Oct 14, 2011 at 2:39 PM, Masood Ahmed <masood24 at gmail.com> wrote:
> > I had successfully installed and configured radiusclient. I do not know
> that
> > what next lines should i do to get users registered on my freeswitch
> using
> > digest authentication with my radius server,
> > Like I have my radsius server and billing system. Now what lines should i
> > add in sofit conf or in dialplans that every time client hits server
> request
> > should be sent to radius server and what lines should i add to register
> them
> > if i get proper return code and how to do that.
>
> Have you looked here?
>
> http://wiki.freeswitch.org/wiki/Mod_xml_radius
>
>
> Best,
> Gabe
>

-- 
*Regards,
Masood Ahmed
*
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From infos at madovsky.org  Sat Oct 22 20:55:30 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Oct 2011 12:55:30 -0400
Subject: [Freeswitch-users] Reloading sofia+distributor
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com><CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com><CAFGRPVpBEgV82zz7o7WFvVtWHg=t+cFJQEmBqMEZ3H3wQULuxw@mail.gmail.com><503994D4609E453E891E2E9F2E5EB307@e1705>
	<CAFGRPVpxOfSM8gVzyStmcibwto3rm-Puw2ecMA+BTBtQwDgwaA@mail.gmail.com>
Message-ID: <665ADE88CDED420C88D15C378B09C48D@e1705>

Ok sorry, your email subject confused me
  ----- Original Message ----- 
  From: David Villasmil 
  To: FreeSWITCH Users Help 
  Sent: Saturday, October 22, 2011 12:40 PM
  Subject: Re: [Freeswitch-users] Reloading sofia+distributor


  Reloading sofia doesn't changes the IP address, for one.
  Relaoding mod_distributor has nothing to do with the GW configuration, only the distribution if the calls amongst the gateways.


  you *must* kill it, then reload it.


  Of course, you can always create a tmp gw, and then rename it, but that sounds horrible to me, not elegant at all.




  David


  On Sat, Oct 22, 2011 at 6:23 PM, Madovsky <infos at madovsky.org> wrote:

    and if you reload the profile that includes these gateways ?
      ----- Original Message ----- 
      From: David Villasmil 
      To: FreeSWITCH Users Help 
      Sent: Saturday, October 22, 2011 12:10 PM
      Subject: Re: [Freeswitch-users] Reloading sofia+distributor


      Hello, 


      Thanks for your answer... 


      But then that's the same as killing it and reloading it, no?


      David


      On Sat, Oct 22, 2011 at 6:03 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

        I was also having issues with changing gateways. A solution I came up with was to have globally unique names and changing a gateway was actually destroying the old one and recreating a new one under new name. The downside is that DB lookup must be performed before new call is initiated.

        Hynek




        On Sat, Oct 22, 2011 at 5:15 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:

          Hello, 


          I understand that if I make a change in a gateway, like IP address I need to first kill the gw because fs doesn't updated the config until all sessions are finished.
          If the gateway has no traffic, that's fine. But the idea would be for the change to take effect starting with the new incoming calls and allow the old ones to end normally.


          Also, when you kill a gw and then reload it, and incoming call between the kill and the reload will NOT be handled... this is just not right for a production environment..


          Is there any other way of doing this WITHOUT actually:


          - not handling new incoming calls?
          - having to kill the gw?


          Thanks!


          David



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From infos at madovsky.org  Sat Oct 22 20:56:42 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 22 Oct 2011 12:56:42 -0400
Subject: [Freeswitch-users] Reloading sofia+distributor
References: <CAFGRPVrBU4=7enL8znHtZ6tkfZxAg55OH1=+990BhnAC+7r_FA@mail.gmail.com><CAJh8XdXLspCU-O1Z2+uN0AhHq-jjxHX_5U_EKvMhpaPzDpdUGA@mail.gmail.com><CAFGRPVpBEgV82zz7o7WFvVtWHg=t+cFJQEmBqMEZ3H3wQULuxw@mail.gmail.com>
	<CAJh8XdWbdLa0Zk05wNDcx4FbHgSJsF5Z1aWf4zda=ZGWLAbYRw@mail.gmail.com>
Message-ID: <C8E05C3FE4DF4C65B5504D29F31F8443@e1705>

you should have unique names in this case
  ----- Original Message ----- 
  From: Hynek Cihlar 
  To: FreeSWITCH Users Help 
  Sent: Saturday, October 22, 2011 12:26 PM
  Subject: Re: [Freeswitch-users] Reloading sofia+distributor


  No. Killing the gateway isn't instantaneous. There is some delay before the gateway disappears completely. Why it works like that or whether it can be tuned I have no idea. For my case, the unique names work just fine.

  Hynek




  On Sat, Oct 22, 2011 at 6:10 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:

    Hello,


    Thanks for your answer... 


    But then that's the same as killing it and reloading it, no?


    David



    On Sat, Oct 22, 2011 at 6:03 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

      I was also having issues with changing gateways. A solution I came up with was to have globally unique names and changing a gateway was actually destroying the old one and recreating a new one under new name. The downside is that DB lookup must be performed before new call is initiated.

      Hynek




      On Sat, Oct 22, 2011 at 5:15 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:

        Hello,


        I understand that if I make a change in a gateway, like IP address I need to first kill the gw because fs doesn't updated the config until all sessions are finished.
        If the gateway has no traffic, that's fine. But the idea would be for the change to take effect starting with the new incoming calls and allow the old ones to end normally.


        Also, when you kill a gw and then reload it, and incoming call between the kill and the reload will NOT be handled... this is just not right for a production environment..


        Is there any other way of doing this WITHOUT actually:


        - not handling new incoming calls?
        - having to kill the gw?


        Thanks!


        David



        FreeSWITCH-users mailing 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 hynek.cihlar at gmail.com  Sat Oct 22 21:39:44 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Sat, 22 Oct 2011 19:39:44 +0200
Subject: [Freeswitch-users] vars.xml in Lua
Message-ID: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>

Are the parameters defined in the XML config files visible to Lua scrips? In
this case it is the script for loading up the directory.

Hynek
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From gustavomarsico at gmail.com  Sun Oct 23 00:07:43 2011
From: gustavomarsico at gmail.com (=?iso-8859-1?Q?Gustavo_M=E1rsico?=)
Date: Sat, 22 Oct 2011 17:07:43 -0300
Subject: [Freeswitch-users] Using different contexts in gateways than the
	profile context
Message-ID: <E0E87306-D908-497B-A9B1-5B95B5833D39@gmail.com>

Hi all

I'm trying to build a useful way to handle the remote gateways in wholesale scenarios. I'm checking the xml_curl due the very complex (and dynamic) rules of wholesale.
As far as I see, besides the IP address of the remote gw, the context could be a good differentiator of different remote entities.

I tried to change the Context of the gateways in the external profile (conf/sip_profiles/external/ws31.xml) to something like:

<include>
    <gateway name="10.8.0.10">
      <param name="register" value="false"/>
      <param name="ping" value="25"/>
      <param name="context" value="tgrp31"/>
    </gateway>
</include>

But for some reason Caller-Context and Hunt-Context are shown as 'public' context, which is the context configured in the external profile (conf/sip_profiles/external.xml).

Also, I tried to remove the context on the external.xml and keep the context in the external/ws31.xml file, to know if profile context configuration has precedence over the gateway context configuration. But this time, Caller-Context and Hunt-Context show 'default' as context (seems like mod_sofia is not using the context defined in the gateway).

Looking the src:

In the file src/endpoints/mod_sofia/sofia.c (parse_gateways function)

            if (!context) {
                context = "default";
            }

Which is perfectly fine and it's according to the behavior I've seen.

Below that, I found where the context is loaded:

                } else if (!strcmp(var, "context")) {
                    context = val;

A few lines after that:

gateway->register_context = switch_core_strdup(gateway->pool, context);

I did some debugs in that function and the context is well recorded.


In the sofia_handle_sip_i_invite function the register_context variable is used again:

if (gateway) {
            context = switch_core_session_strdup(session, gateway->register_context);

But the following comparison:

if (gw_name || gw_param_name) {

always return false, because a "gw+" is expected as part of the destination number.

if (strstr(destination_number, "gw+")) {


Just asking, shouldn't be a gateway treated as a gateway no matter if the destination contains certain characters or not?

I'm using FreeSWITCH Version 1.0.head (git-918d97b 2011-10-13 10-51-34 -0500)


Is this the expected behavior, am I looking this in the wrong way or am I missing something?

Best regards

Gustavo





From masood24 at gmail.com  Sun Oct 23 00:48:22 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Sat, 22 Oct 2011 23:48:22 +0300
Subject: [Freeswitch-users] Digest Authentication issue
In-Reply-To: <CAMd3iJLP5a8J+zB6xpJ=+B+0DsJtiO_t=_kG=CJfgyQTZ47jTw@mail.gmail.com>
References: <CAMd3iJLP5a8J+zB6xpJ=+B+0DsJtiO_t=_kG=CJfgyQTZ47jTw@mail.gmail.com>
Message-ID: <CAMd3iJJeaq2FHSJcz0URQQk96TkbQvoZpeM6figbPKy8YTe8VQ@mail.gmail.com>

Dear Gabriel,

I saw the link what is given for Digest Auhtnetication, There which I
understood is that.
1 construct and XML and place it in directory named "Directory" where switch
ususally looks for users, here at this point my question is that ok i
constructed an XML and placed there, but what liens and where would I call
the RADIUS client which will get the values from user who requested Register
request and will send it to that constructed XML.

Regards
Masood,


post for directory lookup there.

On Sat, Oct 22, 2011 at 7:52 PM, Masood Ahmed <masood24 at gmail.com> wrote:

> Dear Gabriel,
> What I am trying to do here is that there should be no entry in Directory
> like there are 1000 and other entries. I only want that every request for
> register or authentication should be processed by radius client. like
> everytime hits come radius client should send request to server and if
> returns code is +ve or 0 then it should register otherwise not.
>
> I have seen the topic link that you sent but still i am not clear by that.
>
> Thanks in advanced for your help.
>
>
>
>>
>>
>> ---------- Forwarded message ----------
>> From: Gabriel Gunderson <gabe at gundy.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 20 Oct 2011 00:45:07 -0600
>> Subject: Re: [Freeswitch-users] Digest Authentication issue
>> On Fri, Oct 14, 2011 at 2:39 PM, Masood Ahmed <masood24 at gmail.com> wrote:
>> > I had successfully installed and configured radiusclient. I do not know
>> that
>> > what next lines should i do to get users registered on my freeswitch
>> using
>> > digest authentication with my radius server,
>> > Like I have my radsius server and billing system. Now what lines should
>> i
>> > add in sofit conf or in dialplans that every time client hits server
>> request
>> > should be sent to radius server and what lines should i add to register
>> them
>> > if i get proper return code and how to do that.
>>
>> Have you looked here?
>>
>> http://wiki.freeswitch.org/wiki/Mod_xml_radius
>>
>>
>> Best,
>> Gabe
>>
>
> --
> *Regards,
> Masood Ahmed
> *
>
>


-- 
*Regards,
Masood Ahmed
*
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From gcd at i.ph  Sun Oct 23 05:59:21 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Sun, 23 Oct 2011 09:59:21 +0800
Subject: [Freeswitch-users] gsmopen compile error
In-Reply-To: <1319299685372-6920361.post@n2.nabble.com>
References: <CAHKqMb7Ff8d5AnWY+qZz9Oan9v82PWrJS+1DFpwx2vzn2W0hkA@mail.gmail.com>
	<1319299685372-6920361.post@n2.nabble.com>
Message-ID: <CAHKqMb7UEwwgbSLFiaJZqq9neyV9dwo6xokhBWJEzpYYS_GtSQ@mail.gmail.com>

i tried a fresh installation to make sure there are no stale files. the FS
build was ok and running w/ no problem.   when i build gsmopen, i encounter
the same error.

btw, i'm on:
Centos 5.4 kernel 2.6.18-238.12.1.el5.
gcc (GCC) 4.1.2 20080704 (Red Hat 4.1.2-48)

-nandy
On Sun, Oct 23, 2011 at 12:08 AM, mazilo <
Nabble_01394 at slickdeals.endjunk.com> wrote:

> I just updated to git 4ceca1890d10787be6c86446599b95f63d60b014 and cross
> compiled mod_gsmopen without a problem.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/gsmopen-compile-error-tp6920349p6920361.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From basit.engg at gmail.com  Sun Oct 23 14:01:19 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Sun, 23 Oct 2011 15:01:19 +0500
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <CAKzWOxV+Ob=u3j=bYSar0942QhhfcU=Ty9JsjhSJ+-3+vHWoMQ@mail.gmail.com>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>
	<CAGQG5Ma1MrSYVWvRcRAWjby9DP2m5WRQ4=6=UNESbNsnyh1aSQ@mail.gmail.com>
	<CAKzWOxV+Ob=u3j=bYSar0942QhhfcU=Ty9JsjhSJ+-3+vHWoMQ@mail.gmail.com>
Message-ID: <CAGQG5MapcNPkqKwzcFpAbMOFE4WvVqU0jq7MunuF8rHvG_27uA@mail.gmail.com>

Well, I figured out the solution.

Actually, my ubuntu sever was most up-to-date with gcc 4.6. FS was failing
while compiling with gcc versions 4.5 and 4.6

I downgraded it to gcc-4.4 and everything started working.

I spend 10 hrs in installing FS on different platforms, changed header files
path in FS code and comparing the installed packages. On centos  5.7 FS
installed perfectly without any modification because that was using gcc-4.1.

No +ve help i got from forum :(
I cannot welcome the older and newer version arguments. FS must work.

--
regards,

Abdul Basit


On Sat, Oct 22, 2011 at 3:33 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Oct 20, 2011 at 8:41 AM, Abdul Basit <basit.engg at gmail.com> wrote:
>
>> yes. but 1.0.6 was quite good as i heard. This might be some simple issue.
>
>
> Whoever told you that was either high or living in the past. :) The latest
> git HEAD is way better than 1.0.6
>
> -MC
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From basit.engg at gmail.com  Sun Oct 23 14:42:06 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Sun, 23 Oct 2011 15:42:06 +0500
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <CAGQG5MapcNPkqKwzcFpAbMOFE4WvVqU0jq7MunuF8rHvG_27uA@mail.gmail.com>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>
	<CAGQG5Ma1MrSYVWvRcRAWjby9DP2m5WRQ4=6=UNESbNsnyh1aSQ@mail.gmail.com>
	<CAKzWOxV+Ob=u3j=bYSar0942QhhfcU=Ty9JsjhSJ+-3+vHWoMQ@mail.gmail.com>
	<CAGQG5MapcNPkqKwzcFpAbMOFE4WvVqU0jq7MunuF8rHvG_27uA@mail.gmail.com>
Message-ID: <CAGQG5MZWo2q4zyL8SeppDBbbBoJCyu-U62vFu+Ku5SURhcWs3g@mail.gmail.com>

Moreover, I tested with latest git HEAD. This has more issues.
Downgrade to gcc-4.4 works as well.


On Sun, Oct 23, 2011 at 3:01 PM, Abdul Basit <basit.engg at gmail.com> wrote:

> Well, I figured out the solution.
>
> Actually, my ubuntu sever was most up-to-date with gcc 4.6. FS was failing
> while compiling with gcc versions 4.5 and 4.6
>
> I downgraded it to gcc-4.4 and everything started working.
>
> I spend 10 hrs in installing FS on different platforms, changed header
> files path in FS code and comparing the installed packages. On centos  5.7
> FS installed perfectly without any modification because that was using
> gcc-4.1.
>
> No +ve help i got from forum :(
> I cannot welcome the older and newer version arguments. FS must work.
>
> --
> regards,
>
> Abdul Basit
>
>
> On Sat, Oct 22, 2011 at 3:33 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Thu, Oct 20, 2011 at 8:41 AM, Abdul Basit <basit.engg at gmail.com>wrote:
>>
>>> yes. but 1.0.6 was quite good as i heard. This might be some simple
>>> issue.
>>
>>
>> Whoever told you that was either high or living in the past. :) The latest
>> git HEAD is way better than 1.0.6
>>
>> -MC
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


-- 
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
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From freeswitch-list at puzzled.xs4all.nl  Sun Oct 23 14:53:10 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sun, 23 Oct 2011 12:53:10 +0200
Subject: [Freeswitch-users] gsmopen compile error
In-Reply-To: <CAHKqMb7UEwwgbSLFiaJZqq9neyV9dwo6xokhBWJEzpYYS_GtSQ@mail.gmail.com>
References: <CAHKqMb7Ff8d5AnWY+qZz9Oan9v82PWrJS+1DFpwx2vzn2W0hkA@mail.gmail.com>	<1319299685372-6920361.post@n2.nabble.com>
	<CAHKqMb7UEwwgbSLFiaJZqq9neyV9dwo6xokhBWJEzpYYS_GtSQ@mail.gmail.com>
Message-ID: <4EA3F216.1080005@puzzled.xs4all.nl>

On 10/23/2011 03:59 AM, Nandy Dagondon wrote:
> i tried a fresh installation to make sure there are no stale files. the
> FS build was ok and running w/ no problem.   when i build gsmopen, i
> encounter the same error.
>
> btw, i'm on:
> Centos 5.4 kernel 2.6.18-238.12.1.el5.
> gcc (GCC) 4.1.2 20080704 (Red Hat 4.1.2-48)

I don't have a solution for your problem. But I strongly recommend that 
you update your CentOS 5.4 server to CentOS 5.7 as soon as possible. 
Your current 5.4 server has a ton of security issues and makes it 
vulnerable to root exploits. And that is never a good idea for a 
FreeSWITCH server.

All you need to do to update your server from 5.4 to 5.7 is run as root:

yum update

If you want to see how many updates this will apply then run as root:

yum list updates

More information available here:
http://docs.redhat.com/docs/en-US/Red_Hat_Enterprise_Linux/5/index.html

Regards,
Patrick


From Nabble_01394 at slickdeals.endjunk.com  Sun Oct 23 16:43:25 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sun, 23 Oct 2011 05:43:25 -0700 (PDT)
Subject: [Freeswitch-users] gsmopen compile error
In-Reply-To: <CAHKqMb7UEwwgbSLFiaJZqq9neyV9dwo6xokhBWJEzpYYS_GtSQ@mail.gmail.com>
References: <CAHKqMb7Ff8d5AnWY+qZz9Oan9v82PWrJS+1DFpwx2vzn2W0hkA@mail.gmail.com>
	<1319299685372-6920361.post@n2.nabble.com>
	<CAHKqMb7UEwwgbSLFiaJZqq9neyV9dwo6xokhBWJEzpYYS_GtSQ@mail.gmail.com>
Message-ID: <1319373805059-6921935.post@n2.nabble.com>

All I can say is mod_gsmopen requires some libraries, i.e. tiff, spandsp,
gsmlib, gsmme, alsa sound, etc. Make sure you have all those libraries
installed.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/gsmopen-compile-error-tp6920349p6921935.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble_01394 at slickdeals.endjunk.com  Sun Oct 23 16:54:33 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sun, 23 Oct 2011 05:54:33 -0700 (PDT)
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <CAGQG5MapcNPkqKwzcFpAbMOFE4WvVqU0jq7MunuF8rHvG_27uA@mail.gmail.com>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>
	<CAGQG5Ma1MrSYVWvRcRAWjby9DP2m5WRQ4=6=UNESbNsnyh1aSQ@mail.gmail.com>
	<CAKzWOxV+Ob=u3j=bYSar0942QhhfcU=Ty9JsjhSJ+-3+vHWoMQ@mail.gmail.com>
	<CAGQG5MapcNPkqKwzcFpAbMOFE4WvVqU0jq7MunuF8rHvG_27uA@mail.gmail.com>
Message-ID: <1319374473059-6921950.post@n2.nabble.com>


Abdul Basit wrote:
> No +ve help i got from forum :(
> I cannot welcome the older and newer version arguments. FS must work.
Sorry to hear that. Perhaps, we all won't like to live in the past? 

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-while-compiling-switch-swigable-cpp-h-tp6913326p6921950.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nic at itecity.com  Sun Oct 23 16:55:31 2011
From: nic at itecity.com (Network Information Centre)
Date: Sun, 23 Oct 2011 08:55:31 -0400
Subject: [Freeswitch-users] Task was executed late - warning messages
Message-ID: <010c01cc9183$0e42bda0$2ac838e0$@itecity.com>

Hello everyone,

 

I am receiving a great number of these messages, and do not know the service
impact, cause or resolution.

They are continuous, with one to several every hour.  The only items I seem
to see are: heartbeat, check_ip and limit_hash_cleanup.

The late times vary from 2 to 5 seconds.

                

 

2011-10-20 13:42:47.131375 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:43:01.983476 [WARNING] switch_scheduler.c:114 Task was
executed late by 4 seconds 2 check_ip (core)

2011-10-20 13:44:12.353875 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:45:17.292117 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:48:22.629781 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 2 check_ip (core)

2011-10-20 14:03:03.753312 [WARNING] switch_scheduler.c:114 Task was
executed late by 4 seconds 3 limit_hash_cleanup (mod_hash)

2011-10-20 14:38:32.358406 [WARNING] switch_scheduler.c:114 Task was
executed late by 4 seconds 1 heartbeat (core)

2011-10-20 15:17:10.501523 [WARNING] switch_scheduler.c:114 Task was
executed late by 3 seconds 2 check_ip (core)

2011-10-20 16:08:35.544968 [WARNING] switch_scheduler.c:114 Task was
executed late by 2 seconds 1 heartbeat (core)

2011-10-20 17:02:20.345734 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 1 heartbeat (core)

2011-10-20 17:35:13.528281 [WARNING] switch_scheduler.c:114 Task was
executed late by 3 seconds 2 check_ip (core)

2011-10-20 20:02:06.848992 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 1 heartbeat (core)

2011-10-20 20:36:09.719593 [WARNING] switch_scheduler.c:114 Task was
executed late by 3 seconds 1 heartbeat (core)

2011-10-20 21:01:17.014476 [WARNING] switch_scheduler.c:114 Task was
executed late by 2 seconds 2 check_ip (core)

2011-10-20 23:44:22.116851 [WARNING] switch_scheduler.c:114 Task was
executed late by 5 seconds 2 check_ip (core)

2011-10-21 00:39:11.946906 [WARNING] switch_scheduler.c:114 Task was
executed late by 2 seconds 1 heartbeat (core)

 

Any assistance is greatly appreciated.

 

Regards,

 

Don Hunter

e-City Solutions Inc.

 

email:                          don.hunter at 44ecity.com

CA/US Toll-Free:     +1-855-IT-ECITY, ext 200

UK National:             +44 870-820-0286

iNum (inum.net):   +883 5100-0827-0154

 

ITeCity.com  |  FoundU.com  |  StoreLocatorService.com  |  go-Postal.ca  |
getZiped.com | FraudCloud.com

WebPageAlerts.com | TurboBrochures.com | DomainStruck.com | EZ-Verify.com |
VOIPMuch.com

 

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From cliff at develix.com  Sun Oct 23 20:25:07 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 09:25:07 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
Message-ID: <1319387107.4593.70.camel@portable-evil>

I've compiled FS on SL6.1 previously without issue.  I'm wondering how I
can get more informative output so I can figure out if I'm missing a
dependency, have a conflicting library installed, or whatever:

Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c...
Compiling /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c...
quiet_libtool: compile:  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c  -fPIC -DPIC -o .libs/mod_xml_rpc.o
quiet_libtool: compile:  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1
Creating mod_xml_rpc.la...
make[6]: Leaving directory `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
make[3]: Leaving directory `/usr/local/src/freeswitch/src'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [current] Error 2

Thanks for any suggestions.

Cliff




From masood24 at gmail.com  Sun Oct 23 20:25:27 2011
From: masood24 at gmail.com (Masood Ahmed)
Date: Sun, 23 Oct 2011 19:25:27 +0300
Subject: [Freeswitch-users] Radius client authentication
Message-ID: <CAMd3iJ+DdVKWiZuxpk7M1gg2uY9tnOPsOmRfWTQNue_fmXXsOw@mail.gmail.com>

Dear Gurus and Gabriel,

I have installed radius client on free switch successfully , whenever any
person tries to send Register request . frees witch looks for directory look
up in directory named as "directory" in /freeswitch/conf/ .

What I want from freeswitch is that whenever any Register request should hit
the freeswitch, It should do the following.

i) Do not look into the directory for user lookup
ii) Instead core should trigger radius client to send request to radius
server and get the results from radius server.
iii) If authentication (like the return code is 0 ) is successful then it
should let the user register.

I am confused on following things.

a) Which lines I should where freeswitch which will trigger the radius
client to collect the user parameters (like username , password in md5
format ) and pass on to radius server instead looking into directory.
b) where would I add those lines.........

-- 
*Regards,
Masood Ahmed
*
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From infos at madovsky.org  Sun Oct 23 20:29:28 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 23 Oct 2011 12:29:28 -0400
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
References: <1319387107.4593.70.camel@portable-evil>
Message-ID: <5EB5C22894064816996044D18E76BC62@e1705>

did you try sh boostrap.sh before compile ?

----- Original Message ----- 
From: "Cliff Wells" <cliff at develix.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, October 23, 2011 12:25 PM
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1


> I've compiled FS on SL6.1 previously without issue.  I'm wondering how I
> can get more informative output so I can figure out if I'm missing a
> dependency, have a conflicting library installed, or whatever:
>
> Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c...
> Compiling 
> /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c...
> quiet_libtool: compile: 
>  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok 
>  -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src 
>  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c  -fPIC 
>  -DPIC -o .libs/mod_xml_rpc.o
> quiet_libtool: compile: 
>  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok 
>  -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src 
>  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c -o 
> mod_xml_rpc.o >/dev/null 2>&1
> Creating mod_xml_rpc.la...
> make[6]: Leaving directory 
> `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> make[5]: Leaving directory 
> `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
> make[3]: Leaving directory `/usr/local/src/freeswitch/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/local/src/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/local/src/freeswitch'
> make: *** [current] Error 2
>
> Thanks for any suggestions.
>
> Cliff
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 basit.engg at gmail.com  Sun Oct 23 20:46:11 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Sun, 23 Oct 2011 21:46:11 +0500
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <5EB5C22894064816996044D18E76BC62@e1705>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
Message-ID: <CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>

Cliff,

I faced the same issue. Please see your gcc version. Install gcc < 4.5
because they have changed something regarding paths.

while using FS version 1.0.6 i faced switch.h problem and on git HEAD i
faced the same problem as yours.

I installed gcc-4.4 and compilation worked well. Also did you follow
./bootstrap.sh and ./configure ?

--
regards,

Abdul Basit




On Sun, Oct 23, 2011 at 9:29 PM, Madovsky <infos at madovsky.org> wrote:

> did you try sh boostrap.sh before compile ?
>
> ----- Original Message -----
> From: "Cliff Wells" <cliff at develix.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, October 23, 2011 12:25 PM
> Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
>
>
> > I've compiled FS on SL6.1 previously without issue.  I'm wondering how I
> > can get more informative output so I can figure out if I'm missing a
> > dependency, have a conflicting library installed, or whatever:
> >
> > Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c...
> > Compiling
> > /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c...
> > quiet_libtool: compile:
> >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
> -I../../../../libs/xmlrpc-c/lib/expat/xmltok
> >  -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include
> -I../../../../libs/xmlrpc-c/lib/abyss/src
> >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__
> -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/xml_int/mod_xml_rpc/mod_xml_rpc.c
>  -fPIC
> >  -DPIC -o .libs/mod_xml_rpc.o
> > quiet_libtool: compile:
> >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
> -I../../../../libs/xmlrpc-c/lib/expat/xmltok
> >  -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include
> -I../../../../libs/xmlrpc-c/lib/abyss/src
> >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__
> -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/xml_int/mod_xml_rpc/mod_xml_rpc.c -o
> > mod_xml_rpc.o >/dev/null 2>&1
> > Creating mod_xml_rpc.la...
> > make[6]: Leaving directory
> > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> > make[5]: Leaving directory
> > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
> > make[3]: Leaving directory `/usr/local/src/freeswitch/src'
> > make[2]: *** [all-recursive] Error 1
> > make[2]: Leaving directory `/usr/local/src/freeswitch'
> > make[1]: *** [all] Error 2
> > make[1]: Leaving directory `/usr/local/src/freeswitch'
> > make: *** [current] Error 2
> >
> > Thanks for any suggestions.
> >
> > Cliff
> >
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Oct 23 20:50:29 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 23 Oct 2011 09:50:29 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
Message-ID: <CAMWiMy_ngAskkyive5vSTeTVc7uA1GkXzTtCaO_iWgr5E=uFKw@mail.gmail.com>

Reverting back to SL 6.0 compiled without an issue. Was it nss that
was giving the problems?

On Sun, Oct 23, 2011 at 9:46 AM, Abdul Basit <basit.engg at gmail.com> wrote:
> Cliff,
> I faced the same issue. Please see your gcc version. Install gcc < 4.5
> because they have changed something regarding paths.
> while using FS version 1.0.6 i faced switch.h problem and on git HEAD i
> faced the same problem as yours.
> I installed gcc-4.4 and compilation worked well. Also did you follow
> ./bootstrap.sh and ./configure ?
> --
> regards,
> Abdul Basit
>
>
>
> On Sun, Oct 23, 2011 at 9:29 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> did you try sh boostrap.sh before compile ?
>>
>> ----- Original Message -----
>> From: "Cliff Wells" <cliff at develix.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Sunday, October 23, 2011 12:25 PM
>> Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
>>
>>
>> > I've compiled FS on SL6.1 previously without issue. ?I'm wondering how I
>> > can get more informative output so I can figure out if I'm missing a
>> > dependency, have a conflicting library installed, or whatever:
>> >
>> > Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c...
>> > Compiling
>> > /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c...
>> > quiet_libtool: compile:
>> > ?gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
>> > -I../../../../libs/xmlrpc-c/lib/expat/xmltok
>> > ?-I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include
>> > -I../../../../libs/xmlrpc-c/lib/abyss/src
>> > ?-I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__
>> > -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/xml_int/mod_xml_rpc/mod_xml_rpc.c
>> > ?-fPIC
>> > ?-DPIC -o .libs/mod_xml_rpc.o
>> > quiet_libtool: compile:
>> > ?gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
>> > -I../../../../libs/xmlrpc-c/lib/expat/xmltok
>> > ?-I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include
>> > -I../../../../libs/xmlrpc-c/lib/abyss/src
>> > ?-I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__
>> > -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/xml_int/mod_xml_rpc/mod_xml_rpc.c -o
>> > mod_xml_rpc.o >/dev/null 2>&1
>> > Creating mod_xml_rpc.la...
>> > make[6]: Leaving directory
>> > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
>> > make[5]: Leaving directory
>> > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
>> > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>> > make[3]: Leaving directory `/usr/local/src/freeswitch/src'
>> > make[2]: *** [all-recursive] Error 1
>> > make[2]: Leaving directory `/usr/local/src/freeswitch'
>> > make[1]: *** [all] Error 2
>> > make[1]: Leaving directory `/usr/local/src/freeswitch'
>> > make: *** [current] Error 2
>> >
>> > Thanks for any suggestions.
>> >
>> > Cliff
>> >
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch-list at puzzled.xs4all.nl  Sun Oct 23 21:02:08 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sun, 23 Oct 2011 19:02:08 +0200
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
References: <1319387107.4593.70.camel@portable-evil>	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
Message-ID: <4EA44890.4060204@puzzled.xs4all.nl>

On 10/23/2011 06:46 PM, Abdul Basit wrote:
> Cliff,
>
> I faced the same issue. Please see your gcc version. Install gcc < 4.5
> because they have changed something regarding paths.

If something changed in gcc which makes FreeSWITCH compilation fail, 
especially on RHEL/CentOS/SL 6.0/6.1 then that should be fixed in 
FreeSWITCH. Deploying an unsupported version of gcc just to be able to 
compile FreeSWITCH does not make sense.

Since the release of RHEL 6 & 6.1 I have come across several software 
packages that no longer compiled. They compiled again after the 
developers got a proper bug report so they could create a fix. It would 
make sense if something similar was done for FreeSWITCH.

Regards,
Patrick


From cliff at develix.com  Sun Oct 23 21:08:10 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 10:08:10 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
Message-ID: <1319389690.4593.82.camel@portable-evil>

On Sun, 2011-10-23 at 21:46 +0500, Abdul Basit wrote:
> Cliff,
> 
> 
> I faced the same issue. Please see your gcc version. Install gcc < 4.5
> because they have changed something regarding paths.

I currently have gcc 4.4.5 installed.. I don't think 4.5 is available in
the SL repo... is there any chance you are misremembering the versions?
 
Also, this is the same version of gcc as on the working system. I'm
doing "make sure" on the working system to see if compilation continues
to work fine there.  

As far as I can tell, both systems are up-to-date with yum and both have
git head.  I'm going to do a diff on the output of rpm -qa to see if
there's some variation in installed packages.

> while using FS version 1.0.6 i faced switch.h problem and on git HEAD
> i faced the same problem as yours.  
> 
> 
> I installed gcc-4.4 and compilation worked well. Also did you
> follow ./bootstrap.sh and ./configure ? 
> 

Yes, I followed standard procedure: ./bootstrap.sh && ./configure &&
make

Regards,
Cliff


> --
> regards,
> 
> 
> Abdul Basit
> 
> 
> 
> 
> 
> 
> On Sun, Oct 23, 2011 at 9:29 PM, Madovsky <infos at madovsky.org> wrote:
>         did you try sh boostrap.sh before compile ?
>         
>         
>         ----- Original Message -----
>         From: "Cliff Wells" <cliff at develix.com>
>         To: <freeswitch-users at lists.freeswitch.org>
>         Sent: Sunday, October 23, 2011 12:25 PM
>         Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
>         
>         
>         > I've compiled FS on SL6.1 previously without issue.  I'm
>         wondering how I
>         > can get more informative output so I can figure out if I'm
>         missing a
>         > dependency, have a conflicting library installed, or
>         whatever:
>         >
>         >
>         Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c...
>         > Compiling
>         > /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c...
>         > quiet_libtool: compile:
>         >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
>         -I../../../../libs/xmlrpc-c/lib/expat/xmltok
>         >  -I../../../../libs/xmlrpc-c
>         -I../../../../libs/xmlrpc-c/include
>         -I../../../../libs/xmlrpc-c/lib/abyss/src
>         >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD
>         -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c  -fPIC
>         >  -DPIC -o .libs/mod_xml_rpc.o
>         > quiet_libtool: compile:
>         >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
>         -I../../../../libs/xmlrpc-c/lib/expat/xmltok
>         >  -I../../../../libs/xmlrpc-c
>         -I../../../../libs/xmlrpc-c/include
>         -I../../../../libs/xmlrpc-c/lib/abyss/src
>         >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD
>         -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c -o
>         > mod_xml_rpc.o >/dev/null 2>&1
>         > Creating mod_xml_rpc.la...
>         > make[6]: Leaving directory
>         > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
>         > make[5]: Leaving directory
>         > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
>         > make[4]: Leaving directory
>         `/usr/local/src/freeswitch/src/mod'
>         > make[3]: Leaving directory `/usr/local/src/freeswitch/src'
>         > make[2]: *** [all-recursive] Error 1
>         > make[2]: Leaving directory `/usr/local/src/freeswitch'
>         > make[1]: *** [all] Error 2
>         > make[1]: Leaving directory `/usr/local/src/freeswitch'
>         > make: *** [current] Error 2
>         >
>         > Thanks for any suggestions.
>         >
>         > Cliff
>         >
>         >
>         >
>         >
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cliff at develix.com  Sun Oct 23 21:11:21 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 10:11:21 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <CAMWiMy_ngAskkyive5vSTeTVc7uA1GkXzTtCaO_iWgr5E=uFKw@mail.gmail.com>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	<CAMWiMy_ngAskkyive5vSTeTVc7uA1GkXzTtCaO_iWgr5E=uFKw@mail.gmail.com>
Message-ID: <1319389881.4593.85.camel@portable-evil>

On Sun, 2011-10-23 at 09:50 -0700, curriegrad2004 wrote:
> Reverting back to SL 6.0 compiled without an issue. Was it nss that
> was giving the problems?

I do have a minor (and weird) NSS issue on the working system: one
particular module (mod_cdr_pg_csv) won't auto-load due to some nss
linking error, but once FS is running, I can load it by hand.

Cliff



From basit.engg at gmail.com  Sun Oct 23 21:22:15 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Sun, 23 Oct 2011 22:22:15 +0500
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <1319389690.4593.82.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	<1319389690.4593.82.camel@portable-evil>
Message-ID: <CAGQG5MaxywkP4J6TLL8XeZXDtE3hOYWJYzOG5HfQWO80PTk5eA@mail.gmail.com>

On Sun, Oct 23, 2011 at 10:08 PM, Cliff Wells <cliff at develix.com> wrote:

> On Sun, 2011-10-23 at 21:46 +0500, Abdul Basit wrote:
> > Cliff,
> >
> >
> > I faced the same issue. Please see your gcc version. Install gcc < 4.5
> > because they have changed something regarding paths.
>
> I currently have gcc 4.4.5 installed.. I don't think 4.5 is available in
> the SL repo... is there any chance you are misremembering the versions?
>
>
I tested on ubuntu server 11.10 after apt-get upgrade that installed gcc
version 4.6.

https://wiki.ubuntu.com/NattyNarwhal/ToolchainTransition

Please test downgrading the gcc version.

This is just a way around. If you get resolve the issue, just post the bug.


> Also, this is the same version of gcc as on the working system. I'm
> doing "make sure" on the working system to see if compilation continues
> to work fine there.
>
> As far as I can tell, both systems are up-to-date with yum and both have
> git head.  I'm going to do a diff on the output of rpm -qa to see if
> there's some variation in installed packages.
>
> > while using FS version 1.0.6 i faced switch.h problem and on git HEAD
> > i faced the same problem as yours.
> >
> >
> > I installed gcc-4.4 and compilation worked well. Also did you
> > follow ./bootstrap.sh and ./configure ?
> >
>
> Yes, I followed standard procedure: ./bootstrap.sh && ./configure &&
> make
>
> Regards,
> Cliff
>
>
> > --
> > regards,
> >
> >
> > Abdul Basit
> >
> >
> >
> >
> >
> >
> > On Sun, Oct 23, 2011 at 9:29 PM, Madovsky <infos at madovsky.org> wrote:
> >         did you try sh boostrap.sh before compile ?
> >
> >
> >         ----- Original Message -----
> >         From: "Cliff Wells" <cliff at develix.com>
> >         To: <freeswitch-users at lists.freeswitch.org>
> >         Sent: Sunday, October 23, 2011 12:25 PM
> >         Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
> >
> >
> >         > I've compiled FS on SL6.1 previously without issue.  I'm
> >         wondering how I
> >         > can get more informative output so I can figure out if I'm
> >         missing a
> >         > dependency, have a conflicting library installed, or
> >         whatever:
> >         >
> >         >
> >         Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c...
> >         > Compiling
> >         >
> /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c...
> >         > quiet_libtool: compile:
> >         >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
> >         -I../../../../libs/xmlrpc-c/lib/expat/xmltok
> >         >  -I../../../../libs/xmlrpc-c
> >         -I../../../../libs/xmlrpc-c/include
> >         -I../../../../libs/xmlrpc-c/lib/abyss/src
> >         >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD
> >         -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c  -fPIC
> >         >  -DPIC -o .libs/mod_xml_rpc.o
> >         > quiet_libtool: compile:
> >         >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
> >         -I../../../../libs/xmlrpc-c/lib/expat/xmltok
> >         >  -I../../../../libs/xmlrpc-c
> >         -I../../../../libs/xmlrpc-c/include
> >         -I../../../../libs/xmlrpc-c/lib/abyss/src
> >         >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD
> >         -D__EXTENSIONS__ -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/xml_int/mod_xml_rpc/mod_xml_rpc.c -o
> >         > mod_xml_rpc.o >/dev/null 2>&1
> >         > Creating mod_xml_rpc.la...
> >         > make[6]: Leaving directory
> >         > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> >         > make[5]: Leaving directory
> >         > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> >         > make[4]: Leaving directory
> >         `/usr/local/src/freeswitch/src/mod'
> >         > make[3]: Leaving directory `/usr/local/src/freeswitch/src'
> >         > make[2]: *** [all-recursive] Error 1
> >         > make[2]: Leaving directory `/usr/local/src/freeswitch'
> >         > make[1]: *** [all] Error 2
> >         > make[1]: Leaving directory `/usr/local/src/freeswitch'
> >         > make: *** [current] Error 2
> >         >
> >         > Thanks for any suggestions.
> >         >
> >         > Cliff
> >         >
> >         >
> >         >
> >         >
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org
> >         >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > http://www.freeswitch.org
> >
> >
> >
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 | +92 32 1416 4196 | +92 30 0841 1445
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From randy.andrade at gmail.com  Sun Oct 23 22:10:29 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Sun, 23 Oct 2011 14:10:29 -0400
Subject: [Freeswitch-users] BLF not working
In-Reply-To: <1C748D48EFD36B4AA0B934E8B4E2998003615A0B@ipi-cc-srv04.ipinfrastructures.com>
References: <1C748D48EFD36B4AA0B934E8B4E29980036157E6@ipi-cc-srv04.ipinfrastructures.com>
	<-5270605825443932569@unknownmsgid>
	<1C748D48EFD36B4AA0B934E8B4E2998003615A0B@ipi-cc-srv04.ipinfrastructures.com>
Message-ID: <CAJZcYrfELoyEv92ptt9-_t=YLfkkag+tFH2WoVqwMdxSwgWZ+g@mail.gmail.com>

Mike,

I finally got the chance to play with my 508's upstairs.. I don't see any
discernable delay with the BLF's. At work we generally have the phones
re-register every 60 seconds, but they're working in a hosted deployment,
and we have an SBC in between the phones and the switch, which can handle
throttling all the requests, soas to not overwhelm the switch. At home
against FS, I have them registering every 120 seconds, but the BLF's are
responding almost instantly.

Randy

On Thu, Oct 20, 2011 at 7:56 AM, Mike Hammond <mhammond at gcicom.net> wrote:

> Hi Randy,
>
> funnily enough I've just upgraded the phone this morning to 7.4.9.a
> after digging around on Cisco's website last night and I have now got
> some functionality on the 504... yay !
>
> I'm now working through some tests to see what works and what doesn't
> with Fusionpbx/FS.
>
> I've started by removing all of the default * feature codes from the
> handset itself and set just **# against the pickup code in the attendant
> screen and I can now intercept a ringing call by hitting the flashing
> blf key which is cool.
>
> I get grn solid for idle.
>
> Red solid when busy on a call.
>
> Red flashing when ringing.
>
> DND doesn't seem to affect the BLF though which is a shame as that would
> be nice.
>
> I assume that's as good as it gets as unplugging one of the monitored
> extensions and waiting for the thing to drop it's registration doesn't
> seem to make any difference ? although I assume there will be some sort
> of subscription timer involved that FS hasn't reached yet ? given you've
> obviously been through the mill with cisco's before, do you have any
> recommendations for timer settings on the handsets for things like NAT
> keep alives and subscriptions etc ?
>
> Thanks, Mike
>
>
>
>
> -----Original Message-----
> From: Randy Andrade [mailto:randy.andrade at gmail.com]
> Sent: 19 October 2011 23:42
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] BLF not working
>
> I'm curious what version of firmware you're running on your 504's. I
> work with them professionally and have found that certain functions
> work better / don't work depending on your version of firmware.
> Personally I've found 7.4.8a to be the best stability / feature
> balance. 7.4.9a was just released and though it fixes a few bugs,
> especiallly related to BLF's, against our servers a lot of more basic
> functionality, such as blind transfers and others which were related
> to the 500S sidecar, but we weren't testing against FreeSwitch, and we
> were testing SPA509G's. Perhaps you'll have better luck with it? Or
> better luck not using it, if you are? Point I'm trying to make is,
> Cisco has been known to break some functionality in virtually every
> recent release of the firmware, so try a couple different revisions,
> because your code looks correct. I use it on my Linksys SPA942's and
> Cisco SPA508's, which both have working BLF's against my FreeSwitch
> box, at home. The SPA508's are running FW v7.4.8a with the BLF's
> working fine.
>
> Randy
>
> Sent from my iPhone
>
> On Oct 19, 2011, at 5:46 PM, Mike Hammond <mhammond at gcicom.net> wrote:
>
> > Hello all,
> >
> > I'm new to FS and I'm having a problem getting BLF working, initially
> I'm trying with a Cisco SPA504G.
> >
> > I've configured a test switch directly on a public IP using the latest
> download (about a week ago) with a fusionpbx front end gui, running on
> an ubuntu 10.04 LTS server.
> >
> > I've set it up for multi tenant and have had some domain names pointed
> at my public IP, I can register and use handsets and gateways without an
> issue on the different tenants offsite from behind nat'd routers.
> >
> > Next I wanted to try BLF, so I've configured a spare key on the Cisco
> handset (extn 204) with fnc=blf+sd+cp;sub=202@$PROXY;ext=202@$PROXY to
> monitor extension 202 which is within the same tenant domain. But the
> key is glowing orange so obviously not happy
> >
> > If set sofia to debug presence and turn on the sip trace I see the
> following messaging
> >
> > http://pastebin.freeswitch.org/17557
> >
> > Can anyone help me with this one as I'm struggling to know what to do
> next other than strip it all back out and start again with clean a
> single domain setup.
> >
> > Many thanks,
> >
> >
> >
> >
> > Mike Hammond
> >
> >
> > GCI Com
> > Cedar Court Office Park
> > Denby Dale Road
> > Calder Grove
> > Wakefield
> > WF4 3QZ
> >
> > Office:   0844 443 3537
> > Fax:      0844 443 3540
> > http://www.gcicom.net/
> >
> >
> > This email has been swept by Webroot for viruses. Any files
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> >
> >
> > GCI Com incorporates the following Group Companies:
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> Infrastructures Ltd Reg. No. 4657026, Invomo Ltd Reg. No. 6267056,
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> All Registered in England and Wales, Registered Office: Global House, 2
> Crofton Close, Lincoln, LN3 4NT
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
> Mike Hammond
>
>
> GCI Com
> Cedar Court Office Park
> Denby Dale Road
> Calder Grove
> Wakefield
> WF4 3QZ
>
> Office:   0844 443 3537
> Fax:      0844 443 3540
> http://www.gcicom.net/
>
>
> 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 peter.olsson at visionutveckling.se  Sun Oct 23 22:14:06 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 23 Oct 2011 20:14:06 +0200
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <4EA44890.4060204@puzzled.xs4all.nl>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>,
	<4EA44890.4060204@puzzled.xs4all.nl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>

I can't find a proper bug report for this on Jira yet, so that would be a good start to get things changed..

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Patrick Lists [freeswitch-list at puzzled.xs4all.nl]
Skickat: den 23 oktober 2011 19:02
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Make error on Scientific Linux 6.1

On 10/23/2011 06:46 PM, Abdul Basit wrote:
> Cliff,
>
> I faced the same issue. Please see your gcc version. Install gcc < 4.5
> because they have changed something regarding paths.

If something changed in gcc which makes FreeSWITCH compilation fail,
especially on RHEL/CentOS/SL 6.0/6.1 then that should be fixed in
FreeSWITCH. Deploying an unsupported version of gcc just to be able to
compile FreeSWITCH does not make sense.

Since the release of RHEL 6 & 6.1 I have come across several software
packages that no longer compiled. They compiled again after the
developers got a proper bug report so they could create a fix. It would
make sense if something similar was done for FreeSWITCH.

Regards,
Patrick


FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4ea447f532766147085031!



From cliff at develix.com  Sun Oct 23 22:27:40 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 11:27:40 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <CAMWiMy_ngAskkyive5vSTeTVc7uA1GkXzTtCaO_iWgr5E=uFKw@mail.gmail.com>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	<CAMWiMy_ngAskkyive5vSTeTVc7uA1GkXzTtCaO_iWgr5E=uFKw@mail.gmail.com>
Message-ID: <1319394460.4593.90.camel@portable-evil>

On Sun, 2011-10-23 at 09:50 -0700, curriegrad2004 wrote:
> Reverting back to SL 6.0 compiled without an issue. Was it nss that
> was giving the problems?

Actually, I think you are right.  I realized that all the modules are
compiling fine, it's the final linking stage of libfreeswitch.la that is
failing.  The actual error had occurred far back in my scroll buffer:

make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/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="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
Making all in .
quiet_libtool: link: 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 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch freeswitch-switch.o  -lm ./.libs/libfreeswitch.so -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -L/usr/lib6464 /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpq libs/apr/.libs/libapr-1.a -lrt -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz -lncurses -pthread -Wl,-rpath -Wl,/usr/local/freeswitch/lib
/usr/lib64/libnss3.so: undefined reference to `PR_FindSymbol'
/usr/lib64/libnss3.so: undefined reference to `PR_RWLock_Rlock'
/usr/lib64/libssl3.so: undefined reference to `PR_OpenAnonFileMap'
/usr/lib64/libssl3.so: undefined reference to `PR_UnloadLibrary'
/usr/lib64/libnss3.so: undefined reference to `PL_InitArenaPool'
/usr/lib64/libnss3.so: undefined reference to `PR_NewRWLock'
/usr/lib64/libnss3.so: undefined reference to `PR_RWLock_Wlock'
/usr/lib64/libnss3.so: undefined reference to `PR_LoadLibrary'
/lib64/libldap_r-2.4.so.2: undefined reference to `PR_GetEnv'
/usr/lib64/libssl3.so: undefined reference to `PR_LoadLibraryWithFlags'
/usr/lib64/libnssutil3.so: undefined reference to `PL_ClearArenaPool'
/usr/lib64/libnss3.so: undefined reference to `PR_DestroyRWLock'
/usr/lib64/libnss3.so: undefined reference to `PR_NewTCPSocket'
/lib64/libldap_r-2.4.so.2: undefined reference to `PR_SetEnv'
/lib64/libldap_r-2.4.so.2: undefined reference to `PR_GetLibraryName'
/usr/lib64/libssl3.so: undefined reference to `PR_ExportFileMapAsString'
/usr/lib64/libssl3.so: undefined reference to `PR_GetLibraryFilePathname'
/usr/lib64/libssl3.so: undefined reference to `PR_FindFunctionSymbol'
/usr/lib64/libnss3.so: undefined reference to `PL_NewHashTable'
/lib64/libldap_r-2.4.so.2: undefined reference to `PR_ErrorToString'
/usr/lib64/libnss3.so: undefined reference to `PR_RWLock_Unlock'
/usr/lib64/libssl3.so: undefined reference to `PR_ImportFileMapFromString'
/lib64/libldap_r-2.4.so.2: undefined reference to `PR_GetDirectorySeparator'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1
Making all in src
Making all in mod


I'm still a bit baffled as to how one SL6.1 box compiles fine and this one does not.

Cliff


> On Sun, Oct 23, 2011 at 9:46 AM, Abdul Basit <basit.engg at gmail.com> wrote:
> > Cliff,
> > I faced the same issue. Please see your gcc version. Install gcc < 4.5
> > because they have changed something regarding paths.
> > while using FS version 1.0.6 i faced switch.h problem and on git HEAD i
> > faced the same problem as yours.
> > I installed gcc-4.4 and compilation worked well. Also did you follow
> > ./bootstrap.sh and ./configure ?
> > --
> > regards,
> > Abdul Basit
> >
> >
> >
> > On Sun, Oct 23, 2011 at 9:29 PM, Madovsky <infos at madovsky.org> wrote:
> >>
> >> did you try sh boostrap.sh before compile ?
> >>
> >> ----- Original Message -----
> >> From: "Cliff Wells" <cliff at develix.com>
> >> To: <freeswitch-users at lists.freeswitch.org>
> >> Sent: Sunday, October 23, 2011 12:25 PM
> >> Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
> >>
> >>
> >> > I've compiled FS on SL6.1 previously without issue.  I'm wondering how I
> >> > can get more informative output so I can figure out if I'm missing a
> >> > dependency, have a conflicting library installed, or whatever:
> >> >
> >> > Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c...
> >> > Compiling
> >> > /usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c...
> >> > quiet_libtool: compile:
> >> >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
> >> > -I../../../../libs/xmlrpc-c/lib/expat/xmltok
> >> >  -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include
> >> > -I../../../../libs/xmlrpc-c/lib/abyss/src
> >> >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__
> >> > -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/xml_int/mod_xml_rpc/mod_xml_rpc.c
> >> >  -fPIC
> >> >  -DPIC -o .libs/mod_xml_rpc.o
> >> > quiet_libtool: compile:
> >> >  gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse
> >> > -I../../../../libs/xmlrpc-c/lib/expat/xmltok
> >> >  -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include
> >> > -I../../../../libs/xmlrpc-c/lib/abyss/src
> >> >  -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__
> >> > -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/xml_int/mod_xml_rpc/mod_xml_rpc.c -o
> >> > mod_xml_rpc.o >/dev/null 2>&1
> >> > Creating mod_xml_rpc.la...
> >> > make[6]: Leaving directory
> >> > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> >> > make[5]: Leaving directory
> >> > `/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_rpc'
> >> > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
> >> > make[3]: Leaving directory `/usr/local/src/freeswitch/src'
> >> > make[2]: *** [all-recursive] Error 1
> >> > make[2]: Leaving directory `/usr/local/src/freeswitch'
> >> > make[1]: *** [all] Error 2
> >> > make[1]: Leaving directory `/usr/local/src/freeswitch'
> >> > make: *** [current] Error 2
> >> >
> >> > Thanks for any suggestions.
> >> >
> >> > Cliff
> >> >
> >> >
> >> >
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cliff at develix.com  Sun Oct 23 22:28:22 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 11:28:22 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
Message-ID: <1319394502.4593.91.camel@portable-evil>

On Sun, 2011-10-23 at 20:14 +0200, Peter Olsson wrote:
> I can't find a proper bug report for this on Jira yet, so that would be a good start to get things changed..
> 
> /Peter


If I can figure out exactly what to report, I'll make one =)

Cliff




From cliff at develix.com  Sun Oct 23 22:42:58 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 11:42:58 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <1319394502.4593.91.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
Message-ID: <1319395378.4593.93.camel@portable-evil>

On Sun, 2011-10-23 at 11:28 -0700, Cliff Wells wrote:
> On Sun, 2011-10-23 at 20:14 +0200, Peter Olsson wrote:
> > I can't find a proper bug report for this on Jira yet, so that would be a good start to get things changed..
> > 
> > /Peter
> 
> 
> If I can figure out exactly what to report, I'll make one =)


This appears to be the same issue:

http://jira.freeswitch.org/browse/FS-3384

but the bug is closed for not being reported against git head.  I guess
I'll open a new one.

Regards,
Cliff



From freeswitch-list at puzzled.xs4all.nl  Sun Oct 23 22:47:58 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sun, 23 Oct 2011 20:47:58 +0200
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <1319394502.4593.91.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>	<5EB5C22894064816996044D18E76BC62@e1705>	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>	,
	<4EA44890.4060204@puzzled.xs4all.nl>	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
Message-ID: <4EA4615E.9090802@puzzled.xs4all.nl>

On 10/23/2011 08:28 PM, Cliff Wells wrote:
> On Sun, 2011-10-23 at 20:14 +0200, Peter Olsson wrote:
>> I can't find a proper bug report for this on Jira yet, so that would be a good start to get things changed..
>>
>> /Peter
>
>
> If I can figure out exactly what to report, I'll make one =)

Seems a similar bug report was filed back in June:

http://jira.freeswitch.org/browse/FS-3384

It was closed with "Not A Bug" but there is no info why.

Regards,
Patrick



From cliff at develix.com  Sun Oct 23 22:51:10 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 11:51:10 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
Message-ID: <1319395870.4593.94.camel@portable-evil>

Here we are:

http://jira.freeswitch.org/browse/FS-3630

Cliff


On Sun, 2011-10-23 at 20:14 +0200, Peter Olsson wrote:
> I can't find a proper bug report for this on Jira yet, so that would be a good start to get things changed..
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Patrick Lists [freeswitch-list at puzzled.xs4all.nl]
> Skickat: den 23 oktober 2011 19:02
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Make error on Scientific Linux 6.1
> 
> On 10/23/2011 06:46 PM, Abdul Basit wrote:
> > Cliff,
> >
> > I faced the same issue. Please see your gcc version. Install gcc < 4.5
> > because they have changed something regarding paths.
> 
> If something changed in gcc which makes FreeSWITCH compilation fail,
> especially on RHEL/CentOS/SL 6.0/6.1 then that should be fixed in
> FreeSWITCH. Deploying an unsupported version of gcc just to be able to
> compile FreeSWITCH does not make sense.
> 
> Since the release of RHEL 6 & 6.1 I have come across several software
> packages that no longer compiled. They compiled again after the
> developers got a proper bug report so they could create a fix. It would
> make sense if something similar was done for FreeSWITCH.
> 
> Regards,
> Patrick
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4ea447f532766147085031!
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From freeswitch-list at puzzled.xs4all.nl  Sun Oct 23 22:56:05 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sun, 23 Oct 2011 20:56:05 +0200
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <1319395378.4593.93.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>	<5EB5C22894064816996044D18E76BC62@e1705>	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>	,
	<4EA44890.4060204@puzzled.xs4all.nl>	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil>
Message-ID: <4EA46345.20203@puzzled.xs4all.nl>

On 10/23/2011 08:42 PM, Cliff Wells wrote:
> This appears to be the same issue:
>
> http://jira.freeswitch.org/browse/FS-3384
>
> but the bug is closed for not being reported against git head.  I guess
> I'll open a new one.

Cliff,

Here's another one: http://jira.freeswitch.org/browse/FS-3393

Have you tried:
./configure --without-libcurl

Regards,
Patrick


From cliff at develix.com  Sun Oct 23 23:04:21 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 12:04:21 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <4EA46345.20203@puzzled.xs4all.nl>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil>
	<4EA46345.20203@puzzled.xs4all.nl>
Message-ID: <1319396661.4593.97.camel@portable-evil>

On Sun, 2011-10-23 at 20:56 +0200, Patrick Lists wrote:
> On 10/23/2011 08:42 PM, Cliff Wells wrote:
> > This appears to be the same issue:
> >
> > http://jira.freeswitch.org/browse/FS-3384
> >
> > but the bug is closed for not being reported against git head.  I guess
> > I'll open a new one.
> 
> Cliff,
> 
> Here's another one: http://jira.freeswitch.org/browse/FS-3393
> 
> Have you tried:
> ./configure --without-libcurl

Trying it now.  Out of curiosity, does FS build with an internal libcurl
if this is specified?  I do need lib_xml_curl.

Cliff




From curriegrad2004 at gmail.com  Sun Oct 23 23:12:08 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 23 Oct 2011 12:12:08 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <1319396661.4593.97.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	<4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil>
	<4EA46345.20203@puzzled.xs4all.nl>
	<1319396661.4593.97.camel@portable-evil>
Message-ID: <CAMWiMy-1tQVuFxqrT+9xbYZwvk-xCYDrJt-qjZh9NjYSLtT8nA@mail.gmail.com>

--without-libcurl tells the build system to use it's own copy of
libcurl and link that statically

On Sun, Oct 23, 2011 at 12:04 PM, Cliff Wells <cliff at develix.com> wrote:
> On Sun, 2011-10-23 at 20:56 +0200, Patrick Lists wrote:
>> On 10/23/2011 08:42 PM, Cliff Wells wrote:
>> > This appears to be the same issue:
>> >
>> > http://jira.freeswitch.org/browse/FS-3384
>> >
>> > but the bug is closed for not being reported against git head. ?I guess
>> > I'll open a new one.
>>
>> Cliff,
>>
>> Here's another one: http://jira.freeswitch.org/browse/FS-3393
>>
>> Have you tried:
>> ./configure --without-libcurl
>
> Trying it now. ?Out of curiosity, does FS build with an internal libcurl
> if this is specified? ?I do need lib_xml_curl.
>
> Cliff
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cliff at develix.com  Sun Oct 23 23:48:49 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 12:48:49 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
In-Reply-To: <1319396661.4593.97.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil>
	<4EA46345.20203@puzzled.xs4all.nl>
	<1319396661.4593.97.camel@portable-evil>
Message-ID: <1319399329.4593.99.camel@portable-evil>

On Sun, 2011-10-23 at 12:04 -0700, Cliff Wells wrote:
> On Sun, 2011-10-23 at 20:56 +0200, Patrick Lists wrote:
> > On 10/23/2011 08:42 PM, Cliff Wells wrote:
> > > This appears to be the same issue:
> > >
> > > http://jira.freeswitch.org/browse/FS-3384
> > >
> > > but the bug is closed for not being reported against git head.  I guess
> > > I'll open a new one.
> > 
> > Cliff,
> > 
> > Here's another one: http://jira.freeswitch.org/browse/FS-3393
> > 
> > Have you tried:
> > ./configure --without-libcurl
> 
> Trying it now.  Out of curiosity, does FS build with an internal libcurl
> if this is specified?  I do need lib_xml_curl.

Issue remains the same. For the record, I deleted the entire tree, and
did a fresh git clone before trying, since the last comment in that
issue indicated this might be required.

Cliff




From infos at madovsky.org  Mon Oct 24 07:55:26 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 23 Oct 2011 23:55:26 -0400
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1
References: <1319387107.4593.70.camel@portable-evil>	<5EB5C22894064816996044D18E76BC62@e1705>	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>	,
	<4EA44890.4060204@puzzled.xs4all.nl>	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper><1319394502.4593.91.camel@portable-evil>
	<4EA4615E.9090802@puzzled.xs4all.nl>
Message-ID: <7FD18732097F44348C3E565D7F6E7A3C@e1705>

it's better to compile witn internal FS libcurl,
as this lib changes version often.

----- Original Message ----- 
From: "Patrick Lists" <freeswitch-list at puzzled.xs4all.nl>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, October 23, 2011 2:47 PM
Subject: Re: [Freeswitch-users] Make error on Scientific Linux 6.1


> On 10/23/2011 08:28 PM, Cliff Wells wrote:
>> On Sun, 2011-10-23 at 20:14 +0200, Peter Olsson wrote:
>>> I can't find a proper bug report for this on Jira yet, so that would be 
>>> a good start to get things changed..
>>>
>>> /Peter
>>
>>
>> If I can figure out exactly what to report, I'll make one =)
>
> Seems a similar bug report was filed back in June:
>
> http://jira.freeswitch.org/browse/FS-3384
>
> It was closed with "Not A Bug" but there is no info why.
>
> Regards,
> Patrick
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cliff at develix.com  Mon Oct 24 08:18:52 2011
From: cliff at develix.com (Cliff Wells)
Date: Sun, 23 Oct 2011 21:18:52 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]
In-Reply-To: <4EA46345.20203@puzzled.xs4all.nl>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil>
	<4EA46345.20203@puzzled.xs4all.nl>
Message-ID: <1319429933.4593.118.camel@portable-evil>

Okay, so I managed to get it built.  For those who care, here's the
magic incantation required:

yum install nspr-devel
env LIBS="-lnss3 -lnspr4 -lplds4" ./configure 
make && make install

Regards,
Cliff

On Sun, 2011-10-23 at 20:56 +0200, Patrick Lists wrote:
> On 10/23/2011 08:42 PM, Cliff Wells wrote:
> > This appears to be the same issue:
> >
> > http://jira.freeswitch.org/browse/FS-3384
> >
> > but the bug is closed for not being reported against git head.  I guess
> > I'll open a new one.
> 
> Cliff,
> 
> Here's another one: http://jira.freeswitch.org/browse/FS-3393
> 
> Have you tried:
> ./configure --without-libcurl
> 
> Regards,
> Patrick
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 brad at tech21.com  Mon Oct 24 10:10:53 2011
From: brad at tech21.com (Brad Mina)
Date: Sun, 23 Oct 2011 23:10:53 -0700
Subject: [Freeswitch-users] Task was executed late - warning messages
In-Reply-To: <010c01cc9183$0e42bda0$2ac838e0$@itecity.com>
References: <010c01cc9183$0e42bda0$2ac838e0$@itecity.com>
Message-ID: <CAGRr058FVJbgLkB54S-g2VGpGciQLwazHL0Qxf+nOjSv73MS3A@mail.gmail.com>

Would you happen to be running on some type of virtual machine?

On Sun, Oct 23, 2011 at 5:55 AM, Network Information Centre <nic at itecity.com
> wrote:

> Hello everyone,****
>
> ** **
>
> I am receiving a great number of these messages, and do not know the
> service impact, cause or resolution.****
>
> They are continuous, with one to several every hour.  The only items I seem
> to see are: heartbeat, check_ip and limit_hash_cleanup.****
>
> The late times vary from 2 to 5 seconds.****
>
>                 ****
>
> ** **
>
> 2011-10-20 13:42:47.131375 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 1 heartbeat (core)****
>
> 2011-10-20 13:43:01.983476 [WARNING] switch_scheduler.c:114 Task was
> executed late by 4 seconds 2 check_ip (core)****
>
> 2011-10-20 13:44:12.353875 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 1 heartbeat (core)****
>
> 2011-10-20 13:45:17.292117 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 1 heartbeat (core)****
>
> 2011-10-20 13:48:22.629781 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 1 heartbeat (core)****
>
> 2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 1 heartbeat (core)****
>
> 2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 2 check_ip (core)****
>
> 2011-10-20 14:03:03.753312 [WARNING] switch_scheduler.c:114 Task was
> executed late by 4 seconds 3 limit_hash_cleanup (mod_hash)****
>
> 2011-10-20 14:38:32.358406 [WARNING] switch_scheduler.c:114 Task was
> executed late by 4 seconds 1 heartbeat (core)****
>
> 2011-10-20 15:17:10.501523 [WARNING] switch_scheduler.c:114 Task was
> executed late by 3 seconds 2 check_ip (core)****
>
> 2011-10-20 16:08:35.544968 [WARNING] switch_scheduler.c:114 Task was
> executed late by 2 seconds 1 heartbeat (core)****
>
> 2011-10-20 17:02:20.345734 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 1 heartbeat (core)****
>
> 2011-10-20 17:35:13.528281 [WARNING] switch_scheduler.c:114 Task was
> executed late by 3 seconds 2 check_ip (core)****
>
> 2011-10-20 20:02:06.848992 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 1 heartbeat (core)****
>
> 2011-10-20 20:36:09.719593 [WARNING] switch_scheduler.c:114 Task was
> executed late by 3 seconds 1 heartbeat (core)****
>
> 2011-10-20 21:01:17.014476 [WARNING] switch_scheduler.c:114 Task was
> executed late by 2 seconds 2 check_ip (core)****
>
> 2011-10-20 23:44:22.116851 [WARNING] switch_scheduler.c:114 Task was
> executed late by 5 seconds 2 check_ip (core)****
>
> 2011-10-21 00:39:11.946906 [WARNING] switch_scheduler.c:114 Task was
> executed late by 2 seconds 1 heartbeat (core)****
>
> ** **
>
> Any assistance is greatly appreciated.****
>
> ** **
>
> Regards,****
>
> ** **
>
> Don Hunter****
>
> e-City Solutions Inc.****
>
> ** **
>
> email:                          don.hunter at 44ecity.com****
>
> CA/US Toll-Free:     +1-855-IT-ECITY, ext 200****
>
> UK National:             +44 870-820-0286****
>
> iNum (inum.net):   +883 5100-0827-0154****
>
> ** **
>
> ITeCity.com  |  FoundU.com  |  StoreLocatorService.com  |  go-Postal.ca  |
> getZiped.com | FraudCloud.com****
>
> WebPageAlerts.com | TurboBrochures.com | DomainStruck.com | EZ-Verify.com |
> VOIPMuch.com****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cliff at develix.com  Mon Oct 24 11:14:37 2011
From: cliff at develix.com (Cliff Wells)
Date: Mon, 24 Oct 2011 00:14:37 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]
In-Reply-To: <1319429933.4593.118.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil>
	<4EA46345.20203@puzzled.xs4all.nl>
	<1319429933.4593.118.camel@portable-evil>
Message-ID: <1319440477.4593.120.camel@portable-evil>

One last bit of info:

This will get you compiled, but various modules won't load at runtime
for the same reason (unresolved symbols in libnss3, etc).  The solution
is to add this to the top of /etc/init.d/freeswitch:

export LD_PRELOAD="/usr/lib64/libnss3.so /usr/lib64/libnspr4.so /usr/lib64/libplds4.so"

At this point it seems to work fine.

Cliff

On Sun, 2011-10-23 at 21:18 -0700, Cliff Wells wrote:
> Okay, so I managed to get it built.  For those who care, here's the
> magic incantation required:
> 
> yum install nspr-devel
> env LIBS="-lnss3 -lnspr4 -lplds4" ./configure 
> make && make install
> 
> Regards,
> Cliff
> 
> On Sun, 2011-10-23 at 20:56 +0200, Patrick Lists wrote:
> > On 10/23/2011 08:42 PM, Cliff Wells wrote:
> > > This appears to be the same issue:
> > >
> > > http://jira.freeswitch.org/browse/FS-3384
> > >
> > > but the bug is closed for not being reported against git head.  I guess
> > > I'll open a new one.
> > 
> > Cliff,
> > 
> > Here's another one: http://jira.freeswitch.org/browse/FS-3393
> > 
> > Have you tried:
> > ./configure --without-libcurl
> > 
> > Regards,
> > Patrick
> > 
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From infos at madovsky.org  Mon Oct 24 11:22:55 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Oct 2011 03:22:55 -0400
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]
References: <1319387107.4593.70.camel@portable-evil><5EB5C22894064816996044D18E76BC62@e1705><CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>,
	<4EA44890.4060204@puzzled.xs4all.nl><549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper><1319394502.4593.91.camel@portable-evil><1319395378.4593.93.camel@portable-evil><4EA46345.20203@puzzled.xs4all.nl><1319429933.4593.118.camel@portable-evil>
	<1319440477.4593.120.camel@portable-evil>
Message-ID: <82A106D6C07C4517925E6E090314EAD0@e1705>

maybe you can also use symbolic links in /usr/lib/
with same results...

----- Original Message ----- 
From: "Cliff Wells" <cliff at develix.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, October 24, 2011 3:14 AM
Subject: Re: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]


> One last bit of info:
>
> This will get you compiled, but various modules won't load at runtime
> for the same reason (unresolved symbols in libnss3, etc).  The solution
> is to add this to the top of /etc/init.d/freeswitch:
>
> export LD_PRELOAD="/usr/lib64/libnss3.so /usr/lib64/libnspr4.so 
> /usr/lib64/libplds4.so"
>
> At this point it seems to work fine.
>
> Cliff
>
> On Sun, 2011-10-23 at 21:18 -0700, Cliff Wells wrote:
>> Okay, so I managed to get it built.  For those who care, here's the
>> magic incantation required:
>>
>> yum install nspr-devel
>> env LIBS="-lnss3 -lnspr4 -lplds4" ./configure
>> make && make install
>>
>> Regards,
>> Cliff
>>
>> On Sun, 2011-10-23 at 20:56 +0200, Patrick Lists wrote:
>> > On 10/23/2011 08:42 PM, Cliff Wells wrote:
>> > > This appears to be the same issue:
>> > >
>> > > http://jira.freeswitch.org/browse/FS-3384
>> > >
>> > > but the bug is closed for not being reported against git head.  I 
>> > > guess
>> > > I'll open a new one.
>> >
>> > Cliff,
>> >
>> > Here's another one: http://jira.freeswitch.org/browse/FS-3393
>> >
>> > Have you tried:
>> > ./configure --without-libcurl
>> >
>> > Regards,
>> > Patrick
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cliff at develix.com  Mon Oct 24 12:31:46 2011
From: cliff at develix.com (Cliff Wells)
Date: Mon, 24 Oct 2011 01:31:46 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]
In-Reply-To: <82A106D6C07C4517925E6E090314EAD0@e1705>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil><4EA46345.20203@puzzled.xs4all.nl>
	<1319429933.4593.118.camel@portable-evil>
	<1319440477.4593.120.camel@portable-evil>
	<82A106D6C07C4517925E6E090314EAD0@e1705>
Message-ID: <1319445106.4593.128.camel@portable-evil>

On Mon, 2011-10-24 at 03:22 -0400, Madovsky wrote:
> maybe you can also use symbolic links in /usr/lib/
> with same results...

Possibly, but since /usr/lib64 is the default system lib dir, I'd think
that if this were the issue, we'd also see things like /usr/lib/libc.so
not being resolved.  It appears to me that libnss3 *is* found
in /usr/lib64, but for some reason, doesn't automatically load the other
two libraries that it depends on, which is where we see the unresolved
symbols.

In any case, I've already put the system in question into production, so
experiments in this regard will have to wait until later this week when
I setup another system.

As an aside, I also spotted a bug in configure that repeatedly passes
-L/usr/lib6464 as an argument to gcc, but fixing this didn't appear to
resolve any issues.

Cliff


> ----- Original Message ----- 
> From: "Cliff Wells" <cliff at develix.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, October 24, 2011 3:14 AM
> Subject: Re: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]
> 
> 
> > One last bit of info:
> >
> > This will get you compiled, but various modules won't load at runtime
> > for the same reason (unresolved symbols in libnss3, etc).  The solution
> > is to add this to the top of /etc/init.d/freeswitch:
> >
> > export LD_PRELOAD="/usr/lib64/libnss3.so /usr/lib64/libnspr4.so 
> > /usr/lib64/libplds4.so"
> >
> > At this point it seems to work fine.
> >
> > Cliff
> >
> > On Sun, 2011-10-23 at 21:18 -0700, Cliff Wells wrote:
> >> Okay, so I managed to get it built.  For those who care, here's the
> >> magic incantation required:
> >>
> >> yum install nspr-devel
> >> env LIBS="-lnss3 -lnspr4 -lplds4" ./configure
> >> make && make install
> >>
> >> Regards,
> >> Cliff
> >>
> >> On Sun, 2011-10-23 at 20:56 +0200, Patrick Lists wrote:
> >> > On 10/23/2011 08:42 PM, Cliff Wells wrote:
> >> > > This appears to be the same issue:
> >> > >
> >> > > http://jira.freeswitch.org/browse/FS-3384
> >> > >
> >> > > but the bug is closed for not being reported against git head.  I 
> >> > > guess
> >> > > I'll open a new one.
> >> >
> >> > Cliff,
> >> >
> >> > Here's another one: http://jira.freeswitch.org/browse/FS-3393
> >> >
> >> > Have you tried:
> >> > ./configure --without-libcurl
> >> >
> >> > Regards,
> >> > Patrick
> >> >
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fieldpeak at gmail.com  Mon Oct 24 12:39:09 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 24 Oct 2011 16:39:09 +0800
Subject: [Freeswitch-users] Advise on FS core dump
Message-ID: <CAOTmbMwx9CSQ=EHC6rsMtHJ09GRMLSH5Q3Sc2v15pidf6CveyA@mail.gmail.com>

Hi Friends,

i'm using mod_nibblebill for billing on FS, i bundled event processing
fucntions when status change,
e.g. /* on_routing */ check_billing_before_routing,
and * on_hangup */ process_hangup, as below,
however, i found if i make a call containing two-legs e.g. user 1001 call
user 1002, then the FS core dump. but if i only make one leg call, e.g. call
888 for IVR, then no problem occure, i found the problem within function
"check_billing_before_routing" (because once i comment all of lines within
this func, the core did not dump). below is the func
"check_billing_before_routing", can anyone help advise what's wrong? how
should i correct it, thanks.

static switch_status_t check_billing_before_routing(switch_core_session_t
*session)
{
    //charles, check billing before routing the call

    char buf[128];
    char cmd[100] = "check_billing_before_answer";
    FILE *pp = NULL;

    const char* in_username = NULL;
    const char* in_calledstation = NULL;
    const char* in_uniqueid = NULL;
    const char* in_network_addr = NULL;
    const char* in_type = "9";
    int is_b;

    //const char* full_cmd;
    int i_continue = 1;
    switch_channel_t *channel = NULL;


    in_uniqueid = switch_core_session_get_uuid(session);

    channel = switch_core_session_get_channel(session);

    /*only billing on a leg*/
    is_b = channel && switch_channel_get_originator_caller_profile(channel);
    printf("is_b is: %d\n", is_b);
    if(is_b)
    {
    return SWITCH_STATUS_SUCCESS;
    }

    /* Lock this session's data for this module while we tinker with it */
    if (globals.mutex) {
        switch_mutex_lock(globals.mutex);
    }


    in_username = switch_channel_get_variable(channel, "caller_id_number");
    in_calledstation = switch_channel_get_variable(channel,
"destination_number");
    in_network_addr = switch_channel_get_variable(channel, "network_addr");


    /* Done checking - release lock */
    if (globals.mutex) {
        switch_mutex_unlock(globals.mutex);
    }


    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
"check_billing_before_routing, in_username: %s, in_calledstation: %s,
in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
in_calledstation, in_uniqueid, in_type, in_network_addr);


    strcat(cmd, " ");
    strcat(cmd, in_username);
    strcat(cmd, " ");
    strcat(cmd, in_calledstation);
    strcat(cmd, " ");
    strcat(cmd, in_uniqueid);
    strcat(cmd, " ");
    strcat(cmd, in_type);
    strcat(cmd, " ");
    strcat(cmd, in_network_addr);

    printf("excute external cmd: ");
    printf(cmd);
    printf("\n");

    if( (pp = popen(cmd, "r")) == NULL )
    {
        printf("popen() error!\n");
        exit(1);
    }

    while(fgets(buf, sizeof buf, pp))
    {
        printf("the result of excuting external cmd: %s", buf);
    }

    if(pp)
    {
        pclose(pp);
    }

    i_continue =  atoi(buf);

    printf("i_continue: %d", i_continue);


    if (!i_continue) {

        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
SWITCH_LOG_INFO, "call was not authorized to continue \n");

        transfer_call(session, globals.nobal_action);
    }




    return SWITCH_STATUS_SUCCESS;

}


Core dump logs:

freeswitch at freeswitch> 2011-10-24 16:30:26.583369 [NOTICE] mod_xml_rpc.c:969
Starting HTTP Port 8787, DocRoot [/usr/local/freeswitch/htdocs]
2011-10-24 16:30:29.454743 [WARNING] sofia_reg.c:1337 SIP auth challenge
(INVITE) on sofia profile 'internal' for [6628 at 224.193.106.204] from ip
223.128.70.10
2011-10-24 16:30:30.254710 [NOTICE] switch_channel.c:897 New Channel
sofia/internal/8068 at 224.193.106.204 [196a50b8-eaa0-4ef0-a674-18602117f5ec]
is_b is: 0
excute external cmd: check_billing_before_answer 8068 6628
196a50b8-eaa0-4ef0-a674-18602117f5ec 9 223.128.70.10
2011-10-24 16:30:30.254710 [INFO] mod_nibblebill.c:1017
check_billing_before_routing, in_username: 8068, in_calledstation: 6628,
in_uniqueid: 196a50b8-eaa0-4ef0-a674-18602117f5ec in_type: 9,
in_network_addr: 223.128.70.10
the result of excuting external cmd:
the result of excuting external cmd: 60
Segmentation fault (core dumped)
[root at freeswitch bin]#

-- 
Regards,
Charles
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From fieldpeak at gmail.com  Mon Oct 24 12:40:22 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 24 Oct 2011 16:40:22 +0800
Subject: [Freeswitch-users] FS -How to identify the A-leg or B-leg
 inside the source code
In-Reply-To: <CAOTmbMzdQ+ZxwHru=bAJXFiJyvt=TyKQA0VF4PZ28ZpvNeR5Pg@mail.gmail.com>
References: <CAOTmbMxQt7OHO-RGM_BGOiYdu9c1OC8Nb+U+bGqTM671Qvxw+A@mail.gmail.com>
	<CALNZuwkZqyr_7dx29c_bHxTXCT=jYUAF83=0=ZeFic4pzHa5Tg@mail.gmail.com>
	<CAOTmbMzdQ+ZxwHru=bAJXFiJyvt=TyKQA0VF4PZ28ZpvNeR5Pg@mail.gmail.com>
Message-ID: <CAOTmbMy5s=yd8emw82vUetgCGARTOvMdcj7tGkEHDm1Poy5GVg@mail.gmail.com>

Hi Avi,

It works well, thanks for your kindly information.

BR,
Charles

2011/10/17 fieldpeak <fieldpeak at gmail.com>

> Hi Avi,
>
> Thanks your valuable info, and i will try later, Thanks.
>
> BR,
> Charles
>
>
> 2011/10/16 Avi Marcus <avi at avimarcus.net>
>
>> Looking at the a_ prefix from mod_xml_cdr.c, it seems like it's this:
>>>
>>> is_b = channel && switch_channel_get_originator_caller_profile(channel);
>>
>>
>> Feel free to check freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c
>>
>> -Avi
>>
>>
>> On Sun, Oct 16, 2011 at 5:41 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Friends,
>>>
>>> i'm using mod_nibblebill for billing on FS, i bundled event processing
>>> fucntions when status change,
>>> e.g. /* on_routing */ check_billing_before_routing,
>>> and * on_hangup */ process_hangup, as below,
>>> however, i found it excuted the processing function on both A-leg and
>>> B-leg, what i want is that either only A-leg or only B-leg (i can
>>> select by coding), how can i realize it?
>>>
>>> P.S. is it possible that i identify the Legs (A or B) inside the
>>> processing function (e.g. check_billing_before_routing,) by intput
>>> parameter *sessionsession?
>>>
>>>
>>> source codes as below,
>>>
>>> switch_state_handler_table_t nibble_state_handler = {
>>>        /* on_init */ NULL,
>>>        /* on_routing */ check_billing_before_routing, /* Need to add a
>>> check
>>> here for anything in their account before routing */
>>>        /* on_execute */ sched_billing,         /* Turn on heartbeat for
>>> this
>>> session and do an initial account check */
>>>        /* on_hangup */ process_hangup,         /* On hangup - most
>>> important place
>>> to go bill */
>>>        /* on_exch_media */ NULL,
>>>        /* on_soft_exec */ NULL,
>>>        /* on_consume_med */ NULL,
>>>        /* on_hibernate */ NULL,
>>>        /* on_reset */ NULL,
>>>        /* on_park */ NULL,
>>>        /* on_reporting */ NULL,
>>>        /* on_destroy */ NULL
>>> };
>>>
>>>
>>> static switch_status_t
>>> check_billing_before_routing(switch_core_session_t *session)
>>> {
>>>        //is it possible to identify the legs (A or B) here?
>>>        ...
>>> }
>>>
>>>        static switch_status_t process_hangup(switch_core_session_t
>>> *session)
>>> {
>>>        ...
>>> }
>>>
>>> BR,
>>> Charles
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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,
> Charles
>
>


-- 
Regards,
Charles
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From erik.dekkers at wvds.nl  Mon Oct 24 13:12:22 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 24 Oct 2011 11:12:22 +0200
Subject: [Freeswitch-users] mod_spidermonkey make failing
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433CE@FS1.wvds.local>

With today?s git mod_spidermoney is failing to build on FreeBSD 8.2, see pastebin post for all the errors

http://pastebin.freeswitch.org/17574

[cid:image825afe.JPG at 3b03edfb.45b85bd8]
ABC Westland 555        Tel:    +31 174 245 727
P.O. Box 90     Fax:    +31 174 248 739
2685 ZH Poeldijk        Mob:    +31 624 423 009
www.wvds.nl<http://www.wvds.nl/>        erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>
________________________________
Bezoek ons op de Horti Fair / Visit us at the Horti Fair
1-4 november 2011 in Amsterdam, stand 11.0302.
[cid:imagea8290c.JPG at 332157fd.4c89ac79]<https://form.n200.com/?s=1dvsast0dsqul&c=2quptgqmwol3s&ac=NTWO000229FIK>
________________________________
Wilk van der Sande is gecertificeerd Philips Led Horti Partner.

DISCLAIMER
Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van toepassing
All our quotations, all orders placed and all contracts concluded with Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the AVAG-Conditions.


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From wstephen80 at gmail.com  Mon Oct 24 13:13:01 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 24 Oct 2011 11:13:01 +0200
Subject: [Freeswitch-users] Advise on FS core dump
In-Reply-To: <CAOTmbMwx9CSQ=EHC6rsMtHJ09GRMLSH5Q3Sc2v15pidf6CveyA@mail.gmail.com>
References: <CAOTmbMwx9CSQ=EHC6rsMtHJ09GRMLSH5Q3Sc2v15pidf6CveyA@mail.gmail.com>
Message-ID: <CAO4LeU3fL0AXFY6uXYhgXgJYDmkLT6tdwj0xqABcgV4cKXDUYQ@mail.gmail.com>

Can you try to change the allocation of cmd as:

   char cmd[100];

....

strcat(cmd, "check_billing_before_answer ");

strcat(cmd, in_username);

strcat(cmd, " ");


....

On Mon, Oct 24, 2011 at 10:39 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Friends,
>
> i'm using mod_nibblebill for billing on FS, i bundled event processing
> fucntions when status change,
> e.g. /* on_routing */ check_billing_before_routing,
> and * on_hangup */ process_hangup, as below,
> however, i found if i make a call containing two-legs e.g. user 1001 call
> user 1002, then the FS core dump. but if i only make one leg call, e.g. call
> 888 for IVR, then no problem occure, i found the problem within function
> "check_billing_before_routing" (because once i comment all of lines within
> this func, the core did not dump). below is the func
> "check_billing_before_routing", can anyone help advise what's wrong? how
> should i correct it, thanks.
>
> static switch_status_t check_billing_before_routing(switch_core_session_t
> *session)
> {
>     //charles, check billing before routing the call
>
>     char buf[128];
>     char cmd[100] = "check_billing_before_answer";
>     FILE *pp = NULL;
>
>     const char* in_username = NULL;
>     const char* in_calledstation = NULL;
>     const char* in_uniqueid = NULL;
>     const char* in_network_addr = NULL;
>     const char* in_type = "9";
>     int is_b;
>
>     //const char* full_cmd;
>     int i_continue = 1;
>     switch_channel_t *channel = NULL;
>
>
>     in_uniqueid = switch_core_session_get_uuid(session);
>
>     channel = switch_core_session_get_channel(session);
>
>     /*only billing on a leg*/
>     is_b = channel &&
> switch_channel_get_originator_caller_profile(channel);
>     printf("is_b is: %d\n", is_b);
>     if(is_b)
>     {
>     return SWITCH_STATUS_SUCCESS;
>     }
>
>     /* Lock this session's data for this module while we tinker with it */
>     if (globals.mutex) {
>         switch_mutex_lock(globals.mutex);
>     }
>
>
>     in_username = switch_channel_get_variable(channel, "caller_id_number");
>     in_calledstation = switch_channel_get_variable(channel,
> "destination_number");
>     in_network_addr = switch_channel_get_variable(channel, "network_addr");
>
>
>     /* Done checking - release lock */
>     if (globals.mutex) {
>         switch_mutex_unlock(globals.mutex);
>     }
>
>
>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
> in_calledstation, in_uniqueid, in_type, in_network_addr);
>
>
>     strcat(cmd, " ");
>     strcat(cmd, in_username);
>     strcat(cmd, " ");
>     strcat(cmd, in_calledstation);
>     strcat(cmd, " ");
>     strcat(cmd, in_uniqueid);
>     strcat(cmd, " ");
>     strcat(cmd, in_type);
>     strcat(cmd, " ");
>     strcat(cmd, in_network_addr);
>
>     printf("excute external cmd: ");
>     printf(cmd);
>     printf("\n");
>
>     if( (pp = popen(cmd, "r")) == NULL )
>     {
>         printf("popen() error!\n");
>         exit(1);
>     }
>
>     while(fgets(buf, sizeof buf, pp))
>     {
>         printf("the result of excuting external cmd: %s", buf);
>     }
>
>     if(pp)
>     {
>         pclose(pp);
>     }
>
>     i_continue =  atoi(buf);
>
>     printf("i_continue: %d", i_continue);
>
>
>     if (!i_continue) {
>
>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>
>         transfer_call(session, globals.nobal_action);
>     }
>
>
>
>
>     return SWITCH_STATUS_SUCCESS;
>
> }
>
>
> Core dump logs:
>
> freeswitch at freeswitch> 2011-10-24 16:30:26.583369 [NOTICE]
> mod_xml_rpc.c:969 Starting HTTP Port 8787, DocRoot
> [/usr/local/freeswitch/htdocs]
> 2011-10-24 16:30:29.454743 [WARNING] sofia_reg.c:1337 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [6628 at 224.193.106.204] from ip
> 223.128.70.10
> 2011-10-24 16:30:30.254710 [NOTICE] switch_channel.c:897 New Channel
> sofia/internal/8068 at 224.193.106.204 [196a50b8-eaa0-4ef0-a674-18602117f5ec]
> is_b is: 0
> excute external cmd: check_billing_before_answer 8068 6628
> 196a50b8-eaa0-4ef0-a674-18602117f5ec 9 223.128.70.10
> 2011-10-24 16:30:30.254710 [INFO] mod_nibblebill.c:1017
> check_billing_before_routing, in_username: 8068, in_calledstation: 6628,
> in_uniqueid: 196a50b8-eaa0-4ef0-a674-18602117f5ec in_type: 9,
> in_network_addr: 223.128.70.10
> the result of excuting external cmd:
> the result of excuting external cmd: 60
> Segmentation fault (core dumped)
> [root at freeswitch bin]#
>
> --
> Regards,
> Charles
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Mon Oct 24 13:23:13 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 24 Oct 2011 11:23:13 +0200
Subject: [Freeswitch-users] mod_spidermonkey make failing
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433CE@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433CE@FS1.wvds.local>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC45A@cooper>

Try to boostrap.sh and rerun configure script, and see if that helps.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Erik Dekkers
Skickat: den 24 oktober 2011 11:12
Till: 'freeswitch-users at lists.freeswitch.org'
?mne: [Freeswitch-users] mod_spidermonkey make failing

With today?s git mod_spidermoney is failing to build on FreeBSD 8.2, see pastebin post for all the errors

http://pastebin.freeswitch.org/17574
[cid:image001.jpg at 01CC923F.50D91260]
ABC Westland 555

Tel:

+31 174 245 727

P.O. Box 90

Fax:

+31 174 248 739

2685 ZH Poeldijk

Mob:

+31 624 423 009

www.wvds.nl<http://www.wvds.nl/>

erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>

________________________________
Bezoek ons op de Horti Fair / Visit us at the Horti Fair
1-4 november 2011 in Amsterdam, stand 11.0302.
[cid:image002.jpg at 01CC923F.50D91260]<https://form.n200.com/?s=1dvsast0dsqul&c=2quptgqmwol3s&ac=NTWO000229FIK>
________________________________
Wilk van der Sande is gecertificeerd Philips Led Horti Partner.

DISCLAIMER
Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van toepassing
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From stkn at freeswitch.org  Mon Oct 24 13:34:28 2011
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Mon, 24 Oct 2011 11:34:28 +0200
Subject: [Freeswitch-users] mod_spidermonkey make failing
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433CE@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433CE@FS1.wvds.local>
Message-ID: <4EA53124.3030608@freeswitch.org>

On 24.10.2011 11:12, Erik Dekkers wrote:
> With today?s git mod_spidermoney is failing to build on FreeBSD 8.2, see pastebin post for all the errors
> 
> http://pastebin.freeswitch.org/17574

gmake is required to build on FreeBSD, those messages look like you used
standard (BSD) make

-- 
-------------------------------------------------------------------------------
Stefan Knoblich                        | Web:   http://www.axsentis.de/
axsentis GmbH                          |        http://oss.axsentis.de/
Eupener Str. 74, 50933 Koeln, Germany  |
Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
UST-ID: DE244977565                    | JID:
s.knoblich at jabber.axsentis.de
-------------------------------------------------------------------------------
Web:   http://stkn.techmage.de/
Email: stkn at freeswitch.org
IRC:   #freeswitch-de @ irc.freenode.net


From fieldpeak at gmail.com  Mon Oct 24 13:53:07 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 24 Oct 2011 17:53:07 +0800
Subject: [Freeswitch-users] Advise on FS core dump
In-Reply-To: <CAO4LeU3fL0AXFY6uXYhgXgJYDmkLT6tdwj0xqABcgV4cKXDUYQ@mail.gmail.com>
References: <CAOTmbMwx9CSQ=EHC6rsMtHJ09GRMLSH5Q3Sc2v15pidf6CveyA@mail.gmail.com>
	<CAO4LeU3fL0AXFY6uXYhgXgJYDmkLT6tdwj0xqABcgV4cKXDUYQ@mail.gmail.com>
Message-ID: <CAOTmbMwAHBrgG57hVAbr+oyS6U6W3HtGniZXDwTnfAcDKQhbRQ@mail.gmail.com>

Hi Stephen,

Thanks for your kindly help, however, i changed it as your advise, the core
still dumped.

below is the updated code,

static switch_status_t check_billing_before_routing(switch_core_session_t
*session)
{
    //charles, check billing before routing the call

    char buf[128];
    char cmd[100];
    FILE *pp = NULL;

    const char* in_username = NULL;
    const char* in_calledstation = NULL;
    const char* in_uniqueid = NULL;
    const char* in_network_addr = NULL;
    const char* in_type = "9";
    int is_b;

    //const char* full_cmd;
    int i_continue = 1;
    switch_channel_t *channel = NULL;


    in_uniqueid = switch_core_session_get_uuid(session);

    channel = switch_core_session_get_channel(session);

    /*only billing on a leg*/
    is_b = channel && switch_channel_get_originator_caller_profile(channel);
    printf("is_b is: %d\n", is_b);
    if(is_b)
    {
    return SWITCH_STATUS_SUCCESS;
    }

    /* Lock this session's data for this module while we tinker with it */
    if (globals.mutex) {
        switch_mutex_lock(globals.mutex);
    }


    in_username = switch_channel_get_variable(channel, "caller_id_number");
    in_calledstation = switch_channel_get_variable(channel,
"destination_number");
    in_network_addr = switch_channel_get_variable(channel, "network_addr");


    /* Done checking - release lock */
    if (globals.mutex) {
        switch_mutex_unlock(globals.mutex);
    }


    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
"check_billing_before_routing, in_username: %s, in_calledstation: %s,
in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
in_calledstation, in_uniqueid, in_type, in_network_addr);

    strcpy(cmd,"");
    strcat(cmd, "check_billing_before_answer");
    strcat(cmd, " ");
    strcat(cmd, in_username);
    strcat(cmd, " ");
    strcat(cmd, in_calledstation);
    strcat(cmd, " ");
    strcat(cmd, in_uniqueid);
    strcat(cmd, " ");
    strcat(cmd, in_type);
    strcat(cmd, " ");
    strcat(cmd, in_network_addr);

    printf("excute external cmd: ");
    printf(cmd);
    printf("\n");

    if( (pp = popen(cmd, "r")) == NULL )
    {
        printf("popen() error!\n");
        exit(1);
    }

    while(fgets(buf, sizeof buf, pp))
    {
        printf("the result of excuting external cmd: %s", buf);
    }

    if(pp)
    {
        pclose(pp);
    }

    i_continue =  atoi(buf);

    printf("i_continue: %d", i_continue);


    if (!i_continue) {

        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
SWITCH_LOG_INFO, "call was not authorized to continue \n");

        transfer_call(session, globals.nobal_action);
    }




    return SWITCH_STATUS_SUCCESS;

}


2011/10/24 Stephen Wilde <wstephen80 at gmail.com>

> Can you try to change the allocation of cmd as:
>
>    char cmd[100];
>
> ....
>
> strcat(cmd, "check_billing_before_answer ");
>
> strcat(cmd, in_username);
>
> strcat(cmd, " ");
>
>
> ....
>
> On Mon, Oct 24, 2011 at 10:39 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Friends,
>>
>> i'm using mod_nibblebill for billing on FS, i bundled event processing
>> fucntions when status change,
>> e.g. /* on_routing */ check_billing_before_routing,
>> and * on_hangup */ process_hangup, as below,
>> however, i found if i make a call containing two-legs e.g. user 1001 call
>> user 1002, then the FS core dump. but if i only make one leg call, e.g. call
>> 888 for IVR, then no problem occure, i found the problem within function
>> "check_billing_before_routing" (because once i comment all of lines within
>> this func, the core did not dump). below is the func
>> "check_billing_before_routing", can anyone help advise what's wrong? how
>> should i correct it, thanks.
>>
>> static switch_status_t check_billing_before_routing(switch_core_session_t
>> *session)
>> {
>>     //charles, check billing before routing the call
>>
>>     char buf[128];
>>     char cmd[100] = "check_billing_before_answer";
>>     FILE *pp = NULL;
>>
>>     const char* in_username = NULL;
>>     const char* in_calledstation = NULL;
>>     const char* in_uniqueid = NULL;
>>     const char* in_network_addr = NULL;
>>     const char* in_type = "9";
>>     int is_b;
>>
>>     //const char* full_cmd;
>>     int i_continue = 1;
>>     switch_channel_t *channel = NULL;
>>
>>
>>     in_uniqueid = switch_core_session_get_uuid(session);
>>
>>     channel = switch_core_session_get_channel(session);
>>
>>     /*only billing on a leg*/
>>     is_b = channel &&
>> switch_channel_get_originator_caller_profile(channel);
>>     printf("is_b is: %d\n", is_b);
>>     if(is_b)
>>     {
>>     return SWITCH_STATUS_SUCCESS;
>>     }
>>
>>     /* Lock this session's data for this module while we tinker with it */
>>     if (globals.mutex) {
>>         switch_mutex_lock(globals.mutex);
>>     }
>>
>>
>>     in_username = switch_channel_get_variable(channel,
>> "caller_id_number");
>>     in_calledstation = switch_channel_get_variable(channel,
>> "destination_number");
>>     in_network_addr = switch_channel_get_variable(channel,
>> "network_addr");
>>
>>
>>     /* Done checking - release lock */
>>     if (globals.mutex) {
>>         switch_mutex_unlock(globals.mutex);
>>     }
>>
>>
>>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
>> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
>> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
>> in_calledstation, in_uniqueid, in_type, in_network_addr);
>>
>>
>>     strcat(cmd, " ");
>>     strcat(cmd, in_username);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_calledstation);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_uniqueid);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_type);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_network_addr);
>>
>>     printf("excute external cmd: ");
>>     printf(cmd);
>>     printf("\n");
>>
>>     if( (pp = popen(cmd, "r")) == NULL )
>>     {
>>         printf("popen() error!\n");
>>         exit(1);
>>     }
>>
>>     while(fgets(buf, sizeof buf, pp))
>>     {
>>         printf("the result of excuting external cmd: %s", buf);
>>     }
>>
>>     if(pp)
>>     {
>>         pclose(pp);
>>     }
>>
>>     i_continue =  atoi(buf);
>>
>>     printf("i_continue: %d", i_continue);
>>
>>
>>     if (!i_continue) {
>>
>>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>>
>>         transfer_call(session, globals.nobal_action);
>>     }
>>
>>
>>
>>
>>     return SWITCH_STATUS_SUCCESS;
>>
>> }
>>
>>
>> Core dump logs:
>>
>> freeswitch at freeswitch> 2011-10-24 16:30:26.583369 [NOTICE]
>> mod_xml_rpc.c:969 Starting HTTP Port 8787, DocRoot
>> [/usr/local/freeswitch/htdocs]
>> 2011-10-24 16:30:29.454743 [WARNING] sofia_reg.c:1337 SIP auth challenge
>> (INVITE) on sofia profile 'internal' for [6628 at 224.193.106.204] from ip
>> 223.128.70.10
>> 2011-10-24 16:30:30.254710 [NOTICE] switch_channel.c:897 New Channel
>> sofia/internal/8068 at 224.193.106.204[196a50b8-eaa0-4ef0-a674-18602117f5ec]
>> is_b is: 0
>> excute external cmd: check_billing_before_answer 8068 6628
>> 196a50b8-eaa0-4ef0-a674-18602117f5ec 9 223.128.70.10
>> 2011-10-24 16:30:30.254710 [INFO] mod_nibblebill.c:1017
>> check_billing_before_routing, in_username: 8068, in_calledstation: 6628,
>> in_uniqueid: 196a50b8-eaa0-4ef0-a674-18602117f5ec in_type: 9,
>> in_network_addr: 223.128.70.10
>> the result of excuting external cmd:
>> the result of excuting external cmd: 60
>> Segmentation fault (core dumped)
>> [root at freeswitch bin]#
>>
>> --
>> Regards,
>> Charles
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Charles
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From wstephen80 at gmail.com  Mon Oct 24 14:10:23 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 24 Oct 2011 12:10:23 +0200
Subject: [Freeswitch-users] Advise on FS core dump
In-Reply-To: <CAOTmbMwAHBrgG57hVAbr+oyS6U6W3HtGniZXDwTnfAcDKQhbRQ@mail.gmail.com>
References: <CAOTmbMwx9CSQ=EHC6rsMtHJ09GRMLSH5Q3Sc2v15pidf6CveyA@mail.gmail.com>
	<CAO4LeU3fL0AXFY6uXYhgXgJYDmkLT6tdwj0xqABcgV4cKXDUYQ@mail.gmail.com>
	<CAOTmbMwAHBrgG57hVAbr+oyS6U6W3HtGniZXDwTnfAcDKQhbRQ@mail.gmail.com>
Message-ID: <CAO4LeU0fCz1PB17muoyUUCLtTLDr_pNuag7jnbXD6rui216eiQ@mail.gmail.com>

You can issue a:

gdb -c core.xxx freeswitch

in freeswitch bin directory where core.xxx is the name of the generated core
dump then, in gdb:

bt full

to show the stack backtrace

On Mon, Oct 24, 2011 at 11:53 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Stephen,
>
> Thanks for your kindly help, however, i changed it as your advise, the core
> still dumped.
>
> below is the updated code,
>
>
> static switch_status_t check_billing_before_routing(switch_core_session_t
> *session)
> {
>     //charles, check billing before routing the call
>
>     char buf[128];
>     char cmd[100];
>
>     FILE *pp = NULL;
>
>     const char* in_username = NULL;
>     const char* in_calledstation = NULL;
>     const char* in_uniqueid = NULL;
>     const char* in_network_addr = NULL;
>     const char* in_type = "9";
>     int is_b;
>
>     //const char* full_cmd;
>     int i_continue = 1;
>     switch_channel_t *channel = NULL;
>
>
>     in_uniqueid = switch_core_session_get_uuid(session);
>
>     channel = switch_core_session_get_channel(session);
>
>     /*only billing on a leg*/
>     is_b = channel &&
> switch_channel_get_originator_caller_profile(channel);
>     printf("is_b is: %d\n", is_b);
>     if(is_b)
>     {
>     return SWITCH_STATUS_SUCCESS;
>     }
>
>     /* Lock this session's data for this module while we tinker with it */
>     if (globals.mutex) {
>         switch_mutex_lock(globals.mutex);
>     }
>
>
>     in_username = switch_channel_get_variable(channel, "caller_id_number");
>     in_calledstation = switch_channel_get_variable(channel,
> "destination_number");
>     in_network_addr = switch_channel_get_variable(channel, "network_addr");
>
>
>     /* Done checking - release lock */
>     if (globals.mutex) {
>         switch_mutex_unlock(globals.mutex);
>     }
>
>
>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
> in_calledstation, in_uniqueid, in_type, in_network_addr);
>
>     strcpy(cmd,"");
>     strcat(cmd, "check_billing_before_answer");
>
>     strcat(cmd, " ");
>     strcat(cmd, in_username);
>     strcat(cmd, " ");
>     strcat(cmd, in_calledstation);
>     strcat(cmd, " ");
>     strcat(cmd, in_uniqueid);
>     strcat(cmd, " ");
>     strcat(cmd, in_type);
>     strcat(cmd, " ");
>     strcat(cmd, in_network_addr);
>
>     printf("excute external cmd: ");
>     printf(cmd);
>     printf("\n");
>
>     if( (pp = popen(cmd, "r")) == NULL )
>     {
>         printf("popen() error!\n");
>         exit(1);
>     }
>
>     while(fgets(buf, sizeof buf, pp))
>     {
>         printf("the result of excuting external cmd: %s", buf);
>     }
>
>     if(pp)
>     {
>         pclose(pp);
>     }
>
>     i_continue =  atoi(buf);
>
>     printf("i_continue: %d", i_continue);
>
>
>     if (!i_continue) {
>
>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>
>         transfer_call(session, globals.nobal_action);
>     }
>
>
>
>
>     return SWITCH_STATUS_SUCCESS;
>
> }
>
>
> 2011/10/24 Stephen Wilde <wstephen80 at gmail.com>
>
>> Can you try to change the allocation of cmd as:
>>
>>    char cmd[100];
>>
>> ....
>>
>> strcat(cmd, "check_billing_before_answer ");
>>
>> strcat(cmd, in_username);
>>
>> strcat(cmd, " ");
>>
>>
>> ....
>>
>> On Mon, Oct 24, 2011 at 10:39 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Friends,
>>>
>>> i'm using mod_nibblebill for billing on FS, i bundled event processing
>>> fucntions when status change,
>>> e.g. /* on_routing */ check_billing_before_routing,
>>> and * on_hangup */ process_hangup, as below,
>>> however, i found if i make a call containing two-legs e.g. user 1001 call
>>> user 1002, then the FS core dump. but if i only make one leg call, e.g. call
>>> 888 for IVR, then no problem occure, i found the problem within function
>>> "check_billing_before_routing" (because once i comment all of lines within
>>> this func, the core did not dump). below is the func
>>> "check_billing_before_routing", can anyone help advise what's wrong? how
>>> should i correct it, thanks.
>>>
>>> static switch_status_t check_billing_before_routing(switch_core_session_t
>>> *session)
>>> {
>>>     //charles, check billing before routing the call
>>>
>>>     char buf[128];
>>>     char cmd[100] = "check_billing_before_answer";
>>>     FILE *pp = NULL;
>>>
>>>     const char* in_username = NULL;
>>>     const char* in_calledstation = NULL;
>>>     const char* in_uniqueid = NULL;
>>>     const char* in_network_addr = NULL;
>>>     const char* in_type = "9";
>>>     int is_b;
>>>
>>>     //const char* full_cmd;
>>>     int i_continue = 1;
>>>     switch_channel_t *channel = NULL;
>>>
>>>
>>>     in_uniqueid = switch_core_session_get_uuid(session);
>>>
>>>     channel = switch_core_session_get_channel(session);
>>>
>>>     /*only billing on a leg*/
>>>     is_b = channel &&
>>> switch_channel_get_originator_caller_profile(channel);
>>>     printf("is_b is: %d\n", is_b);
>>>     if(is_b)
>>>     {
>>>     return SWITCH_STATUS_SUCCESS;
>>>     }
>>>
>>>     /* Lock this session's data for this module while we tinker with it
>>> */
>>>     if (globals.mutex) {
>>>         switch_mutex_lock(globals.mutex);
>>>     }
>>>
>>>
>>>     in_username = switch_channel_get_variable(channel,
>>> "caller_id_number");
>>>     in_calledstation = switch_channel_get_variable(channel,
>>> "destination_number");
>>>     in_network_addr = switch_channel_get_variable(channel,
>>> "network_addr");
>>>
>>>
>>>     /* Done checking - release lock */
>>>     if (globals.mutex) {
>>>         switch_mutex_unlock(globals.mutex);
>>>     }
>>>
>>>
>>>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
>>> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
>>> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
>>> in_calledstation, in_uniqueid, in_type, in_network_addr);
>>>
>>>
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_username);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_calledstation);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_uniqueid);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_type);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_network_addr);
>>>
>>>     printf("excute external cmd: ");
>>>     printf(cmd);
>>>     printf("\n");
>>>
>>>     if( (pp = popen(cmd, "r")) == NULL )
>>>     {
>>>         printf("popen() error!\n");
>>>         exit(1);
>>>     }
>>>
>>>     while(fgets(buf, sizeof buf, pp))
>>>     {
>>>         printf("the result of excuting external cmd: %s", buf);
>>>     }
>>>
>>>     if(pp)
>>>     {
>>>         pclose(pp);
>>>     }
>>>
>>>     i_continue =  atoi(buf);
>>>
>>>     printf("i_continue: %d", i_continue);
>>>
>>>
>>>     if (!i_continue) {
>>>
>>>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>>>
>>>         transfer_call(session, globals.nobal_action);
>>>     }
>>>
>>>
>>>
>>>
>>>     return SWITCH_STATUS_SUCCESS;
>>>
>>> }
>>>
>>>
>>> Core dump logs:
>>>
>>> freeswitch at freeswitch> 2011-10-24 16:30:26.583369 [NOTICE]
>>> mod_xml_rpc.c:969 Starting HTTP Port 8787, DocRoot
>>> [/usr/local/freeswitch/htdocs]
>>> 2011-10-24 16:30:29.454743 [WARNING] sofia_reg.c:1337 SIP auth challenge
>>> (INVITE) on sofia profile 'internal' for [6628 at 224.193.106.204] from ip
>>> 223.128.70.10
>>> 2011-10-24 16:30:30.254710 [NOTICE] switch_channel.c:897 New Channel
>>> sofia/internal/8068 at 224.193.106.204[196a50b8-eaa0-4ef0-a674-18602117f5ec]
>>> is_b is: 0
>>> excute external cmd: check_billing_before_answer 8068 6628
>>> 196a50b8-eaa0-4ef0-a674-18602117f5ec 9 223.128.70.10
>>> 2011-10-24 16:30:30.254710 [INFO] mod_nibblebill.c:1017
>>> check_billing_before_routing, in_username: 8068, in_calledstation: 6628,
>>> in_uniqueid: 196a50b8-eaa0-4ef0-a674-18602117f5ec in_type: 9,
>>> in_network_addr: 223.128.70.10
>>> the result of excuting external cmd:
>>> the result of excuting external cmd: 60
>>> Segmentation fault (core dumped)
>>> [root at freeswitch bin]#
>>>
>>> --
>>> Regards,
>>> Charles
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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,
> Charles
>
>
>
> FreeSWITCH-users mailing 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 erik.dekkers at wvds.nl  Mon Oct 24 14:24:33 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Mon, 24 Oct 2011 12:24:33 +0200
Subject: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on
 internal calls
In-Reply-To: <CAKbxfG_JR3wD+UyNziiXvuLZYR29RckcUq_10Mk9TPdESAAArg@mail.gmail.com>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433C5@FS1.wvds.local>
	<CAKbxfG_JR3wD+UyNziiXvuLZYR29RckcUq_10Mk9TPdESAAArg@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE07B2D433D4@FS1.wvds.local>

Hi Anthony,

Thank you for your reply.  I can be wrong, but could it be this behavior changed lately?
I do get this since last week while the dialplan+directory hasn?t changed.

Also I?ve made a fresh/clean installation with the default config and still getting OUTBOUND CALL in my screen. Phone firmware didn?t changed either.
I?ve made a new dump: http://pastebin.freeswitch.org/17576

If there?s anything I can do to change this behavior back please tell

Thank you,

Erik

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Anthony Minessale
Verzonden: vrijdag 21 oktober 2011 18:59
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] P-Asserted-Identity: "Outbound Call" on internal calls

if you don't know who you are calling that is the display update you will get.
if you do, set it in the bridge with origination_callee_id_name

{origination_callee_id_name='Fred Smith'}user/1234


On Fri, Oct 21, 2011 at 8:17 AM, Erik Dekkers <erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>> wrote:
Hey list,

Recently (at least since two days) freeswitch is sending UPDATE messages with  P-Asserted-Identity: "Outbound Call" <sip:NUMBER at DOMAIN>  after a call is established on the callers phone.
It looks like it?s not updating the callee?s name with this UPDATE message since the callers name is displayed correctly.

This is the dialplan I?m using for internal call routing:

    <extension name="Local_Extension_SIP">
      <condition field="destination_number" expression="^([1-4]\d{2})$">
            <action application="bridge" data="user/$1"/>
      </condition>
    </extension>

Full debug + siptrace: http://pastebin.freeswitch.org/17568

If there?s anything I can test please let me know

Regards,

Erik Dekkers (wvds-nl)
[cid:image001.jpg at 01CC9247.E2C1B260]
ABC Westland 555

Tel:

+31 174 245 727<tel:%2B31%20174%20245%20727>

P.O. Box 90

Fax:

+31 174 248 739<tel:%2B31%20174%20248%20739>

2685 ZH Poeldijk

Mob:

+31 624 423 009<tel:%2B31%20624%20423%20009>

www.wvds.nl<http://www.wvds.nl/>

erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>

________________________________
Bezoek ons op de Horti Fair / Visit us at the Horti Fair
1-4 november 2011 in Amsterdam, stand 11.0302.
[cid:image002.jpg at 01CC9247.E2C1B260]<https://form.n200.com/?s=1dvsast0dsqul&c=2quptgqmwol3s&ac=NTWO000229FIK>
________________________________
Wilk van der Sande is gecertificeerd Philips Led Horti Partner.

DISCLAIMER
Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van toepassing
All our quotations, all orders placed and all contracts concluded with Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the AVAG-Conditions.




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

[cid:imagea020bc.JPG at 2f22a9a5.4d8f6054]
ABC Westland 555        Tel:    +31 174 245 727
P.O. Box 90     Fax:    +31 174 248 739
2685 ZH Poeldijk        Mob:    +31 624 423 009
www.wvds.nl<http://www.wvds.nl/>        erik.dekkers at wvds.nl<mailto:erik.dekkers at wvds.nl>
________________________________
Bezoek ons op de Horti Fair / Visit us at the Horti Fair
1-4 november 2011 in Amsterdam, stand 11.0302.
[cid:imageb390f3.JPG at b0754e1b.48adb0e9]<https://form.n200.com/?s=1dvsast0dsqul&c=2quptgqmwol3s&ac=NTWO000229FIK>
________________________________
Wilk van der Sande is gecertificeerd Philips Led Horti Partner.

DISCLAIMER
Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van toepassing
All our quotations, all orders placed and all contracts concluded with Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the AVAG-Conditions.


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From faisal.rehman22 at hotmail.com  Mon Oct 24 14:39:43 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Mon, 24 Oct 2011 15:39:43 +0500
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
Message-ID: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>


Hi Everyone,
I want to write a script that reads the Master.csv & generate an alert if the ASR & ACD is low from the desired value. Anybody has written such script, please help me write it or provide any sample.

Thanks

Faisal Rehman 		 	   		  
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From freeswitch-list at puzzled.xs4all.nl  Mon Oct 24 16:04:48 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 24 Oct 2011 14:04:48 +0200
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]
In-Reply-To: <1319445106.4593.128.camel@portable-evil>
References: <1319387107.4593.70.camel@portable-evil>	<5EB5C22894064816996044D18E76BC62@e1705>	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>	,
	<4EA44890.4060204@puzzled.xs4all.nl>	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>	<1319394502.4593.91.camel@portable-evil>	<1319395378.4593.93.camel@portable-evil><4EA46345.20203@puzzled.xs4all.nl>	<1319429933.4593.118.camel@portable-evil>	<1319440477.4593.120.camel@portable-evil>	<82A106D6C07C4517925E6E090314EAD0@e1705>
	<1319445106.4593.128.camel@portable-evil>
Message-ID: <4EA55460.5060203@puzzled.xs4all.nl>

On 10/24/2011 10:31 AM, Cliff Wells wrote:
> As an aside, I also spotted a bug in configure that repeatedly passes
> -L/usr/lib6464 as an argument to gcc, but fixing this didn't appear to
> resolve any issues.

Do you mind sharing a patch?

Regards,
Patrick


From Joshua.Foshee at LogixCom.com  Mon Oct 24 17:48:08 2011
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Mon, 24 Oct 2011 08:48:08 -0500
Subject: [Freeswitch-users] Linksys SPA-942 with G729
Message-ID: <06502C073AD9394AADB3CA7FD94931BC07E75EFA@okc1x1.Logixcom.com>

I am trying to get the SPA-942 to send with G729. I have turned on the
Com G729 Mod and have 10 license on the server. My gateway support G729.
So in theory it should pass the codec though and not have to do
transcoding. G729 works fine with my carrier but not on the phone end.

 

Below is my Vars.xml

 

http://pastebin.com/Fsww9i21

 

Below is my sip trace when I have G729 turned on the phones when
connecting to the box. If I have G711u turned on it works fine but uses
one of my encoding/decoding licenses.

 

http://pastebin.com/B7nSrszx

 

Does anyone have the Linksys SPA-942 working with G729 or any
suggestions?

 

Thanks,

Josh

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From curriegrad2004 at gmail.com  Mon Oct 24 18:16:04 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 24 Oct 2011 07:16:04 -0700
Subject: [Freeswitch-users] Linksys SPA-942 with G729
In-Reply-To: <06502C073AD9394AADB3CA7FD94931BC07E75EFA@okc1x1.Logixcom.com>
References: <06502C073AD9394AADB3CA7FD94931BC07E75EFA@okc1x1.Logixcom.com>
Message-ID: <CAMWiMy_US_mB_hOL8XY_aefcCRQPqqL=YKmgKOjQwimoOmyyMw@mail.gmail.com>

on your internal sofia profile, add this to the param section:

 <param name="NDLB-allow-bad-iananame" value="true"/>

That should make the FS box to allow the bad IANA name of G729A to
work on it. This is a real common problem with the Linksys SPA's out
there.

On Mon, Oct 24, 2011 at 6:48 AM, Joshua Foshee
<Joshua.Foshee at logixcom.com> wrote:
> I am trying to get the SPA-942 to send with G729. I have turned on the Com
> G729 Mod and have 10 license on the server. My gateway support G729. So in
> theory it should pass the codec though and not have to do transcoding. G729
> works fine with my carrier but not on the phone end.
>
>
>
> Below is my Vars.xml
>
>
>
> http://pastebin.com/Fsww9i21
>
>
>
> Below is my sip trace when I have G729 turned on the phones when connecting
> to the box. If I have G711u turned on it works fine but uses one of my
> encoding/decoding licenses.
>
>
>
> http://pastebin.com/B7nSrszx
>
>
>
> Does anyone have the Linksys SPA-942 working with G729 or any suggestions?
>
>
>
> Thanks,
>
> Josh
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cliff at develix.com  Mon Oct 24 18:46:03 2011
From: cliff at develix.com (Cliff Wells)
Date: Mon, 24 Oct 2011 07:46:03 -0700
Subject: [Freeswitch-users] Make error on Scientific Linux 6.1 [SOLVED]
In-Reply-To: <4EA55460.5060203@puzzled.xs4all.nl>
References: <1319387107.4593.70.camel@portable-evil>
	<5EB5C22894064816996044D18E76BC62@e1705>
	<CAGQG5MZTjdfTkxjSPDP+dK5=mKYYPtiL1CLWLcJEj8D-P_cJLw@mail.gmail.com>
	, <4EA44890.4060204@puzzled.xs4all.nl>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF175@cooper>
	<1319394502.4593.91.camel@portable-evil>
	<1319395378.4593.93.camel@portable-evil><4EA46345.20203@puzzled.xs4all.nl>
	<1319429933.4593.118.camel@portable-evil>
	<1319440477.4593.120.camel@portable-evil>
	<82A106D6C07C4517925E6E090314EAD0@e1705>
	<1319445106.4593.128.camel@portable-evil>
	<4EA55460.5060203@puzzled.xs4all.nl>
Message-ID: <1319467563.4593.144.camel@portable-evil>

On Mon, 2011-10-24 at 14:04 +0200, Patrick Lists wrote:
> On 10/24/2011 10:31 AM, Cliff Wells wrote:
> > As an aside, I also spotted a bug in configure that repeatedly passes
> > -L/usr/lib6464 as an argument to gcc, but fixing this didn't appear to
> > resolve any issues.
> 
> Do you mind sharing a patch?

Ha, yes:

for f in `find . -type f -print0 | xargs -0 grep -l "lib6464"`; do 
    sed -i s/lib6464/lib64/g $f
done

;-)

I may take an actual look at fixing it now that my major issue is out of
the way (I had to have that box up for production this morning).


Cliff




From eagle.antonio at gmail.com  Mon Oct 24 19:07:47 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 24 Oct 2011 15:07:47 +0000
Subject: [Freeswitch-users] DTMF Problems
Message-ID: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>

Hello.

This questions is not specific to FS but to all the SIP Gateways.
We currently work with several Voice Providers , including the big carriers
BT , etc  and this situation happens from time to time and some more
frequently then other.

We call phone A , the person answers and presses some keys and voila RTP
Events Received , all goes well.

We disconnect  and call that exact phone and DTMF is not working anymore
,... We disconnect the call from another provider and we have again the DTMF
Tones.

Now we try another carrier and the same situation happens .

Do anyone of you guys have anything to say on this a best practice , should
i use RFC2833 and in band at the same time trying to catch any missing tones
?


Regards
Antonio Teixeira
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From steveayre at gmail.com  Mon Oct 24 19:20:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Oct 2011 16:20:01 +0100
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
Message-ID: <151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>

"use RFC2833 and in band at the same time trying to catch any missing tones"

That's likely to cause you even more problems (duplicate digits).

Steve on iPhone


On 24 Oct 2011, at 16:07, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

> use RFC2833 and in band at the same time trying to catch any missing tones


From eagle.antonio at gmail.com  Mon Oct 24 19:25:03 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 24 Oct 2011 15:25:03 +0000
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
Message-ID: <CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>

Yes but i can deal with duplicated digits since the scripts will detected
double digits :)
Let's say i can survive with that :)

2011/10/24 Steven Ayre <steveayre at gmail.com>

> "use RFC2833 and in band at the same time trying to catch any missing
> tones"
>
> That's likely to cause you even more problems (duplicate digits).
>
> Steve on iPhone
>
>
> On 24 Oct 2011, at 16:07, Antonio Teixeira <eagle.antonio at gmail.com>
> wrote:
>
> > use RFC2833 and in band at the same time trying to catch any missing
> tones
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hynek.cihlar at gmail.com  Mon Oct 24 19:28:44 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Mon, 24 Oct 2011 17:28:44 +0200
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
Message-ID: <-5393121003428480786@unknownmsgid>

Btw, how do u handle duplicates? Short time interval? If so, how short?

Sent from my mobile device

On Oct 24, 2011, at 17:26, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

Yes but i can deal with duplicated digits since the scripts will detected
double digits :)
Let's say i can survive with that :)

2011/10/24 Steven Ayre <steveayre at gmail.com>

> "use RFC2833 and in band at the same time trying to catch any missing
> tones"
>
> That's likely to cause you even more problems (duplicate digits).
>
> Steve on iPhone
>
>
> On 24 Oct 2011, at 16:07, Antonio Teixeira <eagle.antonio at gmail.com>
> wrote:
>
> > use RFC2833 and in band at the same time trying to catch any missing
> tones
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Mon Oct 24 19:38:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Oct 2011 16:38:47 +0100
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <-5393121003428480786@unknownmsgid>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
	<-5393121003428480786@unknownmsgid>
Message-ID: <FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>

Yes, how do you plan the difference between the user dialling 2 and it being duplicated and them pressing 2 twice in quick succession?

Steve on iPhone

On 24 Oct 2011, at 16:28, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

> Btw, how do u handle duplicates? Short time interval? If so, how short?
> 
> Sent from my mobile device
> 
> On Oct 24, 2011, at 17:26, Antonio Teixeira <eagle.antonio at gmail.com> wrote:
> 
>> Yes but i can deal with duplicated digits since the scripts will detected double digits :)
>> Let's say i can survive with that :)
>> 
>> 2011/10/24 Steven Ayre <steveayre at gmail.com>
>> "use RFC2833 and in band at the same time trying to catch any missing tones"
>> 
>> That's likely to cause you even more problems (duplicate digits).
>> 
>> Steve on iPhone
>> 
>> 
>> On 24 Oct 2011, at 16:07, Antonio Teixeira <eagle.antonio at gmail.com> wrote:
>> 
>> > use RFC2833 and in band at the same time trying to catch any missing tones
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 24 20:07:11 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 24 Oct 2011 16:07:11 +0000
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
	<-5393121003428480786@unknownmsgid>
	<FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
Message-ID: <CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>

Well guys it depends on your application , generally  i solve this with
regexp , a buffer and a way to clear the buffer of invalid/duplicated data
I'm extending it to support with min delay between keys.
This is not as hard as you think when using min/max :) and your data is
predictable :)

Also in ivr menus humans don't generally guess whats the next menu so if
len(dtmf) > 1: drop ...

I play with this in our test environment not in our production and it works
:)

2011/10/24 Steven Ayre <steveayre at gmail.com>

> Yes, how do you plan the difference between the user dialling 2 and it
> being duplicated and them pressing 2 twice in quick succession?
>
> Steve on iPhone
>
> On 24 Oct 2011, at 16:28, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:
>
> Btw, how do u handle duplicates? Short time interval? If so, how short?
>
> Sent from my mobile device
>
> On Oct 24, 2011, at 17:26, Antonio Teixeira <eagle.antonio at gmail.com>
> wrote:
>
> Yes but i can deal with duplicated digits since the scripts will detected
> double digits :)
> Let's say i can survive with that :)
>
> 2011/10/24 Steven Ayre < <steveayre at gmail.com>steveayre at gmail.com>
>
>> "use RFC2833 and in band at the same time trying to catch any missing
>> tones"
>>
>> That's likely to cause you even more problems (duplicate digits).
>>
>> Steve on iPhone
>>
>>
>> On 24 Oct 2011, at 16:07, Antonio Teixeira < <eagle.antonio at gmail.com>
>> eagle.antonio at gmail.com> wrote:
>>
>> > use RFC2833 and in band at the same time trying to catch any missing
>> tones
>>
>>
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> 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
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Oct 24 19:50:36 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 24 Oct 2011 15:50:36 +0000
Subject: [Freeswitch-users] event socket
Message-ID: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>

Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,

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From msc at freeswitch.org  Mon Oct 24 20:20:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Oct 2011 09:20:39 -0700
Subject: [Freeswitch-users] error while compiling switch_swigable_cpp.h
In-Reply-To: <CAGQG5MZWo2q4zyL8SeppDBbbBoJCyu-U62vFu+Ku5SURhcWs3g@mail.gmail.com>
References: <CAGQG5MYHacwGEAd1u0VAdymo3W5QnufbVcB78LKXSa60LkrPtg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2AEBD78@cooper>
	<CAGQG5Ma1MrSYVWvRcRAWjby9DP2m5WRQ4=6=UNESbNsnyh1aSQ@mail.gmail.com>
	<CAKzWOxV+Ob=u3j=bYSar0942QhhfcU=Ty9JsjhSJ+-3+vHWoMQ@mail.gmail.com>
	<CAGQG5MapcNPkqKwzcFpAbMOFE4WvVqU0jq7MunuF8rHvG_27uA@mail.gmail.com>
	<CAGQG5MZWo2q4zyL8SeppDBbbBoJCyu-U62vFu+Ku5SURhcWs3g@mail.gmail.com>
Message-ID: <CAKzWOxV6xeT_UbTjyBaqNHMBF7iK7cmVuRU8zA+0ZiM9HYUHbg@mail.gmail.com>

On Sun, Oct 23, 2011 at 3:42 AM, Abdul Basit <basit.engg at gmail.com> wrote:

> Moreover, I tested with latest git HEAD. This has more issues.
> Downgrade to gcc-4.4 works as well.
>

The GCC guys do this all the time - something that compiles perfectly fine
in the previous version now becomes a warning in the new version of gcc.
Downgrading to gcc 4.4 is the best choice under the circumstances.

-MC
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From hynek.cihlar at gmail.com  Mon Oct 24 20:55:30 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Mon, 24 Oct 2011 18:55:30 +0200
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
	<-5393121003428480786@unknownmsgid>
	<FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
	<CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
Message-ID: <CAJh8XdWScd94NuwNRESkFv3=n1h=55u5q9GzV4xTdZzWgms2dA@mail.gmail.com>

Will you disclose to the community your optimal time interval for which it
is safe to consider the succeeding DTMF entry a duplicate? ;-)

Hynek



On Mon, Oct 24, 2011 at 6:07 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Well guys it depends on your application , generally  i solve this with
> regexp , a buffer and a way to clear the buffer of invalid/duplicated data
> I'm extending it to support with min delay between keys.
> This is not as hard as you think when using min/max :) and your data is
> predictable :)
>
> Also in ivr menus humans don't generally guess whats the next menu so if
> len(dtmf) > 1: drop ...
>
> I play with this in our test environment not in our production and it works
> :)
>
>
> 2011/10/24 Steven Ayre <steveayre at gmail.com>
>
>> Yes, how do you plan the difference between the user dialling 2 and it
>> being duplicated and them pressing 2 twice in quick succession?
>>
>> Steve on iPhone
>>
>> On 24 Oct 2011, at 16:28, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:
>>
>> Btw, how do u handle duplicates? Short time interval? If so, how short?
>>
>> Sent from my mobile device
>>
>> On Oct 24, 2011, at 17:26, Antonio Teixeira <eagle.antonio at gmail.com>
>> wrote:
>>
>> Yes but i can deal with duplicated digits since the scripts will detected
>> double digits :)
>> Let's say i can survive with that :)
>>
>> 2011/10/24 Steven Ayre < <steveayre at gmail.com>steveayre at gmail.com>
>>
>>> "use RFC2833 and in band at the same time trying to catch any missing
>>> tones"
>>>
>>> That's likely to cause you even more problems (duplicate digits).
>>>
>>> Steve on iPhone
>>>
>>>
>>> On 24 Oct 2011, at 16:07, Antonio Teixeira < <eagle.antonio at gmail.com>
>>> eagle.antonio at gmail.com> wrote:
>>>
>>> > use RFC2833 and in band at the same time trying to catch any missing
>>> tones
>>>
>>>
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 24 21:09:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Oct 2011 10:09:29 -0700
Subject: [Freeswitch-users] vars.xml in Lua
In-Reply-To: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
References: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
Message-ID: <CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>

Yes, any global variable defined in vars.xml is available in a Lua script.
-MC

On Sat, Oct 22, 2011 at 10:39 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> Are the parameters defined in the XML config files visible to Lua scrips?
> In this case it is the script for loading up the directory.
>
> Hynek
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Oct 24 21:14:12 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Oct 2011 13:14:12 -0400
Subject: [Freeswitch-users] DTMF Problems
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com><151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com><CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com><-5393121003428480786@unknownmsgid><FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com><CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
	<CAJh8XdWScd94NuwNRESkFv3=n1h=55u5q9GzV4xTdZzWgms2dA@mail.gmail.com>
Message-ID: <3F28640FBEAC43159EEB24D1BB484AE2@e1705>

If you played old arcade games, so my record of type as fast as possible
2 times the same key was 9ms...maybe consider a double under 20ms ? :)
  ----- Original Message ----- 
  From: Hynek Cihlar 
  To: FreeSWITCH Users Help 
  Sent: Monday, October 24, 2011 12:55 PM
  Subject: Re: [Freeswitch-users] DTMF Problems


  Will you disclose to the community your optimal time interval for which it is safe to consider the succeeding DTMF entry a duplicate? ;-)

  Hynek




  On Mon, Oct 24, 2011 at 6:07 PM, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

    Well guys it depends on your application , generally  i solve this with regexp , a buffer and a way to clear the buffer of invalid/duplicated data I'm extending it to support with min delay between keys.
    This is not as hard as you think when using min/max :) and your data is predictable :)

    Also in ivr menus humans don't generally guess whats the next menu so if len(dtmf) > 1: drop ... 

    I play with this in our test environment not in our production and it works :)



    2011/10/24 Steven Ayre <steveayre at gmail.com>

      Yes, how do you plan the difference between the user dialling 2 and it being duplicated and them pressing 2 twice in quick succession?


      Steve on iPhone

      On 24 Oct 2011, at 16:28, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:


        Btw, how do u handle duplicates? Short time interval? If so, how short?

        Sent from my mobile device

        On Oct 24, 2011, at 17:26, Antonio Teixeira <eagle.antonio at gmail.com> wrote:


          Yes but i can deal with duplicated digits since the scripts will detected double digits :)
          Let's say i can survive with that :)


          2011/10/24 Steven Ayre <steveayre at gmail.com>

            "use RFC2833 and in band at the same time trying to catch any missing tones"


            That's likely to cause you even more problems (duplicate digits).

            Steve on iPhone



            On 24 Oct 2011, at 16:07, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

            > use RFC2833 and in band at the same time trying to catch any missing tones



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




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


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



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From msc at freeswitch.org  Mon Oct 24 21:14:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Oct 2011 10:14:09 -0700
Subject: [Freeswitch-users] event socket
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
Message-ID: <CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>

Please confirm: are you sending the call to the "socket" application in your
dialplan? If so, then your ESL app needs to be listening on the TCP port
specified. See the examples here:
http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC

On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <
NuwanW at unifybusiness.co.uk> wrote:

> Hi,****
>
> ** **
>
> I?m trying to build an IVR application around freeswtich. I?m using
> freeswitch event socket in C#. I?m intercepting inbound calls in the dial
> plan and sending them to my application via outbound socket. Could someone
> please explain how to execute ?answer? application through event socket. *
> ***
>
> ** **
>
> Eg ? I?m sending the command as below and it?s not working.****
>
> ** **
>
> ESLConnection eslConnection = new ESLConnection(server, port, password)***
> *
>
> ESLEvent eventHandle = eslConnection.SendRecv(?event plain ALL?);****
>
> ** **
>
> eventHandle = eslConnection.RecvEvent();****
>
> string uniqueCallID = evenHandle.GetHeader(?Unique-ID?);****
>
> ** **
>
> *eslConnection.Execute(?answer?, string.Empty, UniqueCallID);*
>
> ** **
>
> ** **
>
> I believe I?m doing something wrong as I can?t find required information.
> Could someone please let me know where to find more information about
> freeswitch event socket in .Net. (Google not giving me required information)
> ****
>
> ** **
>
> Thank you,****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hynek.cihlar at gmail.com  Mon Oct 24 21:32:19 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Mon, 24 Oct 2011 19:32:19 +0200
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <3F28640FBEAC43159EEB24D1BB484AE2@e1705>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
	<-5393121003428480786@unknownmsgid>
	<FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
	<CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
	<CAJh8XdWScd94NuwNRESkFv3=n1h=55u5q9GzV4xTdZzWgms2dA@mail.gmail.com>
	<3F28640FBEAC43159EEB24D1BB484AE2@e1705>
Message-ID: <1793760444750845603@unknownmsgid>

Do you account for system load?

Sent from my mobile device

On Oct 24, 2011, at 19:14, Madovsky <infos at madovsky.org> wrote:

If you played old arcade games, so my record of type as fast as possible
2 times the same key was 9ms...maybe consider a double under 20ms ? :)

----- Original Message -----
*From:* Hynek Cihlar <hynek.cihlar at gmail.com>
*To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Sent:* Monday, October 24, 2011 12:55 PM
*Subject:* Re: [Freeswitch-users] DTMF Problems

Will you disclose to the community your optimal time interval for which it
is safe to consider the succeeding DTMF entry a duplicate? ;-)

Hynek



On Mon, Oct 24, 2011 at 6:07 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Well guys it depends on your application , generally  i solve this with
> regexp , a buffer and a way to clear the buffer of invalid/duplicated data
> I'm extending it to support with min delay between keys.
> This is not as hard as you think when using min/max :) and your data is
> predictable :)
>
> Also in ivr menus humans don't generally guess whats the next menu so if
> len(dtmf) > 1: drop ...
>
> I play with this in our test environment not in our production and it works
> :)
>
>
> 2011/10/24 Steven Ayre <steveayre at gmail.com>
>
>>  Yes, how do you plan the difference between the user dialling 2 and it
>> being duplicated and them pressing 2 twice in quick succession?
>>
>> Steve on iPhone
>>
>> On 24 Oct 2011, at 16:28, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:
>>
>>   Btw, how do u handle duplicates? Short time interval? If so, how short?
>>
>> Sent from my mobile device
>>
>> On Oct 24, 2011, at 17:26, Antonio Teixeira <eagle.antonio at gmail.com>
>> wrote:
>>
>>  Yes but i can deal with duplicated digits since the scripts will
>> detected double digits :)
>> Let's say i can survive with that :)
>>
>> 2011/10/24 Steven Ayre < <steveayre at gmail.com>steveayre at gmail.com>
>>
>>> "use RFC2833 and in band at the same time trying to catch any missing
>>> tones"
>>>
>>> That's likely to cause you even more problems (duplicate digits).
>>>
>>> Steve on iPhone
>>>
>>>
>>> On 24 Oct 2011, at 16:07, Antonio Teixeira < <eagle.antonio at gmail.com>
>>> eagle.antonio at gmail.com> wrote:
>>>
>>> > use RFC2833 and in band at the same time trying to catch any missing
>>> tones
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From hynek.cihlar at gmail.com  Mon Oct 24 21:34:07 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Mon, 24 Oct 2011 19:34:07 +0200
Subject: [Freeswitch-users] vars.xml in Lua
In-Reply-To: <CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>
References: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
	<CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>
Message-ID: <-8152858329743577435@unknownmsgid>

Tx for the reply.

Are they simply declared as global variables or do they need to be accessed
in a special way?

Sent from my mobile device

On Oct 24, 2011, at 19:10, Michael Collins <msc at freeswitch.org> wrote:

Yes, any global variable defined in vars.xml is available in a Lua script.
-MC

On Sat, Oct 22, 2011 at 10:39 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> Are the parameters defined in the XML config files visible to Lua scrips?
> In this case it is the script for loading up the directory.
>
> Hynek
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Mon Oct 24 21:38:34 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Oct 2011 13:38:34 -0400
Subject: [Freeswitch-users] DTMF Problems
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com><151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com><CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com><-5393121003428480786@unknownmsgid><FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com><CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com><CAJh8XdWScd94NuwNRESkFv3=n1h=55u5q9GzV4xTdZzWgms2dA@mail.gmail.com><3F28640FBEAC43159EEB24D1BB484AE2@e1705>
	<1793760444750845603@unknownmsgid>
Message-ID: <D222539C3F534F92B1C8BD04AEA639EA@e1705>

ha, I thought DTMF was also considering timestamp in rfc2833... my bad...
  ----- Original Message ----- 
  From: Hynek Cihlar 
  To: FreeSWITCH Users Help 
  Sent: Monday, October 24, 2011 1:32 PM
  Subject: Re: [Freeswitch-users] DTMF Problems


  Do you account for system load?

  Sent from my mobile device

  On Oct 24, 2011, at 19:14, Madovsky <infos at madovsky.org> wrote:


    If you played old arcade games, so my record of type as fast as possible
    2 times the same key was 9ms...maybe consider a double under 20ms ? :)
      ----- Original Message ----- 
      From: Hynek Cihlar 
      To: FreeSWITCH Users Help 
      Sent: Monday, October 24, 2011 12:55 PM
      Subject: Re: [Freeswitch-users] DTMF Problems


      Will you disclose to the community your optimal time interval for which it is safe to consider the succeeding DTMF entry a duplicate? ;-)

      Hynek




      On Mon, Oct 24, 2011 at 6:07 PM, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

        Well guys it depends on your application , generally  i solve this with regexp , a buffer and a way to clear the buffer of invalid/duplicated data I'm extending it to support with min delay between keys.
        This is not as hard as you think when using min/max :) and your data is predictable :)

        Also in ivr menus humans don't generally guess whats the next menu so if len(dtmf) > 1: drop ... 

        I play with this in our test environment not in our production and it works :) 



        2011/10/24 Steven Ayre <steveayre at gmail.com>

          Yes, how do you plan the difference between the user dialling 2 and it being duplicated and them pressing 2 twice in quick succession?


          Steve on iPhone

          On 24 Oct 2011, at 16:28, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:


            Btw, how do u handle duplicates? Short time interval? If so, how short?

            Sent from my mobile device

            On Oct 24, 2011, at 17:26, Antonio Teixeira <eagle.antonio at gmail.com> wrote:


              Yes but i can deal with duplicated digits since the scripts will detected double digits :)
              Let's say i can survive with that :)


              2011/10/24 Steven Ayre <steveayre at gmail.com>

                "use RFC2833 and in band at the same time trying to catch any missing tones"


                That's likely to cause you even more problems (duplicate digits).

                Steve on iPhone



                On 24 Oct 2011, at 16:07, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

                > use RFC2833 and in band at the same time trying to catch any missing tones



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From msc at freeswitch.org  Mon Oct 24 23:34:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Oct 2011 12:34:21 -0700
Subject: [Freeswitch-users] vars.xml in Lua
In-Reply-To: <-8152858329743577435@unknownmsgid>
References: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
	<CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>
	<-8152858329743577435@unknownmsgid>
Message-ID: <CAKzWOxUd_imRAab6X7BwRe6n1tqMgniu0rwC-gMim59Ht2zr4A@mail.gmail.com>

Nothing special. If you are calling a dialplan script then it's this:

my_val = session:getVariable('Variable_Name')

If this is a non-session kind of script then you can do the API trick:

api = freeswitch.API()
my_val = api:execute('global_getvar','Variable_Name')

-MC

On Mon, Oct 24, 2011 at 10:34 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> Tx for the reply.
>
> Are they simply declared as global variables or do they need to be accessed
> in a special way?
>
> Sent from my mobile device
>
> On Oct 24, 2011, at 19:10, Michael Collins <msc at freeswitch.org> wrote:
>
> Yes, any global variable defined in vars.xml is available in a Lua script.
> -MC
>
> On Sat, Oct 22, 2011 at 10:39 AM, Hynek Cihlar < <hynek.cihlar at gmail.com>
> hynek.cihlar at gmail.com> wrote:
>
>> Are the parameters defined in the XML config files visible to Lua scrips?
>> In this case it is the script for loading up the directory.
>>
>> Hynek
>>
>>
>>
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> 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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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Oct 25 00:06:16 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 24 Oct 2011 21:06:16 +0100
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <D222539C3F534F92B1C8BD04AEA639EA@e1705>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
	<-5393121003428480786@unknownmsgid>
	<FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
	<CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
	<CAJh8XdWScd94NuwNRESkFv3=n1h=55u5q9GzV4xTdZzWgms2dA@mail.gmail.com>
	<3F28640FBEAC43159EEB24D1BB484AE2@e1705>
	<1793760444750845603@unknownmsgid>
	<D222539C3F534F92B1C8BD04AEA639EA@e1705>
Message-ID: <CAF+3j-sYhKRctpr=i8KOF8wOPt7SJkBOS9wX8MtAZqtof6ppPA@mail.gmail.com>

Hynek it really depends on the application and user behaviour despite its
simple a POC not a product application.

And yes you can type as fast as you can if the application drops invalid
keys ... Anyway back to the actual topic :)

2011/10/24 Madovsky <infos at madovsky.org>

> **
> ha, I thought DTMF was also considering timestamp in rfc2833... my bad...
>
> ----- Original Message -----
> *From:* Hynek Cihlar <hynek.cihlar at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, October 24, 2011 1:32 PM
> *Subject:* Re: [Freeswitch-users] DTMF Problems
>
> Do you account for system load?
>
> Sent from my mobile device
>
> On Oct 24, 2011, at 19:14, Madovsky <infos at madovsky.org> wrote:
>
>   If you played old arcade games, so my record of type as fast as possible
> 2 times the same key was 9ms...maybe consider a double under 20ms ? :)
>
> ----- Original Message -----
> *From:* Hynek Cihlar <hynek.cihlar at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, October 24, 2011 12:55 PM
> *Subject:* Re: [Freeswitch-users] DTMF Problems
>
> Will you disclose to the community your optimal time interval for which it
> is safe to consider the succeeding DTMF entry a duplicate? ;-)
>
> Hynek
>
>
>
> On Mon, Oct 24, 2011 at 6:07 PM, Antonio Teixeira <<eagle.antonio at gmail.com>
> eagle.antonio at gmail.com> wrote:
>
>> Well guys it depends on your application , generally  i solve this with
>> regexp , a buffer and a way to clear the buffer of invalid/duplicated data
>> I'm extending it to support with min delay between keys.
>> This is not as hard as you think when using min/max :) and your data is
>> predictable :)
>>
>> Also in ivr menus humans don't generally guess whats the next menu so if
>> len(dtmf) > 1: drop ...
>>
>> I play with this in our test environment not in our production and it
>> works :)
>>
>>
>> 2011/10/24 Steven Ayre < <steveayre at gmail.com>steveayre at gmail.com>
>>
>>>  Yes, how do you plan the difference between the user dialling 2 and it
>>> being duplicated and them pressing 2 twice in quick succession?
>>>
>>> Steve on iPhone
>>>
>>> On 24 Oct 2011, at 16:28, Hynek Cihlar < <hynek.cihlar at gmail.com>
>>> hynek.cihlar at gmail.com> wrote:
>>>
>>>   Btw, how do u handle duplicates? Short time interval? If so, how
>>> short?
>>>
>>> Sent from my mobile device
>>>
>>> On Oct 24, 2011, at 17:26, Antonio Teixeira < <eagle.antonio at gmail.com>
>>> eagle.antonio at gmail.com> wrote:
>>>
>>>  Yes but i can deal with duplicated digits since the scripts will
>>> detected double digits :)
>>> Let's say i can survive with that :)
>>>
>>> 2011/10/24 Steven Ayre < <steveayre at gmail.com> <steveayre at gmail.com>
>>> steveayre at gmail.com>
>>>
>>>> "use RFC2833 and in band at the same time trying to catch any missing
>>>> tones"
>>>>
>>>> That's likely to cause you even more problems (duplicate digits).
>>>>
>>>> Steve on iPhone
>>>>
>>>>
>>>> On 24 Oct 2011, at 16:07, Antonio Teixeira < <eagle.antonio at gmail.com><eagle.antonio at gmail.com>
>>>> eagle.antonio at gmail.com> wrote:
>>>>
>>>> > use RFC2833 and in band at the same time trying to catch any missing
>>>> tones
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> <FreeSWITCH-users at lists.freeswitch.org><FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users><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://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://www.freeswitch.org> <http://www.freeswitch.org>
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> 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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>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
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>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>>
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>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> <FreeSWITCH-users at lists.freeswitch.org>
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>>
>>
>  ------------------------------
>
>
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>  ------------------------------
>
>
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From fcintrono at gmail.com  Tue Oct 25 01:28:56 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Mon, 24 Oct 2011 16:28:56 -0500
Subject: [Freeswitch-users] How to know who is calling me...
Message-ID: <CANtw_ZPhmgMvaPUnLuBSnMPywmYctn1oz=Vn7HuEFjU0fyQDxw@mail.gmail.com>

Hello from Guadalajara, M?xico. :)

For the last month I have been reading the fs book and  playing with fs.
I am pretty conviced that It would be perfect as my soho office.

I have this configuration:

4 normal internal phones.
2 pots lines.

to interface freeswitch with these devices I purchased a CISCO SPA8800.
This unit has 8 ports:

4 for normal phones.
4 for pots lines.

I also purchased a cheap IP phone: linksys SPA921.

With the help of fs wiki site I got how to configure the SPA8800.

The only thing I don't know is how to show in the LCD display of the SPA921
who is calling or the number from which is the call.


Thanks in advance for your kind help.
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From fs-list at communicatefreely.net  Tue Oct 25 01:36:36 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 24 Oct 2011 17:36:36 -0400
Subject: [Freeswitch-users] High load on database server
Message-ID: <4EA5DA64.1060503@communicatefreely.net>

Hello,

I'm having some issues with the number of queries that Freeswitch generates per-second.

Currently, I have about 160 phones registered to FS.  These are almost all behind NAT, and 
all of them have to share presence (there are 2 profiles depending on which network they 
are on), as well as get options pings and that sort of thing.

Database is MySQL 5.5.  I'm seeing 18 queries per second, mostly complicated internal 
joins on tables like sip_dialogs and sip_subscriptions.

Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) and hardly breaking 
a sweat, while the database server is at 100% CPU load.

If I stop the internal profiles, the database server goes right back down.  When I start 
up the profiles again, the load slowly climbs back up to near 100%.

The net result of this is that sofia doesn't always return it's options ping, (so other 
circuits think it's down), and we get slow / no call setup.

Any suggestions as to how I can reduce this?

-Tim


From cliff at develix.com  Tue Oct 25 02:11:23 2011
From: cliff at develix.com (Cliff Wells)
Date: Mon, 24 Oct 2011 15:11:23 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <4EA5DA64.1060503@communicatefreely.net>
References: <4EA5DA64.1060503@communicatefreely.net>
Message-ID: <1319494283.4593.160.camel@portable-evil>

On Mon, 2011-10-24 at 17:36 -0400, Tim St. Pierre wrote:

> Database is MySQL 5.5.  I'm seeing 18 queries per second, mostly complicated internal 
> joins on tables like sip_dialogs and sip_subscriptions.
> 
> Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) and hardly breaking 
> a sweat, while the database server is at 100% CPU load.


You need to optimize your database. Most likely you need to index tables
that are used in joins (I doubt further optimization is needed for such
a small dataset and such a light load).  I don't use MySQL, but I recall
it has a way to log slow queries.  You should turn this on and use the
output to figure out the right queries to optimize.  Don't blindly
create needless indexes as this will actually slow it more.

Cliff



From msc at freeswitch.org  Tue Oct 25 04:24:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Oct 2011 17:24:11 -0700
Subject: [Freeswitch-users] How to know who is calling me...
In-Reply-To: <CANtw_ZPhmgMvaPUnLuBSnMPywmYctn1oz=Vn7HuEFjU0fyQDxw@mail.gmail.com>
References: <CANtw_ZPhmgMvaPUnLuBSnMPywmYctn1oz=Vn7HuEFjU0fyQDxw@mail.gmail.com>
Message-ID: <CAKzWOxX2XUAqFEEFyY_640K4cHRO9osjzwsmBu37ZT9_76UXYQ@mail.gmail.com>

Does Mexico have caller ID service on analog phone lines?
-MC

On Mon, Oct 24, 2011 at 2:28 PM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Hello from Guadalajara, M?xico. :)
>
> For the last month I have been reading the fs book and  playing with fs.
> I am pretty conviced that It would be perfect as my soho office.
>
> I have this configuration:
>
> 4 normal internal phones.
> 2 pots lines.
>
> to interface freeswitch with these devices I purchased a CISCO SPA8800.
> This unit has 8 ports:
>
> 4 for normal phones.
> 4 for pots lines.
>
> I also purchased a cheap IP phone: linksys SPA921.
>
> With the help of fs wiki site I got how to configure the SPA8800.
>
> The only thing I don't know is how to show in the LCD display of the SPA921
> who is calling or the number from which is the call.
>
>
> Thanks in advance for your kind help.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nic at itecity.com  Tue Oct 25 04:35:17 2011
From: nic at itecity.com (Network Information Centre)
Date: Mon, 24 Oct 2011 20:35:17 -0400
Subject: [Freeswitch-users] Task was executed late - warning messages
In-Reply-To: <CAGRr058FVJbgLkB54S-g2VGpGciQLwazHL0Qxf+nOjSv73MS3A@mail.gmail.com>
References: <010c01cc9183$0e42bda0$2ac838e0$@itecity.com>
	<CAGRr058FVJbgLkB54S-g2VGpGciQLwazHL0Qxf+nOjSv73MS3A@mail.gmail.com>
Message-ID: <008a01cc92ad$fe870200$fb950600$@itecity.com>

Yes, Windows Hyper-V w/1 core and 1 GB Ram.

Are there recommendations for running on a VM?

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brad Mina
Sent: October-24-11 2:11 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Task was executed late - warning messages

 

Would you happen to be running on some type of virtual machine?

On Sun, Oct 23, 2011 at 5:55 AM, Network Information Centre <nic at itecity.com> wrote:

Hello everyone,

 

I am receiving a great number of these messages, and do not know the service impact, cause or resolution.

They are continuous, with one to several every hour.  The only items I seem to see are: heartbeat, check_ip and limit_hash_cleanup.

The late times vary from 2 to 5 seconds.

                

 

2011-10-20 13:42:47.131375 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:43:01.983476 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core)

2011-10-20 13:44:12.353875 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:45:17.292117 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:48:22.629781 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)

2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 2 check_ip (core)

2011-10-20 14:03:03.753312 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 3 limit_hash_cleanup (mod_hash)

2011-10-20 14:38:32.358406 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 1 heartbeat (core)

2011-10-20 15:17:10.501523 [WARNING] switch_scheduler.c:114 Task was executed late by 3 seconds 2 check_ip (core)

2011-10-20 16:08:35.544968 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 1 heartbeat (core)

2011-10-20 17:02:20.345734 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)

2011-10-20 17:35:13.528281 [WARNING] switch_scheduler.c:114 Task was executed late by 3 seconds 2 check_ip (core)

2011-10-20 20:02:06.848992 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)

2011-10-20 20:36:09.719593 [WARNING] switch_scheduler.c:114 Task was executed late by 3 seconds 1 heartbeat (core)

2011-10-20 21:01:17.014476 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 2 check_ip (core)

2011-10-20 23:44:22.116851 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 2 check_ip (core)

2011-10-21 00:39:11.946906 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 1 heartbeat (core)

 

Any assistance is greatly appreciated.

 

Regards,

 

Don Hunter

e-City Solutions Inc.

 

email:                          don.hunter at 44ecity.com

CA/US Toll-Free:     +1-855-IT-ECITY, ext 200

UK National:             +44 870-820-0286 <tel:%2B44%20870-820-0286> 

iNum (inum.net):   +883 5100-0827-0154

 

ITeCity.com  |  FoundU.com  |  StoreLocatorService.com  |  go-Postal.ca  |  getZiped.com | FraudCloud.com

WebPageAlerts.com | TurboBrochures.com | DomainStruck.com | EZ-Verify.com | VOIPMuch.com

 



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

 

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From steveu at coppice.org  Tue Oct 25 05:01:11 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 25 Oct 2011 09:01:11 +0800
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>	<-5393121003428480786@unknownmsgid>	<FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
	<CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
Message-ID: <4EA60A57.4050108@coppice.org>

On 10/25/2011 12:07 AM, Antonio Teixeira wrote:
> Well guys it depends on your application , generally  i solve this 
> with regexp , a buffer and a way to clear the buffer of 
> invalid/duplicated data I'm extending it to support with min delay 
> between keys.
> This is not as hard as you think when using min/max :) and your data 
> is predictable :)
Is a properly engineered solution so unattainable that such guesswork is 
the only solution?
>
> Also in ivr menus humans don't generally guess whats the next menu so 
> if len(dtmf) > 1: drop ...
>
> I play with this in our test environment not in our production and it 
> works :)
Try playing with real users. They *always* type ahead by their second or 
third call to the same IVR. Handling type-ahead well - accepting the 
string of entered digits, skipping the playback of prompts so you don't 
slow things down, etc. - is one of the key tests of a good IVR.

Steve



From wasim at convergence.pk  Tue Oct 25 06:32:35 2011
From: wasim at convergence.pk (Wasim Baig)
Date: Tue, 25 Oct 2011 07:32:35 +0500
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
Message-ID: <CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>

Faisal:

Load up the CDR into a db (MySQL or otherwise) ...

For ACD you'll want a SQL query akin

   - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
   SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
   disposition=1;

For ASR you'll need another query

   - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
   - ASR is calls_answered/calls_total

There is sample (read: poor) code in
http://jira.freeswitch.org/browse/FS-1981
which a google for "freeswitch ACD ASR" would have helped you get to in the
first place.

-wasim

On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com>wrote:

>  Hi Everyone,
>
> I want to write a script that reads the Master.csv & generate an alert if
> the ASR & ACD is low from the desired value. Anybody has written such
> script, please help me write it or provide any sample.
>
>
> Thanks
>
>
> Faisal Rehman
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From fieldpeak at gmail.com  Tue Oct 25 06:38:26 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 25 Oct 2011 10:38:26 +0800
Subject: [Freeswitch-users] Advise on FS core dump
In-Reply-To: <CAO4LeU0fCz1PB17muoyUUCLtTLDr_pNuag7jnbXD6rui216eiQ@mail.gmail.com>
References: <CAOTmbMwx9CSQ=EHC6rsMtHJ09GRMLSH5Q3Sc2v15pidf6CveyA@mail.gmail.com>
	<CAO4LeU3fL0AXFY6uXYhgXgJYDmkLT6tdwj0xqABcgV4cKXDUYQ@mail.gmail.com>
	<CAOTmbMwAHBrgG57hVAbr+oyS6U6W3HtGniZXDwTnfAcDKQhbRQ@mail.gmail.com>
	<CAO4LeU0fCz1PB17muoyUUCLtTLDr_pNuag7jnbXD6rui216eiQ@mail.gmail.com>
Message-ID: <CAOTmbMwt4Ewz8tLG3Tp+83V3qyvxT4yP2Arws-UoNHZNbBu3LA@mail.gmail.com>

Hi Stephen,

Thanks for your advise, i tried it, i issued below commands, however, i did
not find any core named files under freeswitch bin directory, and below is
the screen showing, i could you identify the root cause, could you please
help, thanks a lot!

gdb -c core.xxx freeswitch
ddb>run
...(fs running up)
gdb>bt full

freeswitch at freeswitch> 2011-10-25 10:30:44.161051 [WARNING] sofia_reg.c:1337
SIP auth challenge (INVITE) on sofia profile 'internal' for [
13910936628 at 124.193.106.104] from ip 222.128.70.10
2011-10-25 10:30:45.081037 [NOTICE] switch_channel.c:897 New Channel
sofia/internal/13580358068 at 124.193.106.104[38946a57-c673-4705-8433-36f4b066d01c]
[New Thread 0xb71d6b90 (LWP 19991)]
is_b is: 0
excute external cmd: check_billing_before_answer 13580358068 13910936628
38946a57-c673-4705-8433-36f4b066d01c 9 222.128.70.10
2011-10-25 10:30:45.081037 [INFO] mod_nibblebill.c:1017
check_billing_before_routing, in_username: 13580358068, in_calledstation:
13910936628, in_uniqueid: 38946a57-c673-4705-8433-36f4b066d01c in_type: 9,
in_network_addr: 222.128.70.10
Detaching after fork from child process 19992.
the result of excuting external cmd:
the result of excuting external cmd: 60

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 0xb71d6b90 (LWP 19991)]
0x008ff698 in switch_core_get_state_handler at plt () from
/usr/local/freeswitch/lib/libfreeswitch.so.1
(gdb) bt full
#0  0x008ff698 in switch_core_get_state_handler at plt ()
   from /usr/local/freeswitch/lib/libfreeswitch.so.1
No symbol table info available.
#1  0x0092d558 in switch_core_session_run (session=0x81bc238) at
src/switch_core_state_machine.c:373
        global_proceed = 1
        proceed = <value optimized out>
        ptr = <value optimized out>
        rstatus = <value optimized out>
        state = <value optimized out>
        endstate = <value optimized out>
        endpoint_interface = <value optimized out>
        driver_state_handler = 0xcb8420
        application_state_handler = <value optimized out>
        new_loops = 60000
        __PRETTY_FUNCTION__ = "switch_core_session_run"
        __func__ = "switch_core_session_run"
#2  0x009275ca in switch_core_session_thread (thread=0x823fdd8,
obj=0x81bc238)
    at src/switch_core_session.c:1309
        session = 0x81bc238
        event = <value optimized out>
        event_str = 0x0
        val = <value optimized out>
        __func__ = "switch_core_session_thread"
        __PRETTY_FUNCTION__ = "switch_core_session_thread"
#3  0x009c2d16 in dummy_worker (opaque=0x823fdd8) at
threadproc/unix/thread.c:138
---Type <return> to continue, or q <return> to quit---
No locals.
#4  0x002af832 in start_thread () from /lib/libpthread.so.0
No symbol table info available.
#5  0x0078145e in clone () from /lib/libc.so.6
No symbol table info available.
(gdb)

2011/10/24 Stephen Wilde <wstephen80 at gmail.com>

> You can issue a:
>
> gdb -c core.xxx freeswitch
>
> in freeswitch bin directory where core.xxx is the name of the generated
> core dump then, in gdb:
>
> bt full
>
> to show the stack backtrace
>
> On Mon, Oct 24, 2011 at 11:53 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Stephen,
>>
>> Thanks for your kindly help, however, i changed it as your advise, the
>> core still dumped.
>>
>> below is the updated code,
>>
>>
>> static switch_status_t check_billing_before_routing(switch_core_session_t
>> *session)
>> {
>>     //charles, check billing before routing the call
>>
>>     char buf[128];
>>     char cmd[100];
>>
>>     FILE *pp = NULL;
>>
>>     const char* in_username = NULL;
>>     const char* in_calledstation = NULL;
>>     const char* in_uniqueid = NULL;
>>     const char* in_network_addr = NULL;
>>     const char* in_type = "9";
>>     int is_b;
>>
>>     //const char* full_cmd;
>>     int i_continue = 1;
>>     switch_channel_t *channel = NULL;
>>
>>
>>     in_uniqueid = switch_core_session_get_uuid(session);
>>
>>     channel = switch_core_session_get_channel(session);
>>
>>     /*only billing on a leg*/
>>     is_b = channel &&
>> switch_channel_get_originator_caller_profile(channel);
>>     printf("is_b is: %d\n", is_b);
>>     if(is_b)
>>     {
>>     return SWITCH_STATUS_SUCCESS;
>>     }
>>
>>     /* Lock this session's data for this module while we tinker with it */
>>     if (globals.mutex) {
>>         switch_mutex_lock(globals.mutex);
>>     }
>>
>>
>>     in_username = switch_channel_get_variable(channel,
>> "caller_id_number");
>>     in_calledstation = switch_channel_get_variable(channel,
>> "destination_number");
>>     in_network_addr = switch_channel_get_variable(channel,
>> "network_addr");
>>
>>
>>     /* Done checking - release lock */
>>     if (globals.mutex) {
>>         switch_mutex_unlock(globals.mutex);
>>     }
>>
>>
>>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
>> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
>> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
>> in_calledstation, in_uniqueid, in_type, in_network_addr);
>>
>>     strcpy(cmd,"");
>>     strcat(cmd, "check_billing_before_answer");
>>
>>     strcat(cmd, " ");
>>     strcat(cmd, in_username);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_calledstation);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_uniqueid);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_type);
>>     strcat(cmd, " ");
>>     strcat(cmd, in_network_addr);
>>
>>     printf("excute external cmd: ");
>>     printf(cmd);
>>     printf("\n");
>>
>>     if( (pp = popen(cmd, "r")) == NULL )
>>     {
>>         printf("popen() error!\n");
>>         exit(1);
>>     }
>>
>>     while(fgets(buf, sizeof buf, pp))
>>     {
>>         printf("the result of excuting external cmd: %s", buf);
>>     }
>>
>>     if(pp)
>>     {
>>         pclose(pp);
>>     }
>>
>>     i_continue =  atoi(buf);
>>
>>     printf("i_continue: %d", i_continue);
>>
>>
>>     if (!i_continue) {
>>
>>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>>
>>         transfer_call(session, globals.nobal_action);
>>     }
>>
>>
>>
>>
>>     return SWITCH_STATUS_SUCCESS;
>>
>> }
>>
>>
>> 2011/10/24 Stephen Wilde <wstephen80 at gmail.com>
>>
>>> Can you try to change the allocation of cmd as:
>>>
>>>    char cmd[100];
>>>
>>> ....
>>>
>>> strcat(cmd, "check_billing_before_answer ");
>>>
>>> strcat(cmd, in_username);
>>>
>>> strcat(cmd, " ");
>>>
>>>
>>> ....
>>>
>>> On Mon, Oct 24, 2011 at 10:39 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hi Friends,
>>>>
>>>> i'm using mod_nibblebill for billing on FS, i bundled event processing
>>>> fucntions when status change,
>>>> e.g. /* on_routing */ check_billing_before_routing,
>>>> and * on_hangup */ process_hangup, as below,
>>>> however, i found if i make a call containing two-legs e.g. user 1001
>>>> call user 1002, then the FS core dump. but if i only make one leg call, e.g.
>>>> call 888 for IVR, then no problem occure, i found the problem within
>>>> function "check_billing_before_routing" (because once i comment all of lines
>>>> within this func, the core did not dump). below is the func
>>>> "check_billing_before_routing", can anyone help advise what's wrong? how
>>>> should i correct it, thanks.
>>>>
>>>> static switch_status_t
>>>> check_billing_before_routing(switch_core_session_t *session)
>>>> {
>>>>     //charles, check billing before routing the call
>>>>
>>>>     char buf[128];
>>>>     char cmd[100] = "check_billing_before_answer";
>>>>     FILE *pp = NULL;
>>>>
>>>>     const char* in_username = NULL;
>>>>     const char* in_calledstation = NULL;
>>>>     const char* in_uniqueid = NULL;
>>>>     const char* in_network_addr = NULL;
>>>>     const char* in_type = "9";
>>>>     int is_b;
>>>>
>>>>     //const char* full_cmd;
>>>>     int i_continue = 1;
>>>>     switch_channel_t *channel = NULL;
>>>>
>>>>
>>>>     in_uniqueid = switch_core_session_get_uuid(session);
>>>>
>>>>     channel = switch_core_session_get_channel(session);
>>>>
>>>>     /*only billing on a leg*/
>>>>     is_b = channel &&
>>>> switch_channel_get_originator_caller_profile(channel);
>>>>     printf("is_b is: %d\n", is_b);
>>>>     if(is_b)
>>>>     {
>>>>     return SWITCH_STATUS_SUCCESS;
>>>>     }
>>>>
>>>>     /* Lock this session's data for this module while we tinker with it
>>>> */
>>>>     if (globals.mutex) {
>>>>         switch_mutex_lock(globals.mutex);
>>>>     }
>>>>
>>>>
>>>>     in_username = switch_channel_get_variable(channel,
>>>> "caller_id_number");
>>>>     in_calledstation = switch_channel_get_variable(channel,
>>>> "destination_number");
>>>>     in_network_addr = switch_channel_get_variable(channel,
>>>> "network_addr");
>>>>
>>>>
>>>>     /* Done checking - release lock */
>>>>     if (globals.mutex) {
>>>>         switch_mutex_unlock(globals.mutex);
>>>>     }
>>>>
>>>>
>>>>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
>>>> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
>>>> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
>>>> in_calledstation, in_uniqueid, in_type, in_network_addr);
>>>>
>>>>
>>>>     strcat(cmd, " ");
>>>>     strcat(cmd, in_username);
>>>>     strcat(cmd, " ");
>>>>     strcat(cmd, in_calledstation);
>>>>     strcat(cmd, " ");
>>>>     strcat(cmd, in_uniqueid);
>>>>     strcat(cmd, " ");
>>>>     strcat(cmd, in_type);
>>>>     strcat(cmd, " ");
>>>>     strcat(cmd, in_network_addr);
>>>>
>>>>     printf("excute external cmd: ");
>>>>     printf(cmd);
>>>>     printf("\n");
>>>>
>>>>     if( (pp = popen(cmd, "r")) == NULL )
>>>>     {
>>>>         printf("popen() error!\n");
>>>>         exit(1);
>>>>     }
>>>>
>>>>     while(fgets(buf, sizeof buf, pp))
>>>>     {
>>>>         printf("the result of excuting external cmd: %s", buf);
>>>>     }
>>>>
>>>>     if(pp)
>>>>     {
>>>>         pclose(pp);
>>>>     }
>>>>
>>>>     i_continue =  atoi(buf);
>>>>
>>>>     printf("i_continue: %d", i_continue);
>>>>
>>>>
>>>>     if (!i_continue) {
>>>>
>>>>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>>> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>>>>
>>>>         transfer_call(session, globals.nobal_action);
>>>>     }
>>>>
>>>>
>>>>
>>>>
>>>>     return SWITCH_STATUS_SUCCESS;
>>>>
>>>> }
>>>>
>>>>
>>>> Core dump logs:
>>>>
>>>> freeswitch at freeswitch> 2011-10-24 16:30:26.583369 [NOTICE]
>>>> mod_xml_rpc.c:969 Starting HTTP Port 8787, DocRoot
>>>> [/usr/local/freeswitch/htdocs]
>>>> 2011-10-24 16:30:29.454743 [WARNING] sofia_reg.c:1337 SIP auth challenge
>>>> (INVITE) on sofia profile 'internal' for [6628 at 224.193.106.204] from ip
>>>> 223.128.70.10
>>>> 2011-10-24 16:30:30.254710 [NOTICE] switch_channel.c:897 New Channel
>>>> sofia/internal/8068 at 224.193.106.204[196a50b8-eaa0-4ef0-a674-18602117f5ec]
>>>> is_b is: 0
>>>> excute external cmd: check_billing_before_answer 8068 6628
>>>> 196a50b8-eaa0-4ef0-a674-18602117f5ec 9 223.128.70.10
>>>> 2011-10-24 16:30:30.254710 [INFO] mod_nibblebill.c:1017
>>>> check_billing_before_routing, in_username: 8068, in_calledstation: 6628,
>>>> in_uniqueid: 196a50b8-eaa0-4ef0-a674-18602117f5ec in_type: 9,
>>>> in_network_addr: 223.128.70.10
>>>> the result of excuting external cmd:
>>>> the result of excuting external cmd: 60
>>>> Segmentation fault (core dumped)
>>>> [root at freeswitch bin]#
>>>>
>>>> --
>>>> Regards,
>>>> Charles
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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,
>> Charles
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Charles
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From fieldpeak at gmail.com  Tue Oct 25 06:57:37 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 25 Oct 2011 10:57:37 +0800
Subject: [Freeswitch-users] Advise on FS core dump
In-Reply-To: <CAOTmbMwt4Ewz8tLG3Tp+83V3qyvxT4yP2Arws-UoNHZNbBu3LA@mail.gmail.com>
References: <CAOTmbMwx9CSQ=EHC6rsMtHJ09GRMLSH5Q3Sc2v15pidf6CveyA@mail.gmail.com>
	<CAO4LeU3fL0AXFY6uXYhgXgJYDmkLT6tdwj0xqABcgV4cKXDUYQ@mail.gmail.com>
	<CAOTmbMwAHBrgG57hVAbr+oyS6U6W3HtGniZXDwTnfAcDKQhbRQ@mail.gmail.com>
	<CAO4LeU0fCz1PB17muoyUUCLtTLDr_pNuag7jnbXD6rui216eiQ@mail.gmail.com>
	<CAOTmbMwt4Ewz8tLG3Tp+83V3qyvxT4yP2Arws-UoNHZNbBu3LA@mail.gmail.com>
Message-ID: <CAOTmbMxrd+7GdGt_XxPVOp5v2UHSYbc3FQ=L0E6rymCC_3=Mkg@mail.gmail.com>

Hi Stephen,

i have resolved this issue, the root cause is that i define the cmd[100],
while the actual string leghth exceed this lenght (178), then caused the
core dump, now everything runs fine.

Appreciate for your kinldy help which encourging me to make it done,
Cheers!! my friend!

BR,
Charles

2011/10/25 fieldpeak <fieldpeak at gmail.com>

> Hi Stephen,
>
> Thanks for your advise, i tried it, i issued below commands, however, i did
> not find any core named files under freeswitch bin directory, and below is
> the screen showing, i could you identify the root cause, could you please
> help, thanks a lot!
>
> gdb -c core.xxx freeswitch
> ddb>run
> ...(fs running up)
> gdb>bt full
>
> freeswitch at freeswitch> 2011-10-25 10:30:44.161051 [WARNING]
> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
> [13910936628 at 124.193.106.104] from ip 222.128.70.10
> 2011-10-25 10:30:45.081037 [NOTICE] switch_channel.c:897 New Channel
> sofia/internal/13580358068 at 124.193.106.104[38946a57-c673-4705-8433-36f4b066d01c]
> [New Thread 0xb71d6b90 (LWP 19991)]
> is_b is: 0
> excute external cmd: check_billing_before_answer 13580358068 13910936628
> 38946a57-c673-4705-8433-36f4b066d01c 9 222.128.70.10
> 2011-10-25 10:30:45.081037 [INFO] mod_nibblebill.c:1017
> check_billing_before_routing, in_username: 13580358068, in_calledstation:
> 13910936628, in_uniqueid: 38946a57-c673-4705-8433-36f4b066d01c in_type: 9,
> in_network_addr: 222.128.70.10
> Detaching after fork from child process 19992.
>
> the result of excuting external cmd:
> the result of excuting external cmd: 60
>
> Program received signal SIGSEGV, Segmentation fault.
> [Switching to Thread 0xb71d6b90 (LWP 19991)]
> 0x008ff698 in switch_core_get_state_handler at plt () from
> /usr/local/freeswitch/lib/libfreeswitch.so.1
> (gdb) bt full
> #0  0x008ff698 in switch_core_get_state_handler at plt ()
>    from /usr/local/freeswitch/lib/libfreeswitch.so.1
> No symbol table info available.
> #1  0x0092d558 in switch_core_session_run (session=0x81bc238) at
> src/switch_core_state_machine.c:373
>         global_proceed = 1
>         proceed = <value optimized out>
>         ptr = <value optimized out>
>         rstatus = <value optimized out>
>         state = <value optimized out>
>         endstate = <value optimized out>
>         endpoint_interface = <value optimized out>
>         driver_state_handler = 0xcb8420
>         application_state_handler = <value optimized out>
>         new_loops = 60000
>         __PRETTY_FUNCTION__ = "switch_core_session_run"
>         __func__ = "switch_core_session_run"
> #2  0x009275ca in switch_core_session_thread (thread=0x823fdd8,
> obj=0x81bc238)
>     at src/switch_core_session.c:1309
>         session = 0x81bc238
>         event = <value optimized out>
>         event_str = 0x0
>         val = <value optimized out>
>         __func__ = "switch_core_session_thread"
>         __PRETTY_FUNCTION__ = "switch_core_session_thread"
> #3  0x009c2d16 in dummy_worker (opaque=0x823fdd8) at
> threadproc/unix/thread.c:138
> ---Type <return> to continue, or q <return> to quit---
> No locals.
> #4  0x002af832 in start_thread () from /lib/libpthread.so.0
> No symbol table info available.
> #5  0x0078145e in clone () from /lib/libc.so.6
> No symbol table info available.
> (gdb)
>
> 2011/10/24 Stephen Wilde <wstephen80 at gmail.com>
>
>> You can issue a:
>>
>> gdb -c core.xxx freeswitch
>>
>> in freeswitch bin directory where core.xxx is the name of the generated
>> core dump then, in gdb:
>>
>> bt full
>>
>> to show the stack backtrace
>>
>> On Mon, Oct 24, 2011 at 11:53 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Stephen,
>>>
>>> Thanks for your kindly help, however, i changed it as your advise, the
>>> core still dumped.
>>>
>>> below is the updated code,
>>>
>>>
>>> static switch_status_t check_billing_before_routing(switch_core_session_t
>>> *session)
>>> {
>>>     //charles, check billing before routing the call
>>>
>>>     char buf[128];
>>>     char cmd[100];
>>>
>>>     FILE *pp = NULL;
>>>
>>>     const char* in_username = NULL;
>>>     const char* in_calledstation = NULL;
>>>     const char* in_uniqueid = NULL;
>>>     const char* in_network_addr = NULL;
>>>     const char* in_type = "9";
>>>     int is_b;
>>>
>>>     //const char* full_cmd;
>>>     int i_continue = 1;
>>>     switch_channel_t *channel = NULL;
>>>
>>>
>>>     in_uniqueid = switch_core_session_get_uuid(session);
>>>
>>>     channel = switch_core_session_get_channel(session);
>>>
>>>     /*only billing on a leg*/
>>>     is_b = channel &&
>>> switch_channel_get_originator_caller_profile(channel);
>>>     printf("is_b is: %d\n", is_b);
>>>     if(is_b)
>>>     {
>>>     return SWITCH_STATUS_SUCCESS;
>>>     }
>>>
>>>     /* Lock this session's data for this module while we tinker with it
>>> */
>>>     if (globals.mutex) {
>>>         switch_mutex_lock(globals.mutex);
>>>     }
>>>
>>>
>>>     in_username = switch_channel_get_variable(channel,
>>> "caller_id_number");
>>>     in_calledstation = switch_channel_get_variable(channel,
>>> "destination_number");
>>>     in_network_addr = switch_channel_get_variable(channel,
>>> "network_addr");
>>>
>>>
>>>     /* Done checking - release lock */
>>>     if (globals.mutex) {
>>>         switch_mutex_unlock(globals.mutex);
>>>     }
>>>
>>>
>>>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
>>> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
>>> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
>>> in_calledstation, in_uniqueid, in_type, in_network_addr);
>>>
>>>     strcpy(cmd,"");
>>>     strcat(cmd, "check_billing_before_answer");
>>>
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_username);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_calledstation);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_uniqueid);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_type);
>>>     strcat(cmd, " ");
>>>     strcat(cmd, in_network_addr);
>>>
>>>     printf("excute external cmd: ");
>>>     printf(cmd);
>>>     printf("\n");
>>>
>>>     if( (pp = popen(cmd, "r")) == NULL )
>>>     {
>>>         printf("popen() error!\n");
>>>         exit(1);
>>>     }
>>>
>>>     while(fgets(buf, sizeof buf, pp))
>>>     {
>>>         printf("the result of excuting external cmd: %s", buf);
>>>     }
>>>
>>>     if(pp)
>>>     {
>>>         pclose(pp);
>>>     }
>>>
>>>     i_continue =  atoi(buf);
>>>
>>>     printf("i_continue: %d", i_continue);
>>>
>>>
>>>     if (!i_continue) {
>>>
>>>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>>>
>>>         transfer_call(session, globals.nobal_action);
>>>     }
>>>
>>>
>>>
>>>
>>>     return SWITCH_STATUS_SUCCESS;
>>>
>>> }
>>>
>>>
>>> 2011/10/24 Stephen Wilde <wstephen80 at gmail.com>
>>>
>>>> Can you try to change the allocation of cmd as:
>>>>
>>>>    char cmd[100];
>>>>
>>>> ....
>>>>
>>>> strcat(cmd, "check_billing_before_answer ");
>>>>
>>>> strcat(cmd, in_username);
>>>>
>>>> strcat(cmd, " ");
>>>>
>>>>
>>>> ....
>>>>
>>>> On Mon, Oct 24, 2011 at 10:39 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>
>>>>> Hi Friends,
>>>>>
>>>>> i'm using mod_nibblebill for billing on FS, i bundled event processing
>>>>> fucntions when status change,
>>>>> e.g. /* on_routing */ check_billing_before_routing,
>>>>> and * on_hangup */ process_hangup, as below,
>>>>> however, i found if i make a call containing two-legs e.g. user 1001
>>>>> call user 1002, then the FS core dump. but if i only make one leg call, e.g.
>>>>> call 888 for IVR, then no problem occure, i found the problem within
>>>>> function "check_billing_before_routing" (because once i comment all of lines
>>>>> within this func, the core did not dump). below is the func
>>>>> "check_billing_before_routing", can anyone help advise what's wrong? how
>>>>> should i correct it, thanks.
>>>>>
>>>>> static switch_status_t
>>>>> check_billing_before_routing(switch_core_session_t *session)
>>>>> {
>>>>>     //charles, check billing before routing the call
>>>>>
>>>>>     char buf[128];
>>>>>     char cmd[100] = "check_billing_before_answer";
>>>>>     FILE *pp = NULL;
>>>>>
>>>>>     const char* in_username = NULL;
>>>>>     const char* in_calledstation = NULL;
>>>>>     const char* in_uniqueid = NULL;
>>>>>     const char* in_network_addr = NULL;
>>>>>     const char* in_type = "9";
>>>>>     int is_b;
>>>>>
>>>>>     //const char* full_cmd;
>>>>>     int i_continue = 1;
>>>>>     switch_channel_t *channel = NULL;
>>>>>
>>>>>
>>>>>     in_uniqueid = switch_core_session_get_uuid(session);
>>>>>
>>>>>     channel = switch_core_session_get_channel(session);
>>>>>
>>>>>     /*only billing on a leg*/
>>>>>     is_b = channel &&
>>>>> switch_channel_get_originator_caller_profile(channel);
>>>>>     printf("is_b is: %d\n", is_b);
>>>>>     if(is_b)
>>>>>     {
>>>>>     return SWITCH_STATUS_SUCCESS;
>>>>>     }
>>>>>
>>>>>     /* Lock this session's data for this module while we tinker with it
>>>>> */
>>>>>     if (globals.mutex) {
>>>>>         switch_mutex_lock(globals.mutex);
>>>>>     }
>>>>>
>>>>>
>>>>>     in_username = switch_channel_get_variable(channel,
>>>>> "caller_id_number");
>>>>>     in_calledstation = switch_channel_get_variable(channel,
>>>>> "destination_number");
>>>>>     in_network_addr = switch_channel_get_variable(channel,
>>>>> "network_addr");
>>>>>
>>>>>
>>>>>     /* Done checking - release lock */
>>>>>     if (globals.mutex) {
>>>>>         switch_mutex_unlock(globals.mutex);
>>>>>     }
>>>>>
>>>>>
>>>>>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO,
>>>>> "check_billing_before_routing, in_username: %s, in_calledstation: %s,
>>>>> in_uniqueid: %s in_type: %s, in_network_addr: %s \n", in_username,
>>>>> in_calledstation, in_uniqueid, in_type, in_network_addr);
>>>>>
>>>>>
>>>>>     strcat(cmd, " ");
>>>>>     strcat(cmd, in_username);
>>>>>     strcat(cmd, " ");
>>>>>     strcat(cmd, in_calledstation);
>>>>>     strcat(cmd, " ");
>>>>>     strcat(cmd, in_uniqueid);
>>>>>     strcat(cmd, " ");
>>>>>     strcat(cmd, in_type);
>>>>>     strcat(cmd, " ");
>>>>>     strcat(cmd, in_network_addr);
>>>>>
>>>>>     printf("excute external cmd: ");
>>>>>     printf(cmd);
>>>>>     printf("\n");
>>>>>
>>>>>     if( (pp = popen(cmd, "r")) == NULL )
>>>>>     {
>>>>>         printf("popen() error!\n");
>>>>>         exit(1);
>>>>>     }
>>>>>
>>>>>     while(fgets(buf, sizeof buf, pp))
>>>>>     {
>>>>>         printf("the result of excuting external cmd: %s", buf);
>>>>>     }
>>>>>
>>>>>     if(pp)
>>>>>     {
>>>>>         pclose(pp);
>>>>>     }
>>>>>
>>>>>     i_continue =  atoi(buf);
>>>>>
>>>>>     printf("i_continue: %d", i_continue);
>>>>>
>>>>>
>>>>>     if (!i_continue) {
>>>>>
>>>>>         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>>>> SWITCH_LOG_INFO, "call was not authorized to continue \n");
>>>>>
>>>>>         transfer_call(session, globals.nobal_action);
>>>>>     }
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>     return SWITCH_STATUS_SUCCESS;
>>>>>
>>>>> }
>>>>>
>>>>>
>>>>> Core dump logs:
>>>>>
>>>>> freeswitch at freeswitch> 2011-10-24 16:30:26.583369 [NOTICE]
>>>>> mod_xml_rpc.c:969 Starting HTTP Port 8787, DocRoot
>>>>> [/usr/local/freeswitch/htdocs]
>>>>> 2011-10-24 16:30:29.454743 [WARNING] sofia_reg.c:1337 SIP auth
>>>>> challenge (INVITE) on sofia profile 'internal' for [
>>>>> 6628 at 224.193.106.204] from ip 223.128.70.10
>>>>> 2011-10-24 16:30:30.254710 [NOTICE] switch_channel.c:897 New Channel
>>>>> sofia/internal/8068 at 224.193.106.204[196a50b8-eaa0-4ef0-a674-18602117f5ec]
>>>>> is_b is: 0
>>>>> excute external cmd: check_billing_before_answer 8068 6628
>>>>> 196a50b8-eaa0-4ef0-a674-18602117f5ec 9 223.128.70.10
>>>>> 2011-10-24 16:30:30.254710 [INFO] mod_nibblebill.c:1017
>>>>> check_billing_before_routing, in_username: 8068, in_calledstation: 6628,
>>>>> in_uniqueid: 196a50b8-eaa0-4ef0-a674-18602117f5ec in_type: 9,
>>>>> in_network_addr: 223.128.70.10
>>>>> the result of excuting external cmd:
>>>>> the result of excuting external cmd: 60
>>>>> Segmentation fault (core dumped)
>>>>> [root at freeswitch bin]#
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Charles
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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,
>>> Charles
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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,
> Charles
>
>


-- 
Regards,
Charles
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From peter.olsson at visionutveckling.se  Tue Oct 25 09:56:39 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 25 Oct 2011 07:56:39 +0200
Subject: [Freeswitch-users] Task was executed late - warning messages
Message-ID: <A6E852AC-43DE-4057-A699-75563E64F58F@visionutveckling.se>

Only one recommendation - don't do it :) At least not for a production system. The problem is usually the lack of accurate timers - and this exactly what you're experiencing here..

If you want to continue to run it virtually, there are a few basic things to check; first make sure you never share more cores to the virtual machines than you have in your physical hardware. Also use a separate NIC for the FS box. You must also make sure you don't have other overload issues, disk IO etc. This way you might be able to get it work "good enough", but in the long run nothing beats real hardware...

/Peter

----- Reply message -----
Fr?n: "Network Information Centre" <nic at itecity.com>
Datum: tis, okt 25, 2011 02:41
Rubrik: [Freeswitch-users] Task was executed late - warning messages
Till: "&apos;FreeSWITCH Users Help&apos;" <freeswitch-users at lists.freeswitch.org>

Yes, Windows Hyper-V w/1 core and 1 GB Ram.
Are there recommendations for running on a VM?


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brad Mina
Sent: October-24-11 2:11 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Task was executed late - warning messages

Would you happen to be running on some type of virtual machine?
On Sun, Oct 23, 2011 at 5:55 AM, Network Information Centre <nic at itecity.com<mailto:nic at itecity.com>> wrote:
Hello everyone,

I am receiving a great number of these messages, and do not know the service impact, cause or resolution.
They are continuous, with one to several every hour.  The only items I seem to see are: heartbeat, check_ip and limit_hash_cleanup.
The late times vary from 2 to 5 seconds.


2011-10-20 13:42:47.131375 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)
2011-10-20 13:43:01.983476 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core)
2011-10-20 13:44:12.353875 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)
2011-10-20 13:45:17.292117 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)
2011-10-20 13:48:22.629781 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)
2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)
2011-10-20 13:49:07.691921 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 2 check_ip (core)
2011-10-20 14:03:03.753312 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 3 limit_hash_cleanup (mod_hash)
2011-10-20 14:38:32.358406 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 1 heartbeat (core)
2011-10-20 15:17:10.501523 [WARNING] switch_scheduler.c:114 Task was executed late by 3 seconds 2 check_ip (core)
2011-10-20 16:08:35.544968 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 1 heartbeat (core)
2011-10-20 17:02:20.345734 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)
2011-10-20 17:35:13.528281 [WARNING] switch_scheduler.c:114 Task was executed late by 3 seconds 2 check_ip (core)
2011-10-20 20:02:06.848992 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 1 heartbeat (core)
2011-10-20 20:36:09.719593 [WARNING] switch_scheduler.c:114 Task was executed late by 3 seconds 1 heartbeat (core)
2011-10-20 21:01:17.014476 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 2 check_ip (core)
2011-10-20 23:44:22.116851 [WARNING] switch_scheduler.c:114 Task was executed late by 5 seconds 2 check_ip (core)
2011-10-21 00:39:11.946906 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 1 heartbeat (core)

Any assistance is greatly appreciated.

Regards,

Don Hunter
e-City Solutions Inc.

email:                          don.hunter at 44ecity.com<mailto:don.hunter at 44ecity.com>
CA/US Toll-Free:     +1-855-IT-ECITY, ext 200
UK National:             +44 870-820-0286<tel:%2B44%20870-820-0286>
iNum (inum.net<http://inum.net>):   +883 5100-0827-0154

ITeCity.com  |  FoundU.com  |  StoreLocatorService.com  |  go-Postal.ca  |  getZiped.com | FraudCloud.com
WebPageAlerts.com | TurboBrochures.com | DomainStruck.com | EZ-Verify.com | VOIPMuch.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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!DSPAM:4ea603d532761804216445!

From eagle.antonio at gmail.com  Tue Oct 25 10:16:08 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 25 Oct 2011 06:16:08 +0000
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <4EA60A57.4050108@coppice.org>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
	<-5393121003428480786@unknownmsgid>
	<FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
	<CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
	<4EA60A57.4050108@coppice.org>
Message-ID: <CAF+3j-t7XMthMk1nNLYQsTruiiU8bRKKZL8ycVCxMxK0R1a+Pw@mail.gmail.com>

Ok but in our case its not the case anyway , people don't like to comeback
to our ivr's ....

So regarding the initial topic getting DTMF/RTP Events from the main trunk
:)


2011/10/25 Steve Underwood <steveu at coppice.org>

> On 10/25/2011 12:07 AM, Antonio Teixeira wrote:
> > Well guys it depends on your application , generally  i solve this
> > with regexp , a buffer and a way to clear the buffer of
> > invalid/duplicated data I'm extending it to support with min delay
> > between keys.
> > This is not as hard as you think when using min/max :) and your data
> > is predictable :)
> Is a properly engineered solution so unattainable that such guesswork is
> the only solution?
> >
> > Also in ivr menus humans don't generally guess whats the next menu so
> > if len(dtmf) > 1: drop ...
> >
> > I play with this in our test environment not in our production and it
> > works :)
> Try playing with real users. They *always* type ahead by their second or
> third call to the same IVR. Handling type-ahead well - accepting the
> string of entered digits, skipping the playback of prompts so you don't
> slow things down, etc. - is one of the key tests of a good IVR.
>
> Steve
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boris at tagnet.ru  Tue Oct 25 10:18:01 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 25 Oct 2011 12:18:01 +0600
Subject: [Freeswitch-users] strange problems
Message-ID: <4EA65499.80704@tagnet.ru>

Hello!

     I get my log file is populated with 1000+ lines of:
2011-10-25 12:15:54.894956 [INFO] sofia.c:4902 Activating T38 Passthru
2011-10-25 12:15:54.914955 [INFO] sofia.c:4902 Activating T38 Passthru
and I can't find what is the problem and what session is responsible for 
this lines. Any help?

FreeSWITCH Version 1.0.head (git-34827a8 2011-09-26 09-15-12 -0500)

-- 
Regards,
Boris




From steveayre at gmail.com  Tue Oct 25 14:02:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 25 Oct 2011 11:02:07 +0100
Subject: [Freeswitch-users] DTMF Problems
In-Reply-To: <4EA60A57.4050108@coppice.org>
References: <CAF+3j-uQY=EO+ZCJ2xwYv1MLyqV-k+RxXGE-Fvw4-ezf8hGFyA@mail.gmail.com>
	<151CF015-A43B-45F7-83F2-6ABD3513F93B@gmail.com>
	<CAF+3j-uRdo8S3b8sL_GFkaC2bY0qvP-q1rTQAVomVK8LqoddAw@mail.gmail.com>
	<-5393121003428480786@unknownmsgid>
	<FFE8277A-AF4A-4299-9E68-67E1988225AD@gmail.com>
	<CAF+3j-tTzKTe5gqNhbfHvaPKLBDSbPnX9AtqAGE9uekafg844w@mail.gmail.com>
	<4EA60A57.4050108@coppice.org>
Message-ID: <970C9193-3F1F-4A9F-9679-09EE35D5F7DF@gmail.com>

> Try playing with real users. They *always* type ahead by their second or 
> third call to the same IVR. Handling type-ahead well - accepting the 
> string of entered digits, skipping the playback of prompts so you don't 
> slow things down, etc. - is one of the key tests of a good IVR.

I'm guilty of this. Hate sitting through a long message that I already know word for word. 

Steve on iPhone

On 25 Oct 2011, at 02:01, Steve Underwood <steveu at coppice.org> wrote:

> On 10/25/2011 12:07 AM, Antonio Teixeira wrote:
>> Well guys it depends on your application , generally  i solve this 
>> with regexp , a buffer and a way to clear the buffer of 
>> invalid/duplicated data I'm extending it to support with min delay 
>> between keys.
>> This is not as hard as you think when using min/max :) and your data 
>> is predictable :)
> Is a properly engineered solution so unattainable that such guesswork is 
> the only solution?
>> 
>> Also in ivr menus humans don't generally guess whats the next menu so 
>> if len(dtmf) > 1: drop ...
>> 
>> I play with this in our test environment not in our production and it 
>> works :)
> Try playing with real users. They *always* type ahead by their second or 
> third call to the same IVR. Handling type-ahead well - accepting the 
> string of entered digits, skipping the playback of prompts so you don't 
> slow things down, etc. - is one of the key tests of a good IVR.
> 
> Steve
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From NuwanW at unifybusiness.co.uk  Tue Oct 25 17:49:31 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Tue, 25 Oct 2011 13:49:31 +0000
Subject: [Freeswitch-users] event socket
In-Reply-To: <CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
Message-ID: <78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>

Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021 async full"/>
                    </condition>
                </extension>

Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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 hynek.cihlar at gmail.com  Tue Oct 25 17:53:38 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Tue, 25 Oct 2011 15:53:38 +0200
Subject: [Freeswitch-users] event socket
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
Message-ID: <CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek



On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <
NuwanW at unifybusiness.co.uk> wrote:

> Hi,****
>
> ** **
>
> Thanks for the reply. Yes my application is listening on the port and I?m
> receiving events. However when I try to execute ?answer? application it?s
> not working. I?m trying to execute ?answer? application when I receive
> ?CHANNEL_CREATE? event on inbound calls. Please see code below. Could you
> Please advise****
>
> ** **
>
> using System;****
>
> using System.Collections.Generic;****
>
> using System.Linq;****
>
> using System.Text;****
>
> using System.Threading;****
>
> using System.IO;****
>
> ** **
>
> namespace ESLTest****
>
> {****
>
>     public class Test****
>
>     {****
>
>         private Dictionary<string, Call> CallsTable;****
>
>         ****
>
>         private Thread ThreadReader;****
>
>         ****
>
>         private ESLconnection eslConnection;****
>
> ** **
>
>         private const int CONNECTED = 1;****
>
> ** **
>
>         private const string server = "localhost";****
>
>         private const string port = "8021";****
>
>         private const string password = "ClueCon";****
>
> ** **
>
>         ****
>
>         public Test()****
>
>         {****
>
>             CallsTable = new Dictionary<string, Call>();****
>
>             Connect();****
>
>         }****
>
> ** **
>
> ** **
>
>         private void Connect()****
>
>         {****
>
>             eslConnection = new ESLconnection(server, port, password);****
>
> ** **
>
>             if (eslConnection.Connected().Equals( CONNECTED))****
>
>             {****
>
>                 Console.WriteLine("Conneected to FreeSwitch");****
>
> ** **
>
>                 ThreadReader = new Thread(new ThreadStart(Read));****
>
>                 ThreadReader.Start();****
>
>             }****
>
>             else****
>
>             {****
>
>                 Console.WriteLine("Conneection Failure");****
>
>             }****
>
>         }****
>
> ** **
>
> ** **
>
> ** **
>
>         private void Read()****
>
>         {****
>
>             ESLevent eventHandle = eslConnection.SendRecv("event plain
> ALL");****
>
> ** **
>
>             if (eventHandle == null)****
>
>             {****
>
>                 Console.WriteLine("Error subscribing to all events");****
>
>                 return;****
>
>             }****
>
>             else****
>
>            {****
>
>                 Console.WriteLine("Subscribed to all events");****
>
>             }****
>
> ** **
>
>             while (true)****
>
>             {****
>
>                 eventHandle = eslConnection.RecvEvent();****
>
> ** **
>
>                 string CallID = eventHandle.GetHeader("Unique-ID");****
>
> ** **
>
>                 if (eventHandle.getType() == "CHANNEL_CREATE")****
>
>                 {****
>
>                     //Inbound call****
>
>                     if (eventHandle.GetHeader("Call-Direction") ==
> "inbound")****
>
>                     {****
>
>                         CallsTable.Add(CallID, new Call(CallID));****
>
>                         eslConnection.Execute("answer", null, null);****
>
>                         Console.WriteLine(string.Format("{0} : Inbound
> CallID : {1}", eventHandle.getType(), CallID));****
>
>                     }****
>
>                 }****
>
>            }****
>
>     }****
>
> }****
>
> ** **
>
> Dialplan entry as follows,****
>
> ** **
>
>                 <extension name="intercept">****
>
>                      <condition field="destination_number"
> expression="^9$">****
>
>                            <action application="socket" data="
> 127.0.0.1:8021 async full"/>****
>
>                     </condition>****
>
>                 </extension>****
>
> ** **
>
> Thank you,****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 24 October 2011 18:14
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] event socket****
>
> ** **
>
> Please confirm: are you sending the call to the "socket" application in
> your dialplan? If so, then your ESL app needs to be listening on the TCP
> port specified. See the examples here:
> http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples****
>
> ** **
>
> -MC****
>
> On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <
> NuwanW at unifybusiness.co.uk> wrote:****
>
> Hi,****
>
>  ****
>
> I?m trying to build an IVR application around freeswtich. I?m using
> freeswitch event socket in C#. I?m intercepting inbound calls in the dial
> plan and sending them to my application via outbound socket. Could someone
> please explain how to execute ?answer? application through event socket. *
> ***
>
>  ****
>
> Eg ? I?m sending the command as below and it?s not working.****
>
>  ****
>
> ESLConnection eslConnection = new ESLConnection(server, port, password)***
> *
>
> ESLEvent eventHandle = eslConnection.SendRecv(?event plain ALL?);****
>
>  ****
>
> eventHandle = eslConnection.RecvEvent();****
>
> string uniqueCallID = evenHandle.GetHeader(?Unique-ID?);****
>
>  ****
>
> *eslConnection.Execute(?answer?, string.Empty, UniqueCallID);*****
>
>  ****
>
>  ****
>
> I believe I?m doing something wrong as I can?t find required information.
> Could someone please let me know where to find more information about
> freeswitch event socket in .Net. (Google not giving me required information)
> ****
>
>  ****
>
> Thank you,****
>
>  ****
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From claudio at clickfono.com  Tue Oct 25 18:08:25 2011
From: claudio at clickfono.com (=?ISO-8859-1?Q?Claudio_=C1lvarez?=)
Date: Tue, 25 Oct 2011 11:08:25 -0300
Subject: [Freeswitch-users] ESL on Ruby 1.9.2
Message-ID: <4EA6C2D9.1000304@clickfono.com>

Hi everyone,

I've been able to compile the ESL wrapper for Ruby 1.9.2, however, I'm 
getting an abnormal behavior with a small script (100% functional with 
Ruby 1.8.7) which basically opens an ESL socket connection and listens 
to channel events:

The event loop looks like the following:

con = ESL::ESLconnection.new('localhost', '8021', 'ClueCon')
con.events('plain', 'CHANNEL_ANSWER CHANNEL_HANGUP_COMPLETE 
CHANNEL_BRIDGE CHANNEL_EXECUTE_COMPLETE')

i = 0
EM.run do
   while (con.connected)
     e = con.recvEvent()
     EM.defer do
       if (!e.nil? && !e.getHeader("variable_medularis_uuid").nil? && 
!e.getHeader("variable_clicktocall").nil?)
         on_event(e, con, i)
         i = i + 1
       end
     end
   end

The problem I'm experiencing is that the event loop seems to be reading 
events from the socket out of sync. For example, when I answer a call, 
nothing happens (i.e. no event is processed), however, when I hangup, 
the CHANNEL_ANSWER event is read from the socket. As you can see, I'm 
using EventMachine, however, I've also tried Ruby threads and I'm 
getting the same behavior.

Any clues on this issue? Has anyone tried Ruby 1.9.2 successfully with ESL?

Thanks!

Best,

-- 
Claudio ?lvarez, Ph.D.
Arquitecto de Sistemas / Systems Architect
Medularis S.p.A. - ClickFono
Tel:  56 2 934 40 74
Cel: 56 9 9079 4951
claudio at clickfono.com

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From NuwanW at unifybusiness.co.uk  Tue Oct 25 18:18:35 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Tue, 25 Oct 2011 14:18:35 +0000
Subject: [Freeswitch-users] event socket
In-Reply-To: <CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
Message-ID: <78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>

I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek


On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021<http://127.0.0.1:8021> async full"/>
                    </condition>
                </extension>

Thank you,

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 Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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 hynek.cihlar at gmail.com  Tue Oct 25 18:23:23 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Tue, 25 Oct 2011 16:23:23 +0200
Subject: [Freeswitch-users] event socket
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
Message-ID: <8256893087499069959@unknownmsgid>

You also have to provide the UUID of the channel, so freeswitch knows which
one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk>
wrote:

I made the changes below, still no progress,



                 if (eventHandle.getType() == "CHANNEL_STATE" &&
eventHandle.GetHeader("Answer-State") == "ringing")

                {

                    eslConnection.Execute("answer", null, null);

                }



Thank you,



*Nuwan Wijerathne*

* *

*Software Engineering Department*

* *

*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*



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hynek Cihlar
*Sent:* 25 October 2011 14:54
*To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] event socket



Answer on event CHANNEL_CALLSTATE and channel call state RINGING.


Hynek


On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <
NuwanW at unifybusiness.co.uk> wrote:

Hi,



Thanks for the reply. Yes my application is listening on the port and I?m
receiving events. However when I try to execute ?answer? application it?s
not working. I?m trying to execute ?answer? application when I receive
?CHANNEL_CREATE? event on inbound calls. Please see code below. Could you
Please advise



using System;

using System.Collections.Generic;

using System.Linq;

using System.Text;

using System.Threading;

using System.IO;



namespace ESLTest

{

    public class Test

    {

        private Dictionary<string, Call> CallsTable;



        private Thread ThreadReader;



        private ESLconnection eslConnection;



        private const int CONNECTED = 1;



        private const string server = "localhost";

        private const string port = "8021";

        private const string password = "ClueCon";





        public Test()

        {

            CallsTable = new Dictionary<string, Call>();

            Connect();

        }





        private void Connect()

        {

            eslConnection = new ESLconnection(server, port, password);



            if (eslConnection.Connected().Equals( CONNECTED))

            {

                Console.WriteLine("Conneected to FreeSwitch");



                ThreadReader = new Thread(new ThreadStart(Read));

                ThreadReader.Start();

            }

            else

            {

                Console.WriteLine("Conneection Failure");

            }

        }







        private void Read()

        {

            ESLevent eventHandle = eslConnection.SendRecv("event plain
ALL");



            if (eventHandle == null)

            {

                Console.WriteLine("Error subscribing to all events");

                return;

            }

            else

           {

                Console.WriteLine("Subscribed to all events");

            }



            while (true)

            {

                eventHandle = eslConnection.RecvEvent();



                string CallID = eventHandle.GetHeader("Unique-ID");



                if (eventHandle.getType() == "CHANNEL_CREATE")

                {

                    //Inbound call

                    if (eventHandle.GetHeader("Call-Direction") ==
"inbound")

                    {

                        CallsTable.Add(CallID, new Call(CallID));

                        eslConnection.Execute("answer", null, null);

                        Console.WriteLine(string.Format("{0} : Inbound
CallID : {1}", eventHandle.getType(), CallID));

                    }

                }

           }

    }

}



Dialplan entry as follows,



                <extension name="intercept">

                     <condition field="destination_number" expression="^9$">

                           <action application="socket"
data="127.0.0.1:8021async full"/>

                    </condition>

                </extension>



Thank you,



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
Collins
*Sent:* 24 October 2011 18:14
*To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] event socket



Please confirm: are you sending the call to the "socket" application in your
dialplan? If so, then your ESL app needs to be listening on the TCP port
specified. See the examples here:
http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples



-MC

On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <
NuwanW at unifybusiness.co.uk> wrote:

Hi,



I?m trying to build an IVR application around freeswtich. I?m using
freeswitch event socket in C#. I?m intercepting inbound calls in the dial
plan and sending them to my application via outbound socket. Could someone
please explain how to execute ?answer? application through event socket.



Eg ? I?m sending the command as below and it?s not working.



ESLConnection eslConnection = new ESLConnection(server, port, password)

ESLEvent eventHandle = eslConnection.SendRecv(?event plain ALL?);



eventHandle = eslConnection.RecvEvent();

string uniqueCallID = evenHandle.GetHeader(?Unique-ID?);



*eslConnection.Execute(?answer?, string.Empty, UniqueCallID);*





I believe I?m doing something wrong as I can?t find required information.
Could someone please let me know where to find more information about
freeswitch event socket in .Net. (Google not giving me required information)



Thank you,





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





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http://www.freeswitch.org




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From NuwanW at unifybusiness.co.uk  Tue Oct 25 18:36:30 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Tue, 25 Oct 2011 14:36:30 +0000
Subject: [Freeswitch-users] event socket
In-Reply-To: <8256893087499069959@unknownmsgid>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
Message-ID: <78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>

            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:23
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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<http://www.unifybusiness.co.uk>

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 Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek

On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021<http://127.0.0.1:8021> async full"/>
                    </condition>
                </extension>

Thank you,

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 Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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 hynek.cihlar at gmail.com  Tue Oct 25 18:42:45 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Tue, 25 Oct 2011 16:42:45 +0200
Subject: [Freeswitch-users] event socket
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
Message-ID: <CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>

You want the Unique-ID variable of the incoming message, not CallID.

Hynek



On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <
NuwanW at unifybusiness.co.uk> wrote:

>             Sorry still no progress, code as follows****
>
> ** **
>
>                if (eventHandle.getType() == "CHANNEL_STATE" &&
> eventHandle.GetHeader("Answer-State") == "ringing")****
>
>                 {****
>
>                     eslConnection.Execute("answer", null, CallID);****
>
>                 }****
>
> Thank you,****
>
>  ****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hynek Cihlar
> *Sent:* 25 October 2011 15:23
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] event socket****
>
> ** **
>
> You also have to provide the UUID of the channel, so freeswitch knows which
> one to answer.
>
> Sent from my mobile device****
>
>
> On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk>
> wrote:****
>
> I made the changes below, still no progress,****
>
>  ****
>
>                  if (eventHandle.getType() == "CHANNEL_STATE" &&
> eventHandle.GetHeader("Answer-State") == "ringing")****
>
>                 {****
>
>                     eslConnection.Execute("answer", null, null);****
>
>                 }****
>
>  ****
>
> Thank you,****
>
>  ****
>
> *Nuwan Wijerathne*****
>
> * *****
>
> *Software Engineering Department*****
>
> * *****
>
> *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*****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hynek Cihlar
> *Sent:* 25 October 2011 14:54
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] event socket****
>
>  ****
>
> Answer on event CHANNEL_CALLSTATE and channel call state RINGING.****
>
>
> Hynek
>
> ****
>
> On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <
> NuwanW at unifybusiness.co.uk> wrote:****
>
> Hi,****
>
>  ****
>
> Thanks for the reply. Yes my application is listening on the port and I?m
> receiving events. However when I try to execute ?answer? application it?s
> not working. I?m trying to execute ?answer? application when I receive
> ?CHANNEL_CREATE? event on inbound calls. Please see code below. Could you
> Please advise****
>
>  ****
>
> using System;****
>
> using System.Collections.Generic;****
>
> using System.Linq;****
>
> using System.Text;****
>
> using System.Threading;****
>
> using System.IO;****
>
>  ****
>
> namespace ESLTest****
>
> {****
>
>     public class Test****
>
>     {****
>
>         private Dictionary<string, Call> CallsTable;****
>
>         ****
>
>         private Thread ThreadReader;****
>
>         ****
>
>         private ESLconnection eslConnection;****
>
>  ****
>
>         private const int CONNECTED = 1;****
>
>  ****
>
>         private const string server = "localhost";****
>
>         private const string port = "8021";****
>
>         private const string password = "ClueCon";****
>
>  ****
>
>         ****
>
>         public Test()****
>
>         {****
>
>             CallsTable = new Dictionary<string, Call>();****
>
>             Connect();****
>
>         }****
>
>  ****
>
>  ****
>
>         private void Connect()****
>
>         {****
>
>             eslConnection = new ESLconnection(server, port, password);****
>
>  ****
>
>             if (eslConnection.Connected().Equals( CONNECTED))****
>
>             {****
>
>                 Console.WriteLine("Conneected to FreeSwitch");****
>
>  ****
>
>                 ThreadReader = new Thread(new ThreadStart(Read));****
>
>                 ThreadReader.Start();****
>
>             }****
>
>             else****
>
>             {****
>
>                 Console.WriteLine("Conneection Failure");****
>
>             }****
>
>         }****
>
>  ****
>
>  ****
>
>  ****
>
>         private void Read()****
>
>         {****
>
>             ESLevent eventHandle = eslConnection.SendRecv("event plain
> ALL");****
>
>  ****
>
>             if (eventHandle == null)****
>
>             {****
>
>                 Console.WriteLine("Error subscribing to all events");****
>
>                 return;****
>
>             }****
>
>             else****
>
>            {****
>
>                 Console.WriteLine("Subscribed to all events");****
>
>             }****
>
>  ****
>
>             while (true)****
>
>             {****
>
>                 eventHandle = eslConnection.RecvEvent();****
>
>  ****
>
>                 string CallID = eventHandle.GetHeader("Unique-ID");****
>
>  ****
>
>                 if (eventHandle.getType() == "CHANNEL_CREATE")****
>
>                 {****
>
>                     //Inbound call****
>
>                     if (eventHandle.GetHeader("Call-Direction") ==
> "inbound")****
>
>                     {****
>
>                         CallsTable.Add(CallID, new Call(CallID));****
>
>                         eslConnection.Execute("answer", null, null);****
>
>                         Console.WriteLine(string.Format("{0} : Inbound
> CallID : {1}", eventHandle.getType(), CallID));****
>
>                     }****
>
>                 }****
>
>            }****
>
>     }****
>
> }****
>
>  ****
>
> Dialplan entry as follows,****
>
>  ****
>
>                 <extension name="intercept">****
>
>                      <condition field="destination_number"
> expression="^9$">****
>
>                            <action application="socket" data="
> 127.0.0.1:8021 async full"/>****
>
>                     </condition>****
>
>                 </extension>****
>
>  ****
>
> Thank you,****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 24 October 2011 18:14
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] event socket****
>
>  ****
>
> Please confirm: are you sending the call to the "socket" application in
> your dialplan? If so, then your ESL app needs to be listening on the TCP
> port specified. See the examples here:
> http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples****
>
>  ****
>
> -MC****
>
> On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <
> NuwanW at unifybusiness.co.uk> wrote:****
>
> Hi,****
>
>  ****
>
> I?m trying to build an IVR application around freeswtich. I?m using
> freeswitch event socket in C#. I?m intercepting inbound calls in the dial
> plan and sending them to my application via outbound socket. Could someone
> please explain how to execute ?answer? application through event socket. *
> ***
>
>  ****
>
> Eg ? I?m sending the command as below and it?s not working.****
>
>  ****
>
> ESLConnection eslConnection = new ESLConnection(server, port, password)***
> *
>
> ESLEvent eventHandle = eslConnection.SendRecv(?event plain ALL?);****
>
>  ****
>
> eventHandle = eslConnection.RecvEvent();****
>
> string uniqueCallID = evenHandle.GetHeader(?Unique-ID?);****
>
>  ****
>
> *eslConnection.Execute(?answer?, string.Empty, UniqueCallID);*****
>
>  ****
>
>  ****
>
> I believe I?m doing something wrong as I can?t find required information.
> Could someone please let me know where to find more information about
> freeswitch event socket in .Net. (Google not giving me required information)
> ****
>
>  ****
>
> Thank you,****
>
>  ****
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>  ****
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>  ****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fcintrono at gmail.com  Tue Oct 25 18:43:22 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Tue, 25 Oct 2011 09:43:22 -0500
Subject: [Freeswitch-users] How to know who is calling me...
In-Reply-To: <CAKzWOxX2XUAqFEEFyY_640K4cHRO9osjzwsmBu37ZT9_76UXYQ@mail.gmail.com>
References: <CANtw_ZPhmgMvaPUnLuBSnMPywmYctn1oz=Vn7HuEFjU0fyQDxw@mail.gmail.com>
	<CAKzWOxX2XUAqFEEFyY_640K4cHRO9osjzwsmBu37ZT9_76UXYQ@mail.gmail.com>
Message-ID: <CANtw_ZNYh67=od5SUkUvXbzQmrYTydsxGNdY_cXnWmJnKgWJSQ@mail.gmail.com>

Mmm, Let me make some calls to my phone provider.
I bet we certainly have this service.

Have a nice day

On Mon, Oct 24, 2011 at 7:24 PM, Michael Collins <msc at freeswitch.org> wrote:

> Does Mexico have caller ID service on analog phone lines?
> -MC
>
> On Mon, Oct 24, 2011 at 2:28 PM, Francisco Javier Cintr?n Olgu?n <
> fcintrono at gmail.com> wrote:
>
>> Hello from Guadalajara, M?xico. :)
>>
>> For the last month I have been reading the fs book and  playing with fs.
>> I am pretty conviced that It would be perfect as my soho office.
>>
>> I have this configuration:
>>
>> 4 normal internal phones.
>> 2 pots lines.
>>
>> to interface freeswitch with these devices I purchased a CISCO SPA8800.
>> This unit has 8 ports:
>>
>> 4 for normal phones.
>> 4 for pots lines.
>>
>> I also purchased a cheap IP phone: linksys SPA921.
>>
>> With the help of fs wiki site I got how to configure the SPA8800.
>>
>> The only thing I don't know is how to show in the LCD display of the
>> SPA921 who is calling or the number from which is the call.
>>
>>
>> Thanks in advance for your kind help.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hynek.cihlar at gmail.com  Tue Oct 25 18:46:00 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Tue, 25 Oct 2011 16:46:00 +0200
Subject: [Freeswitch-users] vars.xml in Lua
In-Reply-To: <CAKzWOxUd_imRAab6X7BwRe6n1tqMgniu0rwC-gMim59Ht2zr4A@mail.gmail.com>
References: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
	<CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>
	<-8152858329743577435@unknownmsgid>
	<CAKzWOxUd_imRAab6X7BwRe6n1tqMgniu0rwC-gMim59Ht2zr4A@mail.gmail.com>
Message-ID: <CAJh8XdXHzGDA03rpeRMoZfjS28D+-c5enRHeiOww2W7k8OwP7Q@mail.gmail.com>

Like a charm! Tx.

Hynek



On Mon, Oct 24, 2011 at 9:34 PM, Michael Collins <msc at freeswitch.org> wrote:

> Nothing special. If you are calling a dialplan script then it's this:
>
> my_val = session:getVariable('Variable_Name')
>
> If this is a non-session kind of script then you can do the API trick:
>
> api = freeswitch.API()
> my_val = api:execute('global_getvar','Variable_Name')
>
> -MC
>
>
> On Mon, Oct 24, 2011 at 10:34 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> Tx for the reply.
>>
>> Are they simply declared as global variables or do they need to be
>> accessed in a special way?
>>
>> Sent from my mobile device
>>
>> On Oct 24, 2011, at 19:10, Michael Collins <msc at freeswitch.org> wrote:
>>
>> Yes, any global variable defined in vars.xml is available in a Lua script.
>> -MC
>>
>> On Sat, Oct 22, 2011 at 10:39 AM, Hynek Cihlar < <hynek.cihlar at gmail.com>
>> hynek.cihlar at gmail.com> wrote:
>>
>>> Are the parameters defined in the XML config files visible to Lua scrips?
>>> In this case it is the script for loading up the directory.
>>>
>>> Hynek
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Oct 25 18:49:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Oct 2011 09:49:02 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1319494283.4593.160.camel@portable-evil>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
Message-ID: <CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>

make sure you have it setup so it can do transactions or multiple
statements.


On Mon, Oct 24, 2011 at 5:11 PM, Cliff Wells <cliff at develix.com> wrote:

> On Mon, 2011-10-24 at 17:36 -0400, Tim St. Pierre wrote:
>
> > Database is MySQL 5.5.  I'm seeing 18 queries per second, mostly
> complicated internal
> > joins on tables like sip_dialogs and sip_subscriptions.
> >
> > Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
> and hardly breaking
> > a sweat, while the database server is at 100% CPU load.
>
>
> You need to optimize your database. Most likely you need to index tables
> that are used in joins (I doubt further optimization is needed for such
> a small dataset and such a light load).  I don't use MySQL, but I recall
> it has a way to log slow queries.  You should turn this on and use the
> output to figure out the right queries to optimize.  Don't blindly
> create needless indexes as this will actually slow it more.
>
> Cliff
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 Oct 25 18:52:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Oct 2011 09:52:55 -0500
Subject: [Freeswitch-users] vars.xml in Lua
In-Reply-To: <CAKzWOxUd_imRAab6X7BwRe6n1tqMgniu0rwC-gMim59Ht2zr4A@mail.gmail.com>
References: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
	<CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>
	<-8152858329743577435@unknownmsgid>
	<CAKzWOxUd_imRAab6X7BwRe6n1tqMgniu0rwC-gMim59Ht2zr4A@mail.gmail.com>
Message-ID: <CAKbxfG8M=K3SyDwkEdhe+Fj8HG3Uq5m1uDgoEs+QFff7cNAOqw@mail.gmail.com>

with no session there is a more recent way

freeswitch.getGlobalVariable("foobar");

On Mon, Oct 24, 2011 at 2:34 PM, Michael Collins <msc at freeswitch.org> wrote:

> Nothing special. If you are calling a dialplan script then it's this:
>
> my_val = session:getVariable('Variable_Name')
>
> If this is a non-session kind of script then you can do the API trick:
>
> api = freeswitch.API()
> my_val = api:execute('global_getvar','Variable_Name')
>
> -MC
>
>
> On Mon, Oct 24, 2011 at 10:34 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> Tx for the reply.
>>
>> Are they simply declared as global variables or do they need to be
>> accessed in a special way?
>>
>> Sent from my mobile device
>>
>> On Oct 24, 2011, at 19:10, Michael Collins <msc at freeswitch.org> wrote:
>>
>> Yes, any global variable defined in vars.xml is available in a Lua script.
>> -MC
>>
>> On Sat, Oct 22, 2011 at 10:39 AM, Hynek Cihlar < <hynek.cihlar at gmail.com>
>> hynek.cihlar at gmail.com> wrote:
>>
>>> Are the parameters defined in the XML config files visible to Lua scrips?
>>> In this case it is the script for loading up the directory.
>>>
>>> Hynek
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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 NuwanW at unifybusiness.co.uk  Tue Oct 25 18:54:28 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Tue, 25 Oct 2011 14:54:28 +0000
Subject: [Freeswitch-users] event socket
In-Reply-To: <CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
	<CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>
Message-ID: <78990CE7CC964442A7C2CA5F4689695E4BA4CD24@BARXB0003.UnifyBusiness.local>

Hi,

CallID is a variable I defined. I'm assigning the call Unique-ID to this variable as follows,

                string CallID = eventHandle.GetHeader("Unique-ID");
My full code is at the bottom of this email,

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You want the Unique-ID variable of the incoming message, not CallID.

Hynek


On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

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 Hynek Cihlar
Sent: 25 October 2011 15:23

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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<http://www.unifybusiness.co.uk>

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 Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek
On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021<http://127.0.0.1:8021> async full"/>
                    </condition>
                </extension>

Thank you,

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 Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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


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


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From hynek.cihlar at gmail.com  Tue Oct 25 18:55:15 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Tue, 25 Oct 2011 16:55:15 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
Message-ID: <CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>

For the transactions, what would be required to change? Any tweaks in the
ODBC driver/DSN settins?

Hynek



On Tue, Oct 25, 2011 at 4:49 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> make sure you have it setup so it can do transactions or multiple
> statements.
>
>
> On Mon, Oct 24, 2011 at 5:11 PM, Cliff Wells <cliff at develix.com> wrote:
>
>> On Mon, 2011-10-24 at 17:36 -0400, Tim St. Pierre wrote:
>>
>> > Database is MySQL 5.5.  I'm seeing 18 queries per second, mostly
>> complicated internal
>> > joins on tables like sip_dialogs and sip_subscriptions.
>> >
>> > Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>> and hardly breaking
>> > a sweat, while the database server is at 100% CPU load.
>>
>>
>> You need to optimize your database. Most likely you need to index tables
>> that are used in joins (I doubt further optimization is needed for such
>> a small dataset and such a light load).  I don't use MySQL, but I recall
>> it has a way to log slow queries.  You should turn this on and use the
>> output to figure out the right queries to optimize.  Don't blindly
>> create needless indexes as this will actually slow it more.
>>
>> Cliff
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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 hynek.cihlar at gmail.com  Tue Oct 25 18:56:11 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Tue, 25 Oct 2011 16:56:11 +0200
Subject: [Freeswitch-users] vars.xml in Lua
In-Reply-To: <CAKbxfG8M=K3SyDwkEdhe+Fj8HG3Uq5m1uDgoEs+QFff7cNAOqw@mail.gmail.com>
References: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
	<CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>
	<-8152858329743577435@unknownmsgid>
	<CAKzWOxUd_imRAab6X7BwRe6n1tqMgniu0rwC-gMim59Ht2zr4A@mail.gmail.com>
	<CAKbxfG8M=K3SyDwkEdhe+Fj8HG3Uq5m1uDgoEs+QFff7cNAOqw@mail.gmail.com>
Message-ID: <CAJh8XdUAyPDSc7xB9fvhzx7vFkcs4h=LMpgDjUyDqYXe6UoWzg@mail.gmail.com>

Great tip!

Hynek



On Tue, Oct 25, 2011 at 4:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> with no session there is a more recent way
>
> freeswitch.getGlobalVariable("foobar");
>
> On Mon, Oct 24, 2011 at 2:34 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Nothing special. If you are calling a dialplan script then it's this:
>>
>> my_val = session:getVariable('Variable_Name')
>>
>> If this is a non-session kind of script then you can do the API trick:
>>
>> api = freeswitch.API()
>> my_val = api:execute('global_getvar','Variable_Name')
>>
>> -MC
>>
>>
>> On Mon, Oct 24, 2011 at 10:34 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>>
>>> Tx for the reply.
>>>
>>> Are they simply declared as global variables or do they need to be
>>> accessed in a special way?
>>>
>>> Sent from my mobile device
>>>
>>> On Oct 24, 2011, at 19:10, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>> Yes, any global variable defined in vars.xml is available in a Lua
>>> script.
>>> -MC
>>>
>>> On Sat, Oct 22, 2011 at 10:39 AM, Hynek Cihlar <<hynek.cihlar at gmail.com>
>>> hynek.cihlar at gmail.com> wrote:
>>>
>>>> Are the parameters defined in the XML config files visible to Lua
>>>> scrips? In this case it is the script for loading up the directory.
>>>>
>>>> Hynek
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>>
>>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> 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 anthony.minessale at gmail.com  Tue Oct 25 19:00:29 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Oct 2011 10:00:29 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
Message-ID: <CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>

I don't particularly like mysql so I don't know the exact answer, but I do
know that by default it disables transactions and multi statements as a
knee-jerk solution to injection and that it's some numeric param deep in
some config file.  So I would just google for "enable transactions mysql" or
something like that.

You can tell if it matters if freeswitch complains about transactions
disabled when it connect to the db for the first time.


On Tue, Oct 25, 2011 at 9:55 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> For the transactions, what would be required to change? Any tweaks in the
> ODBC driver/DSN settins?
>
> Hynek
>
>
>
>
> On Tue, Oct 25, 2011 at 4:49 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> make sure you have it setup so it can do transactions or multiple
>> statements.
>>
>>
>> On Mon, Oct 24, 2011 at 5:11 PM, Cliff Wells <cliff at develix.com> wrote:
>>
>>> On Mon, 2011-10-24 at 17:36 -0400, Tim St. Pierre wrote:
>>>
>>> > Database is MySQL 5.5.  I'm seeing 18 queries per second, mostly
>>> complicated internal
>>> > joins on tables like sip_dialogs and sip_subscriptions.
>>> >
>>> > Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>> and hardly breaking
>>> > a sweat, while the database server is at 100% CPU load.
>>>
>>>
>>> You need to optimize your database. Most likely you need to index tables
>>> that are used in joins (I doubt further optimization is needed for such
>>> a small dataset and such a light load).  I don't use MySQL, but I recall
>>> it has a way to log slow queries.  You should turn this on and use the
>>> output to figure out the right queries to optimize.  Don't blindly
>>> create needless indexes as this will actually slow it more.
>>>
>>> Cliff
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From hynek.cihlar at gmail.com  Tue Oct 25 19:06:05 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Tue, 25 Oct 2011 17:06:05 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
Message-ID: <CAJh8XdVn+oozWAuEsc0h--sBE_B5_DYqhumdVU4YEeTYiAaoTw@mail.gmail.com>

Of course that MySQL supports and enables transaction by default. It is the
SQL statements that must tell the DB when to start a transaction, commit,
rollback.

I had an issue with batch statements to be enabled on the ODBC connection,
once on, it works ok.

Hynek



On Tue, Oct 25, 2011 at 5:00 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I don't particularly like mysql so I don't know the exact answer, but I do
> know that by default it disables transactions and multi statements as a
> knee-jerk solution to injection and that it's some numeric param deep in
> some config file.  So I would just google for "enable transactions mysql" or
> something like that.
>
> You can tell if it matters if freeswitch complains about transactions
> disabled when it connect to the db for the first time.
>
>
> On Tue, Oct 25, 2011 at 9:55 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> For the transactions, what would be required to change? Any tweaks in the
>> ODBC driver/DSN settins?
>>
>> Hynek
>>
>>
>>
>>
>> On Tue, Oct 25, 2011 at 4:49 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> make sure you have it setup so it can do transactions or multiple
>>> statements.
>>>
>>>
>>> On Mon, Oct 24, 2011 at 5:11 PM, Cliff Wells <cliff at develix.com> wrote:
>>>
>>>> On Mon, 2011-10-24 at 17:36 -0400, Tim St. Pierre wrote:
>>>>
>>>> > Database is MySQL 5.5.  I'm seeing 18 queries per second, mostly
>>>> complicated internal
>>>> > joins on tables like sip_dialogs and sip_subscriptions.
>>>> >
>>>> > Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>>>> and hardly breaking
>>>> > a sweat, while the database server is at 100% CPU load.
>>>>
>>>>
>>>> You need to optimize your database. Most likely you need to index tables
>>>> that are used in joins (I doubt further optimization is needed for such
>>>> a small dataset and such a light load).  I don't use MySQL, but I recall
>>>> it has a way to log slow queries.  You should turn this on and use the
>>>> output to figure out the right queries to optimize.  Don't blindly
>>>> create needless indexes as this will actually slow it more.
>>>>
>>>> Cliff
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Hector.Geraldino at ip-soft.net  Tue Oct 25 19:13:08 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 25 Oct 2011 11:13:08 -0400
Subject: [Freeswitch-users] event socket
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695E4BA4CD24@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
	<CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CD24@BARXB0003.UnifyBusiness.local>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02249DB62D@NY1-EXMB-01.ip-soft.net>

What Type is returned by the .Execute() method call?

Maybe you should create a new variable and assign the returned value to it, so you can inspect the result of the method call, something like:
                EslMessage response = eslConnection.Execute("answer", null, CallID);
String isOk = response.getHeaderValue("Reply-Text");
so you can inspect the response returned by FreeSwitch.

BTW, this is mostly Java code so I'm not sure that these Types/properties are available on the C# assemblies.

And finally, try to perform this operation after receiving the CHANNEL_DATA event instead of CHANNEL_STATE.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 10:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Hi,

CallID is a variable I defined. I'm assigning the call Unique-ID to this variable as follows,

                string CallID = eventHandle.GetHeader("Unique-ID");
My full code is at the bottom of this email,

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You want the Unique-ID variable of the incoming message, not CallID.

Hynek

On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

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 Hynek Cihlar
Sent: 25 October 2011 15:23

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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<http://www.unifybusiness.co.uk>

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 Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek
On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021<http://127.0.0.1:8021> async full"/>
                    </condition>
                </extension>

Thank you,

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 Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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 NuwanW at unifybusiness.co.uk  Tue Oct 25 19:57:17 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Tue, 25 Oct 2011 15:57:17 +0000
Subject: [Freeswitch-users] event socket
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02249DB62D@NY1-EXMB-01.ip-soft.net>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
	<CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CD24@BARXB0003.UnifyBusiness.local>
	<6A6B4C284AD15042B429EB9D904544AD02249DB62D@NY1-EXMB-01.ip-soft.net>
Message-ID: <78990CE7CC964442A7C2CA5F4689695E4BA4D0CC@BARXB0003.UnifyBusiness.local>

I'm receiving "+OK" from .Execute(). I'm not receiving a "CHANNEL_DATA" event.

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino
Sent: 25 October 2011 16:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

What Type is returned by the .Execute() method call?

Maybe you should create a new variable and assign the returned value to it, so you can inspect the result of the method call, something like:
                EslMessage response = eslConnection.Execute("answer", null, CallID);
String isOk = response.getHeaderValue("Reply-Text");
so you can inspect the response returned by FreeSwitch.

BTW, this is mostly Java code so I'm not sure that these Types/properties are available on the C# assemblies.

And finally, try to perform this operation after receiving the CHANNEL_DATA event instead of CHANNEL_STATE.


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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 10:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Hi,

CallID is a variable I defined. I'm assigning the call Unique-ID to this variable as follows,

                string CallID = eventHandle.GetHeader("Unique-ID");
My full code is at the bottom of this email,

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You want the Unique-ID variable of the incoming message, not CallID.

Hynek
On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

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 Hynek Cihlar
Sent: 25 October 2011 15:23

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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<http://www.unifybusiness.co.uk>

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 Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek
On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021<http://127.0.0.1:8021> async full"/>
                    </condition>
                </extension>

Thank you,

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 Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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 nbhatti at gmail.com  Tue Oct 25 20:17:14 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 25 Oct 2011 19:17:14 +0300
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
Message-ID: <CACak_yxZrwDoZxKBovVx+94dBK68SLmsc30-5dWz_N4tN-wfOw@mail.gmail.com>

Tuning MySQL would be the first thing I would do. Download and run
this small script, http://mysqltuner.pl and it would show you some
configurable parameters of MySQL which may greatly enhance your DB
performance.

On Tue, Oct 25, 2011 at 5:49 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> make sure you have it setup so it can do transactions or multiple
> statements.
>
> On Mon, Oct 24, 2011 at 5:11 PM, Cliff Wells <cliff at develix.com> wrote:
>>
>> On Mon, 2011-10-24 at 17:36 -0400, Tim St. Pierre wrote:
>>
>> > Database is MySQL 5.5. ?I'm seeing 18 queries per second, mostly
>> > complicated internal
>> > joins on tables like sip_dialogs and sip_subscriptions.
>> >
>> > Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500)
>> > and hardly breaking
>> > a sweat, while the database server is at 100% CPU load.
>>
>>
>> You need to optimize your database. Most likely you need to index tables
>> that are used in joins (I doubt further optimization is needed for such
>> a small dataset and such a light load). ?I don't use MySQL, but I recall
>> it has a way to log slow queries. ?You should turn this on and use the
>> output to figure out the right queries to optimize. ?Don't blindly
>> create needless indexes as this will actually slow it more.
>>
>> Cliff
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sinan.goo at artcom.de  Tue Oct 25 13:27:28 2011
From: sinan.goo at artcom.de (srg)
Date: Tue, 25 Oct 2011 02:27:28 -0700 (PDT)
Subject: [Freeswitch-users] SDP rewriting
Message-ID: <1319534848836-6928152.post@n2.nabble.com>

i am trying to rewrite the video codec in the SDP header from a SIP client A.
It works only partially. 
Client A sends invite with the SDP header to FS trying to call client B and
FS changes the video codes due to:
<action application="set" data="switch_r_sdp=$1m=video 9078 RTP/AVP
102&#x0A;a=rtpmap:102 H264/90000&#x0A;a=fmtp:102
profile-level-id=42C014&#x0A;"/>
in my dialplan/default.xml file.
I know that this first change by FS works correctly because i can see the
SDP header via wireshark.

Client B then answers with 200 ok and a SDP header
but FS then does not send the SDP header from client B to client A
but a mixture of the original SDP header from client A and the SDP header
from B's answer.

>From added logging to the FS code i see that the SDP header which is given
to nua_respond(..) in mod_sofia.c when FS sends the 200 okay to client A is
still as expected.

So from where comes the mixed SDP header ?
Am i missing some more settings in the dialplan/default.xml file ?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SDP-rewriting-tp6928152p6928152.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue Oct 25 20:18:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Oct 2011 09:18:02 -0700
Subject: [Freeswitch-users] vars.xml in Lua
In-Reply-To: <CAJh8XdUAyPDSc7xB9fvhzx7vFkcs4h=LMpgDjUyDqYXe6UoWzg@mail.gmail.com>
References: <CAJh8XdX=by1SrJvxFiZwSO2UH3xGx=GLWAfR445uK6e8Jpx+ww@mail.gmail.com>
	<CAKzWOxV6CLV8sfb1ESwLNZtkxKjaVvCMkzuv0ObzQ7vFNJQWkA@mail.gmail.com>
	<-8152858329743577435@unknownmsgid>
	<CAKzWOxUd_imRAab6X7BwRe6n1tqMgniu0rwC-gMim59Ht2zr4A@mail.gmail.com>
	<CAKbxfG8M=K3SyDwkEdhe+Fj8HG3Uq5m1uDgoEs+QFff7cNAOqw@mail.gmail.com>
	<CAJh8XdUAyPDSc7xB9fvhzx7vFkcs4h=LMpgDjUyDqYXe6UoWzg@mail.gmail.com>
Message-ID: <CAKzWOxW7dkorXyUeTwSY89-KOgMK0uHsrLiLqKS3oejzapJNGg@mail.gmail.com>

I just added this to the wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.getGlobalVariable

-MC

On Tue, Oct 25, 2011 at 7:56 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> Great tip!
>
> Hynek
>
>
>
>
> On Tue, Oct 25, 2011 at 4:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> with no session there is a more recent way
>>
>> freeswitch.getGlobalVariable("foobar");
>>
>> On Mon, Oct 24, 2011 at 2:34 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Nothing special. If you are calling a dialplan script then it's this:
>>>
>>> my_val = session:getVariable('Variable_Name')
>>>
>>> If this is a non-session kind of script then you can do the API trick:
>>>
>>> api = freeswitch.API()
>>> my_val = api:execute('global_getvar','Variable_Name')
>>>
>>> -MC
>>>
>>>
>>> On Mon, Oct 24, 2011 at 10:34 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>>>
>>>> Tx for the reply.
>>>>
>>>> Are they simply declared as global variables or do they need to be
>>>> accessed in a special way?
>>>>
>>>> Sent from my mobile device
>>>>
>>>> On Oct 24, 2011, at 19:10, Michael Collins <msc at freeswitch.org> wrote:
>>>>
>>>> Yes, any global variable defined in vars.xml is available in a Lua
>>>> script.
>>>> -MC
>>>>
>>>> On Sat, Oct 22, 2011 at 10:39 AM, Hynek Cihlar <<hynek.cihlar at gmail.com>
>>>> hynek.cihlar at gmail.com> wrote:
>>>>
>>>>> Are the parameters defined in the XML config files visible to Lua
>>>>> scrips? In this case it is the script for loading up the directory.
>>>>>
>>>>> Hynek
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct 25 20:23:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Oct 2011 09:23:07 -0700
Subject: [Freeswitch-users] Italian voice files
Message-ID: <CAKzWOxVaATe54kxm1frs_LDB6wy8pQZsVWC=gBZw8s1nLQ2doA@mail.gmail.com>

Ciao!

I am looking for an Italian set of sound files for FreeSWITCH. Does anyone
know where those might be or even if they exist?

Thanks,
Michael
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From msc at freeswitch.org  Tue Oct 25 20:24:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Oct 2011 09:24:06 -0700
Subject: [Freeswitch-users] How to know who is calling me...
In-Reply-To: <CANtw_ZNYh67=od5SUkUvXbzQmrYTydsxGNdY_cXnWmJnKgWJSQ@mail.gmail.com>
References: <CANtw_ZPhmgMvaPUnLuBSnMPywmYctn1oz=Vn7HuEFjU0fyQDxw@mail.gmail.com>
	<CAKzWOxX2XUAqFEEFyY_640K4cHRO9osjzwsmBu37ZT9_76UXYQ@mail.gmail.com>
	<CANtw_ZNYh67=od5SUkUvXbzQmrYTydsxGNdY_cXnWmJnKgWJSQ@mail.gmail.com>
Message-ID: <CAKzWOxVXBj5MLtFd0DvwGj9rnU-Pceg9D836iyi9VWwM80Xt1Q@mail.gmail.com>

If your phone provider offers analog caller ID then hopefully the SPA8800
can detect it. If so, it should pass the caller ID information over to
FreeSWITCH.

-MC

On Tue, Oct 25, 2011 at 7:43 AM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> Mmm, Let me make some calls to my phone provider.
> I bet we certainly have this service.
>
> Have a nice day
>
>
> On Mon, Oct 24, 2011 at 7:24 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Does Mexico have caller ID service on analog phone lines?
>> -MC
>>
>> On Mon, Oct 24, 2011 at 2:28 PM, Francisco Javier Cintr?n Olgu?n <
>> fcintrono at gmail.com> wrote:
>>
>>> Hello from Guadalajara, M?xico. :)
>>>
>>> For the last month I have been reading the fs book and  playing with fs.
>>> I am pretty conviced that It would be perfect as my soho office.
>>>
>>> I have this configuration:
>>>
>>> 4 normal internal phones.
>>> 2 pots lines.
>>>
>>> to interface freeswitch with these devices I purchased a CISCO SPA8800.
>>> This unit has 8 ports:
>>>
>>> 4 for normal phones.
>>> 4 for pots lines.
>>>
>>> I also purchased a cheap IP phone: linksys SPA921.
>>>
>>> With the help of fs wiki site I got how to configure the SPA8800.
>>>
>>> The only thing I don't know is how to show in the LCD display of the
>>> SPA921 who is calling or the number from which is the call.
>>>
>>>
>>> Thanks in advance for your kind help.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 25 21:26:23 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 25 Oct 2011 19:26:23 +0200
Subject: [Freeswitch-users] Italian voice files
In-Reply-To: <CAKzWOxVaATe54kxm1frs_LDB6wy8pQZsVWC=gBZw8s1nLQ2doA@mail.gmail.com>
References: <CAKzWOxVaATe54kxm1frs_LDB6wy8pQZsVWC=gBZw8s1nLQ2doA@mail.gmail.com>
Message-ID: <CALXCt0rdpgoMjBddz40cxQ5bbn=_CE4cinsaXmSwagWyfh=3pQ@mail.gmail.com>

mmmmmh, I don't think they exists.

I had almost convinced a customer to order the entire set, but then he
ended up using completely custom messages (and a small number of
them)... too bad :(.

As soon as I'll have some customer paying for it I'll order the set
from GM voices...

Let me know how it goes.

_giovanni

On Tue, Oct 25, 2011 at 6:23 PM, Michael Collins <msc at freeswitch.org> wrote:
> Ciao!
> I am looking for an Italian set of sound files for FreeSWITCH. Does anyone
> know where those might be or even if they exist?
> Thanks,
> Michael
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ljjimenez at gmail.com  Tue Oct 25 21:50:02 2011
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Tue, 25 Oct 2011 17:50:02 +0000
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
Message-ID: <543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>

Hi Anthony, which db engine do you prefer, this as recommendation for us, for better performance with FS?

Thanks in advance
Luis Jim?nez

Luis Jimenez

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Tue, 25 Oct 2011 10:00:29 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] High load on database server


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


From edpimentl at gmail.com  Tue Oct 25 22:19:14 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 25 Oct 2011 14:19:14 -0400
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <4EA5DA64.1060503@communicatefreely.net>
References: <4EA5DA64.1060503@communicatefreely.net>
Message-ID: <CANMSB1ZH5wzfPbtTKbEeQ-MJSQ6KH=TVsN6dcb02GJRHi72x_Q@mail.gmail.com>

Use Memcache
The recommended approach for using memcached and MySQL is:
Write scale the database by sharding. Partition data across multiple
servers so more data can be written in parallel. This avoids a single
server becoming the bottleneck.
Front MySQL with a memcached farm to scale reads. Applications access
memcached first for data and if the data is not in memcached then the
application tries the database. This removes a great deal of the load
on a database so it can continue to perform it's transactional duties
for writes. In this architecture the database is still the system of
record for the true value of data.
Use MySQL replication for reliability and read query scaling. There's
an effective limit to the number of slaves that can be supported so
just adding slaves won't work as scaling strategy for larger sites.

Using this approach you get scalable reads and writes along with high
availability.

Given that MySQL has a cache, why is memcached needed at all?
The MySQL cache is associated with just one instance. This limits the
cache to the maximum address of one server. If your system is larger
than the memory for one server then using the MySQL cache won't work.
And if the same object is read from another instance its not cached.
The query cache invalidates on writes. You build up all that cache and
it goes away when someone writes to it. Your cache may not be much of
a cache at all depending on usage patterns.
The query cache is row based. Memcached can cache any type of data you
want and it isn't limited to caching database rows. Memcached can
cache complex complex objects that are directly usable without a join.


From cliff at develix.com  Tue Oct 25 22:32:10 2011
From: cliff at develix.com (Cliff Wells)
Date: Tue, 25 Oct 2011 11:32:10 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
	<543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>
Message-ID: <1319567530.2234.16.camel@portable-evil>

On Tue, 2011-10-25 at 17:50 +0000, Luis Jimenez wrote:
> Hi Anthony, which db engine do you prefer, this as recommendation for us, for better performance with FS?

We use PostgreSQL 8.4.  Our system currently processes around 350
concurrent calls during peak hours and PostgreSQL barely notices (every
call accesses the database to get the dialplan for around 5000 numbers).
We also access the same database for CDR logging and
application-specific data, so the number of concurrent database
connections is actually somewhat higher than that.

That being said, unixODBC 2.2 will often segfault under that load, so be
wary of that (it appears reliable up to about 300 connections, seems to
reliably fall over at 330).  We're in the process of testing 2.3.0 which
is reportedly better in this regard.

Anyway, I don't think choice of database is the issue here.  I don't
care for MySQL, but I don't think it should be any sort of bottleneck
for your application.  If you have poorly written or unindexed
queries/joins, they will be slow on whatever database you throw at it.

Also, does MySQL run on the same system as FreeSWITCH?  Have you checked
for basic issues such as memory pressure (or the inverse, failing to
tune MySQL to utilize available resources), slow disk throughput, etc?
Questions about database performance have no "one answer fits all"
solution, so more data would need to be gathered than you've provided
here.

Regards,
Cliff



From Hector.Geraldino at ip-soft.net  Tue Oct 25 23:19:10 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 25 Oct 2011 15:19:10 -0400
Subject: [Freeswitch-users] event socket
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695E4BA4D0CC@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
	<CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CD24@BARXB0003.UnifyBusiness.local>
	<6A6B4C284AD15042B429EB9D904544AD02249DB62D@NY1-EXMB-01.ip-soft.net>
	<78990CE7CC964442A7C2CA5F4689695E4BA4D0CC@BARXB0003.UnifyBusiness.local>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02249DB697@NY1-EXMB-01.ip-soft.net>

I just figured out something when reading your code.

By looking at your dialplan it looks like you're willing to take control over the incoming calls in your application by using the 'socket outbound' connection mode. In this case your application MUST act as a server and listen for socket connections on the designated port (you need a Server Socket class or its equivalent in .NET).

However according to your dialplan it looks like you're trying to bind the incoming call back to freeswitch, using the default port that FS uses for inbound connections (8021) - please correct me if I'm wrong. If you actually want to give the control of the incoming call to your application, you should use a different port, or change the FS default port on the configuration.  I don't know if the eslConnection class in .NET allows you to listen for incoming socket connection, but it looks to me more like a socket client.

Probably you should double check the Mod_event_socket documentation on the wiki to get more info about the .NET client library, and have a better understanding of the inbound/outbound connection modes. In the outbound mode you need to listen for new connections from your application, and you configure this mode by modifying the dialplan in the way you've done. In the inbound mode you can have a regular dialplan, and receive events for the calls that are handled by freeswitch dialplans.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 11:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

I'm receiving "+OK" from .Execute(). I'm not receiving a "CHANNEL_DATA" event.

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino
Sent: 25 October 2011 16:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

What Type is returned by the .Execute() method call?

Maybe you should create a new variable and assign the returned value to it, so you can inspect the result of the method call, something like:
                EslMessage response = eslConnection.Execute("answer", null, CallID);
String isOk = response.getHeaderValue("Reply-Text");
so you can inspect the response returned by FreeSwitch.

BTW, this is mostly Java code so I'm not sure that these Types/properties are available on the C# assemblies.

And finally, try to perform this operation after receiving the CHANNEL_DATA event instead of CHANNEL_STATE.


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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 10:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Hi,

CallID is a variable I defined. I'm assigning the call Unique-ID to this variable as follows,

                string CallID = eventHandle.GetHeader("Unique-ID");
My full code is at the bottom of this email,

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You want the Unique-ID variable of the incoming message, not CallID.

Hynek
On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

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 Hynek Cihlar
Sent: 25 October 2011 15:23

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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<http://www.unifybusiness.co.uk>

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 Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek
On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021<http://127.0.0.1:8021> async full"/>
                    </condition>
                </extension>

Thank you,

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 Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From chrisg.lists at gmail.com  Wed Oct 26 00:50:22 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Tue, 25 Oct 2011 22:50:22 +0200
Subject: [Freeswitch-users] Manipulate the SDP origin field
Message-ID: <CAMKRqpgdUVyfzp0+KA=hidZBi7n7K_gCW=hfanScB+kMoNrQTQ@mail.gmail.com>

Hi All,

I have an issue whereby my cluster is in a mixed nat and not-nat
environment. I would like to manipulate the o= field on a per call
basis.

In short I would like to achieve the below two scenarios, based on the dialplan
Internal offer:
Owner/Creator, Session Id (o): root 4267 4267 IN IP4 192.168.1.1

External offer:
Owner/Creator, Session Id (o): root 4267 4267 IN IP4 208.73.210.48



I have tried the below, with no success to the origin field.

dialplan.xml
<action application="bridge"
data="{sip_from_uri=sip:${caller_id_number}@208.73.210.48}${lcr_auto_route}"/>

vars.xml
<X-PRE-PROCESS cmd="set" data="bind_server_ip=208.73.210.48"/>

internal.xml
<param name="ext-rtp-ip" value="$${local_ip_v4}"/>
 <param name="ext-sip-ip" value="$${local_ip_v4}"/>


external.xml
<param name="ext-rtp-ip" value="208.73.210.48"/>
<param name="ext-sip-ip" value="208.73.210.48"/>

Thanks in advance for any assistance,
Chris G


From mgg at giagnocavo.net  Wed Oct 26 01:55:13 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 25 Oct 2011 17:55:13 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <28590.1315944949@ccs.covici.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
	<CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
	<30786.1315879273@ccs.covici.com>
	<CAAtQP54za1xEuT9HzT-sWOjbzzuKcV-FiotHta_ZXEw7y9vFgQ@mail.gmail.com>
	<28590.1315944949@ccs.covici.com>
Message-ID: <03351FCC6082174C8534AB714B8258A5E4D7CF83@mse17be1.mse17.exchange.ms>

Hey there,

	If anyone has problems with mod_managed, please email me directly. I'm finally maintaining it again.

	I just uploaded a patch:
http://jira.freeswitch.org/secure/attachment/14903/mono_update.patch

	This integrates the 2.8 mono patch as well as fixes a configuration issue when starting mono. This should solve the issue of having to use the mod directory for the current directory, or having it not find libc, etc. 

	I've tried it on Ubuntu 11 and SL 6.1 with Mono 2.10. I'm installing Ubuntu 10 LTS to try it out there.

Thanks,
Michael
	
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Tuesday, September 13, 2011 2:16 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] problem using mod_managed under linux

OK, it was actually in mono that the error occurred, but last couple of times, it did work, but I didn't change very much.  I hope it get fixed soon.  The end of the stack was libc.so which has always been suspect for me.

Thanks again for your response.



R H <buscom123+fs at gmail.com> wrote:

> Hey Covici,
> 
> I currently have a DLL that uses 3 IAppPlugin interfaces, 2 IApiPlugin 
> interfaces, and 2 ILoadNotificationPlugin interfaces. I build the dll 
> on a separate machine but using this system I am able to deploy the 
> dll to a running FreeSwitch and have it unload/reload the dll on the 
> fly without any problems.Its been working great for a week or two. A 
> few things you may want to keep in mind:
> 
> 1) DLL's deployed in the managed directory are not exactly going to 
> work well if they have dependencies on each-other as you never know 
> which one will get loaded first. (Well, if you only drop them in 1 at 
> a time it would probably work)
> 
> 2) Any crash in FreeSwitch I have seen is because of the above issue, 
> or because of some uncaught exception thrown from my code. If you look 
> at your FreeSwitch logs or start FreeSwitch by hand and watch the cli 
> you should see the exception that killed it.
> 
> When I run FreeSwitch as a service even my script in /etc/init.d/ has 
> to cd into the mod directory and then start the server. Hope this helps!
> 
> Ryan
> 
> 
> On Mon, Sep 12, 2011 at 8:01 PM, <covici at ccs.covici.com> wrote:
> 
> > Well, I spoke too soon.  It works, but if I make a change and try to 
> > reload the .dll, it kills fs altogether -- makes it not too usable.
> >
> > R H <buscom123+fs at gmail.com> wrote:
> >
> > > Hey Guys,
> > >
> > > I just noticed this thread in the email queue so this may be 
> > > coming too late. I have the same problem and sent an email about 
> > > it to this group
> > with
> > > the subject line: *Mod_Managed Mono is not following the system LD 
> > > Path.*
> > > *
> > > *
> > > That said,
> > > I have mod_managed working but it is a bit of a hack to do so. I 
> > > simply
> > CD
> > > into the mod directory and then start the server from there.
> > >
> > > EG.
> > >
> > > cd /usr/local/freeswitch/mod
> > > ../bin/freeswitch
> > >
> > > This makes that particular problem go away. I have tried changing 
> > > the LD Path, configured mono per the instructions on the wiki, 
> > > etc? None of that has worked for me but cd'ing into the mod 
> > > directory and then starting the server at least gets me running.
> > >
> > > This might be useful to everyone while we wait for this to be 
> > > resolved. I have yet to add this as a bug in Jira. I was 
> > > attempting to get some
> > response
> > > here before doing so.
> > >
> > > Ryan
> > >
> > > On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:
> > >
> > > > Thanks so much for working on this.
> > > >
> > > > Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> > > >
> > > > > Yea I have been meaning to get around to it. I'll put it on my 
> > > > > todo
> > and
> > > > see if I can figure it out. Sorry for the inconvenience.
> > > > >
> > > > > "covici at ccs.covici.com" <covici at ccs.covici.com> wrote:
> > > > >
> > > > >
> > > > > Any chance of you fixing the bug?  I think you must be right 
> > > > > because
> > in
> > > > > the server stack trace I see a lot of things involving serialization.
> > > > >
> > > > > Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> > > > >
> > > > > > It's probably more related to some 
> > > > > > cross-appdomain/serialization
> > stuff
> > > > that's specific to Mono. I wrote mod_managed against Mono 2.4 or 
> > > > so,
> > and I
> > > > think last time I ran it was 2.6, and only on CentOS 5. Probably
> > something
> > > > changed and is triggering a bug in mod_managed in the newer builds.
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at 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
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > sers
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users 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

--
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


FreeSWITCH-users mailing list
FreeSWITCH-users at 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 rsaavedra at ecogizmos.com  Wed Oct 26 06:16:26 2011
From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com)
Date: Tue, 25 Oct 2011 21:16:26 -0500
Subject: [Freeswitch-users] mod_rtmp login
Message-ID: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com>

Hello,

I'm trying to use mod_rtmp but everytime I tried on the console I see User
not authorized.
I was trying with user:1000 and password:1234.

In the file rtmp.conf.xml I have all by default:
 <param name="bind-address" value="0.0.0.0:1935" />
<param name="context" value="public" />
<param name="dialplan" value="XML" />
<param name="auth-calls" value="true" />
<param name="buffer-len" value="50" />
<param name="chunksize" value="512" />

Best regards,

Ricardo Saavedra



From jack at livecall.com  Wed Oct 26 07:17:37 2011
From: jack at livecall.com (Jack)
Date: Tue, 25 Oct 2011 20:17:37 -0700
Subject: [Freeswitch-users] mod_rtmp login
In-Reply-To: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com>
References: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com>
Message-ID: <4EA77BD1.7080302@livecall.com>

Ricardo,
If your 1000 is default it is probably set up with
<variable name="*user_context*"value="*default*"/>


if you set the param in rtmp.conf.xml to match the context  and then do 
a reloadxml or restart freeswitch it may authorize for you.
<param name="context" value="default" />

Hope that helps!
Jack



On 10/25/2011 7:16 PM, rsaavedra at ecogizmos.com wrote:
> Hello,
>
> I'm trying to use mod_rtmp but everytime I tried on the console I see User
> not authorized.
> I was trying with user:1000 and password:1234.
>
> In the file rtmp.conf.xml I have all by default:
>   <param name="bind-address" value="0.0.0.0:1935" />
> <param name="context" value="public" />
> <param name="dialplan" value="XML" />
> <param name="auth-calls" value="true" />
> <param name="buffer-len" value="50" />
> <param name="chunksize" value="512" />
>
> Best regards,
>
> Ricardo Saavedra
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fcintrono at gmail.com  Wed Oct 26 07:22:43 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Tue, 25 Oct 2011 22:22:43 -0500
Subject: [Freeswitch-users] How to know who is calling me...
In-Reply-To: <CAKzWOxVXBj5MLtFd0DvwGj9rnU-Pceg9D836iyi9VWwM80Xt1Q@mail.gmail.com>
References: <CANtw_ZPhmgMvaPUnLuBSnMPywmYctn1oz=Vn7HuEFjU0fyQDxw@mail.gmail.com>
	<CAKzWOxX2XUAqFEEFyY_640K4cHRO9osjzwsmBu37ZT9_76UXYQ@mail.gmail.com>
	<CANtw_ZNYh67=od5SUkUvXbzQmrYTydsxGNdY_cXnWmJnKgWJSQ@mail.gmail.com>
	<CAKzWOxVXBj5MLtFd0DvwGj9rnU-Pceg9D836iyi9VWwM80Xt1Q@mail.gmail.com>
Message-ID: <CANtw_ZN6+amSRdESWf8gFKbfU6tKLj8wS+XC5nxY-fhKo-MaMQ@mail.gmail.com>

OK, Yes both lines have caller ID. If I understood SPA8800 should pass
caller ID to fs. Well let me make a test, I read in fs book how to print
debug data on fs_cli.

thank you for your help.

On Tue, Oct 25, 2011 at 11:24 AM, Michael Collins <msc at freeswitch.org>wrote:

> If your phone provider offers analog caller ID then hopefully the SPA8800
> can detect it. If so, it should pass the caller ID information over to
> FreeSWITCH.
>
> -MC
>
>
> On Tue, Oct 25, 2011 at 7:43 AM, Francisco Javier Cintr?n Olgu?n <
> fcintrono at gmail.com> wrote:
>
>> Mmm, Let me make some calls to my phone provider.
>> I bet we certainly have this service.
>>
>> Have a nice day
>>
>>
>> On Mon, Oct 24, 2011 at 7:24 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Does Mexico have caller ID service on analog phone lines?
>>> -MC
>>>
>>> On Mon, Oct 24, 2011 at 2:28 PM, Francisco Javier Cintr?n Olgu?n <
>>> fcintrono at gmail.com> wrote:
>>>
>>>> Hello from Guadalajara, M?xico. :)
>>>>
>>>> For the last month I have been reading the fs book and  playing with
>>>> fs.
>>>> I am pretty conviced that It would be perfect as my soho office.
>>>>
>>>> I have this configuration:
>>>>
>>>> 4 normal internal phones.
>>>> 2 pots lines.
>>>>
>>>> to interface freeswitch with these devices I purchased a CISCO SPA8800.
>>>> This unit has 8 ports:
>>>>
>>>> 4 for normal phones.
>>>> 4 for pots lines.
>>>>
>>>> I also purchased a cheap IP phone: linksys SPA921.
>>>>
>>>> With the help of fs wiki site I got how to configure the SPA8800.
>>>>
>>>> The only thing I don't know is how to show in the LCD display of the
>>>> SPA921 who is calling or the number from which is the call.
>>>>
>>>>
>>>> Thanks in advance for your kind help.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From liuyp2 at asiainfo-linkage.com  Wed Oct 26 07:41:33 2011
From: liuyp2 at asiainfo-linkage.com (liuyp2)
Date: Wed, 26 Oct 2011 11:41:33 +0800
Subject: [Freeswitch-users] event socket
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>,
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>,
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>,
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>,
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>,
	<8256893087499069959@unknownmsgid>
Message-ID: <201110261141318128062@asiainfo-linkage.com>

if (eventHandle.getType() == "CHANNEL_CALLSTATE" && eventHandle.GetHeader("Channel-Call-State") == "RINGING")
                {
                    eslConnection.Execute("answer", null, CallID);
                }


Best Regards!
2011-10-26  11:35:19



youpingl
  



???????? Nuwan Wijerathne
?????????? 2011-10-25 22:36:30
???????? FreeSWITCH Users Help
?????? 
?????? Re: [Freeswitch-users] event socket

            Sorry still no progress, code as follows
 
               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:23
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket
 
You also have to provide the UUID of the channel, so freeswitch knows which one to answer. 

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
I made the changes below, still no progress,
 
                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }
 
Thank you,
 
Nuwan Wijerathne
 
Software Engineering Department
 
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
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket
 
Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek


On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
Hi,
 
Thanks for the reply. Yes my application is listening on the port and I??m receiving events. However when I try to execute ??answer?? application it??s not working. I??m trying to execute ??answer?? application when I receive ??CHANNEL_CREATE?? event on inbound calls. Please see code below. Could you Please advise
 
using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;
 
namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;
        
        private Thread ThreadReader;
        
        private ESLconnection eslConnection;
 
        private const int CONNECTED = 1;
 
        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";
 
        
        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }
 
 
        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);
 
            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");
 
                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }
 
 
 
        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");
 
            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }
 
            while (true)
            {
                eventHandle = eslConnection.RecvEvent();
 
                string CallID = eventHandle.GetHeader("Unique-ID");
 
                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}
 
Dialplan entry as follows,
 
                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021 async full"/>
                    </condition>
                </extension>
 
Thank you,
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket
 
Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples
 
-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
Hi,
 
I??m trying to build an IVR application around freeswtich. I??m using freeswitch event socket in C#. I??m intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute ??answer?? application through event socket. 
 
Eg ?C I??m sending the command as below and it??s not working.
 
ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv(??event plain ALL??);
 
eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader(??Unique-ID??);
 
eslConnection.Execute(??answer??, string.Empty, UniqueCallID);
 
 
I believe I??m doing something wrong as I can??t find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)
 
Thank you,
 


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


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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From gcd at i.ph  Wed Oct 26 08:11:34 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 26 Oct 2011 12:11:34 +0800
Subject: [Freeswitch-users] How to know who is calling me...
In-Reply-To: <CANtw_ZN6+amSRdESWf8gFKbfU6tKLj8wS+XC5nxY-fhKo-MaMQ@mail.gmail.com>
References: <CANtw_ZPhmgMvaPUnLuBSnMPywmYctn1oz=Vn7HuEFjU0fyQDxw@mail.gmail.com>
	<CAKzWOxX2XUAqFEEFyY_640K4cHRO9osjzwsmBu37ZT9_76UXYQ@mail.gmail.com>
	<CANtw_ZNYh67=od5SUkUvXbzQmrYTydsxGNdY_cXnWmJnKgWJSQ@mail.gmail.com>
	<CAKzWOxVXBj5MLtFd0DvwGj9rnU-Pceg9D836iyi9VWwM80Xt1Q@mail.gmail.com>
	<CANtw_ZN6+amSRdESWf8gFKbfU6tKLj8wS+XC5nxY-fhKo-MaMQ@mail.gmail.com>
Message-ID: <CAHKqMb56H=Ri1y-Yciv1+fG8R7cxZphoUafuPL4Cvf5ec3b2xA@mail.gmail.com>

the SPA8800 should be configured w/c protocol to pass the CallerID. check ur
manual.
-nandy

On Wed, Oct 26, 2011 at 11:22 AM, Francisco Javier Cintr?n Olgu?n <
fcintrono at gmail.com> wrote:

> OK, Yes both lines have caller ID. If I understood SPA8800 should pass
> caller ID to fs. Well let me make a test, I read in fs book how to print
> debug data on fs_cli.
>
> thank you for your help.
>
> On Tue, Oct 25, 2011 at 11:24 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> If your phone provider offers analog caller ID then hopefully the SPA8800
>> can detect it. If so, it should pass the caller ID information over to
>> FreeSWITCH.
>>
>> -MC
>>
>>
>> On Tue, Oct 25, 2011 at 7:43 AM, Francisco Javier Cintr?n Olgu?n <
>> fcintrono at gmail.com> wrote:
>>
>>> Mmm, Let me make some calls to my phone provider.
>>> I bet we certainly have this service.
>>>
>>> Have a nice day
>>>
>>>
>>> On Mon, Oct 24, 2011 at 7:24 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Does Mexico have caller ID service on analog phone lines?
>>>> -MC
>>>>
>>>> On Mon, Oct 24, 2011 at 2:28 PM, Francisco Javier Cintr?n Olgu?n <
>>>> fcintrono at gmail.com> wrote:
>>>>
>>>>> Hello from Guadalajara, M?xico. :)
>>>>>
>>>>> For the last month I have been reading the fs book and  playing with
>>>>> fs.
>>>>> I am pretty conviced that It would be perfect as my soho office.
>>>>>
>>>>> I have this configuration:
>>>>>
>>>>> 4 normal internal phones.
>>>>> 2 pots lines.
>>>>>
>>>>> to interface freeswitch with these devices I purchased a CISCO
>>>>> SPA8800.
>>>>> This unit has 8 ports:
>>>>>
>>>>> 4 for normal phones.
>>>>> 4 for pots lines.
>>>>>
>>>>> I also purchased a cheap IP phone: linksys SPA921.
>>>>>
>>>>> With the help of fs wiki site I got how to configure the SPA8800.
>>>>>
>>>>> The only thing I don't know is how to show in the LCD display of the
>>>>> SPA921 who is calling or the number from which is the call.
>>>>>
>>>>>
>>>>> Thanks in advance for your kind help.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hynek.cihlar at gmail.com  Wed Oct 26 08:49:18 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Wed, 26 Oct 2011 06:49:18 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1319567530.2234.16.camel@portable-evil>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
	<543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>
	<1319567530.2234.16.camel@portable-evil>
Message-ID: <CAJh8XdXVUkfMH5Ud9NaHCXV18UyBf3RgwBtgWxVihz6ASmFccQ@mail.gmail.com>

>
> That being said, unixODBC 2.2 will often segfault under that load, so be
> wary of that (it appears reliable up to about 300 connections, seems to
> reliably fall over at 330).  We're in the process of testing 2.3.0 which
> is reportedly better in this regard.
>

This is unfortunate! I was planning to cluster several freeswitch instances
through the DB server. But if the unixODBC crashes under high load, it is a
major issue in this regard as there are not many options left to implement
the cluster.

Hynek
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From gmaruzz at gmail.com  Wed Oct 26 13:50:28 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 26 Oct 2011 11:50:28 +0200
Subject: [Freeswitch-users] Italian voice files
In-Reply-To: <4EA7D42C.70301@pmsitalia.it>
References: <CAKzWOxVaATe54kxm1frs_LDB6wy8pQZsVWC=gBZw8s1nLQ2doA@mail.gmail.com>
	<CALXCt0rdpgoMjBddz40cxQ5bbn=_CE4cinsaXmSwagWyfh=3pQ@mail.gmail.com>
	<4EA7D42C.70301@pmsitalia.it>
Message-ID: <CALXCt0p9eG=D9syA5anfToz6QyYxmRjbKc9NN-wUDVMbjsdNgg@mail.gmail.com>

Hello Iesoua,

I believe it will cost around 2500$ (roughly) to have it
professionally produced, recorded, equalized, etc from GM Voices.

That's an order of magnitude, you better check for actual prices (I'm
sure they'll apply a discount if they know it will be put in standard
FS distribution, because it will bring them more business for
additional customized prompts in the future).

Would be great if you can then put it in public domain and add it to
the standard FreeSWITCH distribution.

Michael Collins (mcollins, mercutioviz on irc) can give you all
details about how to interact with GM Voices (or your preferred prompt
provider) and how to put the collection into FreeSWITCH distribution.

-giovanni


On Wed, Oct 26, 2011 at 11:34 AM, iesoua - pms italia
<iesoua.semeco at pmsitalia.it> wrote:
> Hello Giovanni,
>
> we're interesting in buying the italian voice pack for freeswitch, could you
> please let me know the price for the package?
>
>
> Thank you
>
> --
> Iesoua Semeco
> ICT Department
> @mail: iesoua.semeco at pmsitalia.it
> Mobile: +39 333 2525230
> Office: +39 02 87241129
> ____________________________
> PMS Italia srl
> Viale Enrico Forlanini, 15
> 20134 MILANO MI - Italy
> Tel: +39 02 872411
> Fax: +39 02 45479990
> ____________________________
>
>
>
> Il 25/10/2011 19:26, Giovanni Maruzzelli ha scritto:
>>
>> mmmmmh, I don't think they exists.
>>
>> I had almost convinced a customer to order the entire set, but then he
>> ended up using completely custom messages (and a small number of
>> them)... too bad :(.
>>
>> As soon as I'll have some customer paying for it I'll order the set
>> from GM voices...
>>
>> Let me know how it goes.
>>
>> _giovanni
>>
>> On Tue, Oct 25, 2011 at 6:23 PM, Michael Collins<msc at freeswitch.org>
>> ?wrote:
>>>
>>> Ciao!
>>> I am looking for an Italian set of sound files for FreeSWITCH. Does
>>> anyone
>>> know where those might be or even if they exist?
>>> Thanks,
>>> Michael
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 freeswitch-list at puzzled.xs4all.nl  Wed Oct 26 14:40:04 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 26 Oct 2011 12:40:04 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAJh8XdXVUkfMH5Ud9NaHCXV18UyBf3RgwBtgWxVihz6ASmFccQ@mail.gmail.com>
References: <4EA5DA64.1060503@communicatefreely.net>	<1319494283.4593.160.camel@portable-evil>	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>	<543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>	<1319567530.2234.16.camel@portable-evil>
	<CAJh8XdXVUkfMH5Ud9NaHCXV18UyBf3RgwBtgWxVihz6ASmFccQ@mail.gmail.com>
Message-ID: <4EA7E384.9000103@puzzled.xs4all.nl>

On 10/26/2011 06:49 AM, Hynek Cihlar wrote:
>     That being said, unixODBC 2.2 will often segfault under that load, so be
>     wary of that (it appears reliable up to about 300 connections, seems to
>     reliably fall over at 330).  We're in the process of testing 2.3.0 which
>     is reportedly better in this regard.
>
>
> This is unfortunate! I was planning to cluster several freeswitch
> instances through the DB server. But if the unixODBC crashes under high
> load, it is a major issue in this regard as there are not many options
> left to implement the cluster.

If you use Red Hat RHEL or CentOS and there is a crash with stock 
unixODBC (2.2) then a bug should be reported at bugzilla.redhat.com. In 
my experience such bugs do get fixed.

Regards,
Patrick


From faisal.rehman22 at hotmail.com  Wed Oct 26 15:59:57 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Wed, 26 Oct 2011 16:59:57 +0500
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>,
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
Message-ID: <BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>


Hi Waseem,
Thanks for your help, I saw the scripts that were on the link that you provided to me but actually the problem is that I don't want to involve the database though it is a bit difficult task but according to my requirement I just want to read the whole Master.csv file & if the ASR or ACD is low on any server, it should send me email, so my question is how can I skip the database from it to interact directly with the Master.csv file.

Thanks in Advance!!
Faisal Rehman

From: wasim at convergence.pk
Date: Tue, 25 Oct 2011 07:32:35 +0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring

Faisal:
Load up the CDR into a db (MySQL or otherwise) ...
For ACD you'll want a SQL query akinSELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec, SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE disposition=1;

For ASR you'll need another querySELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;ASR is calls_answered/calls_totalThere is sample (read: poor) code in http://jira.freeswitch.org/browse/FS-1981

which a google for "freeswitch ACD ASR" would have helped you get to in the first place.
-wasim
On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:







Hi Everyone,
I want to write a script that reads the Master.csv & generate an alert if the ASR & ACD is low from the desired value. Anybody has written such script, please help me write it or provide any sample.



Thanks

Faisal Rehman 		 	   		  



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-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | peace be upon you ...



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From michaelt at callall.co.za  Wed Oct 26 16:00:42 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Wed, 26 Oct 2011 14:00:42 +0200
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
Message-ID: <CAFmdODi1wWM-xtJ+Qag3V7Tjjc1G_vSXwXrm8k3S59su=CXfBg@mail.gmail.com>

Hi,

 When initiating a call from Freeswitch to a provider, Freeswitch is putting
'0'/ zero in the media port.  This looks like a bug to me.:

   m=audio 0 RTP/AVP 96 101
Media Description, name and address (m): audio 0 RTP/AVP 96 101


 Txt based SIP trace attached (not sure how to scrub pcap files of IPs).

Thanks,

Michael
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-------------- next part --------------
INVITE sip:0836805611 at sip.host.com:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 1 INVITE
Max-Forwards: 70
Contact: 27816011943 <sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266>
Expires: 240
User-Agent: iTelMobileDialer-2.10
Content-Length: 229
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS
Content-Type: application/sdp

v=0
o=- 22469580 22469580 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=iTelMobileDialer-2.10
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 27611 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:40
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -0500
Content-Length: 0

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>;tag=mUpmv98p6vyXj
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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
Proxy-Authenticate: Digest realm="sip.host.com", nonce="e992fce4-ffc4-11e0-b0a5-851ab5c8ac90", algorithm=MD5, qop="auth"
Content-Length: 0

ACK sip:0836805611 at sip.host.com:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;rport;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>;tag=mUpmv98p6vyXj
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: iTelMobileDialer-2.10
Content-Length: 0

INVITE sip:0836805611 at sip.host.com:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 2 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="27816011943", realm="sip.host.com", nonce="e992fce4-ffc4-11e0-b0a5-851ab5c8ac90", uri="sip:0836805611 at ZZZ.ZZZ.ZZZ.ZZZ:290", response="035851f4bfd2cc2567477538f178b946"
Contact: 27816011943 <sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266>
Expires: 240
User-Agent: iTelMobileDialer-2.10
Content-Length: 229
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS
Content-Type: application/sdp

v=0
o=- 22469580 22469580 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=iTelMobileDialer-2.10
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 27611 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:40
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 2 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -0500
Content-Length: 0

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>;tag=N4FDy4St35mge
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 2 INVITE
Contact: <sip:0836805611 at YYY.YYY.YYY.YYY:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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-Type: application/sdp
Content-Disposition: session
Content-Length: 202
Remote-Party-ID: "0836805611" <sip:0836805611 at YYY.YYY.YYY.YYY>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 4288871279 4288871280 IN IP4 YYY.YYY.YYY.YYY
s=FreeSWITCH
c=IN IP4 YYY.YYY.YYY.YYY
t=0 0
m=audio 0 RTP/AVP 96 101
a=rtpmap:96 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 200 OK
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>;tag=N4FDy4St35mge
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 2 INVITE
Contact: <sip:0836805611 at YYY.YYY.YYY.YYY:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -0500
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-Type: application/sdp
Content-Disposition: session
Content-Length: 202
Remote-Party-ID: "Outbound Call" <sip:0836805611 at YYY.YYY.YYY.YYY>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 4288871279 4288871280 IN IP4 YYY.YYY.YYY.YYY
s=FreeSWITCH
c=IN IP4 YYY.YYY.YYY.YYY
t=0 0
m=audio 0 RTP/AVP 96 101
a=rtpmap:96 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
ACK sip:0836805611 at YYY.YYY.YYY.YYY:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;rport;branch=z9hG4bK450900027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:0836805611 at sip.host.com:5060>;tag=N4FDy4St35mge
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 2 ACK
Contact: <sip:27816011943 at sip.host.com:5060>
Max-Forwards: 70
User-Agent: iTelMobileDialer-2.10
Content-Length: 0

REGISTER sip:sip.host.com:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;rport;branch=z9hG4bK2100027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:27816011943 at sip.host.com:5060>
Call-ID: 27816011943945901363 at XXX.XXX.XXX.XXX
CSeq: 17 REGISTER
Contact: 27816011943 <sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266>;expires=60
Authorization: Digest username="27816011943", realm="sip.host.com", nonce="e992fce4-ffc4-11e0-b0a5-851ab5c8ac90", uri="sip:ZZZ.ZZZ.ZZZ.ZZZ:290",algorithm=MD5,response="601346080001c830ccd4c3d1ca67e86a"
Max-Forwards: 70
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;rport=266;branch=z9hG4bK2100027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:27816011943 at sip.host.com:5060>;tag=XctKcX0pcD75a
Call-ID: 27816011943945901363 at XXX.XXX.XXX.XXX
CSeq: 17 REGISTER
Contact: <sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266>;expires=60
Date: Wed, 26 Oct 2011 11:23:53 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK7t9DrDeBgyj3K
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=yNKcerHt9NXrp
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
CSeq: 19478052 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

BYE sip:0836805611 at YYY.YYY.YYY.YYY:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:3961;branch=z9hG4bK21400
From:;tag=N4FDy4St35mge
To:Call-ID: 1920643962 at XXX.XXX.XXX.XXX
CSeq: 3 BYE
Max-Forwards: 70

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK21000
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=111000
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478052 NOTIFY
Content-Length: 0

BYE sip:0836805611 at YYY.YYY.YYY.YYY:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:3961;branch=z9hG4bK21400
From:;tag=N4FDy4St35mge
To:Call-ID: 1920643962 at XXX.XXX.XXX.XXX
CSeq: 3 BYE
Max-Forwards: 70

REGISTER sip:sip.host.com:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;rport;branch=z9hG4bK2101027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:27816011943 at sip.host.com:5060>
Call-ID: 27816011943945901363 at XXX.XXX.XXX.XXX
CSeq: 18 REGISTER
Contact: 27816011943 <sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266>;expires=60
Authorization: Digest username="27816011943", realm="sip.host.com", nonce="e992fce4-ffc4-11e0-b0a5-851ab5c8ac90", uri="sip:ZZZ.ZZZ.ZZZ.ZZZ:290",algorithm=MD5,response="601346080001c830ccd4c3d1ca67e86a"
Max-Forwards: 70
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:266;rport=266;branch=z9hG4bK2101027816011943
From: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
To: <sip:27816011943 at sip.host.com:5060>;tag=Zyc5FK2X6yKBj
Call-ID: 27816011943945901363 at XXX.XXX.XXX.XXX
CSeq: 18 REGISTER
Contact: <sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266>;expires=60
Date: Wed, 26 Oct 2011 11:23:54 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK7t9DrDeBgyj3K
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=yNKcerHt9NXrp
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
CSeq: 19478052 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK8326S8yeD78NF
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=075XHeK1379XD
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
CSeq: 19478053 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK21010
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=111000
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478052 NOTIFY
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK21010
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=111000
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478053 NOTIFY
Content-Length: 0

BYE sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK9cvZU3FjagZ8a
Max-Forwards: 70
From: <sip:0836805611 at sip.host.com:5060>;tag=N4FDy4St35mge
To: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
CSeq: 19478053 BYE
Contact: <sip:0836805611 at YYY.YYY.YYY.YYY:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK9cvZU3FjagZ8a
From: <sip:0836805611 at sip.host.com:5060>;tag=N4FDy4St35mge
To: 27816011943 <sip:27816011943 at sip.host.com:5060>;tag=11100027816011943
Call-ID: 278160119431920643962 at XXX.XXX.XXX.XXX
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478053 BYE
Content-Length: 0

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK8326S8yeD78NF
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=075XHeK1379XD
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
CSeq: 19478053 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK9cvZU3FjagZ8a
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478053 NOTIFY
Content-Length: 0

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK7t9DrDeBgyj3K
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=yNKcerHt9NXrp
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
CSeq: 19478052 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK9cvZU3FjagZ8a
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478052 NOTIFY
Content-Length: 0

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK8326S8yeD78NF
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=075XHeK1379XD
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
CSeq: 19478053 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK9cvZU3FjagZ8a
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478053 NOTIFY
Content-Length: 0

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK7t9DrDeBgyj3K
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=yNKcerHt9NXrp
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
CSeq: 19478052 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK9cvZU3FjagZ8a
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4089a6f-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478052 NOTIFY
Content-Length: 0

NOTIFY sip:27816011943 at ZZZ.ZZZ.ZZZ.ZZZ:266 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK8326S8yeD78NF
Max-Forwards: 70
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=075XHeK1379XD
To: <sip:27816011943 at YYY.YYY.YYY.YYY>
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
CSeq: 19478053 NOTIFY
Contact: <sip:mod_sofia at YYY.YYY.YYY.YYY:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-10d2e80 2011-09-08 22-35-20 -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: 69

Messages-Waiting: no
Message-Account: sip:27816011943 at YYY.YYY.YYY.YYY

SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;rport;branch=z9hG4bK9cvZU3FjagZ8a
From: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
To: <sip:27816011943 at YYY.YYY.YYY.YYY>;tag=N4FDy4St35mge
Call-ID: d4a1cdaf-7a67-122f-0391-0014221086ec
Contact: 27816011943 <sip:27816011943 at XXX.XXX.XXX.XXX:3961>
Max-Forwards: 70
CSeq: 19478053 NOTIFY
Content-Length: 0


From wasim at convergence.pk  Wed Oct 26 16:48:27 2011
From: wasim at convergence.pk (Wasim Baig)
Date: Wed, 26 Oct 2011 17:48:27 +0500
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
Message-ID: <CABHdwCpsg92crnZMay6XpB7K3WK8jOA_8-D9RxoZS-kda60aZQ@mail.gmail.com>

Ofcourse, but you have to parse the CSV or XML file, which is beyond the
scope of this mailing list.

-wasim

On Wed, Oct 26, 2011 at 16:59, Faisal Rehman <faisal.rehman22 at hotmail.com>wrote:

>  *Hi Waseem,*
>
> Thanks for your help, I saw the scripts that were on the link that you
> provided to me but actually the problem is that I don't want to involve the
> database though it is a bit difficult task but according to my requirement I
> just want to read the whole Master.csv file & if the ASR or ACD is low on
> any server, it should send me email, so my question is how can I skip the
> database from it to interact directly with the Master.csv file.
>
>
> Thanks in Advance!!
>
> *Faisal Rehman
> *
> ------------------------------
> From: wasim at convergence.pk
> Date: Tue, 25 Oct 2011 07:32:35 +0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
>
> Faisal:
>
> Load up the CDR into a db (MySQL or otherwise) ...
>
> For ACD you'll want a SQL query akin
>
>    - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
>    SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
>    disposition=1;
>
> For ASR you'll need another query
>
>    - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
>    - ASR is calls_answered/calls_total
>
> There is sample (read: poor) code in
> http://jira.freeswitch.org/browse/FS-1981
> which a google for "freeswitch ACD ASR" would have helped you get to in the
> first place.
>
> -wasim
>
> On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com>wrote:
>
>  Hi Everyone,
>
> I want to write a script that reads the Master.csv & generate an alert if
> the ASR & ACD is low from the desired value. Anybody has written such
> script, please help me write it or provide any sample.
>
>
> Thanks
>
>
> Faisal Rehman
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From eagle.antonio at gmail.com  Wed Oct 26 17:44:28 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 26 Oct 2011 13:44:28 +0000
Subject: [Freeswitch-users] LCR and Continue On Fail
Message-ID: <CAF+3j-sOex2MtO4oZ=bBZPFDKBryMpgRNGBAvYtRFRgiiYmPdg@mail.gmail.com>

Hello Guys.

Logs & Dialplan
http://pastebin.freeswitch.org/17613

I'm currently trying to connect LCR with the Fall back provided with
Continue On Fail.
I could be misunderstading the use of continue on fail i read several posts
on the mailing and in theory this should work :)

Anyway my objective is that
If failure _cause == UNALLOCATED_NUMBER
   continue using the lcr reach
else:
   hangup the call.

So this is me or could be a bug ??

Regards & Thanks for the help :)
Antonio Teixeira
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From moises.silva at gmail.com  Wed Oct 26 18:06:01 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 26 Oct 2011 10:06:01 -0400
Subject: [Freeswitch-users] mod_unicall
In-Reply-To: <1318205874399-6875427.post@n2.nabble.com>
References: <1318205874399-6875427.post@n2.nabble.com>
Message-ID: <CAA4nhyAkeg11Bg7gSLZTqpDSTuQVbh5V2rFLr5B9xsysk0GP1w@mail.gmail.com>

On Sun, Oct 9, 2011 at 8:17 PM, mazilo
<Nabble_01394 at slickdeals.endjunk.com>wrote:

> What is the current status of mod_unicall, i.e. usable, compilable, etc.?
>
>
Considering the last commit to that module was more than a year ago and I
have not heard anyone using it, most likely its development is stalled. I
have not seen Steve Underwood around lately.

On the other hand, what signaling were you looking to use with mod_unicall?
Unicall was mostly widely used for MFCR2, which is now integrated in FreeTDM
(mod_freetdm) and works pretty well. FreeTDM also supports Analog, PRI/BRI,
SS7 and soon GSM.

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

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



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

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

                VegaStream is now part of
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                Ask us about both Gateway
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From moises.silva at gmail.com  Wed Oct 26 18:11:54 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 26 Oct 2011 10:11:54 -0400
Subject: [Freeswitch-users] conference with pin code and esl
In-Reply-To: <FE619B1E9F9144D48432D48BFF3FA30B@e1705>
References: <8E20DFE641574516B5C6891BA3181F1B@e1705>
	<CAKzWOxUO7vNZezeWhdBn5QmAtgzOH4v8bN-b1kb2oeQRHJ_Rug@mail.gmail.com>
	<FD6ED18F454249B497CE21B79AA6A44D@e1705>
	<CAKzWOxVq+-d+N_aZ_98nkR74JG_26ECp+wZ8RbFyU1LhVb17OA@mail.gmail.com>
	<FE619B1E9F9144D48432D48BFF3FA30B@e1705>
Message-ID: <CAA4nhyAQ_XZ+T4VsBv-xEoHGYWJMdcTtOzQypLWP7M1k2=p4wg@mail.gmail.com>

On Mon, Oct 17, 2011 at 9:59 PM, Madovsky <infos at madovsky.org> wrote:

> **
> ok deosn't matter.
> thanks anyway
>
>

Moderator PIN support was already merged:
http://jira.freeswitch.org/browse/FS-3493

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

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



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

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>

                VegaStream is now part of
Sangoma!<http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acquires_All_Key_Assets_of_VegaStream.html>

                Ask us about both Gateway
Appliances<http://sangoma.com/products/media_gateways/vega.html>
 and Internal Gateways<http://sangoma.com/products/software_products/netborder_suite/netborder_express.html>
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From NuwanW at unifybusiness.co.uk  Wed Oct 26 18:50:42 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Wed, 26 Oct 2011 14:50:42 +0000
Subject: [Freeswitch-users] event socket
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02249DB697@NY1-EXMB-01.ip-soft.net>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
	<CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CD24@BARXB0003.UnifyBusiness.local>
	<6A6B4C284AD15042B429EB9D904544AD02249DB62D@NY1-EXMB-01.ip-soft.net>
	<78990CE7CC964442A7C2CA5F4689695E4BA4D0CC@BARXB0003.UnifyBusiness.local>
	<6A6B4C284AD15042B429EB9D904544AD02249DB697@NY1-EXMB-01.ip-soft.net>
Message-ID: <78990CE7CC964442A7C2CA5F4689695E55498132@BARXB0003.UnifyBusiness.local>

Thank you for the reply. Could you please answer couple of questions,


1.       Can an external application in inbound mode can answer inbound calls. For example extension 200 dials 9. Is it possible to answer this call by an external application (inbound mode) ? If it is possible what is the dial plan entry for this ? I read event socket inbound and outbound mode and I'm under the impression that I can't use an external application in inbound mode to answer calls.

2.       Could you please direct me to a source where I can find all supported features/functionality in event socket inbound and outbound mode.


Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino
Sent: 25 October 2011 20:19
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

I just figured out something when reading your code.

By looking at your dialplan it looks like you're willing to take control over the incoming calls in your application by using the 'socket outbound' connection mode. In this case your application MUST act as a server and listen for socket connections on the designated port (you need a Server Socket class or its equivalent in .NET).

However according to your dialplan it looks like you're trying to bind the incoming call back to freeswitch, using the default port that FS uses for inbound connections (8021) - please correct me if I'm wrong. If you actually want to give the control of the incoming call to your application, you should use a different port, or change the FS default port on the configuration.  I don't know if the eslConnection class in .NET allows you to listen for incoming socket connection, but it looks to me more like a socket client.

Probably you should double check the Mod_event_socket documentation on the wiki to get more info about the .NET client library, and have a better understanding of the inbound/outbound connection modes. In the outbound mode you need to listen for new connections from your application, and you configure this mode by modifying the dialplan in the way you've done. In the inbound mode you can have a regular dialplan, and receive events for the calls that are handled by freeswitch dialplans.


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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 11:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

I'm receiving "+OK" from .Execute(). I'm not receiving a "CHANNEL_DATA" event.

Nuwan Wijerathne

Software Engineering Department

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

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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Hector Geraldino
Sent: 25 October 2011 16:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

What Type is returned by the .Execute() method call?

Maybe you should create a new variable and assign the returned value to it, so you can inspect the result of the method call, something like:
                EslMessage response = eslConnection.Execute("answer", null, CallID);
String isOk = response.getHeaderValue("Reply-Text");
so you can inspect the response returned by FreeSwitch.

BTW, this is mostly Java code so I'm not sure that these Types/properties are available on the C# assemblies.

And finally, try to perform this operation after receiving the CHANNEL_DATA event instead of CHANNEL_STATE.


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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 10:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Hi,

CallID is a variable I defined. I'm assigning the call Unique-ID to this variable as follows,

                string CallID = eventHandle.GetHeader("Unique-ID");
My full code is at the bottom of this email,

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You want the Unique-ID variable of the incoming message, not CallID.

Hynek
On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

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 Hynek Cihlar
Sent: 25 October 2011 15:23

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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<http://www.unifybusiness.co.uk>

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 Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek
On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021<http://127.0.0.1:8021> async full"/>
                    </condition>
                </extension>

Thank you,

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 Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk<mailto:NuwanW at unifybusiness.co.uk>> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From kris at kriskinc.com  Wed Oct 26 19:58:09 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 26 Oct 2011 11:58:09 -0400
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <CAFmdODi1wWM-xtJ+Qag3V7Tjjc1G_vSXwXrm8k3S59su=CXfBg@mail.gmail.com>
References: <CAFmdODi1wWM-xtJ+Qag3V7Tjjc1G_vSXwXrm8k3S59su=CXfBg@mail.gmail.com>
Message-ID: <CAKDfjgck+A_bQfHtHo2p=tUo6wwO-S5kir6u+mnh86CeSCa0ww@mail.gmail.com>

Can you post a console log with siptrace enabled:

sofia global siptrace on

It looks like FreeSWITCH is getting that SDP from somewhere else.  0
as the port number, G729 as dynamic payload type 96...  It's
completely messed up and I've never seen FreeSWITCH generate an SDP
like this.

On Wed, Oct 26, 2011 at 8:00 AM, Michael Toop <michaelt at callall.co.za> wrote:
> Hi,
> ?When initiating a call from Freeswitch to a provider, Freeswitch is putting
> '0'/ zero in the media port. ?This looks like a bug to me.:
> ? ?m=audio 0 RTP/AVP 96 101
> Media Description, name and address (m): audio 0 RTP/AVP 96 101
>
> ?Txt based SIP trace attached (not sure how to scrub pcap files of IPs).
> Thanks,
> Michael
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Wed Oct 26 20:12:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Oct 2011 11:12:52 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <4EA7E384.9000103@puzzled.xs4all.nl>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
	<543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>
	<1319567530.2234.16.camel@portable-evil>
	<CAJh8XdXVUkfMH5Ud9NaHCXV18UyBf3RgwBtgWxVihz6ASmFccQ@mail.gmail.com>
	<4EA7E384.9000103@puzzled.xs4all.nl>
Message-ID: <CAKbxfG_X_-eQ8jNand_E9yZfroPDMaiotNk9Vr4MB+rPAf=oyg@mail.gmail.com>

if you have older unixodbc edit /etc/odbcinst.ini and add the param
Threading=0 to each db entry.
I like postgres better but that's mostly a personal preference from watching
both of them grow up over the years.

The only thing that crashes in ODBC is bad drivers typically.  With mysql
you often need to specify the "threadsafe" one by editing odbc.ini and
adding in the _r to the name of the mysql driver .so file.

In general postgres to me appears to care more about multithreaded
environments and with the above threading=0 change works pretty well.





On Wed, Oct 26, 2011 at 5:40 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 10/26/2011 06:49 AM, Hynek Cihlar wrote:
> >     That being said, unixODBC 2.2 will often segfault under that load, so
> be
> >     wary of that (it appears reliable up to about 300 connections, seems
> to
> >     reliably fall over at 330).  We're in the process of testing 2.3.0
> which
> >     is reportedly better in this regard.
> >
> >
> > This is unfortunate! I was planning to cluster several freeswitch
> > instances through the DB server. But if the unixODBC crashes under high
> > load, it is a major issue in this regard as there are not many options
> > left to implement the cluster.
>
> If you use Red Hat RHEL or CentOS and there is a crash with stock
> unixODBC (2.2) then a bug should be reported at bugzilla.redhat.com. In
> my experience such bugs do get fixed.
>
> Regards,
> Patrick
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 rsaavedra at ecogizmos.com  Wed Oct 26 21:04:47 2011
From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com)
Date: Wed, 26 Oct 2011 12:04:47 -0500
Subject: [Freeswitch-users] mod_rtmp login
In-Reply-To: <4EA77BD1.7080302@livecall.com>
References: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com>
	<4EA77BD1.7080302@livecall.com>
Message-ID: <1f1fc7140c13295f010812918da115bf.squirrel@emailmg.globat.com>

Thank you,

so in the flex client I will write the user like 1000 at 1.2.3.4

Best regards,

Ricardo

> Ricardo,
> If your 1000 is default it is probably set up with
> <variable name="*user_context*"value="*default*"/>
>
>
> if you set the param in rtmp.conf.xml to match the context  and then do
> a reloadxml or restart freeswitch it may authorize for you.
> <param name="context" value="default" />
>
> Hope that helps!
> Jack
>
>
>
> On 10/25/2011 7:16 PM, rsaavedra at ecogizmos.com wrote:
>> Hello,
>>
>> I'm trying to use mod_rtmp but everytime I tried on the console I see
>> User
>> not authorized.
>> I was trying with user:1000 and password:1234.
>>
>> In the file rtmp.conf.xml I have all by default:
>>   <param name="bind-address" value="0.0.0.0:1935" />
>> <param name="context" value="public" />
>> <param name="dialplan" value="XML" />
>> <param name="auth-calls" value="true" />
>> <param name="buffer-len" value="50" />
>> <param name="chunksize" value="512" />
>>
>> Best regards,
>>
>> Ricardo Saavedra
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From david.villasmil.work at gmail.com  Wed Oct 26 21:48:57 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 26 Oct 2011 19:48:57 +0200
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
Message-ID: <CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>

Hello,

That's a faily simple bash script.

"cat" the log file to an awk script like so;

cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{
FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT:
"CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR;
if(ASR<100){ system("echo send email");} }'

the "system" is where you put the mail command

also, bear in mind that the master can get VERY large if you don't rotate
it.

david

On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman
<faisal.rehman22 at hotmail.com>wrote:

>  *Hi Waseem,*
>
> Thanks for your help, I saw the scripts that were on the link that you
> provided to me but actually the problem is that I don't want to involve the
> database though it is a bit difficult task but according to my requirement I
> just want to read the whole Master.csv file & if the ASR or ACD is low on
> any server, it should send me email, so my question is how can I skip the
> database from it to interact directly with the Master.csv file.
>
>
> Thanks in Advance!!
>
> *Faisal Rehman
> *
> ------------------------------
> From: wasim at convergence.pk
> Date: Tue, 25 Oct 2011 07:32:35 +0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
>
> Faisal:
>
> Load up the CDR into a db (MySQL or otherwise) ...
>
> For ACD you'll want a SQL query akin
>
>    - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
>    SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
>    disposition=1;
>
> For ASR you'll need another query
>
>    - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
>    - ASR is calls_answered/calls_total
>
> There is sample (read: poor) code in
> http://jira.freeswitch.org/browse/FS-1981
> which a google for "freeswitch ACD ASR" would have helped you get to in the
> first place.
>
> -wasim
>
> On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com>wrote:
>
>  Hi Everyone,
>
> I want to write a script that reads the Master.csv & generate an alert if
> the ASR & ACD is low from the desired value. Anybody has written such
> script, please help me write it or provide any sample.
>
>
> Thanks
>
>
> Faisal Rehman
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble_01394 at slickdeals.endjunk.com  Wed Oct 26 23:18:44 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Wed, 26 Oct 2011 12:18:44 -0700 (PDT)
Subject: [Freeswitch-users] mod_unicall
In-Reply-To: <CAA4nhyAkeg11Bg7gSLZTqpDSTuQVbh5V2rFLr5B9xsysk0GP1w@mail.gmail.com>
References: <1318205874399-6875427.post@n2.nabble.com>
	<CAA4nhyAkeg11Bg7gSLZTqpDSTuQVbh5V2rFLr5B9xsysk0GP1w@mail.gmail.com>
Message-ID: <1319656724818-6933928.post@n2.nabble.com>

ATM, I have no use of this mod_unicall. What prompted me to post asking such
a question was the corss compilation on mod_unicall spits out error
messages. If this module is no longer maintained, then I will just ignore
and move on.

Thanks for your response.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-unicall-tp6875427p6933928.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jack at livecall.com  Wed Oct 26 23:48:36 2011
From: jack at livecall.com (Jack)
Date: Wed, 26 Oct 2011 12:48:36 -0700
Subject: [Freeswitch-users] mod_rtmp login
In-Reply-To: <1f1fc7140c13295f010812918da115bf.squirrel@emailmg.globat.com>
References: <0e6a9dcb6a5cb714cd6a130ec6497142.squirrel@emailmg.globat.com>
	<4EA77BD1.7080302@livecall.com>
	<1f1fc7140c13295f010812918da115bf.squirrel@emailmg.globat.com>
Message-ID: <4EA86414.3040903@livecall.com>

That would be correct if your FreeSwitch server IP address is 1.2.3.4


On 10/26/2011 10:04 AM, rsaavedra at ecogizmos.com wrote:
> Thank you,
>
> so in the flex client I will write the user like 1000 at 1.2.3.4
>
> Best regards,
>
> Ricardo
>
>> Ricardo,
>> If your 1000 is default it is probably set up with
>> <variable name="*user_context*"value="*default*"/>
>>
>>
>> if you set the param in rtmp.conf.xml to match the context  and then do
>> a reloadxml or restart freeswitch it may authorize for you.
>> <param name="context" value="default" />
>>
>> Hope that helps!
>> Jack
>>
>>
>>
>> On 10/25/2011 7:16 PM, rsaavedra at ecogizmos.com wrote:
>>> Hello,
>>>
>>> I'm trying to use mod_rtmp but everytime I tried on the console I see
>>> User
>>> not authorized.
>>> I was trying with user:1000 and password:1234.
>>>
>>> In the file rtmp.conf.xml I have all by default:
>>>    <param name="bind-address" value="0.0.0.0:1935" />
>>> <param name="context" value="public" />
>>> <param name="dialplan" value="XML" />
>>> <param name="auth-calls" value="true" />
>>> <param name="buffer-len" value="50" />
>>> <param name="chunksize" value="512" />
>>>
>>> Best regards,
>>>
>>> Ricardo Saavedra
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ip-soft.net  Wed Oct 26 23:49:16 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Wed, 26 Oct 2011 15:49:16 -0400
Subject: [Freeswitch-users] event socket
In-Reply-To: <78990CE7CC964442A7C2CA5F4689695E55498132@BARXB0003.UnifyBusiness.local>
References: <78990CE7CC964442A7C2CA5F4689695E4BA4C70E@BARXB0003.UnifyBusiness.local>
	<CAKzWOxWkmav+-UHh+8LAXP1HTrNe8g450V63-fW0kYVhHWCCQA@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CC42@BARXB0003.UnifyBusiness.local>
	<CAJh8XdUD+6iqVgfKtpiNuJrx4vP-g_JCSgrNnQqhtF-xzhk0kw@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCA6@BARXB0003.UnifyBusiness.local>
	<8256893087499069959@unknownmsgid>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CCDE@BARXB0003.UnifyBusiness.local>
	<CAJh8XdVoa0cEc08hQiTo3TE3T_LFOWuEE29JQv=BYu_vPHjc=g@mail.gmail.com>
	<78990CE7CC964442A7C2CA5F4689695E4BA4CD24@BARXB0003.UnifyBusiness.local>
	<6A6B4C284AD15042B429EB9D904544AD02249DB62D@NY1-EXMB-01.ip-soft.net>
	<78990CE7CC964442A7C2CA5F4689695E4BA4D0CC@BARXB0003.UnifyBusiness.local>
	<6A6B4C284AD15042B429EB9D904544AD02249DB697@NY1-EXMB-01.ip-soft.net>
	<78990CE7CC964442A7C2CA5F4689695E55498132@BARXB0003.UnifyBusiness.local>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02249DB7C7@NY1-EXMB-01.ip-soft.net>

Hi,

The best source is the Mod_event_socket page on the wiki.

        http://wiki.freeswitch.org/wiki/Event_Socket
        http://wiki.freeswitch.org/wiki/Event_Socket_Outbound


You can answer a call by using a socket inbound connection to freeswitch, and you don't need to write any special instructions on the dialplan to do so, just listen for events and send api commands. But in my particular experience, if you want to control an incoming call (a call that hits an extension managed by freeswitch) you'll be better served by using a socket outbound connection. Off course this is from a design perspective, not because there are limitations when using an inbound connection.

Think about a socket outbound connection as a substitution of part of your dialplan. You take complete control over the incoming call, and you can answer it, play messages, capture dtmf tones, bridge it to another extension, etc. in a more cleaner way.

The socket inbound connection in my experience is better suited for situations where you need to create (and control) a new call from freeswitch to another extension, either controlled by freeswitch or another platform through a gateway. A simple use case scenario: an IVR app that bridges an incoming call to another leg:

1) PSTN -> FS -> Outbound socket app (C#)
2) C# app answers the call, plays a message and wait for DTMF tones
3) User presses 1 on the dialpad, an event is sent to the C# app
4) The C# app creates a new inbound connection to FS, and using this new connection:
        a) Originates a call to another extension (leg B)
        b) bridge the two calls, using uuid_bridge


In resume what I'm trying to say is: use outbound mode to control incoming calls as a way to replace/enhance the actions you can execute in a diaplan, and use inbound mode when you want freeswitch to make new calls, check status, etc.

Again, you can use the inbound mode to perform the same tasks as with outbound, this is just a guideline based on my experience.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Wednesday, October 26, 2011 10:51 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Thank you for the reply. Could you please answer couple of questions,

1. Can an external application in inbound mode can answer inbound calls. For example extension 200 dials 9. Is it possible to answer this call by an external application (inbound mode) ? If it is possible what is the dial plan entry for this ? I read event socket inbound and outbound mode and I'm under the impression that I can't use an external application in inbound mode to answer calls.
2. Could you please direct me to a source where I can find all supported features/functionality in event socket inbound and outbound mode.

Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino
Sent: 25 October 2011 20:19
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

I just figured out something when reading your code.

By looking at your dialplan it looks like you're willing to take control over the incoming calls in your application by using the 'socket outbound' connection mode. In this case your application MUST act as a server and listen for socket connections on the designated port (you need a Server Socket class or its equivalent in .NET).

However according to your dialplan it looks like you're trying to bind the incoming call back to freeswitch, using the default port that FS uses for inbound connections (8021) - please correct me if I'm wrong. If you actually want to give the control of the incoming call to your application, you should use a different port, or change the FS default port on the configuration.  I don't know if the eslConnection class in .NET allows you to listen for incoming socket connection, but it looks to me more like a socket client.

Probably you should double check the Mod_event_socket documentation on the wiki to get more info about the .NET client library, and have a better understanding of the inbound/outbound connection modes. In the outbound mode you need to listen for new connections from your application, and you configure this mode by modifying the dialplan in the way you've done. In the inbound mode you can have a regular dialplan, and receive events for the calls that are handled by freeswitch dialplans.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 11:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

I'm receiving "+OK" from .Execute(). I'm not receiving a "CHANNEL_DATA" event.

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino
Sent: 25 October 2011 16:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

What Type is returned by the .Execute() method call?

Maybe you should create a new variable and assign the returned value to it, so you can inspect the result of the method call, something like:
                EslMessage response = eslConnection.Execute("answer", null, CallID);
       String isOk = response.getHeaderValue("Reply-Text");
so you can inspect the response returned by FreeSwitch.

BTW, this is mostly Java code so I'm not sure that these Types/properties are available on the C# assemblies.

And finally, try to perform this operation after receiving the CHANNEL_DATA event instead of CHANNEL_STATE.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 10:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Hi,

CallID is a variable I defined. I'm assigning the call Unique-ID to this variable as follows,

                string CallID = eventHandle.GetHeader("Unique-ID");
My full code is at the bottom of this email,

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You want the Unique-ID variable of the incoming message, not CallID.

Hynek
On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:23

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek
On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021 async full"/>
                    </condition>
                </extension>

Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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From jack at livecall.com  Wed Oct 26 23:57:53 2011
From: jack at livecall.com (Jack)
Date: Wed, 26 Oct 2011 12:57:53 -0700
Subject: [Freeswitch-users] Web conference interface
Message-ID: <4EA86641.2000705@livecall.com>

I would like to display the conference participants on my web page along 
with my flex phone like the freeswitch conference page.
Can anyone point me at the right components to make it happen?
Thanks
Jack Loranger



From NuwanW at unifybusiness.co.uk  Thu Oct 27 00:14:23 2011
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Wed, 26 Oct 2011 20:14:23 +0000
Subject: [Freeswitch-users] event socket
Message-ID: <mDHfqlQ1L6Nh@4kLkkBMV>

Thanks a lot. This helps and solve my issues. Thank you again.

--- original message ---
From: "Hector Geraldino" <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] event socket
Date: 26th October 2011
Time: 8:55:00 pm


Hi,

The best source is the Mod_event_socket page on the wiki.

        http://wiki.freeswitch.org/wiki/Event_Socket
        http://wiki.freeswitch.org/wiki/Event_Socket_Outbound


You can answer a call by using a socket inbound connection to freeswitch, and you don't need to write any special instructions on the dialplan to do so, just listen for events and send api commands. But in my particular experience, if you want to control an incoming call (a call that hits an extension managed by freeswitch) you'll be better served by using a socket outbound connection. Off course this is from a design perspective, not because there are limitations when using an inbound connection.

Think about a socket outbound connection as a substitution of part of your dialplan. You take complete control over the incoming call, and you can answer it, play messages, capture dtmf tones, bridge it to another extension, etc. in a more cleaner way.

The socket inbound connection in my experience is better suited for situations where you need to create (and control) a new call from freeswitch to another extension, either controlled by freeswitch or another platform through a gateway. A simple use case scenario: an IVR app that bridges an incoming call to another leg:

1) PSTN -> FS -> Outbound socket app (C#)
2) C# app answers the call, plays a message and wait for DTMF tones
3) User presses 1 on the dialpad, an event is sent to the C# app
4) The C# app creates a new inbound connection to FS, and using this new connection:
        a) Originates a call to another extension (leg B)
        b) bridge the two calls, using uuid_bridge


In resume what I'm trying to say is: use outbound mode to control incoming calls as a way to replace/enhance the actions you can execute in a diaplan, and use inbound mode when you want freeswitch to make new calls, check status, etc.

Again, you can use the inbound mode to perform the same tasks as with outbound, this is just a guideline based on my experience.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Wednesday, October 26, 2011 10:51 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Thank you for the reply. Could you please answer couple of questions,

1. Can an external application in inbound mode can answer inbound calls. For example extension 200 dials 9. Is it possible to answer this call by an external application (inbound mode) ? If it is possible what is the dial plan entry for this ? I read event socket inbound and outbound mode and I'm under the impression that I can't use an external application in inbound mode to answer calls.
2. Could you please direct me to a source where I can find all supported features/functionality in event socket inbound and outbound mode.

Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino
Sent: 25 October 2011 20:19
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

I just figured out something when reading your code.

By looking at your dialplan it looks like you're willing to take control over the incoming calls in your application by using the 'socket outbound' connection mode. In this case your application MUST act as a server and listen for socket connections on the designated port (you need a Server Socket class or its equivalent in .NET).

However according to your dialplan it looks like you're trying to bind the incoming call back to freeswitch, using the default port that FS uses for inbound connections (8021) - please correct me if I'm wrong. If you actually want to give the control of the incoming call to your application, you should use a different port, or change the FS default port on the configuration.  I don't know if the eslConnection class in .NET allows you to listen for incoming socket connection, but it looks to me more like a socket client.

Probably you should double check the Mod_event_socket documentation on the wiki to get more info about the .NET client library, and have a better understanding of the inbound/outbound connection modes. In the outbound mode you need to listen for new connections from your application, and you configure this mode by modifying the dialplan in the way you've done. In the inbound mode you can have a regular dialplan, and receive events for the calls that are handled by freeswitch dialplans.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 11:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

I'm receiving "+OK" from .Execute(). I'm not receiving a "CHANNEL_DATA" event.

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino
Sent: 25 October 2011 16:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

What Type is returned by the .Execute() method call?

Maybe you should create a new variable and assign the returned value to it, so you can inspect the result of the method call, something like:
                EslMessage response = eslConnection.Execute("answer", null, CallID);
       String isOk = response.getHeaderValue("Reply-Text");
so you can inspect the response returned by FreeSwitch.

BTW, this is mostly Java code so I'm not sure that these Types/properties are available on the C# assemblies.

And finally, try to perform this operation after receiving the CHANNEL_DATA event instead of CHANNEL_STATE.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nuwan Wijerathne
Sent: Tuesday, October 25, 2011 10:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Hi,

CallID is a variable I defined. I'm assigning the call Unique-ID to this variable as follows,

                string CallID = eventHandle.GetHeader("Unique-ID");
My full code is at the bottom of this email,

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You want the Unique-ID variable of the incoming message, not CallID.

Hynek
On Tue, Oct 25, 2011 at 4:36 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
            Sorry still no progress, code as follows

               if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, CallID);
                }
Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 15:23

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

You also have to provide the UUID of the channel, so freeswitch knows which one to answer.

Sent from my mobile device

On Oct 25, 2011, at 16:19, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
I made the changes below, still no progress,

                 if (eventHandle.getType() == "CHANNEL_STATE" && eventHandle.GetHeader("Answer-State") == "ringing")
                {
                    eslConnection.Execute("answer", null, null);
                }

Thank you,

Nuwan Wijerathne

Software Engineering Department

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

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hynek Cihlar
Sent: 25 October 2011 14:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Answer on event CHANNEL_CALLSTATE and channel call state RINGING.

Hynek
On Tue, Oct 25, 2011 at 3:49 PM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
Hi,

Thanks for the reply. Yes my application is listening on the port and I'm receiving events. However when I try to execute "answer" application it's not working. I'm trying to execute "answer" application when I receive "CHANNEL_CREATE" event on inbound calls. Please see code below. Could you Please advise

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading;
using System.IO;

namespace ESLTest
{
    public class Test
    {
        private Dictionary<string, Call> CallsTable;

        private Thread ThreadReader;

        private ESLconnection eslConnection;

        private const int CONNECTED = 1;

        private const string server = "localhost";
        private const string port = "8021";
        private const string password = "ClueCon";


        public Test()
        {
            CallsTable = new Dictionary<string, Call>();
            Connect();
        }


        private void Connect()
        {
            eslConnection = new ESLconnection(server, port, password);

            if (eslConnection.Connected().Equals( CONNECTED))
            {
                Console.WriteLine("Conneected to FreeSwitch");

                ThreadReader = new Thread(new ThreadStart(Read));
                ThreadReader.Start();
            }
            else
            {
                Console.WriteLine("Conneection Failure");
            }
        }



        private void Read()
        {
            ESLevent eventHandle = eslConnection.SendRecv("event plain ALL");

            if (eventHandle == null)
            {
                Console.WriteLine("Error subscribing to all events");
                return;
            }
            else
           {
                Console.WriteLine("Subscribed to all events");
            }

            while (true)
            {
                eventHandle = eslConnection.RecvEvent();

                string CallID = eventHandle.GetHeader("Unique-ID");

                if (eventHandle.getType() == "CHANNEL_CREATE")
                {
                    //Inbound call
                    if (eventHandle.GetHeader("Call-Direction") == "inbound")
                    {
                        CallsTable.Add(CallID, new Call(CallID));
                        eslConnection.Execute("answer", null, null);
                        Console.WriteLine(string.Format("{0} : Inbound CallID : {1}", eventHandle.getType(), CallID));
                    }
                }
           }
    }
}

Dialplan entry as follows,

                <extension name="intercept">
                     <condition field="destination_number" expression="^9$">
                           <action application="socket" data="127.0.0.1:8021 async full"/>
                    </condition>
                </extension>

Thank you,

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 24 October 2011 18:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] event socket

Please confirm: are you sending the call to the "socket" application in your dialplan? If so, then your ESL app needs to be listening on the TCP port specified. See the examples here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Examples

-MC
On Mon, Oct 24, 2011 at 8:50 AM, Nuwan Wijerathne <NuwanW at unifybusiness.co.uk> wrote:
Hi,

I'm trying to build an IVR application around freeswtich. I'm using freeswitch event socket in C#. I'm intercepting inbound calls in the dial plan and sending them to my application via outbound socket. Could someone please explain how to execute "answer" application through event socket.

Eg - I'm sending the command as below and it's not working.

ESLConnection eslConnection = new ESLConnection(server, port, password)
ESLEvent eventHandle = eslConnection.SendRecv("event plain ALL");

eventHandle = eslConnection.RecvEvent();
string uniqueCallID = evenHandle.GetHeader("Unique-ID");

eslConnection.Execute("answer", string.Empty, UniqueCallID);


I believe I'm doing something wrong as I can't find required information. Could someone please let me know where to find more information about freeswitch event socket in .Net. (Google not giving me required information)

Thank you,



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From michaelt at callall.co.za  Thu Oct 27 02:48:06 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Thu, 27 Oct 2011 00:48:06 +0200
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <CAKDfjgck+A_bQfHtHo2p=tUo6wwO-S5kir6u+mnh86CeSCa0ww@mail.gmail.com>
References: <CAFmdODi1wWM-xtJ+Qag3V7Tjjc1G_vSXwXrm8k3S59su=CXfBg@mail.gmail.com>
	<CAKDfjgck+A_bQfHtHo2p=tUo6wwO-S5kir6u+mnh86CeSCa0ww@mail.gmail.com>
Message-ID: <CAFmdODjcZ8tw1js5gZbSDYz1jyQfYsoRu+j64hS2z8bdubdX0A@mail.gmail.com>

Hi Kristian,

 Thanks for your input, please find trace attached as requested with sofia
SIP trace on.

 I have gone through it again & don't see the SDP arriving wrong.

 Let me know what else I can do to assist.

Kind Regards,

Michael

On Wed, Oct 26, 2011 at 5:58 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Can you post a console log with siptrace enabled:
>
> sofia global siptrace on
>
> It looks like FreeSWITCH is getting that SDP from somewhere else.  0
> as the port number, G729 as dynamic payload type 96...  It's
> completely messed up and I've never seen FreeSWITCH generate an SDP
> like this.
>
> On Wed, Oct 26, 2011 at 8:00 AM, Michael Toop <michaelt at callall.co.za>
> wrote:
> > Hi,
> >  When initiating a call from Freeswitch to a provider, Freeswitch is
> putting
> > '0'/ zero in the media port.  This looks like a bug to me.:
> >    m=audio 0 RTP/AVP 96 101
> > Media Description, name and address (m): audio 0 RTP/AVP 96 101
> >
> >  Txt based SIP trace attached (not sure how to scrub pcap files of IPs).
> > Thanks,
> > Michael
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From etiennedip at gmail.com  Wed Oct 26 14:20:05 2011
From: etiennedip at gmail.com (ET1.D)
Date: Wed, 26 Oct 2011 03:20:05 -0700 (PDT)
Subject: [Freeswitch-users] Get Fax2Mail to work on ubuntu11.04
Message-ID: <1319624405574-6932179.post@n2.nabble.com>

Good day,

Currently I am trying to set up a Fax to mail solution, I am using the
method that is posted on the wiki with the python script, currently the
message I am getting is 

2011-10-26 12:01:23.564498 [INFO] mod_dialplan_xml.c:336 Processing 1616
Etienne <2787xxxxxx>->9178 in context default
2011-10-26 12:01:23.564498 [ERR] switch_core_session.c:2062 Invalid
Application python
2011-10-26 12:01:23.564498 [NOTICE] switch_core_session.c:2063 Hangup
sofia/external/27878051616 at 172.16.1.2:5060 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2011-10-26 12:01:23.564498 [NOTICE] switch_core_session.c:1369 Session 14
(sofia/external/27878051616 at 172.16.1.2:5060) Ended

I have uncommented the mod_python module in modules.conf.xml and restarted
freeswitch, but I still get the same message. Can you please assist.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Get-Fax2Mail-to-work-on-ubuntu11-04-tp6932179p6932179.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From miha at softnet.si  Wed Oct 26 17:34:23 2011
From: miha at softnet.si (Miha Zoubek)
Date: Wed, 26 Oct 2011 15:34:23 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
Message-ID: <4EA80C5F.6020501@softnet.si>

Hi,

I am new Freeswitch user. I have read freeswitch book. I registered 
first few phones and make few inbound calls.
I am trying to get outgoing calls working but having no luck.
I have configured my gw and set up calling plan. When I try to dial 
number 051357952 I get this:

reeswitch at localhost.localdomain> 2011-10-26 15:27:25.919562 [NOTICE] 
switch_channel.c:669 New Channel sofia/internal/1030 at 172.31.1.157 
[f2132149-dc40-4601-bd34-c22e5b06b696]
2011-10-26 15:27:25.955550 [INFO] mod_dialplan_xml.c:418 Processing 
1030->9051357952 in context default
2011-10-26 15:27:25.960548 [NOTICE] switch_channel.c:669 New Channel 
sofia/external/051357952 [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
2011-10-26 15:27:26.095518 [NOTICE] sofia.c:4789 Hangup 
sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2011-10-26 15:27:26.098534 [INFO] mod_dptools.c:2355 Originate Failed.  
Cause: NO_ROUTE_DESTINATION
2011-10-26 15:27:26.098534 [NOTICE] mod_dptools.c:2418 Hangup 
sofia/internal/1030 at 172.31.1.157 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1182 Session 
49 (sofia/external/051357952) Ended
2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/external/051357952 [CS_DESTROY]
2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1182 Session 
48 (sofia/internal/1030 at 172.31.1.157) Ended
2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1030 at 172.31.1.157 [CS_DESTROY]

GW:

<include>
<gateway name="IP800">
<!-- Your username, which is not your DID -->
<param name="username" value="Demo2"/>
<param name="password" value="1234"/>
<param name="register" value="true"/>
<param name="realm" value="172.31.1.200"/>
</gateway>
</include>

This is my dial plan:

<include>
<extension name="Dial Out Custom Gateway">
<condition field="destination_number" expression="^9(\d{9})$">
<action application="bridge" data="sofia/gateway/IP800/$1"/>
</condition>
</extension>
</include>


sofia status

                      Name          
Type                                       Data      State
=================================================================================================
                  external       profile            
sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
     external::example.com       gateway                    
sip:joeuser at example.com      NOREG
           external::IP800       gateway                     
sip:Demo2 at 172.31.1.200      REGED
                  internal       profile            
sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
              172.31.1.157         
alias                                   internal      ALIASED
=================================================================================================
2 profiles 1 alias


Please help me why I am getting not route to destination.

Thank you very much for all your help!

BR,
Miha
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From nirc at fring.com  Wed Oct 26 18:30:06 2011
From: nirc at fring.com (Nir Cohen)
Date: Wed, 26 Oct 2011 16:30:06 +0200
Subject: [Freeswitch-users] Phantom Calls
Message-ID: <754CEFADFC65654BAC484896D31C2C030566AC0D76@exchangesrv2007.ONEFONE.MSDOMAIN>

Hello,

I'm new to freeswitch. Even though I've been using it for quite a while, I hadn't had the chance to really dig into the config/monitoring/debugging.

I have a freeswitch which runs flawlessly with a large number of calls but a small percentage of them do not disconnect at all until the freeswitch service is restarted.
I was wondering if there's some timeout which I need to define or if there's a way to see a direct reason for this to happen.


-          I know they are stuck because I did "show calls" and saw calls from different times during the time it was up (see below)

-          I removed the freeswitch from the DNS to make sure it's not receiving anymore calls.

-          I opened wireshark and saw that the same amount of calls are referred to as "session in progress" - that is, my fs sends a "Status: 183 session progress..." and the app server returns: (ICMP) "destination unreachable".

Thank you.










call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid
2011-10-09 15:10:08,1318165808,switch_ivr_signal_bridge,1001,1001,r0018095215036,sofia/external/1001 at 172.16.100.155,edac000a-bc5a-8642-aba8-ed3be16cd762,1001,fring,18095215036,sofia/external/18095215036,487fbf53-4d6c-134e-93bc-31cc3ee1baf8
2011-10-09 23:34:57,1318196097,switch_ivr_signal_bridge,1001,1001,r006498287601,sofia/external/1001 at 172.16.100.155,5d4b4b67-def4-2c44-addc-73f3366d6f83,1001,fring,6498287601,sofia/external/6498287601,6087cfe3-e9b6-de44-b38e-bc86d83ff95a
2011-10-09 23:38:30,1318196310,switch_ivr_signal_bridge,1001,1001,r0020472321728,sofia/external/1001 at 172.16.100.155,9913ff13-33ac-9d4b-b601-d761e49b81b6,1001,fring,20472321728,sofia/external/20472321728,954fff0a-4b7c-8444-8821-306dc6729e42
2011-10-09 23:41:09,1318196469,switch_ivr_signal_bridge,1001,1001,r0020108866522,sofia/external/1001 at 172.16.100.155,f6590d4f-bda0-2146-a309-f2e485ff231c,1001,fring,20108866522,sofia/external/20108866522,ba1f72d8-49fc-5145-8376-87e9f4a3b3b4
2011-10-10 02:16:36,1318205796,switch_ivr_signal_bridge,1001,1001,r0050687102330,sofia/external/1001 at 172.16.100.155,c48e48c1-590d-f84a-a34a-e2da40574d59,1001,fring,50687102330,sofia/external/50687102330,95c4c053-c925-5c42-9ac8-87cace55694b
2011-10-10 06:09:18,1318219758,switch_ivr_signal_bridge,1001,1001,r00919744973603,sofia/external/1001 at 172.16.100.155,04127975-c3f9-b949-907e-f915197bf4fc,1001,fring,919744973603,sofia/external/919744973603,f86302d2-e7ab-0940-b927-b673544c7241
2011-10-10 06:21:24,1318220484,switch_ivr_signal_bridge,1001,1001,r00917204674056,sofia/external/1001 at 172.16.100.155,c835ad85-9e04-b84d-993e-426261686cc7,1001,fring,917204674056,sofia/external/917204674056,1304eae6-4def-c743-9214-89991e6f3350
2011-10-10 06:53:04,1318222384,switch_ivr_signal_bridge,1001,1001,r00919715155054,sofia/external/1001 at 172.16.100.155,c4082f14-7cae-e846-9b05-65a519320619,1001,fring,919715155054,sofia/external/919715155054,16fa4e97-eb15-d24b-907d-f39a7741b493
2011-10-10 07:26:37,1318224397,switch_ivr_signal_bridge,1001,1001,r00919215900646,sofia/external/1001 at 172.16.100.155,40a79a56-cf31-824f-bd47-0006f4abb59b,1001,fring,919215900646,sofia/external/919215900646,f2a02bce-d25a-f448-af39-fa60f3509f51
2011-10-10 09:38:45,1318232325,switch_ivr_signal_bridge,1001,1001,r00919779210235,sofia/external/1001 at 172.16.100.155,3249aa91-d99f-8947-bcd4-749e199ecfbb,1001,fring,919779210235,sofia/external/919779210235,6b55ec96-6fe2-d047-a349-efd44df00c67
2011-10-10 13:22:42,1318245762,switch_ivr_signal_bridge,1001,1001,r00919312238801,sofia/external/1001 at 172.16.100.155,1226ffc9-f338-8b42-be8a-93868dc1ed0a,1001,fring,919312238801,sofia/external/919312238801,f63b997f-2371-2441-b0f3-cfcf0330c87a
2011-10-10 13:37:31,1318246651,switch_ivr_signal_bridge,1001,1001,r0091978999666,sofia/external/1001 at 172.16.100.155,3bf75841-ef52-8244-9c18-5ff783ad5bd4,1001,fring,91978999666,sofia/external/91978999666,8de43ba6-ff36-bf40-941f-a71d3574a95d
2011-10-10 14:15:44,1318248944,switch_ivr_signal_bridge,1001,1001,r00919855283870,sofia/external/1001 at 172.16.100.155,a1844ff2-c9f7-eb4b-a9ac-4e31d5c5a591,1001,fring,919855283870,sofia/external/919855283870,5c73749d-3f35-aa46-a13e-9c1b5d78c592
2011-10-10 14:56:30,1318251390,switch_ivr_signal_bridge,1001,1001,r00919891882161,sofia/external/1001 at 172.16.100.155,9b087d32-2a87-db4f-ac66-3bb91d7afdeb,1001,fring,919891882161,sofia/external/919891882161,0972c900-805d-f949-b47a-fb1398a5739d
2011-10-10 16:31:56,1318257116,switch_ivr_signal_bridge,1001,1001,r00919810399899,sofia/external/1001 at 172.16.100.155,c8b382a4-81ce-6740-bb5d-ebc9ab37d388,1001,fring,919810399899,sofia/external/919810399899,84313caf-fe39-5f45-bf4a-d1cc5e575eff
2011-10-10 16:47:27,1318258047,switch_ivr_signal_bridge,1001,1001,r0096597284481,sofia/external/1001 at 172.16.100.155,b8d93c17-97ee-074e-84c9-103e5259ddb1,1001,fring,96597284481,sofia/external/96597284481,1dee8a27-b3f1-e246-99f6-baca4cea0dd6
2011-10-10 20:11:30,1318270290,switch_ivr_signal_bridge,1001,1001,r00967771771821,sofia/external/1001 at 172.16.100.155,16462169-cb2f-6b42-bf01-64e43a8e4311,1001,fring,967771771821,sofia/external/967771771821,0a98e975-6bd1-4948-8a99-2b682935e893
2011-10-11 01:37:56,1318289876,switch_ivr_signal_bridge,1001,1001,r00584120208712,sofia/external/1001 at 172.16.100.155,7566612f-e7f5-a74a-848b-fa52db667bc1,1001,fring,584120208712,sofia/external/584120208712,bec24956-f8cc-9e46-a1b1-e5f75f827306
2011-10-11 02:23:53,1318292633,switch_ivr_signal_bridge,1001,1001,r00919815288834,sofia/external/1001 at 172.16.100.155,0c7c07a8-ed17-3e43-9d00-5928b998ac48,1001,fring,919815288834,sofia/external/919815288834,11c3f91e-4e84-7d4c-93d9-fbadbc0e0d69
2011-10-11 04:30:11,1318300211,switch_ivr_signal_bridge,1001,1001,r00919924392230,sofia/external/1001 at 172.16.100.155,63514e60-511f-5748-b1ed-6ebcc41d5878,1001,fring,919924392230,sofia/external/919924392230,b465f0f0-f0f5-fb4c-bb0f-93b23b59a4cf
2011-10-11 05:05:39,1318302339,switch_ivr_signal_bridge,1001,1001,r00919703570642,sofia/external/1001 at 172.16.100.155,d6181187-6d71-844a-a4cd-b8add2393c0b,1001,fring,919703570642,sofia/external/919703570642,ed78294e-33f5-7d4d-af4b-40e0efbc9d17
2011-10-11 07:24:57,1318310697,switch_ivr_signal_bridge,1001,1001,r00919845045625,sofia/external/1001 at 172.16.100.155,781823e0-1496-2f46-b6bc-a98cee121887,1001,fring,919845045625,sofia/external/919845045625,bd19f919-4b3c-3342-a23f-53088baedc8f
2011-10-11 07:55:46,1318312546,switch_ivr_signal_bridge,1001,1001,r0084090,sofia/external/1001 at 172.16.100.155,8edd5195-a573-4042-adf7-2c7d093b8178,1001,fring,84090,sofia/external/84090,7e83eb35-d155-384a-ab22-4545e0bb62bb
2011-10-11 10:11:50,1318320710,switch_ivr_signal_bridge,1001,1001,r009197845827258,sofia/external/1001 at 172.16.100.155,0dbff49d-e8cd-f44d-ab68-f0d5babd61a8,1001,fring,9197845827258,sofia/external/9197845827258,d11b3020-6def-2849-aac0-7454c25d5077
2011-10-11 11:28:26,1318325306,switch_ivr_signal_bridge,1001,1001,r00996550675305,sofia/external/1001 at 172.16.100.155,9354bc8e-100d-3e4e-972a-5814fde9ce4c,1001,fring,996550675305,sofia/external/996550675305,cf2662b9-3be1-a440-91f5-7d3088d574bb
2011-10-11 16:43:55,1318344235,switch_ivr_signal_bridge,1001,1001,r00919895419251,sofia/external/1001 at 172.16.100.155,69dab1ec-9d20-4c4d-a8b8-036cc2be4382,1001,fring,919895419251,sofia/external/919895419251,db22ba2d-dff2-1544-a67e-578886921592
2011-10-11 17:48:39,1318348119,switch_ivr_signal_bridge,1001,1001,r00919773212368,sofia/external/1001 at 172.16.100.155,7a35e17e-9604-4743-b131-d339e08245d0,1001,fring,919773212368,sofia/external/919773212368,0cce04b4-3776-4c4f-adf8-e42a1555d33a
2011-10-12 01:57:34,1318377454,switch_ivr_signal_bridge,1001,1001,r0023794311238,sofia/external/1001 at 172.16.100.155,4a09bce4-5342-3943-9b36-a9bd7c51d8bc,1001,fring,23794311238,sofia/external/23794311238,ede10bc2-45b2-1240-a20c-9b3e65868823
2011-10-12 05:39:54,1318390794,switch_ivr_signal_bridge,1001,1001,r00918121275024,sofia/external/1001 at 172.16.100.155,4c991cad-f71d-e644-9946-8fff485c9a40,1001,fring,918121275024,sofia/external/918121275024,4b3e9c9f-bb26-814e-89e5-0685c05692df
2011-10-12 06:39:05,1318394345,switch_ivr_signal_bridge,1001,1001,r00919947389305,sofia/external/1001 at 172.16.100.155,d2a0d36e-182e-bc47-8fe2-83032467eff3,1001,fring,919947389305,sofia/external/919947389305,e5d52ab2-2c36-1843-bea8-0cfdce391e15
2011-10-12 06:59:34,1318395574,switch_ivr_signal_bridge,1001,1001,r00865395760139,sofia/external/1001 at 172.16.100.155,03a603a0-6418-7a46-a1d4-5d4d2357bf23,1001,fring,865395760139,sofia/external/865395760139,ef325309-716c-1246-8f01-513c7e9a93bb
2011-10-12 08:49:22,1318402162,switch_ivr_signal_bridge,1001,1001,r00919944731940,sofia/external/1001 at 172.16.100.155,024d66fc-6582-b04b-a962-56c4db8cb013,1001,fring,919944731940,sofia/external/919944731940,1de9cb6b-4eed-3d4d-89bc-6b533b07849a
2011-10-12 09:37:03,1318405023,switch_ivr_signal_bridge,1001,1001,r00919916780545,sofia/external/1001 at 172.16.100.155,379b8c6e-a998-2649-a577-5f8f1b6b3c4e,1001,fring,919916780545,sofia/external/919916780545,097c1a46-1e65-6749-aa14-592b1cae6fd5
2011-10-12 10:17:08,1318407428,switch_ivr_signal_bridge,1001,1001,r00914562224442,sofia/external/1001 at 172.16.100.155,f91ec6f2-e8a1-b549-86b5-cbb1d17039b3,1001,fring,914562224442,sofia/external/914562224442,3c2ced4c-7b98-4948-8520-8bc357e4a437
2011-10-12 13:54:07,1318420447,switch_ivr_signal_bridge,1001,1001,r00919746980086,sofia/external/1001 at 172.16.100.155,6f1a25be-891e-ee4f-8c70-4ae79694a651,1001,fring,919746980086,sofia/external/919746980086,d0fdad10-f3a2-5347-ab1f-95d4383089b7
2011-10-12 21:04:21,1318446261,switch_ivr_signal_bridge,1001,1001,r0096597280025,sofia/external/1001 at 172.16.100.155,3349eb67-9f4b-f64f-98af-854dd9a07c23,1001,fring,96597280025,sofia/external/96597280025,14d7da2f-d2e6-084c-b5d3-375377400832
2011-10-13 08:38:46,1318487926,switch_ivr_signal_bridge,1001,1001,r00639214286751,sofia/external/1001 at 172.16.100.155,9b909f83-2d31-a145-82bb-1c0dd1f46695,1001,fring,639214286751,sofia/external/639214286751,6bd8bfbf-4391-1e48-ae3d-16c7f81c74a9
2011-10-13 09:28:42,1318490922,switch_ivr_signal_bridge,1001,1001,r00919855874737,sofia/external/1001 at 172.16.100.155,ff74debc-509c-2040-bf50-42313aa6629d,1001,fring,919855874737,sofia/external/919855874737,579ecfbf-d00c-7947-9c2f-6029dbf9b507
2011-10-13 10:44:23,1318495463,switch_ivr_signal_bridge,1001,1001,r00919966842195,sofia/external/1001 at 172.16.100.155,16e5bb3f-5ff1-0743-b787-78fbfb032583,1001,fring,919966842195,sofia/external/919966842195,77d980c9-4ccb-1c40-a901-3e6fd6960a2d
2011-10-13 11:54:10,1318499650,switch_ivr_signal_bridge,1001,1001,r00919751864788,sofia/external/1001 at 172.16.100.155,84725b33-97ab-d64f-8a36-b7f10a2b8723,1001,fring,919751864788,sofia/external/919751864788,f7e86569-01a2-5540-9c9b-3e300f1a0797
2011-10-13 14:26:04,1318508764,switch_ivr_signal_bridge,1001,1001,r00919922440249,sofia/external/1001 at 172.16.100.155,ad0c00ab-17c2-4949-880b-9ff164941f1b,1001,fring,919922440249,sofia/external/919922440249,1d139c8d-82eb-9248-8e81-e9d6315ba50d
2011-10-13 16:10:48,1318515048,switch_ivr_signal_bridge,1001,1001,r0020197596666,sofia/external/1001 at 172.16.100.155,ad64dcd2-232c-d543-8d01-2a513c3dc315,1001,fring,20197596666,sofia/external/20197596666,7f9b2f6c-41db-3840-ab78-e0052c56a885
2011-10-13 20:09:02,1318529342,switch_ivr_signal_bridge,1001,1001,r00551158958193,sofia/external/1001 at 172.16.100.155,0730806a-137f-924e-ace6-9852c72bb185,1001,fring,551158958193,sofia/external/551158958193,de18fc0d-d46f-014c-92f1-ff5ac3d69e47
2011-10-13 21:11:25,1318533085,switch_ivr_signal_bridge,1001,1001,r0020123946125,sofia/external/1001 at 172.16.100.155,74c80972-2dd4-c443-b831-d76003497612,1001,fring,20123946125,sofia/external/20123946125,be927ca2-99cc-4840-aa86-8c9a85e32f80
2011-10-13 22:43:24,1318538604,switch_ivr_signal_bridge,1001,1001,r00919769404307,sofia/external/1001 at 172.16.100.155,16b1cdf4-c7de-2642-a3d1-a901c7168829,1001,fring,919769404307,sofia/external/919769404307,74860ee3-d592-8441-a738-3b82103033a1
2011-10-14 03:50:54,1318557054,switch_ivr_signal_bridge,1001,1001,r00919988274760,sofia/external/1001 at 172.16.100.155,378a4ecc-7646-6147-9086-381a2ffc5570,1001,fring,919988274760,sofia/external/919988274760,49650a00-03dc-054a-ae56-2debe46897a4
2011-10-14 04:13:23,1318558403,switch_ivr_signal_bridge,1001,1001,r00919944010360,sofia/external/1001 at 172.16.100.155,5aec4e1e-ed85-ef41-b9fe-7f7e9666b0da,1001,fring,919944010360,sofia/external/919944010360,39aa0999-6817-1148-98a2-dc8b1cf768c8
2011-10-14 05:28:12,1318562892,switch_ivr_signal_bridge,1001,1001,r00919944191129,sofia/external/1001 at 172.16.100.155,f4e6f3dc-543a-0a4e-b81d-58c8e5434b0a,1001,fring,919944191129,sofia/external/919944191129,b702994a-2237-a442-89c3-1026966a1feb
2011-10-14 06:14:54,1318565694,switch_ivr_signal_bridge,1001,1001,r0060148144168,sofia/external/1001 at 172.16.100.155,95f21383-70db-2848-a1a4-bf1e279970cb,1001,fring,60148144168,sofia/external/60148144168,f9c3747a-4c39-aa4d-a655-7cea1222ada9
2011-10-14 06:40:42,1318567242,switch_ivr_signal_bridge,1001,1001,r008615952583367,sofia/external/1001 at 172.16.100.155,d4976a2b-d218-a14a-ab73-ed5252d23223,1001,fring,8615952583367,sofia/external/8615952583367,d514508a-e4c8-f14f-82fc-43b71560f565
2011-10-14 13:30:22,1318591822,switch_ivr_signal_bridge,1001,1001,r00919902730542,sofia/external/1001 at 172.16.100.155,535a6926-431e-154b-bed1-bea1ea8240a0,1001,fring,919902730542,sofia/external/919902730542,45a49ed1-b426-5f4d-8da6-5d7ae5934dc9
2011-10-14 13:52:34,1318593154,switch_ivr_signal_bridge,1001,1001,r008613986690504,sofia/external/1001 at 172.16.100.155,39dd44b8-1854-be40-8cd4-81c34ebbcc11,1001,fring,8613986690504,sofia/external/8613986690504,7d641511-0cd1-c54c-8058-568f7f0f67f0
2011-10-14 14:44:34,1318596274,switch_ivr_signal_bridge,1001,1001,r0020101,sofia/external/1001 at 172.16.100.155,08408567-1407-ea4a-be73-ce0219738962,1001,fring,20101,sofia/external/20101,aa1cbd89-f2f1-114a-a0a4-2d18fb647a1d
2011-10-14 19:30:55,1318613455,switch_ivr_signal_bridge,1001,1001,r00919953239015,sofia/external/1001 at 172.16.100.155,931bd3ec-a4a9-1b47-b6a3-7226cf83dc3b,1001,fring,919953239015,sofia/external/919953239015,37b98044-fbe2-e940-a368-fe9556fb810f
2011-10-15 01:42:58,1318635778,switch_ivr_signal_bridge,1001,1001,r00919032587213,sofia/external/1001 at 172.16.100.155,11537992-801d-eb42-ba4e-763fbff0f381,1001,fring,919032587213,sofia/external/919032587213,4d2e98a0-d4b4-d046-aa6c-11c7d7137c63
2011-10-15 05:32:32,1318649552,switch_ivr_signal_bridge,1001,1001,r00919885688639,sofia/external/1001 at 172.16.100.155,b2b70ad1-0754-8c49-82eb-c0ae5d051618,1001,fring,919885688639,sofia/external/919885688639,a35b33e7-08c0-f34f-b664-2c1b68be3e95
2011-10-15 06:40:56,1318653656,switch_ivr_signal_bridge,1001,1001,r00919886690120,sofia/external/1001 at 172.16.100.155,13a08888-c378-d949-8625-95e2239bac45,1001,fring,919886690120,sofia/external/919886690120,ff62df61-8354-0641-a1ce-cba74fe53448
2011-10-15 09:51:20,1318665080,switch_ivr_signal_bridge,1001,1001,r00265888891340,sofia/external/1001 at 172.16.100.155,41c4263a-18f2-d147-8b7a-3a14739eeb4f,1001,fring,265888891340,sofia/external/265888891340,1c19dbaa-d35c-3e4d-a913-355d2f9b6c0d
2011-10-15 10:06:41,1318666001,switch_ivr_signal_bridge,1001,1001,r00919820158704,sofia/external/1001 at 172.16.100.155,c1363ab8-2905-2549-876c-dd3cbc05563f,1001,fring,919820158704,sofia/external/919820158704,660b2675-974e-a249-8a24-349020a7449a
2011-10-15 11:19:21,1318670361,switch_ivr_signal_bridge,1001,1001,r0020197105643,sofia/external/1001 at 172.16.100.155,cc2f1ede-7d0c-8845-b455-4eb97ed2b12f,1001,fring,20197105643,sofia/external/20197105643,81cb54ff-b63d-6c4e-afa9-bfea68a553a9
2011-10-15 12:14:13,1318673653,switch_ivr_signal_bridge,1001,1001,r00918870183328,sofia/external/1001 at 172.16.100.155,81ba52e8-d157-4441-bfe1-c314e4f1b4c7,1001,fring,918870183328,sofia/external/918870183328,72306599-6017-c54a-a291-9b3eb1650d83
2011-10-15 14:15:42,1318680942,switch_ivr_signal_bridge,1001,1001,r00919539499709,sofia/external/1001 at 172.16.100.155,b24a486b-b798-c048-b0c5-9c624b0ebade,1001,fring,919539499709,sofia/external/919539499709,456681fa-c2bf-004f-875a-4d4b18ea1e99
2011-10-15 23:23:12,1318713792,switch_ivr_signal_bridge,1001,1001,r0020140088523,sofia/external/1001 at 172.16.100.155,c2f98b57-0c7b-1b41-a77c-05a34ab99b79,1001,fring,20140088523,sofia/external/20140088523,f681a50f-d263-a942-a48f-ab7eb2adff38
2011-10-16 02:16:44,1318724204,switch_ivr_signal_bridge,1001,1001,r00919886859264,sofia/external/1001 at 172.16.100.155,764923b2-29b4-fd4d-9de8-cd89f2654a9f,1001,fring,919886859264,sofia/external/919886859264,ecbe5234-bb44-1a40-a93c-4fd36307f3e9
2011-10-16 06:50:47,1318740647,switch_ivr_signal_bridge,1001,1001,r00919915919879,sofia/external/1001 at 172.16.100.155,27d55f39-86e3-f146-ad9b-2f77938ad868,1001,fring,919915919879,sofia/external/919915919879,2df4fa27-2896-6f4d-81ce-97532b2dee0c
2011-10-16 10:02:03,1318752123,switch_ivr_signal_bridge,1001,1001,r00965555751124,sofia/external/1001 at 172.16.100.155,7d69745c-89e1-ad49-8f57-ca146ecc769a,1001,fring,965555751124,sofia/external/965555751124,973c1c3b-0f50-684a-873b-b61bdd21a245
2011-10-16 11:37:30,1318757850,switch_ivr_signal_bridge,1001,1001,r00201001301440,sofia/external/1001 at 172.16.100.155,22fb4d57-3207-4d4c-8be1-145655dad6a5,1001,fring,201001301440,sofia/external/201001301440,397ee409-504c-244c-b11b-b693070ff7dd
2011-10-16 14:55:05,1318769705,switch_ivr_signal_bridge,1001,1001,r00919742055333,sofia/external/1001 at 172.16.100.155,1817b98a-e036-1746-bd7d-437152ed6b0a,1001,fring,919742055333,sofia/external/919742055333,ab2b04ce-a454-a041-a40e-7ee5ddbb29a4
2011-10-16 16:20:52,1318774852,switch_ivr_signal_bridge,1001,1001,r00998662245737,sofia/external/1001 at 172.16.100.155,986c8dc4-831b-0041-9b3c-1bee42818ab5,1001,fring,998662245737,sofia/external/998662245737,a1eaeb3c-6b1a-ea4c-a3b1-f3144572bf72
2011-10-16 16:51:24,1318776684,switch_ivr_signal_bridge,1001,1001,r008801712054706,sofia/external/1001 at 172.16.100.155,84a34d92-8be5-1644-a9cd-487023eee234,1001,fring,8801712054706,sofia/external/8801712054706,1930550c-d3e3-f645-bda9-e4a0d39a6397
2011-10-16 18:14:22,1318781662,switch_ivr_signal_bridge,1001,1001,r00919390177288,sofia/external/1001 at 172.16.100.155,2878943a-56ee-be47-a5fc-632a92c3734a,1001,fring,919390177288,sofia/external/919390177288,4a8de922-86ab-ba41-8e9c-05a79087a79a
2011-10-16 20:06:31,1318788391,switch_ivr_signal_bridge,1001,1001,r00201224955898,sofia/external/1001 at 172.16.100.155,fc68f803-781d-c347-a6dc-f50a8651f61f,1001,fring,201224955898,sofia/external/201224955898,1e873400-c1ca-da42-be32-837931d2c845
2011-10-16 22:52:28,1318798348,switch_ivr_signal_bridge,1001,1001,r00914567242736,sofia/external/1001 at 172.16.100.155,d8b4385a-bcf9-ad4a-ae00-0539fe7c56f8,1001,fring,914567242736,sofia/external/914567242736,5eb0ca19-155a-3047-a480-8f951f26993f
2011-10-16 23:06:09,1318799169,switch_ivr_signal_bridge,1001,1001,r00966552321650,sofia/external/1001 at 172.16.100.155,f9998988-f7f1-9543-afd5-1cd76e035594,1001,fring,966552321650,sofia/external/966552321650,c01c89f6-212b-5a48-90ce-c79587ecdbac
2011-10-17 00:38:08,1318804688,switch_ivr_signal_bridge,1001,1001,r00911821277451,sofia/external/1001 at 172.16.100.155,54365caa-8f25-8545-9010-db155e4bdde8,1001,fring,911821277451,sofia/external/911821277451,8df3edfb-3b35-a047-8c8d-51d57eeb56b3
2011-10-17 06:02:00,1318824120,switch_ivr_signal_bridge,1001,1001,r00919746008642,sofia/external/1001 at 172.16.100.155,ba807588-4536-d94f-81a7-e1d0e8434e05,1001,fring,919746008642,sofia/external/919746008642,6252a3b1-a0a6-ff42-8188-69e771094104
2011-10-17 06:30:17,1318825817,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/1001 at 172.16.100.155,2e099d56-5e90-2f48-816f-d66e410fc8b9,1001,fring,919824645363,sofia/external/919824645363,7d18d6e9-dfb9-c949-b00b-5a9e62ecc21c
2011-10-17 07:42:55,1318830175,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/1001 at 172.16.100.155,54960c11-5352-e34b-ae35-672923413dae,1001,fring,919824645363,sofia/external/919824645363,bbc5cdc4-7a02-2945-b685-28baae5fe280
2011-10-17 10:11:26,1318839086,switch_ivr_signal_bridge,1001,1001,r00919711534925,sofia/external/1001 at 172.16.100.155,583ffded-6229-6345-bd46-4ef09652a8aa,1001,fring,919711534925,sofia/external/919711534925,9d101a1a-0e4e-7144-bff1-a132378a3d9f
2011-10-17 17:50:55,1318866655,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,b6cd78b7-7da4-b44e-a4f4-60e404fafca3,1001,fring,919988175489,sofia/external/919988175489,602bd41d-3f60-d144-b2bb-36d4da232ddc
2011-10-17 22:52:13,1318884733,switch_ivr_signal_bridge,1001,1001,r00919999233886,sofia/external/1001 at 172.16.100.155,eefd784e-9c04-084f-888b-b0a033a5ee5b,1001,fring,919999233886,sofia/external/919999233886,389a824e-e4c5-b74e-ac8a-9315d9821117
2011-10-17 23:24:02,1318886642,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/1001 at 172.16.100.155,fcf32cd6-9148-cc40-bc90-534145199ba9,1001,fring,919313467728,sofia/external/919313467728,7d13a986-8454-be4d-a056-c47276ba9f5b
2011-10-18 00:41:19,1318891279,switch_ivr_signal_bridge,1001,1001,r00919747072485,sofia/external/1001 at 172.16.100.155,25f36536-4606-1942-825f-5bfaf28a66ca,1001,fring,919747072485,sofia/external/919747072485,ee6298c7-a6d4-3547-9578-7c95f96ede0f
2011-10-18 01:05:08,1318892708,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/1001 at 172.16.100.155,8c89657f-9ce7-404a-9d5c-a3559f66b141,1001,fring,919313467728,sofia/external/919313467728,b6a3eb92-6419-4643-930f-e7bebeabb33a
2011-10-18 03:25:04,1318901104,switch_ivr_signal_bridge,1001,1001,r006596112485,sofia/external/1001 at 172.16.100.155,b5189c01-0df2-4e47-af11-8a3e96e247a8,1001,fring,6596112485,sofia/external/6596112485,a834eb93-3744-be49-8165-3b39966c9502
2011-10-18 06:25:34,1318911934,switch_ivr_signal_bridge,1001,1001,r00919876855863,sofia/external/1001 at 172.16.100.155,d6589f30-ba06-5c45-9d20-1e99664a10d8,1001,fring,919876855863,sofia/external/919876855863,4999eac4-e560-fe40-b795-fc5d5cf8ac44
2011-10-18 09:41:35,1318923695,switch_ivr_signal_bridge,1001,1001,r00919922965741,sofia/external/1001 at 172.16.100.155,57335341-bda9-3943-8f0d-59c7a7903c15,1001,fring,919922965741,sofia/external/919922965741,d201c220-2c1f-bb4e-b725-bada9895035b
2011-10-18 12:52:37,1318935157,switch_ivr_signal_bridge,1001,1001,r00919967359493,sofia/external/1001 at 172.16.100.155,66fadc05-a756-a848-bb84-5d39c9998fc5,1001,fring,919967359493,sofia/external/919967359493,71bcd983-d4a4-de4c-b615-f0dbf1095739
2011-10-18 16:41:02,1318948862,switch_ivr_signal_bridge,1001,1001,r00919831094384,sofia/external/1001 at 172.16.100.155,2a96ebc1-a6be-8e49-b508-8b3993b62f9a,1001,fring,919831094384,sofia/external/919831094384,86e74690-fca2-6247-9d6a-c7732445a1be
2011-10-18 21:01:18,1318964478,switch_ivr_signal_bridge,1001,1001,r00919729947506,sofia/external/1001 at 172.16.100.155,96995af7-e82c-3e40-addb-1b62c1f05747,1001,fring,919729947506,sofia/external/919729947506,6a82a0ec-e4a2-be4f-85ee-87b34b4ea3e2
2011-10-19 00:20:24,1318976424,switch_ivr_signal_bridge,1001,1001,r00919605053363,sofia/external/1001 at 172.16.100.155,8f9d6670-2aff-c242-b01e-3d9f0e493d12,1001,fring,919605053363,sofia/external/919605053363,7fd8cee5-cd6b-7b44-848f-8d756f26fd4a
2011-10-19 04:30:30,1318991430,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/1001 at 172.16.100.155,1a6895e1-57a5-564c-bf08-5780df1d79c7,1001,fring,919814950511,sofia/external/919814950511,595fe521-1673-ea44-8bc3-c9d69c2f2de6
2011-10-19 04:40:27,1318992027,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/1001 at 172.16.100.155,905be5af-844a-d44f-bbfd-a160ff1f1759,1001,fring,919814950511,sofia/external/919814950511,8a41744e-b19e-954a-a68f-7d16e29976e5
2011-10-19 06:23:43,1318998223,switch_ivr_signal_bridge,1001,1001,r00919990249401,sofia/external/1001 at 172.16.100.155,d009a643-b1b3-3f46-b194-c971e2a17b64,1001,fring,919990249401,sofia/external/919990249401,f26ae13f-de35-2f40-92cb-232449391e3e
2011-10-19 09:38:43,1319009923,switch_ivr_signal_bridge,1001,1001,r00919761333537,sofia/external/1001 at 172.16.100.155,353ede21-8d68-b74e-8711-0ddce2a82928,1001,fring,919761333537,sofia/external/919761333537,d0567650-c2f4-9542-9a20-d8bc1718cc80
2011-10-19 10:59:56,1319014796,switch_ivr_signal_bridge,1001,1001,r00918144890121,sofia/external/1001 at 172.16.100.155,8593dc8b-8d4f-bc40-8e3a-daf85d7c7702,1001,fring,918144890121,sofia/external/918144890121,ec3fda7c-cc8e-e041-b4a0-171928ade343
2011-10-19 13:00:41,1319022041,switch_ivr_signal_bridge,1001,1001,r00201060198003,sofia/external/1001 at 172.16.100.155,bba15b1f-785a-674f-a023-bb8f64f57c53,1001,fring,201060198003,sofia/external/201060198003,2c90f16e-a4ec-6d42-9bd1-4f06bf7c2da9
2011-10-19 13:45:11,1319024711,switch_ivr_signal_bridge,1001,1001,r00914312341722,sofia/external/1001 at 172.16.100.155,845903b7-2770-6847-a116-3d0a8cb7b096,1001,fring,914312341722,sofia/external/914312341722,aa20b33f-eb4d-714a-b236-59afbfe158e7
2011-10-19 19:11:01,1319044261,switch_ivr_signal_bridge,1001,1001,r009845967072,sofia/external/1001 at 172.16.100.155,349ef121-64b9-dc44-a22d-3d12acbb6ece,1001,fring,9845967072,sofia/external/9845967072,d89904a7-daf7-ea45-bae3-57c914e88b48
2011-10-20 02:36:13,1319070973,switch_ivr_signal_bridge,1001,1001,r0060166298037,sofia/external/1001 at 172.16.100.155,f6ff096b-612e-0544-a733-9a566b08fc6d,1001,fring,60166298037,sofia/external/60166298037,1c35f10e-3695-2845-9f4c-dd5a517dd6f7
2011-10-20 03:15:43,1319073343,switch_ivr_signal_bridge,1001,1001,r00919988498802,sofia/external/1001 at 172.16.100.155,1a9da86a-bc16-3e49-9b08-94facc657843,1001,fring,919988498802,sofia/external/919988498802,9430553a-ee00-ed4f-b109-41eabdd4dc7e
2011-10-20 04:35:24,1319078124,switch_ivr_signal_bridge,1001,1001,r00919976131008,sofia/external/1001 at 172.16.100.155,12d0429e-a539-6944-831e-a862183d905b,1001,fring,919976131008,sofia/external/919976131008,f672283c-a37e-e947-b45a-b7e66ef7881a
2011-10-20 05:22:23,1319080943,switch_ivr_signal_bridge,1001,1001,r00919973840848,sofia/external/1001 at 172.16.100.155,42e29646-c0c4-d64a-9cf5-7c37aeabc1a6,1001,fring,919973840848,sofia/external/919973840848,1d2ef437-17aa-db4b-86d9-6d3d5310e280
2011-10-20 05:37:34,1319081854,switch_ivr_signal_bridge,1001,1001,r00919840288799,sofia/external/1001 at 172.16.100.155,f48fed29-8041-3545-bffc-5050bc19cbc6,1001,fring,919840288799,sofia/external/919840288799,df29a0df-3ef8-4b43-807e-73e0c0e25f3b
2011-10-20 10:00:26,1319097626,switch_ivr_signal_bridge,1001,1001,r0091955445666,sofia/external/1001 at 172.16.100.155,931102bc-ac35-3140-acab-988224d502b0,1001,fring,91955445666,sofia/external/91955445666,9283eb63-bd0a-6d43-9d8a-7985ec48e23c
2011-10-20 12:16:27,1319105787,switch_ivr_signal_bridge,1001,1001,r00919855120931,sofia/external/1001 at 172.16.100.155,db0f66aa-69f1-fe47-85af-de610a77d517,1001,fring,919855120931,sofia/external/919855120931,ceab5ae5-9b8f-3d4c-b149-c248567ab5d4
2011-10-20 12:42:02,1319107322,switch_ivr_signal_bridge,1001,1001,r00918872855601,sofia/external/1001 at 172.16.100.155,95249509-e793-0d42-9948-5ef53898b942,1001,fring,918872855601,sofia/external/918872855601,8c9fc203-8d54-aa4f-983f-889fbf0ca883
2011-10-20 13:45:10,1319111110,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,7b645b14-5f89-a64c-b059-2574b2e4e69d,1001,fring,919988175489,sofia/external/919988175489,55f50da5-5679-fe4f-9a7f-b96111f3cd67
2011-10-20 14:49:50,1319114990,switch_ivr_signal_bridge,1001,1001,r00919663195202,sofia/external/1001 at 172.16.100.155,3601f4b8-c0ff-1a4c-9f76-47e4fa95b549,1001,fring,919663195202,sofia/external/919663195202,a22e2d11-0463-c845-a953-a12b49cfe863
2011-10-20 15:10:47,1319116247,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/1001 at 172.16.100.155,e09d3de0-c759-fc4b-96f1-f01353a4297d,1001,fring,919988175489,sofia/external/919988175489,23d4b490-cbb7-7a45-86af-705c45d07fc7
2011-10-20 20:52:17,1319136737,switch_ivr_signal_bridge,1001,1001,r009888560750,sofia/external/1001 at 172.16.100.155,ff8ab1f6-f1c6-224f-928b-1c042b0ed908,1001,fring,9888560750,sofia/external/9888560750,c37b8ebe-2264-934f-a187-6b14c7dcb308
2011-10-20 22:32:38,1319142758,switch_ivr_signal_bridge,1001,1001,r008801816630096,sofia/external/1001 at 172.16.100.155,a38182cb-b661-854d-b243-a830ab0bf992,1001,fring,8801816630096,sofia/external/8801816630096,f88dc530-303e-d041-83cb-eca118780ca2
2011-10-20 22:34:32,1319142872,switch_ivr_signal_bridge,1001,1001,r00919161209997,sofia/external/1001 at 172.16.100.155,0add3c1c-7fb5-2447-a9d6-d0622886fcf9,1001,fring,919161209997,sofia/external/919161209997,2b91f5de-4668-0244-ac76-d031fe827a25
2011-10-20 23:56:16,1319147776,switch_ivr_signal_bridge,1001,1001,r00985468962,sofia/external/1001 at 172.16.100.155,f7137d11-0a87-6a4b-b5b3-b43eeeb2b462,1001,fring,985468962,sofia/external/985468962,025f197b-c5aa-5b4b-8606-20fb6f17e888
2011-10-21 04:36:00,1319164560,switch_ivr_signal_bridge,1001,1001,r00917597234252,sofia/external/1001 at 172.16.100.155,edd2f16b-9a6a-f749-bd62-4d303bc09442,1001,fring,917597234252,sofia/external/917597234252,1c55b21b-9c27-b647-8ba5-1ffce398c747
2011-10-21 04:47:03,1319165223,switch_ivr_signal_bridge,1001,1001,r00919541622040,sofia/external/1001 at 172.16.100.155,336c146b-c96d-494e-b378-c024bcfc0c94,1001,fring,919541622040,sofia/external/919541622040,746877e1-c509-8a45-934e-f3f4b7f82db2
2011-10-21 07:59:54,1319176794,switch_ivr_signal_bridge,1001,1001,r009613235498,sofia/external/1001 at 172.16.100.155,16575979-c92a-5842-879f-e9f6448c7d54,1001,fring,9613235498,sofia/external/9613235498,bb05fd5a-4a6f-3248-81a9-a81074a9eacd
2011-10-21 08:02:05,1319176925,switch_ivr_signal_bridge,1001,1001,r00912026336917,sofia/external/1001 at 172.16.100.155,ebb234aa-8edf-2e4b-b641-602478b25782,1001,fring,912026336917,sofia/external/912026336917,f6a0e4b1-69f7-dc49-a5bf-41e6d8c7c96d
2011-10-21 08:51:57,1319179917,switch_ivr_signal_bridge,1001,1001,r00919372875359,sofia/external/1001 at 172.16.100.155,b090dc91-932e-1246-abb8-8fc949131ae6,1001,fring,919372875359,sofia/external/919372875359,3a4702a6-e7cc-cb41-ad99-defa80b14680
2011-10-21 09:15:47,1319181347,switch_ivr_signal_bridge,1001,1001,r00919915709069,sofia/external/1001 at 172.16.100.155,ae308194-ce89-4945-b8de-6923ce6d0dc0,1001,fring,919915709069,sofia/external/919915709069,502ffe82-0fb9-b14a-9421-4acd25a0b4df
2011-10-21 11:35:16,1319189716,switch_ivr_signal_bridge,1001,1001,r00919847023725,sofia/external/1001 at 172.16.100.155,c7cb7640-7313-d742-b756-a7713273ccea,1001,fring,919847023725,sofia/external/919847023725,4ab50403-04ac-bc40-a759-40f2a62f1699
2011-10-21 14:48:46,1319201326,switch_ivr_signal_bridge,1001,1001,r00919878738103,sofia/external/1001 at 172.16.100.155,91626ac0-9e14-a34d-94f5-b1fbfe7e7629,1001,fring,919878738103,sofia/external/919878738103,39c58424-b087-6749-9ba9-c9a821f46895
2011-10-21 17:41:28,1319211688,switch_ivr_signal_bridge,1001,1001,r009843648082,sofia/external/1001 at 172.16.100.155,3929dc6b-b7de-c648-8eae-3d749f0ac0fa,1001,fring,9843648082,sofia/external/9843648082,6653a9ed-d23b-564e-98e5-9a56a40e499c
2011-10-22 05:07:21,1319252841,switch_ivr_signal_bridge,1001,1001,r00919036392951,sofia/external/1001 at 172.16.100.155,7cd56510-2983-554a-9fa2-a27c3fb1f9e6,1001,fring,919036392951,sofia/external/919036392951,e63642ec-d43a-c243-9291-0ea6725d04bb
2011-10-22 06:42:56,1319258576,switch_ivr_signal_bridge,1001,1001,r00918870521722,sofia/external/1001 at 172.16.100.155,7dd85bc6-e00e-c44e-bd42-89c6d438485e,1001,fring,918870521722,sofia/external/918870521722,8d7e1c17-2ad9-2447-b91b-d2f2b2e27e4c
2011-10-22 08:36:26,1319265386,switch_ivr_signal_bridge,1001,1001,r00911874289543,sofia/external/1001 at 172.16.100.155,1214a817-f235-1a4e-a21c-d2363e9e5760,1001,fring,911874289543,sofia/external/911874289543,13f85a5d-eb37-0c4a-a15e-2c86dbc34817
2011-10-22 12:14:32,1319278472,switch_ivr_signal_bridge,1001,1001,r00918988262666,sofia/external/1001 at 172.16.100.155,1b5cb871-e043-f244-9942-251a7a492136,1001,fring,918988262666,sofia/external/918988262666,21f328fb-0dfe-2643-957b-302f6a80f17f
2011-10-22 16:28:35,1319293715,switch_ivr_signal_bridge,1001,1001,r00918722222400,sofia/external/1001 at 172.16.100.155,912e1b07-a99b-bf4d-9e2f-0b5990a68821,1001,fring,918722222400,sofia/external/918722222400,8b417629-3161-f141-93d7-bb0ddd7982d5
2011-10-22 17:32:10,1319297530,switch_ivr_signal_bridge,1001,1001,r00919839884297,sofia/external/1001 at 172.16.100.155,ec7decdd-c378-6c45-a888-9b962e4ecece,1001,fring,919839884297,sofia/external/919839884297,f12896b7-9605-9b40-bdc9-face2181146b
2011-10-23 06:26:42,1319344002,switch_ivr_signal_bridge,1001,1001,r00919567382898,sofia/external/1001 at 172.16.100.155,db2e199b-7b9f-fe4d-9742-7415dded69c7,1001,fring,919567382898,sofia/external/919567382898,dd8bb627-c804-2f49-8b5b-5acf5b3423d4
2011-10-23 07:21:29,1319347289,switch_ivr_signal_bridge,1001,1001,r00966537646047,sofia/external/1001 at 172.16.100.155,e97ba53e-1ef6-2944-b53b-b3abe5dfef36,1001,fring,966537646047,sofia/external/966537646047,9eea7d3f-5092-de4b-9291-e7f0a37db2a1
2011-10-23 08:12:33,1319350353,switch_ivr_signal_bridge,1001,1001,r00914942588715,sofia/external/1001 at 172.16.100.155,30ca9d2d-5cae-6645-a16c-6ae8735b6316,1001,fring,914942588715,sofia/external/914942588715,6774d23c-f131-7d41-854d-2fcff77c2cf6
2011-10-23 13:04:33,1319367873,switch_ivr_signal_bridge,1001,1001,r00919625833433,sofia/external/1001 at 172.16.100.155,284691f9-dd3b-cc4a-97c7-9465bd3a6b6c,1001,fring,919625833433,sofia/external/919625833433,4efb7446-11f2-c84c-88c4-10ab97eddedf
2011-10-23 14:35:20,1319373320,switch_ivr_signal_bridge,1001,1001,r00919888880401,sofia/external/1001 at 172.16.100.155,475abe11-2137-c94a-ada7-825ad8524bd8,1001,fring,919888880401,sofia/external/919888880401,5e06efb5-2c72-ba4f-9443-c15568ad2608
2011-10-23 15:59:20,1319378360,switch_ivr_signal_bridge,1001,1001,r009878539161,sofia/external/1001 at 172.16.100.155,aa5c4f8f-5a69-e947-a239-75f4a584b9f0,1001,fring,9878539161,sofia/external/9878539161,366b56d1-3ce4-fa43-9d3d-872c04f5229f
2011-10-23 20:52:23,1319395943,switch_ivr_signal_bridge,1001,1001,r0017787079272,sofia/external/1001 at 172.16.100.155,3f9c8b11-2e7a-e641-8cfc-3b7dc65e7118,1001,fring,17787079272,sofia/external/17787079272,d5b4a2e0-ee48-7048-b54c-f6ea968428af
2011-10-24 10:23:29,1319444609,switch_ivr_signal_bridge,1001,1001,r00919881375673,sofia/external/1001 at 172.16.100.155,527d1ca5-3f05-a54b-8b0c-192b7515eebb,1001,fring,919881375673,sofia/external/919881375673,4d1da2aa-87c1-7d41-b57d-05034e503ede
2011-10-24 10:29:44,1319444984,switch_ivr_signal_bridge,1001,1001,r0061430594924,sofia/external/1001 at 172.16.100.155,6ce52143-219d-ed40-a27f-fc519eef5092,1001,fring,61430594924,sofia/external/61430594924,11ef495a-d651-564e-9479-94fa579a5286
2011-10-24 10:56:32,1319446592,switch_ivr_signal_bridge,1001,1001,r00919003574979,sofia/external/1001 at 172.16.100.155,8d83282b-f96b-0a43-ac58-1f3d3e4e26f0,1001,fring,919003574979,sofia/external/919003574979,2bd6e807-c5ef-8c47-807f-1e5171bcbc3d
2011-10-24 13:58:59,1319457539,switch_ivr_signal_bridge,1001,1001,r00919876395511,sofia/external/1001 at 172.16.100.155,68fdbb19-f76f-0a40-b46b-e0296eed8690,1001,fring,919876395511,sofia/external/919876395511,6cf77cf5-8cd9-f340-98c0-bb896304dbf9
2011-10-24 14:16:45,1319458605,switch_ivr_signal_bridge,1001,1001,r00919988170711,sofia/external/1001 at 172.16.100.155,57e0187b-9d25-044c-bffb-9d11431c251a,1001,fring,919988170711,sofia/external/919988170711,b71d6822-289f-614b-8e5b-2bc20225d9ef
2011-10-25 00:16:09,1319494569,switch_ivr_signal_bridge,1001,1001,r005016630551,sofia/external/1001 at 172.16.100.155,f5e81b48-8f86-4644-b206-cb066e15418d,1001,fring,5016630551,sofia/external/5016630551,b1467ed2-56e3-bb42-ba6c-cdc272b2533b
2011-10-25 02:30:30,1319502630,switch_ivr_signal_bridge,1001,1001,r00919729325377,sofia/external/1001 at 172.16.100.155,f6eb7c48-cc1d-9b42-bbac-46ac759de0e0,1001,fring,919729325377,sofia/external/919729325377,d8eb68b9-ab39-814b-a70d-915347e2afe7
2011-10-25 09:35:05,1319528105,switch_ivr_signal_bridge,1001,1001,r00966554656169,sofia/external/1001 at 172.16.100.155,03c9a95e-4d36-8847-a2a2-3545c38e233e,1001,fring,966554656169,sofia/external/966554656169,1d2927f5-7202-914c-a463-6dcd00588cfc
2011-10-25 12:53:58,1319540038,switch_ivr_signal_bridge,1001,1001,r00201094811534,sofia/external/1001 at 172.16.100.155,3bfe5634-dcf9-c84d-ba72-26f8fee4a8eb,1001,fring,201094811534,sofia/external/201094811534,d0c58bb9-53a3-0549-9b78-4ca5f14fcb63
2011-10-26 04:22:12,1319595732,switch_ivr_signal_bridge,1001,1001,r00913871398117,sofia/external/1001 at 172.16.100.155,1479c10c-9922-854c-8030-6f2a6ce4f605,1001,fring,913871398117,sofia/external/913871398117,fbdc8cf2-e898-0340-bc48-b4ad7c54ac0e
2011-10-26 10:01:27,1319616087,switch_ivr_signal_bridge,1001,1001,r00201113961426,sofia/external/1001 at 172.16.100.155,db18777e-aefa-c248-84ea-06150fe6989f,1001,fring,201113961426,sofia/external/201113961426,0a5572c6-54b7-1d40-bf4e-918e6880b3ce

147 total.

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From dmj at azteknetworks.com  Wed Oct 26 21:51:19 2011
From: dmj at azteknetworks.com (Dean Johnson)
Date: Wed, 26 Oct 2011 12:51:19 -0500
Subject: [Freeswitch-users] incomplete metaswitch voicemail configuration
Message-ID: <C366D60B11A899478D3463C78EE7E8A701F2AD8A7965@AUSP01VMBX24.collaborationhost.net>

When the Metaswitch tries to forward a call to the freeswitch, it gets back a 480. and the freeswitch log shows that it breaks off its side.

We have been following the http://wiki.freeswitch.org/wiki/FreeSwitch_MetaSwitch_HowTo with the idea of setting up freeswitch in our test lab to be a voicemail application server for our Metaswitch 2510.

We successfully compiled, installed and are running freeswitch on Centos6.
Voiper can call and leave voicemail, as well as go retreive the voicemail.
The Metaswitch 2510 and freeswitch "see" each other without error messages.
We are new to freeswitch and are wondering if we have left out some configuration that assigns freeswitch a correct telephone number or range of numbers that it can service for voicemail.

What information can I provide so I can learn what I have omitted?


______________________________________

Dean Johnson

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From brian.wiese.freeswitch at gmail.com  Thu Oct 27 05:03:51 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Wed, 26 Oct 2011 20:03:51 -0500
Subject: [Freeswitch-users] SIP/2.0 500 No session set by user
Message-ID: <CADEQS5xrxgdqF-=E2+TL+oR53zYUvL65adigrrCLxGfMGQw3pw@mail.gmail.com>

Hello again everyone!

I'm getting a strange result when calling from a Polycom endpoint on FS
server "A" to FS server "B":  About one out of ten calls won't complete with
the error "SIP/2.0 500 No session set by user".  If I immediately redial the
number again it works.

Here's my log:  http://pastebin.freeswitch.org/17622
Note the response to the endpoint from server "A" on line 452.

Important IP addresses:
Polycom endpoint:  10.15.100.250
Server "A":  10.15.100.20
Server "B":  10.15.100.10

I'm on the latest revision (git-2f786a0 2011-10-26 12-57-59 -0500) as well.

I appreciate your help.

~Brian
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From rhuddleston at gmail.com  Thu Oct 27 05:34:37 2011
From: rhuddleston at gmail.com (Robert)
Date: Wed, 26 Oct 2011 21:34:37 -0400
Subject: [Freeswitch-users] Get Fax2Mail to work on ubuntu11.04
In-Reply-To: <1319624405574-6932179.post@n2.nabble.com>
References: <1319624405574-6932179.post@n2.nabble.com>
Message-ID: <3E504D46-0896-46EC-8BD9-206709EB3122@gmail.com>

I am successfully running the LUA example and quite pleased

Sent from my iPad

On Oct 26, 2011, at 6:20 AM, "ET1.D" <etiennedip at gmail.com> wrote:

> Good day,
> 
> Currently I am trying to set up a Fax to mail solution, I am using the
> method that is posted on the wiki with the python script, currently the
> message I am getting is 
> 
> 2011-10-26 12:01:23.564498 [INFO] mod_dialplan_xml.c:336 Processing 1616
> Etienne <2787xxxxxx>->9178 in context default
> 2011-10-26 12:01:23.564498 [ERR] switch_core_session.c:2062 Invalid
> Application python
> 2011-10-26 12:01:23.564498 [NOTICE] switch_core_session.c:2063 Hangup
> sofia/external/27878051616 at 172.16.1.2:5060 [CS_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
> 2011-10-26 12:01:23.564498 [NOTICE] switch_core_session.c:1369 Session 14
> (sofia/external/27878051616 at 172.16.1.2:5060) Ended
> 
> I have uncommented the mod_python module in modules.conf.xml and restarted
> freeswitch, but I still get the same message. Can you please assist.
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Get-Fax2Mail-to-work-on-ubuntu11-04-tp6932179p6932179.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From cliff at develix.com  Thu Oct 27 10:02:46 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 26 Oct 2011 23:02:46 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG_X_-eQ8jNand_E9yZfroPDMaiotNk9Vr4MB+rPAf=oyg@mail.gmail.com>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
	<543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>
	<1319567530.2234.16.camel@portable-evil>
	<CAJh8XdXVUkfMH5Ud9NaHCXV18UyBf3RgwBtgWxVihz6ASmFccQ@mail.gmail.com>
	<4EA7E384.9000103@puzzled.xs4all.nl>
	<CAKbxfG_X_-eQ8jNand_E9yZfroPDMaiotNk9Vr4MB+rPAf=oyg@mail.gmail.com>
Message-ID: <1319695366.2234.19.camel@portable-evil>

On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote:
> if you have older unixodbc edit /etc/odbcinst.ini and add the param
> Threading=0 to each db entry.

> In general postgres to me appears to care more about multithreaded
> environments and with the above threading=0 change works pretty well.

I tried this (after googling a bit and coming across an older thread you
responded to) and it didn't change anything.  Crashes at ~330 concurrent
calls like clockwork.  This was with unixODBC 2.14 and PostgreSQL 8.4
odbc driver.

Regards,
Cliff




From infos at madovsky.org  Thu Oct 27 10:15:25 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 02:15:25 -0400
Subject: [Freeswitch-users] High load on database server
References: <4EA5DA64.1060503@communicatefreely.net><1319494283.4593.160.camel@portable-evil><CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com><CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com><CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com><543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry><1319567530.2234.16.camel@portable-evil><CAJh8XdXVUkfMH5Ud9NaHCXV18UyBf3RgwBtgWxVihz6ASmFccQ@mail.gmail.com><4EA7E384.9000103@puzzled.xs4all.nl><CAKbxfG_X_-eQ8jNand_E9yZfroPDMaiotNk9Vr4MB+rPAf=oyg@mail.gmail.com>
	<1319695366.2234.19.camel@portable-evil>
Message-ID: <1D58D1DAD92C4E8D9FBFDE2F44F266A3@e1705>

try the version 2.3.0 of ODBC (compiled from sources)
I can't tell you if it will work because I never had more than 20 
councurrent calls
but it should be better anyeway

----- Original Message ----- 
From: "Cliff Wells" <cliff at develix.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, October 27, 2011 2:02 AM
Subject: Re: [Freeswitch-users] High load on database server


> On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote:
>> if you have older unixodbc edit /etc/odbcinst.ini and add the param
>> Threading=0 to each db entry.
>
>> In general postgres to me appears to care more about multithreaded
>> environments and with the above threading=0 change works pretty well.
>
> I tried this (after googling a bit and coming across an older thread you
> responded to) and it didn't change anything.  Crashes at ~330 concurrent
> calls like clockwork.  This was with unixODBC 2.14 and PostgreSQL 8.4
> odbc driver.
>
> Regards,
> Cliff
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Thu Oct 27 10:21:02 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 02:21:02 -0400
Subject: [Freeswitch-users] High load on database server
Message-ID: <A49B2950111D4F7CA5CBBC805B21B970@e1705>

possible that you have somwhere redundant SQL request from your script or 
dialplan...

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, October 27, 2011 2:15 AM
Subject: Re: [Freeswitch-users] High load on database server


> try the version 2.3.0 of ODBC (compiled from sources)
> I can't tell you if it will work because I never had more than 20 
> councurrent calls
> but it should be better anyeway
>
> ----- Original Message ----- 
> From: "Cliff Wells" <cliff at develix.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, October 27, 2011 2:02 AM
> Subject: Re: [Freeswitch-users] High load on database server
>
>
>> On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote:
>>> if you have older unixodbc edit /etc/odbcinst.ini and add the param
>>> Threading=0 to each db entry.
>>
>>> In general postgres to me appears to care more about multithreaded
>>> environments and with the above threading=0 change works pretty well.
>>
>> I tried this (after googling a bit and coming across an older thread you
>> responded to) and it didn't change anything.  Crashes at ~330 concurrent
>> calls like clockwork.  This was with unixODBC 2.14 and PostgreSQL 8.4
>> odbc driver.
>>
>> Regards,
>> Cliff
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 cliff at develix.com  Thu Oct 27 10:32:29 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 26 Oct 2011 23:32:29 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1D58D1DAD92C4E8D9FBFDE2F44F266A3@e1705>
References: <4EA5DA64.1060503@communicatefreely.net>
	<1319494283.4593.160.camel@portable-evil>
	<CAKbxfG8-z9HbfRE6E3fSiHW-xreF9Gaubx5E3KAEi9SxF+t7xQ@mail.gmail.com>
	<CAJh8XdVE1TmgDsy02ZmydFMBJXMbbXnBd4KujMYjQ+2Wh5Thxg@mail.gmail.com>
	<CAKbxfG-bhSusFBwPvo9JvvOMGjpB7C8dmHY1VshynQFNeNximg@mail.gmail.com>
	<543606611-1319565003-cardhu_decombobulator_blackberry.rim.net-1845868528-@b13.c27.bise6.blackberry>
	<1319567530.2234.16.camel@portable-evil>
	<CAJh8XdXVUkfMH5Ud9NaHCXV18UyBf3RgwBtgWxVihz6ASmFccQ@mail.gmail.com>
	<4EA7E384.9000103@puzzled.xs4all.nl>
	<CAKbxfG_X_-eQ8jNand_E9yZfroPDMaiotNk9Vr4MB+rPAf=oyg@mail.gmail.com>
	<1319695366.2234.19.camel@portable-evil>
	<1D58D1DAD92C4E8D9FBFDE2F44F266A3@e1705>
Message-ID: <1319697149.2234.23.camel@portable-evil>

On Thu, 2011-10-27 at 02:15 -0400, Madovsky wrote:
> try the version 2.3.0 of ODBC (compiled from sources)

Actually, I went ahead and built an rpm for 2.3.0 (I needed to satisfy
the dependency for postgresql-odbc rpm).  Unfortunately I haven't yet
been able to test up to 330 concurrent calls since then because we ended
up splitting the traffic across multiple servers to resolve the issue so
no server is seeing more than 80-100 concurrent calls at this point.

Cliff



From cliff at develix.com  Thu Oct 27 10:34:28 2011
From: cliff at develix.com (Cliff Wells)
Date: Wed, 26 Oct 2011 23:34:28 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
Message-ID: <1319697268.2234.25.camel@portable-evil>

On Thu, 2011-10-27 at 02:21 -0400, Madovsky wrote:
> possible that you have somwhere redundant SQL request from your script or 
> dialplan...

As I mentioned previously, we also have some applications (mod_lua) that
utilize the internal odbc pool, so the concurrency is somewhat higher
(but not a lot, maybe another 20%).

Regards,
Cliff



From avi at avimarcus.net  Thu Oct 27 10:51:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 27 Oct 2011 08:51:46 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1319697268.2234.25.camel@portable-evil>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<1319697268.2234.25.camel@portable-evil>
Message-ID: <CALNZuwnTLhL2F8as5Ragm5sPVMbNFXL-1Rx+3ONGdukndR=XqQ@mail.gmail.com>

You should be able to test the DB connections w/o live traffic..
-Avi


On Thu, Oct 27, 2011 at 8:34 AM, Cliff Wells <cliff at develix.com> wrote:

> On Thu, 2011-10-27 at 02:21 -0400, Madovsky wrote:
> > possible that you have somwhere redundant SQL request from your script or
> > dialplan...
>
> As I mentioned previously, we also have some applications (mod_lua) that
> utilize the internal odbc pool, so the concurrency is somewhat higher
> (but not a lot, maybe another 20%).
>
> Regards,
> Cliff
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chrisbware at interfree.it  Thu Oct 27 11:26:37 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 27 Oct 2011 07:26:37 -0000
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
Message-ID: <20111027072637.26979.qmail@community22.interfree.it>

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From hynek.cihlar at gmail.com  Thu Oct 27 11:46:26 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Thu, 27 Oct 2011 09:46:26 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
Message-ID: <2934644557544732041@unknownmsgid>

What does it mean? That I cannot access the DB from scripts?

Sent from my mobile device

On Oct 27, 2011, at 8:21, "Madovsky" <infos at madovsky.org> wrote:

> possible that you have somwhere redundant SQL request from your script or
> dialplan...
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, October 27, 2011 2:15 AM
> Subject: Re: [Freeswitch-users] High load on database server
>
>
>> try the version 2.3.0 of ODBC (compiled from sources)
>> I can't tell you if it will work because I never had more than 20
>> councurrent calls
>> but it should be better anyeway
>>
>> ----- Original Message -----
>> From: "Cliff Wells" <cliff at develix.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, October 27, 2011 2:02 AM
>> Subject: Re: [Freeswitch-users] High load on database server
>>
>>
>>> On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote:
>>>> if you have older unixodbc edit /etc/odbcinst.ini and add the param
>>>> Threading=0 to each db entry.
>>>
>>>> In general postgres to me appears to care more about multithreaded
>>>> environments and with the above threading=0 change works pretty well.
>>>
>>> I tried this (after googling a bit and coming across an older thread you
>>> responded to) and it didn't change anything.  Crashes at ~330 concurrent
>>> calls like clockwork.  This was with unixODBC 2.14 and PostgreSQL 8.4
>>> odbc driver.
>>>
>>> Regards,
>>> Cliff
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 faisal.rehman22 at hotmail.com  Thu Oct 27 12:02:19 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Thu, 27 Oct 2011 13:02:19 +0500
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>,
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>,
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>,
	<CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>
Message-ID: <BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>


Hi David,
Thanks for sharing the script but will that work too for Asterisk's Master.csv if I needed the ASR & ACD records from that file?

Regards,
Faisal Rehman

From: david.villasmil.work at gmail.com
Date: Wed, 26 Oct 2011 19:48:57 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring

Hello,
That's a faily simple bash script. 
"cat" the log file to an awk script like so;
cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{ FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT: "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR; if(ASR<100){ system("echo send email");} }' 


the "system" is where you put the mail command
also, bear in mind that the master can get VERY large if you don't rotate it.
david



On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:







Hi Waseem,
Thanks for your help, I saw the scripts that were on the link that you provided to me but actually the problem is that I don't want to involve the database though it is a bit difficult task but according to my requirement I just want to read the whole Master.csv file & if the ASR or ACD is low on any server, it should send me email, so my question is how can I skip the database from it to interact directly with the Master.csv file.



Thanks in Advance!!
Faisal Rehman

From: wasim at convergence.pk
Date: Tue, 25 Oct 2011 07:32:35 +0500


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring



Faisal:
Load up the CDR into a db (MySQL or otherwise) ...
For ACD you'll want a SQL query akinSELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec, SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE disposition=1;



For ASR you'll need another querySELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;ASR is calls_answered/calls_totalThere is sample (read: poor) code in http://jira.freeswitch.org/browse/FS-1981



which a google for "freeswitch ACD ASR" would have helped you get to in the first place.
-wasim
On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:









Hi Everyone,
I want to write a script that reads the Master.csv & generate an alert if the ASR & ACD is low from the desired value. Anybody has written such script, please help me write it or provide any sample.





Thanks

Faisal Rehman 		 	   		  



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-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | peace be upon you ...





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From miha at softnet.si  Thu Oct 27 12:26:48 2011
From: miha at softnet.si (Miha Zoubek)
Date: Thu, 27 Oct 2011 10:26:48 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <4EA80C5F.6020501@softnet.si>
References: <4EA80C5F.6020501@softnet.si>
Message-ID: <4EA915C8.80809@softnet.si>

On 10/26/2011 3:34 PM, Miha Zoubek wrote:
> Hi,
>
> I am new Freeswitch user. I have read freeswitch book. I registered 
> first few phones and make few inbound calls.
> I am trying to get outgoing calls working but having no luck.
> I have configured my gw and set up calling plan. When I try to dial 
> number 051357952 I get this:
>
> reeswitch at localhost.localdomain> 2011-10-26 15:27:25.919562 [NOTICE] 
> switch_channel.c:669 New Channel sofia/internal/1030 at 172.31.1.157 
> [f2132149-dc40-4601-bd34-c22e5b06b696]
> 2011-10-26 15:27:25.955550 [INFO] mod_dialplan_xml.c:418 Processing 
> 1030->9051357952 in context default
> 2011-10-26 15:27:25.960548 [NOTICE] switch_channel.c:669 New Channel 
> sofia/external/051357952 [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
> 2011-10-26 15:27:26.095518 [NOTICE] sofia.c:4789 Hangup 
> sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
> 2011-10-26 15:27:26.098534 [INFO] mod_dptools.c:2355 Originate 
> Failed.  Cause: NO_ROUTE_DESTINATION
> 2011-10-26 15:27:26.098534 [NOTICE] mod_dptools.c:2418 Hangup 
> sofia/internal/1030 at 172.31.1.157 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1182 Session 
> 49 (sofia/external/051357952) Ended
> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1184 Close 
> Channel sofia/external/051357952 [CS_DESTROY]
> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1182 Session 
> 48 (sofia/internal/1030 at 172.31.1.157) Ended
> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1184 Close 
> Channel sofia/internal/1030 at 172.31.1.157 [CS_DESTROY]
>
> GW:
>
> <include>
> <gateway name="IP800">
> <!-- Your username, which is not your DID -->
> <param name="username" value="Demo2"/>
> <param name="password" value="1234"/>
> <param name="register" value="true"/>
> <param name="realm" value="172.31.1.200"/>
> </gateway>
> </include>
>
> This is my dial plan:
>
> <include>
> <extension name="Dial Out Custom Gateway">
> <condition field="destination_number" expression="^9(\d{9})$">
> <action application="bridge" data="sofia/gateway/IP800/$1"/>
> </condition>
> </extension>
> </include>
>
>
> sofia status
>
>                      Name          
> Type                                       Data      State
> =================================================================================================
>                  external       profile 
> sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>     external::example.com       gateway sip:joeuser at example.com      NOREG
>           external::IP800       gateway sip:Demo2 at 172.31.1.200      REGED
>                  internal       profile 
> sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>              172.31.1.157         
> alias                                   internal      ALIASED
> =================================================================================================
> 2 profiles 1 alias
>
>
> Please help me why I am getting not route to destination.
>
> Thank you very much for all your help!
>
> BR,
> Miha
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Please, could any one just point me on right direction:)

Thanks!

BR,
Miha
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From steve.kurzeja at gmail.com  Thu Oct 27 13:10:56 2011
From: steve.kurzeja at gmail.com (Steve Kurzeja)
Date: Thu, 27 Oct 2011 22:10:56 +1300
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <CAFmdODjcZ8tw1js5gZbSDYz1jyQfYsoRu+j64hS2z8bdubdX0A@mail.gmail.com>
References: <CAFmdODi1wWM-xtJ+Qag3V7Tjjc1G_vSXwXrm8k3S59su=CXfBg@mail.gmail.com>
	<CAKDfjgck+A_bQfHtHo2p=tUo6wwO-S5kir6u+mnh86CeSCa0ww@mail.gmail.com>
	<CAFmdODjcZ8tw1js5gZbSDYz1jyQfYsoRu+j64hS2z8bdubdX0A@mail.gmail.com>
Message-ID: <CAENkwGtj7ka7e0xDSD7=NRFXOfpwWimqV2TV_mgoux2Gq0fZKA@mail.gmail.com>

Hi Michael, this is the behavior I've seen if there's a bad IANA name in the
rtpmap for G729 and from your logs it looks like this is the case - G729a is
being presented (a=rtpmap:18 G729a/8000)

Turn on bad IANA name support (
http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-bad-iananame)  in your SIP
profile(s).

<param name="NDLB-allow-bad-iananame" value="true"/>

Regards,
Steve


On Thu, Oct 27, 2011 at 11:48 AM, Michael Toop <michaelt at callall.co.za>wrote:

> Hi Kristian,
>
>  Thanks for your input, please find trace attached as requested with sofia
> SIP trace on.
>
>  I have gone through it again & don't see the SDP arriving wrong.
>
>  Let me know what else I can do to assist.
>
> Kind Regards,
>
> Michael
>
>
> On Wed, Oct 26, 2011 at 5:58 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> Can you post a console log with siptrace enabled:
>>
>> sofia global siptrace on
>>
>> It looks like FreeSWITCH is getting that SDP from somewhere else.  0
>> as the port number, G729 as dynamic payload type 96...  It's
>> completely messed up and I've never seen FreeSWITCH generate an SDP
>> like this.
>>
>> On Wed, Oct 26, 2011 at 8:00 AM, Michael Toop <michaelt at callall.co.za>
>> wrote:
>> > Hi,
>> >  When initiating a call from Freeswitch to a provider, Freeswitch is
>> putting
>> > '0'/ zero in the media port.  This looks like a bug to me.:
>> >    m=audio 0 RTP/AVP 96 101
>> > Media Description, name and address (m): audio 0 RTP/AVP 96 101
>> >
>> >  Txt based SIP trace attached (not sure how to scrub pcap files of IPs).
>> > Thanks,
>> > Michael
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gcd at i.ph  Thu Oct 27 13:19:09 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 27 Oct 2011 17:19:09 +0800
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <4EA80C5F.6020501@softnet.si>
References: <4EA80C5F.6020501@softnet.si>
Message-ID: <CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>

hi,

what is IP800? is this a FXO gateway? or a SIP provider?

u hv to dial 9051357952 to match the dialplan.

-nandy

On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek <miha at softnet.si> wrote:

>  Hi,
>
> I am new Freeswitch user. I have read freeswitch book. I registered first
> few phones and make few inbound calls.
> I am trying to get outgoing calls working but having no luck.
> I have configured my gw and set up calling plan. When I try to dial number
> 051357952 I get this:
>
> reeswitch at localhost.localdomain> 2011-10-26 15:27:25.919562 [NOTICE]
> switch_channel.c:669 New Channel sofia/internal/1030 at 172.31.1.157[f2132149-dc40-4601-bd34-c22e5b06b696]
> 2011-10-26 15:27:25.955550 [INFO] mod_dialplan_xml.c:418 Processing 1030->
> 9051357952 in context default
> 2011-10-26 15:27:25.960548 [NOTICE] switch_channel.c:669 New Channel sofia/external/051357952
> [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
> 2011-10-26 15:27:26.095518 [NOTICE] sofia.c:4789 Hangup sofia/external/051357952
> [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
> 2011-10-26 15:27:26.098534 [INFO] mod_dptools.c:2355 Originate Failed.
> Cause: NO_ROUTE_DESTINATION
> 2011-10-26 15:27:26.098534 [NOTICE] mod_dptools.c:2418 Hangup sofia
> /internal/1030 at 172.31.1.157 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1182 Session 49
> (sofia/external/051357952) Ended
> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/external/051357952 [CS_DESTROY]
> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1182 Session 48
> (sofia/internal/1030 at 172.31.1.157) Ended
> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1030 at 172.31.1.157 [CS_DESTROY]
>
> GW:
>
> <include>
>   <gateway name="IP800">
>     <!-- Your username, which is not your DID -->
>     <param name="username" value="Demo2"/>
>     <param name="password" value="1234"/>
>     <param name="register" value="true"/>
>     <param name="realm" value="172.31.1.200"/>
>  </gateway>
> </include>
>
> This is my dial plan:
>
> <include>
>         <extension name="Dial Out Custom Gateway">
>                 <condition field="destination_number"
> expression="^9(\d{9})$">
>         <action application="bridge" data="sofia/gateway/IP800/$1"/>
>                 </condition>
>         </extension>
> </include>
>
>
> sofia status
>
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>                  external       profile
> sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com      NOREG
>           external::IP800       gateway
> sip:Demo2 at 172.31.1.200      REGED
>                  internal       profile
> sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>              172.31.1.157         alias
> internal      ALIASED
>
> =================================================================================================
> 2 profiles 1 alias
>
>
> Please help me why I am getting not route to destination.
>
> Thank you very much for all your help!
>
> BR,
> Miha
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pkelly at gmail.com  Thu Oct 27 13:25:30 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Thu, 27 Oct 2011 10:25:30 +0100
Subject: [Freeswitch-users] ./configure fails on vanilla checkout
Message-ID: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>

Hi

I just checked out freeswitch from Git (git clone git://
git.freeswitch.org/freeswitch.git), did ./bootstrap.sh, commented out the
sangoma codec module from modules.conf and ran ./configure.

The ./configure fails when it gets to libs/iksemel/configure +11073

It complains about the syntax of the PKG_CHECK_MODULE statement.

I commented out the whole if statement and it configures fine.

Pete
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From michaelt at callall.co.za  Thu Oct 27 13:34:44 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Thu, 27 Oct 2011 11:34:44 +0200
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <20111027072637.26979.qmail@community22.interfree.it>
References: <20111027072637.26979.qmail@community22.interfree.it>
Message-ID: <CAFmdODgcPWv2T5LmaYtxHRDMeHsQzTdfCmNOW4exZQ1LMyaihw@mail.gmail.com>

Hi,

 Start & stop RTP ports are commented out in switch.conf so must be the
defaults, 16384 & 32768.

 I have also tried the NDLB-allow-bad-iananame suggestion & have the same
results, see SIP trace here: http://pastebin.com/vdztr7cf

 What is very strange about this issue it only happens on certain routes
i.e. the SDP of the B leg of the call is somehow breaking things.  Really at
a loss here.

Cheers,

Michael



On Thu, Oct 27, 2011 at 9:26 AM, <chrisbware at interfree.it> wrote:

> Even payload type 96 for G729 sounds strange to me.
> How did you set  *rtp-start-portand ***
> ***rtp-end-port ?*
> **
> ***
> *
> **
> ***
> *
> -----Messaggio originale-----
> Da: Michael Toop <michaelt at callall.co.za>
> Inviato il: 26 Ott 2011 - 05:00
> A: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
>
> Hi,
>  When initiating a call from Freeswitch to a provider, Freeswitch is
> putting '0'/ zero in the media port.  This looks like a bug to me.:
>    m=audio 0 RTP/AVP 96 101
> Media Description, name and address (m): audio 0 RTP/AVP 96 101
>  Txt based SIP trace attached (not sure how to scrub pcap files of IPs).
> Thanks,
> Michael
>
>
> -------------------------------------------------------------------------------
> Valore legale alle tue mail
> InterfreePEC - la tua Posta Elettronica Certificata
> http://pec.interfree.it
>
> -------------------------------------------------------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miha at softnet.si  Thu Oct 27 14:07:05 2011
From: miha at softnet.si (Miha Zoubek)
Date: Thu, 27 Oct 2011 12:07:05 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
Message-ID: <4EA92D49.5030909@softnet.si>

Hi,

I have dialed 9051357952. IP800 is our GW. I created on it one SIP 
account and connect freeswitch with it.

BR,
Miha

On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
> hi,
>
> what is IP800? is this a FXO gateway? or a SIP provider?
>
> u hv to dial 9051357952 to match the dialplan.
>
> -nandy
>
> On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     Hi,
>
>     I am new Freeswitch user. I have read freeswitch book. I
>     registered first few phones and make few inbound calls.
>     I am trying to get outgoing calls working but having no luck.
>     I have configured my gw and set up calling plan. When I try to
>     dial number 051357952 I get this:
>
>     reeswitch at localhost.localdomain> 2011-10-26 <tel:2011-10-26>
>     15:27:25.919562 <tel:25.919562> [NOTICE] switch_channel.c:669 New
>     Channel sofia/internal/1030 at 172.31.1.157
>     <mailto:1030 at 172.31.1.157> [f2132149-dc40-4601-bd34-c22e5b06b696]
>     2011-10-26 <tel:2011-10-26> 15:27:25.955550 <tel:25.955550> [INFO]
>     mod_dialplan_xml.c:418 Processing 1030->9051357952
>     <tel:9051357952> in context default
>     2011-10-26 <tel:2011-10-26> 15:27:25.960548 <tel:25.960548>
>     [NOTICE] switch_channel.c:669 New Channel sofia/external/051357952
>     [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>     2011-10-26 <tel:2011-10-26> 15:27:26.095518 <tel:26.095518>
>     [NOTICE] sofia.c:4789 Hangup sofia/external/051357952
>     [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>     2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534> [INFO]
>     mod_dptools.c:2355 Originate Failed.  Cause: NO_ROUTE_DESTINATION
>     2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>     [NOTICE] mod_dptools.c:2418 Hangup
>     sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>
>     [CS_EXECUTE] [NO_ROUTE_DESTINATION]
>     2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>     [NOTICE] switch_core_session.c:1182 Session 49
>     (sofia/external/051357952) Ended
>     2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>     [NOTICE] switch_core_session.c:1184 Close Channel
>     sofia/external/051357952 [CS_DESTROY]
>     2011-10-26 <tel:2011-10-26> 15:27:26.140506 <tel:26.140506>
>     [NOTICE] switch_core_session.c:1182 Session 48
>     (sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>) Ended
>     2011-10-26 <tel:2011-10-26> 15:27:26.140506 <tel:26.140506>
>     [NOTICE] switch_core_session.c:1184 Close Channel
>     sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>
>     [CS_DESTROY]
>
>     GW:
>
>     <include>
>     <gateway name="IP800">
>     <!-- Your username, which is not your DID -->
>     <param name="username" value="Demo2"/>
>     <param name="password" value="1234"/>
>     <param name="register" value="true"/>
>     <param name="realm" value="172.31.1.200"/>
>     </gateway>
>     </include>
>
>     This is my dial plan:
>
>     <include>
>     <extension name="Dial Out Custom Gateway">
>     <condition field="destination_number" expression="^9(\d{9})$">
>     <action application="bridge" data="sofia/gateway/IP800/$1"/>
>     </condition>
>     </extension>
>     </include>
>
>
>     sofia status
>
>                          Name         
>     Type                                       Data      State
>     =================================================================================================
>                      external       profile
>     sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>         external::example.com <http://example.com>       gateway
>     sip:joeuser at example.com      NOREG
>               external::IP800       gateway
>     sip:Demo2 at 172.31.1.200      REGED
>                      internal       profile
>     sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>                  172.31.1.157        
>     alias                                   internal      ALIASED
>     =================================================================================================
>     2 profiles 1 alias
>
>
>     Please help me why I am getting not route to destination.
>
>     Thank you very much for all your help!
>
>     BR,
>     Miha
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gcd at i.ph  Thu Oct 27 16:15:22 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 27 Oct 2011 20:15:22 +0800
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <4EA92D49.5030909@softnet.si>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
	<4EA92D49.5030909@softnet.si>
Message-ID: <CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>

what is the full path/filename of your dialplan?
your debug level is not detailed enough.  set the log level to 7.
$fs_cli -l 7


On Thu, Oct 27, 2011 at 6:07 PM, Miha Zoubek <miha at softnet.si> wrote:

>  Hi,
>
> I have dialed 9051357952. IP800 is our GW. I created on it one SIP account
> and connect freeswitch with it.
>
> BR,
> Miha
>
>
> On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
>
>  hi,
>
>  what is IP800? is this a FXO gateway? or a SIP provider?
>
>  u hv to dial 9051357952 to match the dialplan.
>
>  -nandy
>
> On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek <miha at softnet.si> wrote:
>
>>  Hi,
>>
>> I am new Freeswitch user. I have read freeswitch book. I registered first
>> few phones and make few inbound calls.
>> I am trying to get outgoing calls working but having no luck.
>> I have configured my gw and set up calling plan. When I try to dial
>> number 051357952 I get this:
>>
>> reeswitch at localhost.localdomain> 2011-10-26 15:27:25.919562 [NOTICE]
>> switch_channel.c:669 New Channel sofia/internal/1030 at 172.31.1.157[f2132149-dc40-4601-bd34-c22e5b06b696]
>> 2011-10-26 15:27:25.955550 [INFO] mod_dialplan_xml.c:418 Processing
>> 1030->9051357952 in context default
>> 2011-10-26 15:27:25.960548 [NOTICE] switch_channel.c:669 New Channel
>> sofia/external/051357952 [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>> 2011-10-26 15:27:26.095518 [NOTICE] sofia.c:4789 Hangup sofia/external/051357952
>> [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>> 2011-10-26 15:27:26.098534 [INFO] mod_dptools.c:2355 Originate Failed.
>> Cause: NO_ROUTE_DESTINATION
>> 2011-10-26 15:27:26.098534 [NOTICE] mod_dptools.c:2418 Hangup sofia
>> /internal/1030 at 172.31.1.157 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
>> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1182 Session 49
>> (sofia/external/051357952) Ended
>> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/external/051357952 [CS_DESTROY]
>> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1182 Session 48
>> (sofia/internal/1030 at 172.31.1.157) Ended
>> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/internal/1030 at 172.31.1.157 [CS_DESTROY]
>>
>> GW:
>>
>> <include>
>>   <gateway name="IP800">
>>     <!-- Your username, which is not your DID -->
>>     <param name="username" value="Demo2"/>
>>     <param name="password" value="1234"/>
>>     <param name="register" value="true"/>
>>     <param name="realm" value="172.31.1.200"/>
>>  </gateway>
>> </include>
>>
>> This is my dial plan:
>>
>> <include>
>>         <extension name="Dial Out Custom Gateway">
>>                 <condition field="destination_number"
>> expression="^9(\d{9})$">
>>         <action application="bridge" data="sofia/gateway/IP800/$1"/>
>>                 </condition>
>>         </extension>
>> </include>
>>
>>
>> sofia status
>>
>>                      Name
>> Type                                       Data      State
>>
>> =================================================================================================
>>                  external       profile
>> sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>>     external::example.com       gateway
>> sip:joeuser at example.com      NOREG
>>           external::IP800       gateway
>> sip:Demo2 at 172.31.1.200      REGED
>>                  internal       profile
>> sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>>              172.31.1.157         alias
>> internal      ALIASED
>>
>> =================================================================================================
>> 2 profiles 1 alias
>>
>>
>> Please help me why I am getting not route to destination.
>>
>> Thank you very much for all your help!
>>
>> BR,
>>  Miha
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miha at softnet.si  Thu Oct 27 16:57:16 2011
From: miha at softnet.si (Miha Zoubek)
Date: Thu, 27 Oct 2011 14:57:16 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
	<4EA92D49.5030909@softnet.si>
	<CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>
Message-ID: <4EA9552C.2030205@softnet.si>

Hi,

path of my dialplan is: /usr/local/freeswitch/conf/dialplan/default/test.xml



I rename IP800 for innovaphone.

Here is my debugging log:

Dialplan: sofia/internal/1030 at 172.31.1.192 Regex (FAIL) [hold_music] 
destination_number(9051357952) =~ /^9999$/ break=on-false
Dialplan: sofia/internal/1030 at 172.31.1.192 parsing [default->Dial Out 
Custom Gateway] continue=false
Dialplan: sofia/internal/1030 at 172.31.1.192 Regex (PASS) [Dial Out Custom 
Gateway] destination_number(9051357952) =~ /^9(\d{9})$/ break=on-false
Dialplan: sofia/internal/1030 at 172.31.1.192 Action 
bridge(sofia/gateway/innovaphone/051357952)
2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:119 
(sofia/internal/1030 at 172.31.1.192) State Change CS_ROUTING -> CS_EXECUTE
2011-10-27 14:54:58.927692 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/internal/1030 at 172.31.1.192 [BREAK]
2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:341 
(sofia/internal/1030 at 172.31.1.192) State ROUTING going to sleep
2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:314 
(sofia/internal/1030 at 172.31.1.192) Running State Change CS_EXECUTE
2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:348 
(sofia/internal/1030 at 172.31.1.192) State EXECUTE
2011-10-27 14:54:58.927692 [DEBUG] mod_sofia.c:226 
sofia/internal/1030 at 172.31.1.192 SOFIA EXECUTE
2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:157 
sofia/internal/1030 at 172.31.1.192 Standard EXECUTE
EXECUTE sofia/internal/1030 at 172.31.1.192 set(open=true)
2011-10-27 14:54:58.927692 [DEBUG] mod_dptools.c:816 
sofia/internal/1030 at 172.31.1.192 SET [open]=[true]
EXECUTE sofia/internal/1030 at 172.31.1.192 
hash(insert/172.31.1.192-spymap/1030/147c8578-500d-4d91-bad5-9391c4400934)
EXECUTE sofia/internal/1030 at 172.31.1.192 
hash(insert/172.31.1.192-last_dial/1030/9051357952)
EXECUTE sofia/internal/1030 at 172.31.1.192 
hash(insert/172.31.1.192-last_dial/global/147c8578-500d-4d91-bad5-9391c4400934)
EXECUTE sofia/internal/1030 at 172.31.1.192 
bridge(sofia/gateway/innovaphone/051357952)
2011-10-27 14:54:58.931685 [NOTICE] switch_channel.c:669 New Channel 
sofia/external/051357952 [c4a81269-04e7-42db-aaf8-ecf528b74d68]
2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:3384 
(sofia/external/051357952) State Change CS_NEW -> CS_INIT
2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/external/051357952 [BREAK]
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/051357952) Running State Change CS_INIT
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:338 
(sofia/external/051357952) State INIT
2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:83 
sofia/external/051357952 SOFIA INIT
2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:117 
(sofia/external/051357952) State Change CS_INIT -> CS_ROUTING
2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/external/051357952 [BREAK]
2011-10-27 14:54:58.933717 [DEBUG] sofia.c:4153 Channel 
sofia/external/051357952 entering state [calling][0]
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:338 
(sofia/external/051357952) State INIT going to sleep
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/051357952) Running State Change CS_ROUTING
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:341 
(sofia/external/051357952) State ROUTING
2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:140 
sofia/external/051357952 SOFIA ROUTING
2011-10-27 14:54:58.933717 [DEBUG] switch_ivr_originate.c:66 
(sofia/external/051357952) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/external/051357952 [BREAK]
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:341 
(sofia/external/051357952) State ROUTING going to sleep
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/051357952) Running State Change CS_CONSUME_MEDIA
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:360 
(sofia/external/051357952) State CONSUME_MEDIA
2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:360 
(sofia/external/051357952) State CONSUME_MEDIA going to sleep
2011-10-27 14:54:58.937691 [DEBUG] sofia.c:4153 Channel 
sofia/external/051357952 entering state [calling][0]
2011-10-27 14:54:59.062721 [DEBUG] sofia.c:4153 Channel 
sofia/external/051357952 entering state [terminated][404]
2011-10-27 14:54:59.062721 [NOTICE] sofia.c:4789 Hangup 
sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2011-10-27 14:54:59.062721 [DEBUG] switch_channel.c:2102 Send signal 
sofia/external/051357952 [KILL]
2011-10-27 14:54:59.062721 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/external/051357952 [BREAK]
2011-10-27 14:54:59.062721 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/051357952) Running State Change CS_HANGUP
2011-10-27 14:54:59.064679 [DEBUG] switch_ivr_originate.c:3228 Originate 
Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
2011-10-27 14:54:59.064679 [INFO] mod_dptools.c:2355 Originate Failed.  
Cause: NO_ROUTE_DESTINATION
2011-10-27 14:54:59.064679 [NOTICE] mod_dptools.c:2418 Hangup 
sofia/internal/1030 at 172.31.1.192 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
2011-10-27 14:54:59.064679 [DEBUG] switch_channel.c:2102 Send signal 
sofia/internal/1030 at 172.31.1.192 [KILL]
2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/internal/1030 at 172.31.1.192 [BREAK]
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:348 
(sofia/internal/1030 at 172.31.1.192) State EXECUTE going to sleep
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:314 
(sofia/internal/1030 at 172.31.1.192) Running State Change CS_HANGUP
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499 
(sofia/internal/1030 at 172.31.1.192) State HANGUP
2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408 
sofia/internal/1030 at 172.31.1.192 Overriding SIP cause 404 with 404 from 
the other leg
2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel 
sofia/internal/1030 at 172.31.1.192 hanging up, cause: NO_ROUTE_DESTINATION
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499 
(sofia/external/051357952) State HANGUP
2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408 
sofia/external/051357952 Overriding SIP cause 404 with 404 from the 
other leg
2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel 
sofia/external/051357952 hanging up, cause: NO_ROUTE_DESTINATION
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:46 
sofia/external/051357952 Standard HANGUP, cause: NO_ROUTE_DESTINATION
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499 
(sofia/external/051357952) State HANGUP going to sleep
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:333 
(sofia/external/051357952) State Change CS_HANGUP -> CS_REPORTING
2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/external/051357952 [BREAK]
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/051357952) Running State Change CS_REPORTING
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:590 
(sofia/external/051357952) State REPORTING
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:53 
sofia/external/051357952 Standard REPORTING, cause: NO_ROUTE_DESTINATION
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:590 
(sofia/external/051357952) State REPORTING going to sleep
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:327 
(sofia/external/051357952) State Change CS_REPORTING -> CS_DESTROY
2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/external/051357952 [BREAK]
2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1164 Session 4 
(sofia/external/051357952) Locked, Waiting on external entities
2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1182 Session 4 
(sofia/external/051357952) Ended
2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/external/051357952 [CS_DESTROY]
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:428 
(sofia/external/051357952) Running State Change CS_DESTROY
2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/051357952) State DESTROY
2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:341 
sofia/external/051357952 SOFIA DESTROY
2011-10-27 14:54:59.066681 [DEBUG] switch_core_state_machine.c:60 
sofia/external/051357952 Standard DESTROY
2011-10-27 14:54:59.066681 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/051357952) State DESTROY going to sleep
2011-10-27 14:54:59.108644 [DEBUG] mod_sofia.c:476 Responding to INVITE 
with: 404
2011-10-27 14:54:59.108644 [DEBUG] switch_core_state_machine.c:46 
sofia/internal/1030 at 172.31.1.192 Standard HANGUP, cause: 
NO_ROUTE_DESTINATION
2011-10-27 14:54:59.108644 [DEBUG] switch_core_state_machine.c:499 
(sofia/internal/1030 at 172.31.1.192) State HANGUP going to sleep
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:333 
(sofia/internal/1030 at 172.31.1.192) State Change CS_HANGUP -> CS_REPORTING
2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/internal/1030 at 172.31.1.192 [BREAK]
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:314 
(sofia/internal/1030 at 172.31.1.192) Running State Change CS_REPORTING
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:590 
(sofia/internal/1030 at 172.31.1.192) State REPORTING
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:53 
sofia/internal/1030 at 172.31.1.192 Standard REPORTING, cause: 
NO_ROUTE_DESTINATION
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:590 
(sofia/internal/1030 at 172.31.1.192) State REPORTING going to sleep
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:327 
(sofia/internal/1030 at 172.31.1.192) State Change CS_REPORTING -> CS_DESTROY
2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/internal/1030 at 172.31.1.192 [BREAK]
2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1164 Session 3 
(sofia/internal/1030 at 172.31.1.192) Locked, Waiting on external entities
2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1182 Session 3 
(sofia/internal/1030 at 172.31.1.192) Ended
2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1030 at 172.31.1.192 [CS_DESTROY]
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:428 
(sofia/internal/1030 at 172.31.1.192) Running State Change CS_DESTROY
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:439 
(sofia/internal/1030 at 172.31.1.192) State DESTROY
2011-10-27 14:54:59.110650 [DEBUG] mod_sofia.c:341 
sofia/internal/1030 at 172.31.1.192 SOFIA DESTROY
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:60 
sofia/internal/1030 at 172.31.1.192 Standard DESTROY
2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:439 
(sofia/internal/1030 at 172.31.1.192) State DESTROY going to slee


BR,
Miha

On 10/27/2011 2:15 PM, Nandy Dagondon wrote:
> what is the full path/filename of your dialplan?
> your debug level is not detailed enough.  set the log level to 7.
> $fs_cli -l 7
>
>
> On Thu, Oct 27, 2011 at 6:07 PM, Miha Zoubek <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     Hi,
>
>     I have dialed 9051357952 <tel:9051357952>. IP800 is our GW. I
>     created on it one SIP account and connect freeswitch with it.
>
>     BR,
>     Miha
>
>
>     On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
>>     hi,
>>
>>     what is IP800? is this a FXO gateway? or a SIP provider?
>>
>>     u hv to dial 9051357952 <tel:9051357952> to match the dialplan.
>>
>>     -nandy
>>
>>     On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek <miha at softnet.si
>>     <mailto:miha at softnet.si>> wrote:
>>
>>         Hi,
>>
>>         I am new Freeswitch user. I have read freeswitch book. I
>>         registered first few phones and make few inbound calls.
>>         I am trying to get outgoing calls working but having no luck.
>>         I have configured my gw and set up calling plan. When I try
>>         to dial number 051357952 I get this:
>>
>>         reeswitch at localhost.localdomain> 2011-10-26 <tel:2011-10-26>
>>         15:27:25.919562 <tel:25.919562> [NOTICE] switch_channel.c:669
>>         New Channel sofia/internal/1030 at 172.31.1.157
>>         <mailto:1030 at 172.31.1.157> [f2132149-dc40-4601-bd34-c22e5b06b696]
>>         2011-10-26 <tel:2011-10-26> 15:27:25.955550 <tel:25.955550>
>>         [INFO] mod_dialplan_xml.c:418 Processing 1030->9051357952
>>         <tel:9051357952> in context default
>>         2011-10-26 <tel:2011-10-26> 15:27:25.960548 <tel:25.960548>
>>         [NOTICE] switch_channel.c:669 New Channel
>>         sofia/external/051357952 [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>>         2011-10-26 <tel:2011-10-26> 15:27:26.095518 <tel:26.095518>
>>         [NOTICE] sofia.c:4789 Hangup sofia/external/051357952
>>         [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>         [INFO] mod_dptools.c:2355 Originate Failed.  Cause:
>>         NO_ROUTE_DESTINATION
>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>         [NOTICE] mod_dptools.c:2418 Hangup
>>         sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>
>>         [CS_EXECUTE] [NO_ROUTE_DESTINATION]
>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>         [NOTICE] switch_core_session.c:1182 Session 49
>>         (sofia/external/051357952) Ended
>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>         [NOTICE] switch_core_session.c:1184 Close Channel
>>         sofia/external/051357952 [CS_DESTROY]
>>         2011-10-26 <tel:2011-10-26> 15:27:26.140506 <tel:26.140506>
>>         [NOTICE] switch_core_session.c:1182 Session 48
>>         (sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>)
>>         Ended
>>         2011-10-26 <tel:2011-10-26> 15:27:26.140506 <tel:26.140506>
>>         [NOTICE] switch_core_session.c:1184 Close Channel
>>         sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>
>>         [CS_DESTROY]
>>
>>         GW:
>>
>>         <include>
>>         <gateway name="IP800">
>>         <!-- Your username, which is not your DID -->
>>         <param name="username" value="Demo2"/>
>>         <param name="password" value="1234"/>
>>         <param name="register" value="true"/>
>>         <param name="realm" value="172.31.1.200"/>
>>         </gateway>
>>         </include>
>>
>>         This is my dial plan:
>>
>>         <include>
>>         <extension name="Dial Out Custom Gateway">
>>         <condition field="destination_number" expression="^9(\d{9})$">
>>         <action application="bridge" data="sofia/gateway/IP800/$1"/>
>>         </condition>
>>         </extension>
>>         </include>
>>
>>
>>         sofia status
>>
>>                              Name         
>>         Type                                       Data      State
>>         =================================================================================================
>>                          external       profile
>>         sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>>             external::example.com <http://example.com>       gateway
>>         sip:joeuser at example.com      NOREG
>>                   external::IP800       gateway
>>         sip:Demo2 at 172.31.1.200      REGED
>>                          internal       profile
>>         sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>>                      172.31.1.157        
>>         alias                                   internal      ALIASED
>>         =================================================================================================
>>         2 profiles 1 alias
>>
>>
>>         Please help me why I am getting not route to destination.
>>
>>         Thank you very much for all your help!
>>
>>         BR,
>>         Miha
>>
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.villasmil.work at gmail.com  Thu Oct 27 17:38:29 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 27 Oct 2011 15:38:29 +0200
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
	<CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>
	<BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>
Message-ID: <CAFGRPVqTuivucF_kptTCF+ufexjKG67ZHzZA5LidSvruYE6ZCg@mail.gmail.com>

Hello,

No, you need to adapt it to asterisk's file format.

David

On Thu, Oct 27, 2011 at 10:02 AM, Faisal Rehman <faisal.rehman22 at hotmail.com
> wrote:

>  Hi David,
>
> Thanks for sharing the script but will that work too for Asterisk's
> Master.csv if I needed the ASR & ACD records from that file?
>
>
> Regards,
>
> Faisal Rehman
>
> ------------------------------
> From: david.villasmil.work at gmail.com
> Date: Wed, 26 Oct 2011 19:48:57 +0200
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
> Hello,
>
> That's a faily simple bash script.
>
> "cat" the log file to an awk script like so;
>
> cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{
> FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT:
> "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR;
> if(ASR<100){ system("echo send email");} }'
>
> the "system" is where you put the mail command
>
> also, bear in mind that the master can get VERY large if you don't rotate
> it.
>
> david
>
> On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman <
> faisal.rehman22 at hotmail.com> wrote:
>
>  *Hi Waseem,*
>
> Thanks for your help, I saw the scripts that were on the link that you
> provided to me but actually the problem is that I don't want to involve the
> database though it is a bit difficult task but according to my requirement I
> just want to read the whole Master.csv file & if the ASR or ACD is low on
> any server, it should send me email, so my question is how can I skip the
> database from it to interact directly with the Master.csv file.
>
>
> Thanks in Advance!!
>
> *Faisal Rehman
> *
> ------------------------------
> From: wasim at convergence.pk
> Date: Tue, 25 Oct 2011 07:32:35 +0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
>
> Faisal:
>
> Load up the CDR into a db (MySQL or otherwise) ...
>
> For ACD you'll want a SQL query akin
>
>    - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
>    SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
>    disposition=1;
>
> For ASR you'll need another query
>
>    - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
>    - ASR is calls_answered/calls_total
>
> There is sample (read: poor) code in
> http://jira.freeswitch.org/browse/FS-1981
> which a google for "freeswitch ACD ASR" would have helped you get to in the
> first place.
>
> -wasim
>
> On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com>wrote:
>
>  Hi Everyone,
>
> I want to write a script that reads the Master.csv & generate an alert if
> the ASR & ACD is low from the desired value. Anybody has written such
> script, please help me write it or provide any sample.
>
>
> Thanks
>
>
> Faisal Rehman
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Oct 27 17:45:52 2011
From: miha at softnet.si (Miha Zoubek)
Date: Thu, 27 Oct 2011 15:45:52 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <4EA9552C.2030205@softnet.si>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
	<4EA92D49.5030909@softnet.si>
	<CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>
	<4EA9552C.2030205@softnet.si>
Message-ID: <4EA96090.2090908@softnet.si>

This log is from cli:

[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1164]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 136
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_session.c
Log-Func: switch_core_session_thread
Log-Line: 1164
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.325859 [DEBUG] switch_core_session.c:1164 Session 8 
(sofia/external/051357952) Locked, Waiting on external entities
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [106]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [5]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_session.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_thread]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1182]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 106
Log-Level: 5
Text-Channel: 3
Log-File: switch_core_session.c
Log-Func: switch_core_session_thread
Log-Line: 1182
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1182 Session 8 
(sofia/external/051357952) Ended
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [115]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [5]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_session.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_thread]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1184]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 115
Log-Level: 5
Text-Channel: 0
Log-File: switch_core_session.c
Log-Func: switch_core_session_thread
Log-Line: 1184
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/external/051357952 [CS_DESTROY]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [126]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_destroy_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[428]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 126
Log-Level: 7
Text-Channel: 0
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_destroy_state
Log-Line: 428
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:428 
(sofia/external/051357952) Running State Change CS_DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [108]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_destroy_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[439]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 108
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_destroy_state
Log-Line: 439
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/051357952) State DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [90]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[mod_sofia.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[sofia_on_destroy]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[341]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 90
Log-Level: 7
Text-Channel: 3
Log-File: mod_sofia.c
Log-Func: sofia_on_destroy
Log-Line: 341
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.328931 [DEBUG] mod_sofia.c:341 
sofia/external/051357952 SOFIA DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [108]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_standard_on_destroy]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[60]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 108
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_standard_on_destroy
Log-Line: 60
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:60 
sofia/external/051357952 Standard DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [123]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_destroy_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[439]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [9240aaac-7466-4a4d-aded-e74998dfeb01]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 123
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_destroy_state
Log-Line: 439
User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01


2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/051357952) State DESTROY going to sleep
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [82]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[mod_sofia.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[sofia_on_hangup]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[476]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 82
Log-Level: 7
Text-Channel: 3
Log-File: mod_sofia.c
Log-Func: sofia_on_hangup
Log-Line: 476
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.355813 [DEBUG] mod_sofia.c:476 Responding to INVITE 
with: 404
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [144]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_standard_on_hangup]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[46]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 144
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_standard_on_hangup
Log-Line: 46
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:46 
sofia/internal/1030 at 172.31.1.192 Standard HANGUP, cause: 
NO_ROUTE_DESTINATION
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [130]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_hangup_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[499]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 130
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_hangup_state
Log-Line: 499
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:499 
(sofia/internal/1030 at 172.31.1.192) State HANGUP going to sleep
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [141]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_run]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[333]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 141
Log-Level: 7
Text-Channel: 0
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_run
Log-Line: 333
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:333 
(sofia/internal/1030 at 172.31.1.192) State Change CS_HANGUP -> CS_REPORTING
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [115]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_session.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_signal_state_change]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1021]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 115
Log-Level: 7
Text-Channel: 0
Log-File: switch_core_session.c
Log-Func: switch_core_session_signal_state_change
Log-Line: 1021
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/internal/1030 at 172.31.1.192 [BREAK]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [136]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_run]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[314]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 136
Log-Level: 7
Text-Channel: 0
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_run
Log-Line: 314
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:314 
(sofia/internal/1030 at 172.31.1.192) Running State Change CS_REPORTING
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [118]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_reporting_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[590]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 118
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_reporting_state
Log-Line: 590
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:590 
(sofia/internal/1030 at 172.31.1.192) State REPORTING
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [147]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_standard_on_reporting]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[53]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 147
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_standard_on_reporting
Log-Line: 53
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:53 
sofia/internal/1030 at 172.31.1.192 Standard REPORTING, cause: 
NO_ROUTE_DESTINATION
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [133]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_reporting_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[590]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 133
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_reporting_state
Log-Line: 590
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:590 
(sofia/internal/1030 at 172.31.1.192) State REPORTING going to sleep
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [142]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_run]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[327]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 142
Log-Level: 7
Text-Channel: 0
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_run
Log-Line: 327
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:327 
(sofia/internal/1030 at 172.31.1.192) State Change CS_REPORTING -> CS_DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [115]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_session.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_signal_state_change]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1021]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 115
Log-Level: 7
Text-Channel: 0
Log-File: switch_core_session.c
Log-Func: switch_core_session_signal_state_change
Log-Line: 1021
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021 Send 
signal sofia/internal/1030 at 172.31.1.192 [BREAK]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [144]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_session.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_thread]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1164]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 144
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_session.c
Log-Func: switch_core_session_thread
Log-Line: 1164
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1164 Session 7 
(sofia/internal/1030 at 172.31.1.192) Locked, Waiting on external entities
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [114]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [5]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_session.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_thread]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1182]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 114
Log-Level: 5
Text-Channel: 3
Log-File: switch_core_session.c
Log-Func: switch_core_session_thread
Log-Line: 1182
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1182 Session 7 
(sofia/internal/1030 at 172.31.1.192) Ended
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [123]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [5]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_session.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_thread]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[1184]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 123
Log-Level: 5
Text-Channel: 0
Log-File: switch_core_session.c
Log-Func: switch_core_session_thread
Log-Line: 1184
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1030 at 172.31.1.192 [CS_DESTROY]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [134]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [0]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_destroy_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[428]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 134
Log-Level: 7
Text-Channel: 0
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_destroy_state
Log-Line: 428
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:428 
(sofia/internal/1030 at 172.31.1.192) Running State Change CS_DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [116]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_destroy_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[439]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 116
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_destroy_state
Log-Line: 439
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:439 
(sofia/internal/1030 at 172.31.1.192) State DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [98]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[mod_sofia.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[sofia_on_destroy]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[341]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 98
Log-Level: 7
Text-Channel: 3
Log-File: mod_sofia.c
Log-Func: sofia_on_destroy
Log-Line: 341
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] mod_sofia.c:341 
sofia/internal/1030 at 172.31.1.192 SOFIA DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [116]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_standard_on_destroy]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[60]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 116
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_standard_on_destroy
Log-Line: 60
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:60 
sofia/internal/1030 at 172.31.1.192 Standard DESTROY
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Content-Length] = [131]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] 
= [7]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER 
[Text-Channel] = [3]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] = 
[switch_core_state_machine.c]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] = 
[switch_core_session_destroy_state]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] = 
[439]
[DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] 
= [962db612-d472-4f8b-88f9-5349f0034660]
[DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 131
Log-Level: 7
Text-Channel: 3
Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_destroy_state
Log-Line: 439
User-Data: 962db612-d472-4f8b-88f9-5349f0034660


2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:439 
(sofia/internal/1030 at 172.31.1.192) State DESTROY going to sleep


On 10/27/2011 2:57 PM, Miha Zoubek wrote:
> Hi,
>
> path of my dialplan is: 
> /usr/local/freeswitch/conf/dialplan/default/test.xml
>
>
>
> I rename IP800 for innovaphone.
>
> Here is my debugging log:
>
> Dialplan: sofia/internal/1030 at 172.31.1.192 Regex (FAIL) [hold_music] 
> destination_number(9051357952) =~ /^9999$/ break=on-false
> Dialplan: sofia/internal/1030 at 172.31.1.192 parsing [default->Dial Out 
> Custom Gateway] continue=false
> Dialplan: sofia/internal/1030 at 172.31.1.192 Regex (PASS) [Dial Out 
> Custom Gateway] destination_number(9051357952) =~ /^9(\d{9})$/ 
> break=on-false
> Dialplan: sofia/internal/1030 at 172.31.1.192 Action 
> bridge(sofia/gateway/innovaphone/051357952)
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:119 
> (sofia/internal/1030 at 172.31.1.192) State Change CS_ROUTING -> CS_EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:341 
> (sofia/internal/1030 at 172.31.1.192) State ROUTING going to sleep
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:314 
> (sofia/internal/1030 at 172.31.1.192) Running State Change CS_EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:348 
> (sofia/internal/1030 at 172.31.1.192) State EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] mod_sofia.c:226 
> sofia/internal/1030 at 172.31.1.192 SOFIA EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:157 
> sofia/internal/1030 at 172.31.1.192 Standard EXECUTE
> EXECUTE sofia/internal/1030 at 172.31.1.192 set(open=true)
> 2011-10-27 14:54:58.927692 [DEBUG] mod_dptools.c:816 
> sofia/internal/1030 at 172.31.1.192 SET [open]=[true]
> EXECUTE sofia/internal/1030 at 172.31.1.192 
> hash(insert/172.31.1.192-spymap/1030/147c8578-500d-4d91-bad5-9391c4400934)
> EXECUTE sofia/internal/1030 at 172.31.1.192 
> hash(insert/172.31.1.192-last_dial/1030/9051357952)
> EXECUTE sofia/internal/1030 at 172.31.1.192 
> hash(insert/172.31.1.192-last_dial/global/147c8578-500d-4d91-bad5-9391c4400934)
> EXECUTE sofia/internal/1030 at 172.31.1.192 
> bridge(sofia/gateway/innovaphone/051357952)
> 2011-10-27 14:54:58.931685 [NOTICE] switch_channel.c:669 New Channel 
> sofia/external/051357952 [c4a81269-04e7-42db-aaf8-ecf528b74d68]
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:3384 
> (sofia/external/051357952) State Change CS_NEW -> CS_INIT
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314 
> (sofia/external/051357952) Running State Change CS_INIT
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:338 
> (sofia/external/051357952) State INIT
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:83 
> sofia/external/051357952 SOFIA INIT
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:117 
> (sofia/external/051357952) State Change CS_INIT -> CS_ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:58.933717 [DEBUG] sofia.c:4153 Channel 
> sofia/external/051357952 entering state [calling][0]
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:338 
> (sofia/external/051357952) State INIT going to sleep
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314 
> (sofia/external/051357952) Running State Change CS_ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:341 
> (sofia/external/051357952) State ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:140 
> sofia/external/051357952 SOFIA ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] switch_ivr_originate.c:66 
> (sofia/external/051357952) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:341 
> (sofia/external/051357952) State ROUTING going to sleep
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314 
> (sofia/external/051357952) Running State Change CS_CONSUME_MEDIA
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:360 
> (sofia/external/051357952) State CONSUME_MEDIA
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:360 
> (sofia/external/051357952) State CONSUME_MEDIA going to sleep
> 2011-10-27 14:54:58.937691 [DEBUG] sofia.c:4153 Channel 
> sofia/external/051357952 entering state [calling][0]
> 2011-10-27 14:54:59.062721 [DEBUG] sofia.c:4153 Channel 
> sofia/external/051357952 entering state [terminated][404]
> 2011-10-27 14:54:59.062721 [NOTICE] sofia.c:4789 Hangup 
> sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
> 2011-10-27 14:54:59.062721 [DEBUG] switch_channel.c:2102 Send signal 
> sofia/external/051357952 [KILL]
> 2011-10-27 14:54:59.062721 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:59.062721 [DEBUG] switch_core_state_machine.c:314 
> (sofia/external/051357952) Running State Change CS_HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] switch_ivr_originate.c:3228 
> Originate Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
> 2011-10-27 14:54:59.064679 [INFO] mod_dptools.c:2355 Originate 
> Failed.  Cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [NOTICE] mod_dptools.c:2418 Hangup 
> sofia/internal/1030 at 172.31.1.192 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_channel.c:2102 Send signal 
> sofia/internal/1030 at 172.31.1.192 [KILL]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:348 
> (sofia/internal/1030 at 172.31.1.192) State EXECUTE going to sleep
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:314 
> (sofia/internal/1030 at 172.31.1.192) Running State Change CS_HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499 
> (sofia/internal/1030 at 172.31.1.192) State HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408 
> sofia/internal/1030 at 172.31.1.192 Overriding SIP cause 404 with 404 
> from the other leg
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel 
> sofia/internal/1030 at 172.31.1.192 hanging up, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499 
> (sofia/external/051357952) State HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408 
> sofia/external/051357952 Overriding SIP cause 404 with 404 from the 
> other leg
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel 
> sofia/external/051357952 hanging up, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:46 
> sofia/external/051357952 Standard HANGUP, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499 
> (sofia/external/051357952) State HANGUP going to sleep
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:333 
> (sofia/external/051357952) State Change CS_HANGUP -> CS_REPORTING
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:314 
> (sofia/external/051357952) Running State Change CS_REPORTING
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:590 
> (sofia/external/051357952) State REPORTING
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:53 
> sofia/external/051357952 Standard REPORTING, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:590 
> (sofia/external/051357952) State REPORTING going to sleep
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:327 
> (sofia/external/051357952) State Change CS_REPORTING -> CS_DESTROY
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1164 Session 
> 4 (sofia/external/051357952) Locked, Waiting on external entities
> 2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1182 Session 
> 4 (sofia/external/051357952) Ended
> 2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1184 Close 
> Channel sofia/external/051357952 [CS_DESTROY]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:428 
> (sofia/external/051357952) Running State Change CS_DESTROY
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:439 
> (sofia/external/051357952) State DESTROY
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:341 
> sofia/external/051357952 SOFIA DESTROY
> 2011-10-27 14:54:59.066681 [DEBUG] switch_core_state_machine.c:60 
> sofia/external/051357952 Standard DESTROY
> 2011-10-27 14:54:59.066681 [DEBUG] switch_core_state_machine.c:439 
> (sofia/external/051357952) State DESTROY going to sleep
> 2011-10-27 14:54:59.108644 [DEBUG] mod_sofia.c:476 Responding to 
> INVITE with: 404
> 2011-10-27 14:54:59.108644 [DEBUG] switch_core_state_machine.c:46 
> sofia/internal/1030 at 172.31.1.192 Standard HANGUP, cause: 
> NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.108644 [DEBUG] switch_core_state_machine.c:499 
> (sofia/internal/1030 at 172.31.1.192) State HANGUP going to sleep
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:333 
> (sofia/internal/1030 at 172.31.1.192) State Change CS_HANGUP -> CS_REPORTING
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:314 
> (sofia/internal/1030 at 172.31.1.192) Running State Change CS_REPORTING
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:590 
> (sofia/internal/1030 at 172.31.1.192) State REPORTING
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:53 
> sofia/internal/1030 at 172.31.1.192 Standard REPORTING, cause: 
> NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:590 
> (sofia/internal/1030 at 172.31.1.192) State REPORTING going to sleep
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:327 
> (sofia/internal/1030 at 172.31.1.192) State Change CS_REPORTING -> CS_DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021 Send 
> signal sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1164 Session 
> 3 (sofia/internal/1030 at 172.31.1.192) Locked, Waiting on external entities
> 2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1182 Session 
> 3 (sofia/internal/1030 at 172.31.1.192) Ended
> 2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1184 Close 
> Channel sofia/internal/1030 at 172.31.1.192 [CS_DESTROY]
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:428 
> (sofia/internal/1030 at 172.31.1.192) Running State Change CS_DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:439 
> (sofia/internal/1030 at 172.31.1.192) State DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] mod_sofia.c:341 
> sofia/internal/1030 at 172.31.1.192 SOFIA DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:60 
> sofia/internal/1030 at 172.31.1.192 Standard DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:439 
> (sofia/internal/1030 at 172.31.1.192) State DESTROY going to slee
>
>
> BR,
> Miha
>
> On 10/27/2011 2:15 PM, Nandy Dagondon wrote:
>> what is the full path/filename of your dialplan?
>> your debug level is not detailed enough.  set the log level to 7.
>> $fs_cli -l 7
>>
>>
>> On Thu, Oct 27, 2011 at 6:07 PM, Miha Zoubek <miha at softnet.si 
>> <mailto:miha at softnet.si>> wrote:
>>
>>     Hi,
>>
>>     I have dialed 9051357952 <tel:9051357952>. IP800 is our GW. I
>>     created on it one SIP account and connect freeswitch with it.
>>
>>     BR,
>>     Miha
>>
>>
>>     On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
>>>     hi,
>>>
>>>     what is IP800? is this a FXO gateway? or a SIP provider?
>>>
>>>     u hv to dial 9051357952 <tel:9051357952> to match the dialplan.
>>>
>>>     -nandy
>>>
>>>     On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek <miha at softnet.si
>>>     <mailto:miha at softnet.si>> wrote:
>>>
>>>         Hi,
>>>
>>>         I am new Freeswitch user. I have read freeswitch book. I
>>>         registered first few phones and make few inbound calls.
>>>         I am trying to get outgoing calls working but having no luck.
>>>         I have configured my gw and set up calling plan. When I try
>>>         to dial number 051357952 I get this:
>>>
>>>         reeswitch at localhost.localdomain> 2011-10-26 <tel:2011-10-26>
>>>         15:27:25.919562 <tel:25.919562> [NOTICE]
>>>         switch_channel.c:669 New Channel
>>>         sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>
>>>         [f2132149-dc40-4601-bd34-c22e5b06b696]
>>>         2011-10-26 <tel:2011-10-26> 15:27:25.955550 <tel:25.955550>
>>>         [INFO] mod_dialplan_xml.c:418 Processing 1030->9051357952
>>>         <tel:9051357952> in context default
>>>         2011-10-26 <tel:2011-10-26> 15:27:25.960548 <tel:25.960548>
>>>         [NOTICE] switch_channel.c:669 New Channel
>>>         sofia/external/051357952 [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>>>         2011-10-26 <tel:2011-10-26> 15:27:26.095518 <tel:26.095518>
>>>         [NOTICE] sofia.c:4789 Hangup sofia/external/051357952
>>>         [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>>         [INFO] mod_dptools.c:2355 Originate Failed.  Cause:
>>>         NO_ROUTE_DESTINATION
>>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>>         [NOTICE] mod_dptools.c:2418 Hangup
>>>         sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>
>>>         [CS_EXECUTE] [NO_ROUTE_DESTINATION]
>>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>>         [NOTICE] switch_core_session.c:1182 Session 49
>>>         (sofia/external/051357952) Ended
>>>         2011-10-26 <tel:2011-10-26> 15:27:26.098534 <tel:26.098534>
>>>         [NOTICE] switch_core_session.c:1184 Close Channel
>>>         sofia/external/051357952 [CS_DESTROY]
>>>         2011-10-26 <tel:2011-10-26> 15:27:26.140506 <tel:26.140506>
>>>         [NOTICE] switch_core_session.c:1182 Session 48
>>>         (sofia/internal/1030 at 172.31.1.157
>>>         <mailto:1030 at 172.31.1.157>) Ended
>>>         2011-10-26 <tel:2011-10-26> 15:27:26.140506 <tel:26.140506>
>>>         [NOTICE] switch_core_session.c:1184 Close Channel
>>>         sofia/internal/1030 at 172.31.1.157 <mailto:1030 at 172.31.1.157>
>>>         [CS_DESTROY]
>>>
>>>         GW:
>>>
>>>         <include>
>>>         <gateway name="IP800">
>>>         <!-- Your username, which is not your DID -->
>>>         <param name="username" value="Demo2"/>
>>>         <param name="password" value="1234"/>
>>>         <param name="register" value="true"/>
>>>         <param name="realm" value="172.31.1.200"/>
>>>         </gateway>
>>>         </include>
>>>
>>>         This is my dial plan:
>>>
>>>         <include>
>>>         <extension name="Dial Out Custom Gateway">
>>>         <condition field="destination_number" expression="^9(\d{9})$">
>>>         <action application="bridge" data="sofia/gateway/IP800/$1"/>
>>>         </condition>
>>>         </extension>
>>>         </include>
>>>
>>>
>>>         sofia status
>>>
>>>                              Name         
>>>         Type                                       Data      State
>>>         =================================================================================================
>>>                          external       profile
>>>         sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>>>             external::example.com <http://example.com>       gateway
>>>         sip:joeuser at example.com      NOREG
>>>                   external::IP800       gateway
>>>         sip:Demo2 at 172.31.1.200      REGED
>>>                          internal       profile
>>>         sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>>>                      172.31.1.157        
>>>         alias                                   internal      ALIASED
>>>         =================================================================================================
>>>         2 profiles 1 alias
>>>
>>>
>>>         Please help me why I am getting not route to destination.
>>>
>>>         Thank you very much for all your help!
>>>
>>>         BR,
>>>         Miha
>>>
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org
>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From etiennedip at gmail.com  Thu Oct 27 10:50:06 2011
From: etiennedip at gmail.com (ET1.D)
Date: Wed, 26 Oct 2011 23:50:06 -0700 (PDT)
Subject: [Freeswitch-users] Get Fax2Mail to work on ubuntu11.04
In-Reply-To: <3E504D46-0896-46EC-8BD9-206709EB3122@gmail.com>
References: <1319624405574-6932179.post@n2.nabble.com>
	<3E504D46-0896-46EC-8BD9-206709EB3122@gmail.com>
Message-ID: <1319698206000-6935271.post@n2.nabble.com>

Ok, I got it working, I had to run the whole make, make install etc process.
Did you manage to get mail2fax working, if you did can you maybe explain to
me how I must go about to get mine working, Don't know where to start and
what to do

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Get-Fax2Mail-to-work-on-ubuntu11-04-tp6932179p6935271.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sos at sokhapkin.dyndns.org  Thu Oct 27 15:51:00 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 27 Oct 2011 07:51:00 -0400
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1319697268.2234.25.camel@portable-evil>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<1319697268.2234.25.camel@portable-evil>
Message-ID: <201110270751.00821.sos@sokhapkin.dyndns.org>

Well, that was the reason why I stopped using lua a while ago. It crashes on 
high load if database queries are involved. I had crashes with both native FS 
odbc support or with luasql.

On Thursday 27 October 2011, Cliff Wells wrote:
> On Thu, 2011-10-27 at 02:21 -0400, Madovsky wrote:
> > possible that you have somwhere redundant SQL request from your script or
> > dialplan...
> 
> As I mentioned previously, we also have some applications (mod_lua) that
> utilize the internal odbc pool, so the concurrency is somewhat higher
> (but not a lot, maybe another 20%).
> 
> Regards,
> Cliff
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Oct 27 19:18:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Oct 2011 08:18:47 -0700
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <4EA96090.2090908@softnet.si>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
	<4EA92D49.5030909@softnet.si>
	<CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>
	<4EA9552C.2030205@softnet.si> <4EA96090.2090908@softnet.si>
Message-ID: <CAKzWOxWRM1xUeJHwYkd7dOwkBELY8jkzjvJaFZFS1gjaQ00Axg@mail.gmail.com>

Okay, a few things. First, let's use pastebin.freeswitch.org since there is
so much information. Second, I recommend turning on sofia sip trace:
sofia global siptrace on

Make another test call and capture the output. Let's see what the other side
sends leading up to the no route message.
-MC

On Thu, Oct 27, 2011 at 6:45 AM, Miha Zoubek <miha at softnet.si> wrote:

>  This log is from cli:
>
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1164]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 136
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_thread
> Log-Line: 1164
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.325859 [DEBUG] switch_core_session.c:1164 Session 8
> (sofia/external/051357952) Locked, Waiting on external entities
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [106]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [5]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_session.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_thread]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1182]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 106
> Log-Level: 5
> Text-Channel: 3
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_thread
> Log-Line: 1182
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1182 Session 8
> (sofia/external/051357952) Ended
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [115]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [5]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_session.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_thread]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1184]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 115
> Log-Level: 5
> Text-Channel: 0
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_thread
> Log-Line: 1184
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/external/051357952 [CS_DESTROY]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [126]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_destroy_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [428]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 126
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_destroy_state
> Log-Line: 428
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:428
> (sofia/external/051357952) Running State Change CS_DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [108]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_destroy_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [439]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 108
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_destroy_state
> Log-Line: 439
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/051357952) State DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [90]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [mod_sofia.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [sofia_on_destroy]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [341]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 90
> Log-Level: 7
> Text-Channel: 3
> Log-File: mod_sofia.c
> Log-Func: sofia_on_destroy
> Log-Line: 341
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.328931 [DEBUG] mod_sofia.c:341 sofia/external/051357952
> SOFIA DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [108]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_standard_on_destroy]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [60]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 108
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_standard_on_destroy
> Log-Line: 60
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:60
> sofia/external/051357952 Standard DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [123]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_destroy_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [439]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [9240aaac-7466-4a4d-aded-e74998dfeb01]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 123
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_destroy_state
> Log-Line: 439
> User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
> 2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/051357952) State DESTROY going to sleep
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [82]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [mod_sofia.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [sofia_on_hangup]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [476]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 82
> Log-Level: 7
> Text-Channel: 3
> Log-File: mod_sofia.c
> Log-Func: sofia_on_hangup
> Log-Line: 476
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.355813 [DEBUG] mod_sofia.c:476 Responding to INVITE
> with: 404
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [144]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_standard_on_hangup]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [46]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 144
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_standard_on_hangup
> Log-Line: 46
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1030 at 172.31.1.192 Standard HANGUP, cause:
> NO_ROUTE_DESTINATION
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [130]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_hangup_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [499]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 130
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_hangup_state
> Log-Line: 499
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:499 (
> sofia/internal/1030 at 172.31.1.192) State HANGUP going to sleep
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [141]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_run]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [333]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 141
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_run
> Log-Line: 333
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:333 (
> sofia/internal/1030 at 172.31.1.192) State Change CS_HANGUP -> CS_REPORTING
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [115]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_session.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_signal_state_change]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1021]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 115
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_signal_state_change
> Log-Line: 1021
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1030 at 172.31.1.192 [BREAK]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [136]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_run]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [314]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 136
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_run
> Log-Line: 314
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1030 at 172.31.1.192) Running State Change CS_REPORTING
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [118]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_reporting_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [590]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 118
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_reporting_state
> Log-Line: 590
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1030 at 172.31.1.192) State REPORTING
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [147]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_standard_on_reporting]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [53]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 147
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_standard_on_reporting
> Log-Line: 53
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1030 at 172.31.1.192 Standard REPORTING, cause:
> NO_ROUTE_DESTINATION
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [133]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_reporting_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [590]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 133
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_reporting_state
> Log-Line: 590
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1030 at 172.31.1.192) State REPORTING going to sleep
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [142]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_run]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [327]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 142
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_run
> Log-Line: 327
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:327 (
> sofia/internal/1030 at 172.31.1.192) State Change CS_REPORTING -> CS_DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [115]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_session.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_signal_state_change]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1021]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 115
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_signal_state_change
> Log-Line: 1021
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1030 at 172.31.1.192 [BREAK]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [144]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_session.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_thread]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1164]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 144
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_thread
> Log-Line: 1164
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1164 Session 7 (
> sofia/internal/1030 at 172.31.1.192) Locked, Waiting on external entities
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [114]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [5]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_session.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_thread]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1182]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 114
> Log-Level: 5
> Text-Channel: 3
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_thread
> Log-Line: 1182
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1182 Session 7 (
> sofia/internal/1030 at 172.31.1.192) Ended
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [123]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [5]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_session.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_thread]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [1184]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 123
> Log-Level: 5
> Text-Channel: 0
> Log-File: switch_core_session.c
> Log-Func: switch_core_session_thread
> Log-Line: 1184
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1030 at 172.31.1.192 [CS_DESTROY]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [134]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_destroy_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [428]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 134
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_destroy_state
> Log-Line: 428
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:428 (
> sofia/internal/1030 at 172.31.1.192) Running State Change CS_DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [116]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_destroy_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [439]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 116
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_destroy_state
> Log-Line: 439
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1030 at 172.31.1.192) State DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [98]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [mod_sofia.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [sofia_on_destroy]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [341]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 98
> Log-Level: 7
> Text-Channel: 3
> Log-File: mod_sofia.c
> Log-Func: sofia_on_destroy
> Log-Line: 341
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] mod_sofia.c:341
> sofia/internal/1030 at 172.31.1.192 SOFIA DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [116]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_standard_on_destroy]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [60]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 116
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_standard_on_destroy
> Log-Line: 60
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1030 at 172.31.1.192 Standard DESTROY
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
> [Content-Length] = [131]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Text-Channel]
> = [3]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-File] =
> [switch_core_state_machine.c]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_core_session_destroy_state]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [Log-Line] =
> [439]
> [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER [User-Data] =
> [962db612-d472-4f8b-88f9-5349f0034660]
> [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 131
> Log-Level: 7
> Text-Channel: 3
> Log-File: switch_core_state_machine.c
> Log-Func: switch_core_session_destroy_state
> Log-Line: 439
> User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
> 2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1030 at 172.31.1.192) State DESTROY going to sleep
>
>
>
> On 10/27/2011 2:57 PM, Miha Zoubek wrote:
>
> Hi,
>
> path of my dialplan is:
> /usr/local/freeswitch/conf/dialplan/default/test.xml
>
>
>
> I rename IP800 for innovaphone.
>
> Here is my debugging log:
>
> Dialplan: sofia/internal/1030 at 172.31.1.192 Regex (FAIL) [hold_music]
> destination_number(9051357952) =~ /^9999$/ break=on-false
> Dialplan: sofia/internal/1030 at 172.31.1.192 parsing [default->Dial Out
> Custom Gateway] continue=false
> Dialplan: sofia/internal/1030 at 172.31.1.192 Regex (PASS) [Dial Out Custom
> Gateway] destination_number(9051357952) =~ /^9(\d{9})$/ break=on-false
> Dialplan: sofia/internal/1030 at 172.31.1.192 Action
> bridge(sofia/gateway/innovaphone/051357952)
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:119 (
> sofia/internal/1030 at 172.31.1.192) State Change CS_ROUTING -> CS_EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:341 (
> sofia/internal/1030 at 172.31.1.192) State ROUTING going to sleep
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1030 at 172.31.1.192) Running State Change CS_EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/1030 at 172.31.1.192) State EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] mod_sofia.c:226
> sofia/internal/1030 at 172.31.1.192 SOFIA EXECUTE
> 2011-10-27 14:54:58.927692 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/1030 at 172.31.1.192 Standard EXECUTE
> EXECUTE sofia/internal/1030 at 172.31.1.192 set(open=true)
> 2011-10-27 14:54:58.927692 [DEBUG] mod_dptools.c:816
> sofia/internal/1030 at 172.31.1.192 SET [open]=[true]
> EXECUTE sofia/internal/1030 at 172.31.1.192hash(insert/172.31.1.192-spymap/1030/147c8578-500d-4d91-bad5-9391c4400934)
> EXECUTE sofia/internal/1030 at 172.31.1.192hash(insert/172.31.1.192-last_dial/1030/9051357952)
> EXECUTE sofia/internal/1030 at 172.31.1.192hash(insert/172.31.1.192-last_dial/global/147c8578-500d-4d91-bad5-9391c4400934)
> EXECUTE sofia/internal/1030 at 172.31.1.192bridge(sofia/gateway/innovaphone/051357952)
> 2011-10-27 14:54:58.931685 [NOTICE] switch_channel.c:669 New Channel
> sofia/external/051357952 [c4a81269-04e7-42db-aaf8-ecf528b74d68]
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:3384
> (sofia/external/051357952) State Change CS_NEW -> CS_INIT
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/051357952) Running State Change CS_INIT
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/051357952) State INIT
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:83 sofia/external/051357952
> SOFIA INIT
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:117
> (sofia/external/051357952) State Change CS_INIT -> CS_ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:58.933717 [DEBUG] sofia.c:4153 Channel
> sofia/external/051357952 entering state [calling][0]
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/051357952) State INIT going to sleep
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/051357952) Running State Change CS_ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/051357952) State ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:140 sofia/external/051357952
> SOFIA ROUTING
> 2011-10-27 14:54:58.933717 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/051357952) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/051357952) State ROUTING going to sleep
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/051357952) Running State Change CS_CONSUME_MEDIA
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/051357952) State CONSUME_MEDIA
> 2011-10-27 14:54:58.933717 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/051357952) State CONSUME_MEDIA going to sleep
> 2011-10-27 14:54:58.937691 [DEBUG] sofia.c:4153 Channel
> sofia/external/051357952 entering state [calling][0]
> 2011-10-27 14:54:59.062721 [DEBUG] sofia.c:4153 Channel
> sofia/external/051357952 entering state [terminated][404]
> 2011-10-27 14:54:59.062721 [NOTICE] sofia.c:4789 Hangup
> sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
> 2011-10-27 14:54:59.062721 [DEBUG] switch_channel.c:2102 Send signal
> sofia/external/051357952 [KILL]
> 2011-10-27 14:54:59.062721 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:59.062721 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/051357952) Running State Change CS_HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] switch_ivr_originate.c:3228 Originate
> Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
> 2011-10-27 14:54:59.064679 [INFO] mod_dptools.c:2355 Originate Failed.
> Cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [NOTICE] mod_dptools.c:2418 Hangup
> sofia/internal/1030 at 172.31.1.192 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_channel.c:2102 Send signal
> sofia/internal/1030 at 172.31.1.192 [KILL]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/1030 at 172.31.1.192) State EXECUTE going to sleep
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1030 at 172.31.1.192) Running State Change CS_HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499 (
> sofia/internal/1030 at 172.31.1.192) State HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408
> sofia/internal/1030 at 172.31.1.192 Overriding SIP cause 404 with 404 from
> the other leg
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
> sofia/internal/1030 at 172.31.1.192 hanging up, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/051357952) State HANGUP
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408 sofia/external/051357952
> Overriding SIP cause 404 with 404 from the other leg
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
> sofia/external/051357952 hanging up, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:46
> sofia/external/051357952 Standard HANGUP, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/051357952) State HANGUP going to sleep
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/051357952) State Change CS_HANGUP -> CS_REPORTING
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/051357952) Running State Change CS_REPORTING
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/051357952) State REPORTING
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:53
> sofia/external/051357952 Standard REPORTING, cause: NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/051357952) State REPORTING going to sleep
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:327
> (sofia/external/051357952) State Change CS_REPORTING -> CS_DESTROY
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/external/051357952 [BREAK]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1164 Session 4
> (sofia/external/051357952) Locked, Waiting on external entities
> 2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1182 Session 4
> (sofia/external/051357952) Ended
> 2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/external/051357952 [CS_DESTROY]
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:428
> (sofia/external/051357952) Running State Change CS_DESTROY
> 2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/051357952) State DESTROY
> 2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:341 sofia/external/051357952
> SOFIA DESTROY
> 2011-10-27 14:54:59.066681 [DEBUG] switch_core_state_machine.c:60
> sofia/external/051357952 Standard DESTROY
> 2011-10-27 14:54:59.066681 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/051357952) State DESTROY going to sleep
> 2011-10-27 14:54:59.108644 [DEBUG] mod_sofia.c:476 Responding to INVITE
> with: 404
> 2011-10-27 14:54:59.108644 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1030 at 172.31.1.192 Standard HANGUP, cause:
> NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.108644 [DEBUG] switch_core_state_machine.c:499 (
> sofia/internal/1030 at 172.31.1.192) State HANGUP going to sleep
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:333 (
> sofia/internal/1030 at 172.31.1.192) State Change CS_HANGUP -> CS_REPORTING
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1030 at 172.31.1.192) Running State Change CS_REPORTING
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1030 at 172.31.1.192) State REPORTING
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1030 at 172.31.1.192 Standard REPORTING, cause:
> NO_ROUTE_DESTINATION
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1030 at 172.31.1.192) State REPORTING going to sleep
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:327 (
> sofia/internal/1030 at 172.31.1.192) State Change CS_REPORTING -> CS_DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1030 at 172.31.1.192 [BREAK]
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1164 Session 3 (
> sofia/internal/1030 at 172.31.1.192) Locked, Waiting on external entities
> 2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1182 Session 3 (
> sofia/internal/1030 at 172.31.1.192) Ended
> 2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1030 at 172.31.1.192 [CS_DESTROY]
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:428 (
> sofia/internal/1030 at 172.31.1.192) Running State Change CS_DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1030 at 172.31.1.192) State DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] mod_sofia.c:341
> sofia/internal/1030 at 172.31.1.192 SOFIA DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1030 at 172.31.1.192 Standard DESTROY
> 2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1030 at 172.31.1.192) State DESTROY going to slee
>
>
> BR,
> Miha
>
> On 10/27/2011 2:15 PM, Nandy Dagondon wrote:
>
>  what is the full path/filename of your dialplan?
> your debug level is not detailed enough.  set the log level to 7.
> $fs_cli -l 7
>
>
> On Thu, Oct 27, 2011 at 6:07 PM, Miha Zoubek <miha at softnet.si> wrote:
>
>>  Hi,
>>
>> I have dialed 9051357952. IP800 is our GW. I created on it one SIP
>> account and connect freeswitch with it.
>>
>> BR,
>>  Miha
>>
>>
>> On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
>>
>>  hi,
>>
>>  what is IP800? is this a FXO gateway? or a SIP provider?
>>
>>  u hv to dial 9051357952 to match the dialplan.
>>
>>  -nandy
>>
>> On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek <miha at softnet.si> wrote:
>>
>>>  Hi,
>>>
>>> I am new Freeswitch user. I have read freeswitch book. I registered
>>> first few phones and make few inbound calls.
>>> I am trying to get outgoing calls working but having no luck.
>>> I have configured my gw and set up calling plan. When I try to dial
>>> number 051357952 I get this:
>>>
>>> reeswitch at localhost.localdomain> 2011-10-26 15:27:25.919562 [NOTICE]
>>> switch_channel.c:669 New Channel sofia/internal/1030 at 172.31.1.157[f2132149-dc40-4601-bd34-c22e5b06b696]
>>> 2011-10-26 15:27:25.955550 [INFO] mod_dialplan_xml.c:418 Processing
>>> 1030->9051357952 in context default
>>> 2011-10-26 15:27:25.960548 [NOTICE] switch_channel.c:669 New Channel
>>> sofia/external/051357952 [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>>> 2011-10-26 15:27:26.095518 [NOTICE] sofia.c:4789 Hangup sofia/external/051357952
>>> [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>>> 2011-10-26 15:27:26.098534 [INFO] mod_dptools.c:2355 Originate Failed.
>>> Cause: NO_ROUTE_DESTINATION
>>> 2011-10-26 15:27:26.098534 [NOTICE] mod_dptools.c:2418 Hangup sofia
>>> /internal/1030 at 172.31.1.157 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
>>> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1182 Session
>>> 49 (sofia/external/051357952) Ended
>>> 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1184 Close
>>> Channel sofia/external/051357952 [CS_DESTROY]
>>> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1182 Session
>>> 48 (sofia/internal/1030 at 172.31.1.157) Ended
>>> 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1184 Close
>>> Channel sofia/internal/1030 at 172.31.1.157 [CS_DESTROY]
>>>
>>> GW:
>>>
>>> <include>
>>>   <gateway name="IP800">
>>>     <!-- Your username, which is not your DID -->
>>>     <param name="username" value="Demo2"/>
>>>     <param name="password" value="1234"/>
>>>     <param name="register" value="true"/>
>>>     <param name="realm" value="172.31.1.200"/>
>>>  </gateway>
>>> </include>
>>>
>>> This is my dial plan:
>>>
>>> <include>
>>>         <extension name="Dial Out Custom Gateway">
>>>                 <condition field="destination_number"
>>> expression="^9(\d{9})$">
>>>         <action application="bridge" data="sofia/gateway/IP800/$1"/>
>>>                 </condition>
>>>         </extension>
>>> </include>
>>>
>>>
>>> sofia status
>>>
>>>                      Name
>>> Type                                       Data      State
>>>
>>> =================================================================================================
>>>                  external       profile
>>> sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>>>     external::example.com       gateway
>>> sip:joeuser at example.com      NOREG
>>>           external::IP800       gateway
>>> sip:Demo2 at 172.31.1.200      REGED
>>>                  internal       profile
>>> sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>>>              172.31.1.157         alias
>>> internal      ALIASED
>>>
>>> =================================================================================================
>>> 2 profiles 1 alias
>>>
>>>
>>> Please help me why I am getting not route to destination.
>>>
>>> Thank you very much for all your help!
>>>
>>> BR,
>>>  Miha
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Oct 27 19:29:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 10:29:51 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <201110270751.00821.sos@sokhapkin.dyndns.org>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<1319697268.2234.25.camel@portable-evil>
	<201110270751.00821.sos@sokhapkin.dyndns.org>
Message-ID: <CAKbxfG_MiCN6oo2hYA_UWL1vAOAvAGkz=7ouM0Oh0ZQwVa-Y+Q@mail.gmail.com>

That is FUD, please refrain.

If properly setup and deployed database stuff is fine.  If some people have
had problems, there is a reason they are just too busy to figure out why and
move on.....



On Thu, Oct 27, 2011 at 6:51 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Well, that was the reason why I stopped using lua a while ago. It crashes
> on
> high load if database queries are involved. I had crashes with both native
> FS
> odbc support or with luasql.
>
> On Thursday 27 October 2011, Cliff Wells wrote:
> > On Thu, 2011-10-27 at 02:21 -0400, Madovsky wrote:
> > > possible that you have somwhere redundant SQL request from your script
> or
> > > dialplan...
> >
> > As I mentioned previously, we also have some applications (mod_lua) that
> > utilize the internal odbc pool, so the concurrency is somewhat higher
> > (but not a lot, maybe another 20%).
> >
> > Regards,
> > Cliff
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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  Thu Oct 27 19:40:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Oct 2011 08:40:12 -0700
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
	<CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>
	<BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>
Message-ID: <CAKzWOxUCgJbP0XmH-+AXyKfcQX=TCf8cQeh0irPP1Fc__HLcvA@mail.gmail.com>

On Thu, Oct 27, 2011 at 1:02 AM, Faisal Rehman
<faisal.rehman22 at hotmail.com>wrote:

>  Hi David,
>
> Thanks for sharing the script but will that work too for Asterisk's
> Master.csv if I needed the ASR & ACD records from that file?
>

Use $14 (instead of $9) for Asterisk's Master.csv file.
-MC


>
>
> Regards,
>
> Faisal Rehman
>
> ------------------------------
> From: david.villasmil.work at gmail.com
> Date: Wed, 26 Oct 2011 19:48:57 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
> Hello,
>
> That's a faily simple bash script.
>
> "cat" the log file to an awk script like so;
>
> cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{
> FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT:
> "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR;
> if(ASR<100){ system("echo send email");} }'
>
> the "system" is where you put the mail command
>
> also, bear in mind that the master can get VERY large if you don't rotate
> it.
>
> david
>
> On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman <
> faisal.rehman22 at hotmail.com> wrote:
>
>  *Hi Waseem,*
>
> Thanks for your help, I saw the scripts that were on the link that you
> provided to me but actually the problem is that I don't want to involve the
> database though it is a bit difficult task but according to my requirement I
> just want to read the whole Master.csv file & if the ASR or ACD is low on
> any server, it should send me email, so my question is how can I skip the
> database from it to interact directly with the Master.csv file.
>
>
> Thanks in Advance!!
>
> *Faisal Rehman
> *
> ------------------------------
> From: wasim at convergence.pk
> Date: Tue, 25 Oct 2011 07:32:35 +0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
>
> Faisal:
>
> Load up the CDR into a db (MySQL or otherwise) ...
>
> For ACD you'll want a SQL query akin
>
>    - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
>    SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
>    disposition=1;
>
> For ASR you'll need another query
>
>    - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
>    - ASR is calls_answered/calls_total
>
> There is sample (read: poor) code in
> http://jira.freeswitch.org/browse/FS-1981
> which a google for "freeswitch ACD ASR" would have helped you get to in the
> first place.
>
> -wasim
>
> On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com>wrote:
>
>  Hi Everyone,
>
> I want to write a script that reads the Master.csv & generate an alert if
> the ASR & ACD is low from the desired value. Anybody has written such
> script, please help me write it or provide any sample.
>
>
> Thanks
>
>
> Faisal Rehman
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Oct 27 19:55:23 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 11:55:23 -0400
Subject: [Freeswitch-users] High load on database server
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<2934644557544732041@unknownmsgid>
Message-ID: <17277C2D60E140D0B41A28F4223062CC@e1705>

I means that depend the quality of your script ;)

----- Original Message ----- 
From: "Hynek Cihlar" <hynek.cihlar at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, October 27, 2011 3:46 AM
Subject: Re: [Freeswitch-users] High load on database server


> What does it mean? That I cannot access the DB from scripts?
>
> Sent from my mobile device
>
> On Oct 27, 2011, at 8:21, "Madovsky" <infos at madovsky.org> wrote:
>
>> possible that you have somwhere redundant SQL request from your script or
>> dialplan...
>>
>> ----- Original Message -----
>> From: "Madovsky" <infos at madovsky.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, October 27, 2011 2:15 AM
>> Subject: Re: [Freeswitch-users] High load on database server
>>
>>
>>> try the version 2.3.0 of ODBC (compiled from sources)
>>> I can't tell you if it will work because I never had more than 20
>>> councurrent calls
>>> but it should be better anyeway
>>>
>>> ----- Original Message -----
>>> From: "Cliff Wells" <cliff at develix.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Thursday, October 27, 2011 2:02 AM
>>> Subject: Re: [Freeswitch-users] High load on database server
>>>
>>>
>>>> On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote:
>>>>> if you have older unixodbc edit /etc/odbcinst.ini and add the param
>>>>> Threading=0 to each db entry.
>>>>
>>>>> In general postgres to me appears to care more about multithreaded
>>>>> environments and with the above threading=0 change works pretty well.
>>>>
>>>> I tried this (after googling a bit and coming across an older thread 
>>>> you
>>>> responded to) and it didn't change anything.  Crashes at ~330 
>>>> concurrent
>>>> calls like clockwork.  This was with unixODBC 2.14 and PostgreSQL 8.4
>>>> odbc driver.
>>>>
>>>> Regards,
>>>> Cliff
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jeff at jefflenk.com  Thu Oct 27 20:13:11 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 27 Oct 2011 09:13:11 -0700 (PDT)
Subject: [Freeswitch-users] ./configure fails on vanilla checkout
In-Reply-To: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>
References: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>
Message-ID: <1319731991828-6936975.post@n2.nabble.com>

You should provide somemore details here such as platform etc. I assume this
does not happen for all platforms?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/configure-fails-on-vanilla-checkout-tp6935612p6936975.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Thu Oct 27 20:18:36 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 27 Oct 2011 18:18:36 +0200
Subject: [Freeswitch-users] ./configure fails on vanilla checkout
In-Reply-To: <1319731991828-6936975.post@n2.nabble.com>
References: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>
	<1319731991828-6936975.post@n2.nabble.com>
Message-ID: <CALNZuwmBvrK0SF1RCAMVMJsOvv-V9-0StO4Fuz6+gNupJMYe8A@mail.gmail.com>

This sounds like something to be jira'd.
-Avi


On Thu, Oct 27, 2011 at 6:13 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> You should provide somemore details here such as platform etc. I assume
> this
> does not happen for all platforms?
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/configure-fails-on-vanilla-checkout-tp6935612p6936975.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From cliff at develix.com  Thu Oct 27 20:20:10 2011
From: cliff at develix.com (Cliff Wells)
Date: Thu, 27 Oct 2011 09:20:10 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CALNZuwnTLhL2F8as5Ragm5sPVMbNFXL-1Rx+3ONGdukndR=XqQ@mail.gmail.com>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<1319697268.2234.25.camel@portable-evil>
	<CALNZuwnTLhL2F8as5Ragm5sPVMbNFXL-1Rx+3ONGdukndR=XqQ@mail.gmail.com>
Message-ID: <1319732410.2234.27.camel@portable-evil>

On Thu, 2011-10-27 at 08:51 +0200, Avi Marcus wrote:
> You should be able to test the DB connections w/o live traffic..

Yes, I plan to, it's a matter of budgeting some time for it.

Cliff




From cliff at develix.com  Thu Oct 27 20:26:39 2011
From: cliff at develix.com (Cliff Wells)
Date: Thu, 27 Oct 2011 09:26:39 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <17277C2D60E140D0B41A28F4223062CC@e1705>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<2934644557544732041@unknownmsgid>
	<17277C2D60E140D0B41A28F4223062CC@e1705>
Message-ID: <1319732799.2234.34.camel@portable-evil>

On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
> I means that depend the quality of your script ;)

Quality isn't the issue here.  The script is too simple to be incorrect
(and it carefully releases the odbc connection back to the pool in a
hangup handler).  I have written poor-quality code plenty of times, but
25 years as a programmer usually allows me the luxury of knowing when
I'm doing it, thanks.

In any case, Lua scripts work fine.  Google will tell you that unixODBC
before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
again, I emphasize, everything WORKS GREAT (been using it for a couple
of years now) until you get to very high concurrency (about 330
concurrent calls on a single system).  If you don't expect to handle
more than 300 concurrent calls, then you do not need to worry about it.
At all.

Regards,
Cliff



From infos at madovsky.org  Thu Oct 27 20:33:16 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 12:33:16 -0400
Subject: [Freeswitch-users] High load on database server
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705><2934644557544732041@unknownmsgid><17277C2D60E140D0B41A28F4223062CC@e1705>
	<1319732799.2234.34.camel@portable-evil>
Message-ID: <29D6A63296F74D059644600F15149129@e1705>

ha ok, good luck so

----- Original Message ----- 
From: "Cliff Wells" <cliff at develix.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, October 27, 2011 12:26 PM
Subject: Re: [Freeswitch-users] High load on database server


> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
>> I means that depend the quality of your script ;)
> 
> Quality isn't the issue here.  The script is too simple to be incorrect
> (and it carefully releases the odbc connection back to the pool in a
> hangup handler).  I have written poor-quality code plenty of times, but
> 25 years as a programmer usually allows me the luxury of knowing when
> I'm doing it, thanks.
> 
> In any case, Lua scripts work fine.  Google will tell you that unixODBC
> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
> again, I emphasize, everything WORKS GREAT (been using it for a couple
> of years now) until you get to very high concurrency (about 330
> concurrent calls on a single system).  If you don't expect to handle
> more than 300 concurrent calls, then you do not need to worry about it.
> At all.
> 
> Regards,
> Cliff
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Thu Oct 27 20:36:39 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 12:36:39 -0400
Subject: [Freeswitch-users] High load on database server
Message-ID: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>

but, is anyone experienced that with shared ODBC ? (managing more than 330 
concurrent calls)

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, October 27, 2011 12:33 PM
Subject: Re: [Freeswitch-users] High load on database server


> ha ok, good luck so
>
> ----- Original Message ----- 
> From: "Cliff Wells" <cliff at develix.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, October 27, 2011 12:26 PM
> Subject: Re: [Freeswitch-users] High load on database server
>
>
>> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
>>> I means that depend the quality of your script ;)
>>
>> Quality isn't the issue here.  The script is too simple to be incorrect
>> (and it carefully releases the odbc connection back to the pool in a
>> hangup handler).  I have written poor-quality code plenty of times, but
>> 25 years as a programmer usually allows me the luxury of knowing when
>> I'm doing it, thanks.
>>
>> In any case, Lua scripts work fine.  Google will tell you that unixODBC
>> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
>> again, I emphasize, everything WORKS GREAT (been using it for a couple
>> of years now) until you get to very high concurrency (about 330
>> concurrent calls on a single system).  If you don't expect to handle
>> more than 300 concurrent calls, then you do not need to worry about it.
>> At all.
>>
>> Regards,
>> Cliff
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Thu Oct 27 20:41:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 11:41:50 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
Message-ID: <CAKbxfG8Nc9Gdk4+AuhmZpJ0Da58sOk_fwNOFdqjhnc-A61e3EA@mail.gmail.com>

Combine the knowledge expressed in this thread and make the right choices
and you can get any load you want within the limits of the box's resources.

On Thu, Oct 27, 2011 at 11:36 AM, Madovsky <infos at madovsky.org> wrote:

> but, is anyone experienced that with shared ODBC ? (managing more than 330
> concurrent calls)
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, October 27, 2011 12:33 PM
> Subject: Re: [Freeswitch-users] High load on database server
>
>
> > ha ok, good luck so
> >
> > ----- Original Message -----
> > From: "Cliff Wells" <cliff at develix.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Thursday, October 27, 2011 12:26 PM
> > Subject: Re: [Freeswitch-users] High load on database server
> >
> >
> >> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
> >>> I means that depend the quality of your script ;)
> >>
> >> Quality isn't the issue here.  The script is too simple to be incorrect
> >> (and it carefully releases the odbc connection back to the pool in a
> >> hangup handler).  I have written poor-quality code plenty of times, but
> >> 25 years as a programmer usually allows me the luxury of knowing when
> >> I'm doing it, thanks.
> >>
> >> In any case, Lua scripts work fine.  Google will tell you that unixODBC
> >> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
> >> again, I emphasize, everything WORKS GREAT (been using it for a couple
> >> of years now) until you get to very high concurrency (about 330
> >> concurrent calls on a single system).  If you don't expect to handle
> >> more than 300 concurrent calls, then you do not need to worry about it.
> >> At all.
> >>
> >> Regards,
> >> Cliff
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 hynek.cihlar at gmail.com  Thu Oct 27 20:42:38 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Thu, 27 Oct 2011 18:42:38 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1319732799.2234.34.camel@portable-evil>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<2934644557544732041@unknownmsgid>
	<17277C2D60E140D0B41A28F4223062CC@e1705>
	<1319732799.2234.34.camel@portable-evil>
Message-ID: <1759442497975160368@unknownmsgid>

Did you try to locate the source of the segmantation fault? At least
the module.

Sent from my mobile device

On Oct 27, 2011, at 18:27, Cliff Wells <cliff at develix.com> wrote:

> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
>> I means that depend the quality of your script ;)
>
> Quality isn't the issue here.  The script is too simple to be incorrect
> (and it carefully releases the odbc connection back to the pool in a
> hangup handler).  I have written poor-quality code plenty of times, but
> 25 years as a programmer usually allows me the luxury of knowing when
> I'm doing it, thanks.
>
> In any case, Lua scripts work fine.  Google will tell you that unixODBC
> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
> again, I emphasize, everything WORKS GREAT (been using it for a couple
> of years now) until you get to very high concurrency (about 330
> concurrent calls on a single system).  If you don't expect to handle
> more than 300 concurrent calls, then you do not need to worry about it.
> At all.
>
> Regards,
> Cliff
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Oct 27 20:42:48 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 27 Oct 2011 12:42:48 -0400
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
Message-ID: <201110271242.48743.sos@sokhapkin.dyndns.org>

What do you mean "shared ODBC"?

On Thursday 27 October 2011, Madovsky wrote:
> but, is anyone experienced that with shared ODBC ? (managing more than 330
> concurrent calls)
> 
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, October 27, 2011 12:33 PM
> Subject: Re: [Freeswitch-users] High load on database server
> 
> > ha ok, good luck so
> > 
> > ----- Original Message -----
> > From: "Cliff Wells" <cliff at develix.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Thursday, October 27, 2011 12:26 PM
> > Subject: Re: [Freeswitch-users] High load on database server
> > 
> >> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
> >>> I means that depend the quality of your script ;)
> >> 
> >> Quality isn't the issue here.  The script is too simple to be incorrect
> >> (and it carefully releases the odbc connection back to the pool in a
> >> hangup handler).  I have written poor-quality code plenty of times, but
> >> 25 years as a programmer usually allows me the luxury of knowing when
> >> I'm doing it, thanks.
> >> 
> >> In any case, Lua scripts work fine.  Google will tell you that unixODBC
> >> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
> >> again, I emphasize, everything WORKS GREAT (been using it for a couple
> >> of years now) until you get to very high concurrency (about 330
> >> concurrent calls on a single system).  If you don't expect to handle
> >> more than 300 concurrent calls, then you do not need to worry about it.
> >> At all.
> >> 
> >> Regards,
> >> Cliff
> >> 
> >> 
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rhuddleston at gmail.com  Thu Oct 27 20:55:10 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Thu, 27 Oct 2011 12:55:10 -0400
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <201110271242.48743.sos@sokhapkin.dyndns.org>
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
	<201110271242.48743.sos@sokhapkin.dyndns.org>
Message-ID: <020401cc94c9$31469b70$93d3d250$@com>

My bologna has a first name - it's O S C A R...

This topic is getting really old... I have to agree with Antm - get out of
the lazy pants and do some research / hard work.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
Okhapkin
Sent: Thursday, October 27, 2011 12:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] High load on database server

What do you mean "shared ODBC"?

On Thursday 27 October 2011, Madovsky wrote:
> but, is anyone experienced that with shared ODBC ? (managing more than 330
> concurrent calls)
> 
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, October 27, 2011 12:33 PM
> Subject: Re: [Freeswitch-users] High load on database server
> 
> > ha ok, good luck so
> > 
> > ----- Original Message -----
> > From: "Cliff Wells" <cliff at develix.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Thursday, October 27, 2011 12:26 PM
> > Subject: Re: [Freeswitch-users] High load on database server
> > 
> >> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
> >>> I means that depend the quality of your script ;)
> >> 
> >> Quality isn't the issue here.  The script is too simple to be incorrect
> >> (and it carefully releases the odbc connection back to the pool in a
> >> hangup handler).  I have written poor-quality code plenty of times, but
> >> 25 years as a programmer usually allows me the luxury of knowing when
> >> I'm doing it, thanks.
> >> 
> >> In any case, Lua scripts work fine.  Google will tell you that unixODBC
> >> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
> >> again, I emphasize, everything WORKS GREAT (been using it for a couple
> >> of years now) until you get to very high concurrency (about 330
> >> concurrent calls on a single system).  If you don't expect to handle
> >> more than 300 concurrent calls, then you do not need to worry about it.
> >> At all.
> >> 
> >> Regards,
> >> Cliff
> >> 
> >> 
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



FreeSWITCH-users mailing list
FreeSWITCH-users at 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 hynek.cihlar at gmail.com  Thu Oct 27 21:24:40 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Thu, 27 Oct 2011 19:24:40 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <020401cc94c9$31469b70$93d3d250$@com>
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
	<201110271242.48743.sos@sokhapkin.dyndns.org>
	<020401cc94c9$31469b70$93d3d250$@com>
Message-ID: <5675366332321559776@unknownmsgid>

Do not agree. Issues happening under extreme cases like high load with
some specific conditions are hard to reproduce, sharing the
information in this case is a lot more efficient, for all.

Sent from my mobile device

On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com> wrote:

> My bologna has a first name - it's O S C A R...
>
> This topic is getting really old... I have to agree with Antm - get out of
> the lazy pants and do some research / hard work.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
> Okhapkin
> Sent: Thursday, October 27, 2011 12:43 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] High load on database server
>
> What do you mean "shared ODBC"?
>
> On Thursday 27 October 2011, Madovsky wrote:
>> but, is anyone experienced that with shared ODBC ? (managing more than 330
>> concurrent calls)
>>
>> ----- Original Message -----
>> From: "Madovsky" <infos at madovsky.org>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, October 27, 2011 12:33 PM
>> Subject: Re: [Freeswitch-users] High load on database server
>>
>>> ha ok, good luck so
>>>
>>> ----- Original Message -----
>>> From: "Cliff Wells" <cliff at develix.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Thursday, October 27, 2011 12:26 PM
>>> Subject: Re: [Freeswitch-users] High load on database server
>>>
>>>> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
>>>>> I means that depend the quality of your script ;)
>>>>
>>>> Quality isn't the issue here.  The script is too simple to be incorrect
>>>> (and it carefully releases the odbc connection back to the pool in a
>>>> hangup handler).  I have written poor-quality code plenty of times, but
>>>> 25 years as a programmer usually allows me the luxury of knowing when
>>>> I'm doing it, thanks.
>>>>
>>>> In any case, Lua scripts work fine.  Google will tell you that unixODBC
>>>> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
>>>> again, I emphasize, everything WORKS GREAT (been using it for a couple
>>>> of years now) until you get to very high concurrency (about 330
>>>> concurrent calls on a single system).  If you don't expect to handle
>>>> more than 300 concurrent calls, then you do not need to worry about it.
>>>> At all.
>>>>
>>>> Regards,
>>>> Cliff
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Thu Oct 27 21:36:53 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 12:36:53 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <5675366332321559776@unknownmsgid>
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
	<201110271242.48743.sos@sokhapkin.dyndns.org>
	<020401cc94c9$31469b70$93d3d250$@com>
	<5675366332321559776@unknownmsgid>
Message-ID: <CAKbxfG_KWB41kEoY+PCR1q3MQ2E_bLrJNp-EMugi2cZaA5r4Qw@mail.gmail.com>

Let's recap

You have 4 moving parts, you have various versions of all 4 to choose from
and even different implementations of 3 of the 4.

ODBC LIB
ODBC DRIVER
DATABASE
FS

You must choose a stable combination of all 4 which may require specific
configuration of each component as well.

My only issue is FUD, I do not want people to advertise that FS does not
work on ODBC under load when the whole story is FS does not work under load
when you use fooODBC with barDRIVER with basBASE database server configured
a particular way.

It should never crash, if it does, its a bug in something.  I can speak for
FS that there is no bug there.  The other 3 are variables.
Remember this next time there is a discussion about using system libs and
why we build our own depends.  The one thing we use system libs for, ODBC,
is a huge PITA.........

Things to remember:

1) try the Threading=0 in the odbcinst.ini, this is a serious problem and
has been made the default in latest versions.
2) If you must use Mysql, enable transactions and make sure your odbc.ini is
referencing the threadsafe version of the lib
     libmyodbc3_r.so <-- note _r
3) Try various combos of drivers and odbc libs, many time newer is not
better, stable versions lie in the past.
4) build the drivers and odbc yourself or get the debug symbols so you can
get a backtrace, you could be finding a bug for them.......
5) Find a test to reproduce your problem so you can try different database
engines and driver combos.
6) Make sure you are on latest FS git so you know you have a stable copy.


Anyway, this is a pain, that's why ppl pay you to do it.
Discuss this all you want here, just minimize any FUD to scare away people
who want to use it.






On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> Do not agree. Issues happening under extreme cases like high load with
> some specific conditions are hard to reproduce, sharing the
> information in this case is a lot more efficient, for all.
>
> Sent from my mobile device
>
> On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com>
> wrote:
>
> > My bologna has a first name - it's O S C A R...
> >
> > This topic is getting really old... I have to agree with Antm - get out
> of
> > the lazy pants and do some research / hard work.
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Sergey
> > Okhapkin
> > Sent: Thursday, October 27, 2011 12:43 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] High load on database server
> >
> > What do you mean "shared ODBC"?
> >
> > On Thursday 27 October 2011, Madovsky wrote:
> >> but, is anyone experienced that with shared ODBC ? (managing more than
> 330
> >> concurrent calls)
> >>
> >> ----- Original Message -----
> >> From: "Madovsky" <infos at madovsky.org>
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Sent: Thursday, October 27, 2011 12:33 PM
> >> Subject: Re: [Freeswitch-users] High load on database server
> >>
> >>> ha ok, good luck so
> >>>
> >>> ----- Original Message -----
> >>> From: "Cliff Wells" <cliff at develix.com>
> >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >>> Sent: Thursday, October 27, 2011 12:26 PM
> >>> Subject: Re: [Freeswitch-users] High load on database server
> >>>
> >>>> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
> >>>>> I means that depend the quality of your script ;)
> >>>>
> >>>> Quality isn't the issue here.  The script is too simple to be
> incorrect
> >>>> (and it carefully releases the odbc connection back to the pool in a
> >>>> hangup handler).  I have written poor-quality code plenty of times,
> but
> >>>> 25 years as a programmer usually allows me the luxury of knowing when
> >>>> I'm doing it, thanks.
> >>>>
> >>>> In any case, Lua scripts work fine.  Google will tell you that
> unixODBC
> >>>> before 2.3.0 was a bit of a mess, so I expect the issue lies there,
> but
> >>>> again, I emphasize, everything WORKS GREAT (been using it for a couple
> >>>> of years now) until you get to very high concurrency (about 330
> >>>> concurrent calls on a single system).  If you don't expect to handle
> >>>> more than 300 concurrent calls, then you do not need to worry about
> it.
> >>>> At all.
> >>>>
> >>>> Regards,
> >>>> Cliff
> >>>>
> >>>>
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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  Thu Oct 27 21:39:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 12:39:47 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG_KWB41kEoY+PCR1q3MQ2E_bLrJNp-EMugi2cZaA5r4Qw@mail.gmail.com>
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
	<201110271242.48743.sos@sokhapkin.dyndns.org>
	<020401cc94c9$31469b70$93d3d250$@com>
	<5675366332321559776@unknownmsgid>
	<CAKbxfG_KWB41kEoY+PCR1q3MQ2E_bLrJNp-EMugi2cZaA5r4Qw@mail.gmail.com>
Message-ID: <CAKbxfG-aDdq8esGx8ewOibncEQr_O=78VC7fP6zD1LjveQ4Y3Q@mail.gmail.com>

BTW I know it's a contradiction to tell you to consider older ODBC and newer
FS but I wrote FS so I can attest to its stability especially the ODBC code
in the core.




On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Let's recap
>
> You have 4 moving parts, you have various versions of all 4 to choose from
> and even different implementations of 3 of the 4.
>
> ODBC LIB
> ODBC DRIVER
> DATABASE
> FS
>
> You must choose a stable combination of all 4 which may require specific
> configuration of each component as well.
>
> My only issue is FUD, I do not want people to advertise that FS does not
> work on ODBC under load when the whole story is FS does not work under load
> when you use fooODBC with barDRIVER with basBASE database server configured
> a particular way.
>
> It should never crash, if it does, its a bug in something.  I can speak for
> FS that there is no bug there.  The other 3 are variables.
> Remember this next time there is a discussion about using system libs and
> why we build our own depends.  The one thing we use system libs for, ODBC,
> is a huge PITA.........
>
> Things to remember:
>
> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem and
> has been made the default in latest versions.
> 2) If you must use Mysql, enable transactions and make sure your odbc.ini
> is referencing the threadsafe version of the lib
>      libmyodbc3_r.so <-- note _r
> 3) Try various combos of drivers and odbc libs, many time newer is not
> better, stable versions lie in the past.
> 4) build the drivers and odbc yourself or get the debug symbols so you can
> get a backtrace, you could be finding a bug for them.......
> 5) Find a test to reproduce your problem so you can try different database
> engines and driver combos.
> 6) Make sure you are on latest FS git so you know you have a stable copy.
>
>
> Anyway, this is a pain, that's why ppl pay you to do it.
> Discuss this all you want here, just minimize any FUD to scare away people
> who want to use it.
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> Do not agree. Issues happening under extreme cases like high load with
>> some specific conditions are hard to reproduce, sharing the
>> information in this case is a lot more efficient, for all.
>>
>> Sent from my mobile device
>>
>> On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com>
>> wrote:
>>
>> > My bologna has a first name - it's O S C A R...
>> >
>> > This topic is getting really old... I have to agree with Antm - get out
>> of
>> > the lazy pants and do some research / hard work.
>> >
>> >
>> > -----Original Message-----
>> > From: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Sergey
>> > Okhapkin
>> > Sent: Thursday, October 27, 2011 12:43 PM
>> > To: FreeSWITCH Users Help
>> > Subject: Re: [Freeswitch-users] High load on database server
>> >
>> > What do you mean "shared ODBC"?
>> >
>> > On Thursday 27 October 2011, Madovsky wrote:
>> >> but, is anyone experienced that with shared ODBC ? (managing more than
>> 330
>> >> concurrent calls)
>> >>
>> >> ----- Original Message -----
>> >> From: "Madovsky" <infos at madovsky.org>
>> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> >> Sent: Thursday, October 27, 2011 12:33 PM
>> >> Subject: Re: [Freeswitch-users] High load on database server
>> >>
>> >>> ha ok, good luck so
>> >>>
>> >>> ----- Original Message -----
>> >>> From: "Cliff Wells" <cliff at develix.com>
>> >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> >>> Sent: Thursday, October 27, 2011 12:26 PM
>> >>> Subject: Re: [Freeswitch-users] High load on database server
>> >>>
>> >>>> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
>> >>>>> I means that depend the quality of your script ;)
>> >>>>
>> >>>> Quality isn't the issue here.  The script is too simple to be
>> incorrect
>> >>>> (and it carefully releases the odbc connection back to the pool in a
>> >>>> hangup handler).  I have written poor-quality code plenty of times,
>> but
>> >>>> 25 years as a programmer usually allows me the luxury of knowing when
>> >>>> I'm doing it, thanks.
>> >>>>
>> >>>> In any case, Lua scripts work fine.  Google will tell you that
>> unixODBC
>> >>>> before 2.3.0 was a bit of a mess, so I expect the issue lies there,
>> but
>> >>>> again, I emphasize, everything WORKS GREAT (been using it for a
>> couple
>> >>>> of years now) until you get to very high concurrency (about 330
>> >>>> concurrent calls on a single system).  If you don't expect to handle
>> >>>> more than 300 concurrent calls, then you do not need to worry about
>> it.
>> >>>> At all.
>> >>>>
>> >>>> Regards,
>> >>>> Cliff
>> >>>>
>> >>>>
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From infos at madovsky.org  Thu Oct 27 21:51:25 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 13:51:25 -0400
Subject: [Freeswitch-users] High load on database server
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705><201110271242.48743.sos@sokhapkin.dyndns.org><020401cc94c9$31469b70$93d3d250$@com><5675366332321559776@unknownmsgid><CAKbxfG_KWB41kEoY+PCR1q3MQ2E_bLrJNp-EMugi2cZaA5r4Qw@mail.gmail.com>
	<CAKbxfG-aDdq8esGx8ewOibncEQr_O=78VC7fP6zD1LjveQ4Y3Q@mail.gmail.com>
Message-ID: <32B0C7B03EDD4BF2BBE142ACDE35EFA0@e1705>

on this subject,
is SIPp can be used to test to reproduce hundreds calls ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: FreeSWITCH Users Help 
  Sent: Thursday, October 27, 2011 1:39 PM
  Subject: Re: [Freeswitch-users] High load on database server


  BTW I know it's a contradiction to tell you to consider older ODBC and newer FS but I wrote FS so I can attest to its stability especially the ODBC code in the core.







  On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

    Let's recap


    You have 4 moving parts, you have various versions of all 4 to choose from and even different implementations of 3 of the 4.


    ODBC LIB
    ODBC DRIVER
    DATABASE
    FS


    You must choose a stable combination of all 4 which may require specific configuration of each component as well.


    My only issue is FUD, I do not want people to advertise that FS does not work on ODBC under load when the whole story is FS does not work under load when you use fooODBC with barDRIVER with basBASE database server configured a particular way.


    It should never crash, if it does, its a bug in something.  I can speak for FS that there is no bug there.  The other 3 are variables. 
    Remember this next time there is a discussion about using system libs and why we build our own depends.  The one thing we use system libs for, ODBC, is a huge PITA.........


    Things to remember:


    1) try the Threading=0 in the odbcinst.ini, this is a serious problem and has been made the default in latest versions.
    2) If you must use Mysql, enable transactions and make sure your odbc.ini is referencing the threadsafe version of the lib 
         libmyodbc3_r.so <-- note _r
    3) Try various combos of drivers and odbc libs, many time newer is not better, stable versions lie in the past.
    4) build the drivers and odbc yourself or get the debug symbols so you can get a backtrace, you could be finding a bug for them.......
    5) Find a test to reproduce your problem so you can try different database engines and driver combos.
    6) Make sure you are on latest FS git so you know you have a stable copy.




    Anyway, this is a pain, that's why ppl pay you to do it.
    Discuss this all you want here, just minimize any FUD to scare away people who want to use it.











    On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

      Do not agree. Issues happening under extreme cases like high load with
      some specific conditions are hard to reproduce, sharing the
      information in this case is a lot more efficient, for all.


      Sent from my mobile device


      On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com> wrote:

      > My bologna has a first name - it's O S C A R...
      >
      > This topic is getting really old... I have to agree with Antm - get out of
      > the lazy pants and do some research / hard work.
      >
      >
      > -----Original Message-----
      > From: freeswitch-users-bounces at lists.freeswitch.org
      > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
      > Okhapkin
      > Sent: Thursday, October 27, 2011 12:43 PM
      > To: FreeSWITCH Users Help
      > Subject: Re: [Freeswitch-users] High load on database server
      >
      > What do you mean "shared ODBC"?
      >
      > On Thursday 27 October 2011, Madovsky wrote:
      >> but, is anyone experienced that with shared ODBC ? (managing more than 330
      >> concurrent calls)
      >>
      >> ----- Original Message -----
      >> From: "Madovsky" <infos at madovsky.org>
      >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
      >> Sent: Thursday, October 27, 2011 12:33 PM
      >> Subject: Re: [Freeswitch-users] High load on database server
      >>
      >>> ha ok, good luck so
      >>>
      >>> ----- Original Message -----
      >>> From: "Cliff Wells" <cliff at develix.com>
      >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
      >>> Sent: Thursday, October 27, 2011 12:26 PM
      >>> Subject: Re: [Freeswitch-users] High load on database server
      >>>
      >>>> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote:
      >>>>> I means that depend the quality of your script ;)
      >>>>
      >>>> Quality isn't the issue here.  The script is too simple to be incorrect
      >>>> (and it carefully releases the odbc connection back to the pool in a
      >>>> hangup handler).  I have written poor-quality code plenty of times, but
      >>>> 25 years as a programmer usually allows me the luxury of knowing when
      >>>> I'm doing it, thanks.
      >>>>
      >>>> In any case, Lua scripts work fine.  Google will tell you that unixODBC
      >>>> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but
      >>>> again, I emphasize, everything WORKS GREAT (been using it for a couple
      >>>> of years now) until you get to very high concurrency (about 330
      >>>> concurrent calls on a single system).  If you don't expect to handle
      >>>> more than 300 concurrent calls, then you do not need to worry about it.
      >>>> At all.
      >>>>
      >>>> Regards,
      >>>> Cliff
      >>>>
      >>>>
      >>>>
      >>>> FreeSWITCH-users mailing list
      >>>> FreeSWITCH-users at lists.freeswitch.org
      >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >>>>
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >>>> http://www.freeswitch.org
      >>
      >> FreeSWITCH-users mailing list
      >> FreeSWITCH-users at lists.freeswitch.org
      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >> http://www.freeswitch.org
      >
      >
      >
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      >
      >
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org


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






    -- 
    Anthony Minessale II

    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






  -- 
  Anthony Minessale II

  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 krice at freeswitch.org  Thu Oct 27 21:54:29 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Oct 2011 12:54:29 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <32B0C7B03EDD4BF2BBE142ACDE35EFA0@e1705>
Message-ID: <CACF0505.211E1%krice@freeswitch.org>

SIPP works fine for this... However keep in mind some of their default
scenario files arent exactly the best in the world and can in many
situations leave a bit to be desired... Check the wiki I think there is more
info on this on there

K


On 10/27/11 12:51 PM, "Madovsky" <infos at madovsky.org> wrote:

> on this subject,
> is SIPp can be used to test to reproduce hundreds calls ?
>>  
>> ----- Original Message -----
>>  
>> From:  Anthony Minessale <mailto:anthony.minessale at gmail.com>
>>  
>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
>>  
>> Sent: Thursday, October 27, 2011 1:39  PM
>>  
>> Subject: Re: [Freeswitch-users] High load  on database server
>>  
>> 
>> BTW I know it's a contradiction to tell you to consider older  ODBC and newer
>> FS but I wrote FS so I can attest to its stability especially  the ODBC code
>> in the core. 
>> 
>>  
>> 
>>  
>> 
>> 
>>  
>> On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale
>> <anthony.minessale at gmail.com>  wrote:
>>  
>>>  
>>> Let's recap
>>>  
>>> 
>>>  
>>> You have 4 moving parts, you have various versions of all 4 to choose  from
>>> and even different implementations of 3 of the 4.
>>>  
>>> 
>>> ODBC LIB 
>>> ODBC DRIVER
>>>  
>>> DATABASE
>>>  
>>> FS
>>>  
>>> 
>>>  
>>> You must choose a stable combination of all 4 which may require  specific
>>> configuration of each component as well.
>>>  
>>> 
>>>  
>>> My only issue is FUD, I do not want people to advertise that FS does  not
>>> work on ODBC under load when the whole story is FS does not work under  load
>>> when you use fooODBC with barDRIVER with basBASE database server  configured
>>> a particular way.
>>>  
>>> 
>>>  
>>> It should never crash, if it does, its a bug in something.  I can  speak for
>>> FS that there is no bug there.  The other 3 are  variables.
>>>  
>>> Remember this next time there is a discussion about using system libs  and
>>> why we build our own depends.  The one thing we use system libs  for, ODBC,
>>> is a huge PITA.........
>>>  
>>> 
>>>  
>>> Things to remember:
>>>  
>>> 
>>>  
>>> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem  and
>>> has been made the default in latest versions.
>>>  
>>> 2) If you must use Mysql, enable transactions and make sure your  odbc.ini
>>> is referencing the threadsafe version of the lib
>>>  
>>>      libmyodbc3_r.so <-- note _r
>>>  
>>> 3) Try various combos of drivers and odbc libs, many time newer is not
>>> better, stable versions lie in the past.
>>>  
>>> 4) build the drivers and odbc yourself or get the debug symbols so you  can
>>> get a backtrace, you could be finding a bug for them.......
>>>  
>>> 5) Find a test to reproduce your problem so you can try different  database
>>> engines and driver combos.
>>>  
>>> 6) Make sure you are on latest FS git so you know you have a stable  copy.
>>>  
>>> 
>>>  
>>> 
>>>  
>>> Anyway, this is a pain, that's why ppl pay you to do it.
>>>  
>>> Discuss this all you want here, just minimize any FUD to scare away  people
>>> who want to use it.
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>>  
>>>  
>>>  
>>> 
>>> 
>>>  
>>> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com>
>>> wrote:
>>>  
>>>> Do not agree. Issues happening under extreme cases like  high load with
>>>> some specific conditions are hard to reproduce, sharing  the
>>>> information in this case is a lot more efficient, for all.
>>>>  
>>>> 
>>>> Sent from my mobile device
>>>> 
>>>>  
>>>>  
>>>>  
>>>> On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com> wrote:
>>>> 
>>>>> > My bologna  has a first name - it's O S C A R...
>>>>> >
>>>>> > This topic is getting  really old... I have to agree with Antm - get out
>>>>> of
>>>>> > the lazy  pants and do some research / hard work.
>>>>> >
>>>>> >
>>>>> >  -----Original Message-----
>>>>> > From: freeswitch-users-bounces at lists.freeswitch.org
>>>>> >  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf  Of
>>>>> Sergey
>>>>> > Okhapkin
>>>>> > Sent: Thursday, October 27, 2011 12:43  PM
>>>>> > To: FreeSWITCH Users Help
>>>>> > Subject: Re:  [Freeswitch-users] High load on database server
>>>>> >
>>>>> > What do  you mean "shared ODBC"?
>>>>> >
>>>>> > On Thursday 27 October 2011,  Madovsky wrote:
>>>>>> >> but, is anyone experienced that with shared  ODBC ? (managing more
>>>>>> than 330
>>>>>> >> concurrent  calls)
>>>>>> >>
>>>>>> >> ----- Original Message -----
>>>>>> >>  From: "Madovsky" <infos at madovsky.org>
>>>>>> >> To: "FreeSWITCH Users  Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> >>  Sent: Thursday, October 27, 2011 12:33 PM
>>>>>> >> Subject: Re:  [Freeswitch-users] High load on database  server
>>>>>> >>
>>>>>>> >>> ha ok, good luck  so
>>>>>>> >>>
>>>>>>> >>> ----- Original Message  -----
>>>>>>> >>> From: "Cliff Wells" <cliff at develix.com>
>>>>>>> >>> To: "FreeSWITCH  Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>> >>>  Sent: Thursday, October 27, 2011 12:26 PM
>>>>>>> >>> Subject: Re:  [Freeswitch-users] High load on database  server
>>>>>>> >>>
>>>>>>>> >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
>>>>>>>>> >>>>> I means that depend the  quality of your script ;)
>>>>>>>> >>>>
>>>>>>>> >>>> Quality  isn't the issue here.  The script is too simple to be
>>>>>>>> incorrect
>>>>>>>> >>>> (and it carefully releases the odbc  connection back to the pool
in a
>>>>>>>> >>>> hangup handler).   I have written poor-quality code plenty of
>>>>>>>> times,  but
>>>>>>>> >>>> 25 years as a programmer usually allows me the  luxury of knowing
when
>>>>>>>> >>>> I'm doing it,  thanks.
>>>>>>>> >>>>
>>>>>>>> >>>> In any case, Lua scripts  work fine.  Google will tell you that
>>>>>>>> unixODBC
>>>>>>>> >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies
>>>>>>>> there,  but
>>>>>>>> >>>> again, I emphasize, everything WORKS GREAT (been  using it for a
>>>>>>>> couple
>>>>>>>> >>>> of years now) until you get to  very high concurrency (about 330
>>>>>>>> >>>> concurrent calls on a  single system).  If you don't expect to
>>>>>>>> handle
>>>>>>>> >>>>  more than 300 concurrent calls, then you do not need to worry
>>>>>>>> about  it.
>>>>>>>> >>>> At all.
>>>>>>>> >>>>
>>>>>>>> >>>>  Regards,
>>>>>>>> >>>>  Cliff
>>>>>>>> >>>>
>>>>>>>> >>>>
>>>>>>>> >>>>
>>>>>>>> >>>>  FreeSWITCH-users mailing list
>>>>>>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>>>
>>>>> >  
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>>>  http://www.freeswitch.org
>>>>>> >>
>>>>>> >>  FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >>  
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >>  http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >
>>>>> >  FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >  
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >  http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >
>>>>> >  FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >  
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >  http://www.freeswitch.org
>>>> 
>>>> 
>>>> FreeSWITCH-users  mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>>  
>>> 
>>>  

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From hynek.cihlar at gmail.com  Thu Oct 27 22:06:00 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Thu, 27 Oct 2011 20:06:00 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG_KWB41kEoY+PCR1q3MQ2E_bLrJNp-EMugi2cZaA5r4Qw@mail.gmail.com>
References: <4B9586DE259E4097AE6C3DADA1FC2A2D@e1705>
	<201110271242.48743.sos@sokhapkin.dyndns.org>
	<020401cc94c9$31469b70$93d3d250$@com>
	<5675366332321559776@unknownmsgid>
	<CAKbxfG_KWB41kEoY+PCR1q3MQ2E_bLrJNp-EMugi2cZaA5r4Qw@mail.gmail.com>
Message-ID: <CAJh8XdXi-kxyd55LExHkVC88XUYohvOWes8721LyWRdmEjKiNQ@mail.gmail.com>

On Thu, Oct 27, 2011 at 7:36 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Let's recap
>
> You have 4 moving parts, you have various versions of all 4 to choose from
> and even different implementations of 3 of the 4.
>
> ODBC LIB
> ODBC DRIVER
>  DATABASE
> FS
>
> You must choose a stable combination of all 4 which may require specific
> configuration of each component as well.
>
> My only issue is FUD, I do not want people to advertise that FS does not
> work on ODBC under load when the whole story is FS does not work under load
> when you use fooODBC with barDRIVER with basBASE database server configured
> a particular way.
>
> It should never crash, if it does, its a bug in something.  I can speak for
> FS that there is no bug there.  The other 3 are variables.
> Remember this next time there is a discussion about using system libs and
> why we build our own depends.  The one thing we use system libs for, ODBC,
> is a huge PITA.........
>
> Things to remember:
>
> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem and
> has been made the default in latest versions.
> 2) If you must use Mysql, enable transactions and make sure your odbc.ini
> is referencing the threadsafe version of the lib
>      libmyodbc3_r.so <-- note _r
>

Enable transactions? What does it mean?

When you experience the segfault,in which module (mem address), if it is
tracable?

Hynek
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From infos at madovsky.org  Thu Oct 27 22:07:25 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 14:07:25 -0400
Subject: [Freeswitch-users] High load on database server
References: <CACF0505.211E1%krice@freeswitch.org>
Message-ID: <D5E62885E535486BB92F94CB5CBBC1A1@e1705>

Re: [Freeswitch-users] High load on database serverThanks Ken. here the link of official SIPP website
http://sipp.sourceforge.net/

I didn't know it was a HP app ! :0)
  ----- Original Message ----- 
  From: Ken Rice 
  To: FreeSWITCH Users Help 
  Sent: Thursday, October 27, 2011 1:54 PM
  Subject: Re: [Freeswitch-users] High load on database server


  SIPP works fine for this... However keep in mind some of their default scenario files arent exactly the best in the world and can in many situations leave a bit to be desired... Check the wiki I think there is more info on this on there

  K


  On 10/27/11 12:51 PM, "Madovsky" <infos at madovsky.org> wrote:


    on this subject,
    is SIPp can be used to test to reproduce hundreds calls ?


      ----- Original Message ----- 
       
      From:  Anthony Minessale <mailto:anthony.minessale at gmail.com>  
       
      To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>   
       
      Sent: Thursday, October 27, 2011 1:39  PM
       
      Subject: Re: [Freeswitch-users] High load  on database server
       

      BTW I know it's a contradiction to tell you to consider older  ODBC and newer FS but I wrote FS so I can attest to its stability especially  the ODBC code in the core. 

       

       


       
      On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <anthony.minessale at gmail.com>  wrote:
       


        Let's recap
         

         
        You have 4 moving parts, you have various versions of all 4 to choose  from and even different implementations of 3 of the 4.
         

        ODBC LIB 
        ODBC DRIVER
         
        DATABASE
         
        FS
         

         
        You must choose a stable combination of all 4 which may require  specific configuration of each component as well.
         

         
        My only issue is FUD, I do not want people to advertise that FS does  not work on ODBC under load when the whole story is FS does not work under  load when you use fooODBC with barDRIVER with basBASE database server  configured a particular way.
         

         
        It should never crash, if it does, its a bug in something.  I can  speak for FS that there is no bug there.  The other 3 are  variables. 
         
        Remember this next time there is a discussion about using system libs  and why we build our own depends.  The one thing we use system libs  for, ODBC, is a huge PITA.........
         

         
        Things to remember:
         

         
        1) try the Threading=0 in the odbcinst.ini, this is a serious problem  and has been made the default in latest versions.
         
        2) If you must use Mysql, enable transactions and make sure your  odbc.ini is referencing the threadsafe version of the lib 
         
             libmyodbc3_r.so <-- note _r
         
        3) Try various combos of drivers and odbc libs, many time newer is not  better, stable versions lie in the past.
         
        4) build the drivers and odbc yourself or get the debug symbols so you  can get a backtrace, you could be finding a bug for them.......
         
        5) Find a test to reproduce your problem so you can try different  database engines and driver combos.
         
        6) Make sure you are on latest FS git so you know you have a stable  copy.
         

         

         
        Anyway, this is a pain, that's why ppl pay you to do it.
         
        Discuss this all you want here, just minimize any FUD to scare away  people who want to use it.
         

         

         

         

         
         
         
         


         
        On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:
         

          Do not agree. Issues happening under extreme cases like  high load with
          some specific conditions are hard to reproduce, sharing  the
          information in this case is a lot more efficient, for all.
           

          Sent from my mobile device

           
           
           
          On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com> wrote:

          > My bologna  has a first name - it's O S C A R...
          >
          > This topic is getting  really old... I have to agree with Antm - get out of
          > the lazy  pants and do some research / hard work.
          >
          >
          >  -----Original Message-----
          > From: freeswitch-users-bounces at lists.freeswitch.org
          >  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf  Of Sergey
          > Okhapkin
          > Sent: Thursday, October 27, 2011 12:43  PM
          > To: FreeSWITCH Users Help
          > Subject: Re:  [Freeswitch-users] High load on database server
          >
          > What do  you mean "shared ODBC"?
          >
          > On Thursday 27 October 2011,  Madovsky wrote:
          >> but, is anyone experienced that with shared  ODBC ? (managing more than 330
          >> concurrent  calls)
          >>
          >> ----- Original Message -----
          >>  From: "Madovsky" <infos at madovsky.org>
          >> To: "FreeSWITCH Users  Help" <freeswitch-users at lists.freeswitch.org>
          >>  Sent: Thursday, October 27, 2011 12:33 PM
          >> Subject: Re:  [Freeswitch-users] High load on database  server
          >>
          >>> ha ok, good luck  so
          >>>
          >>> ----- Original Message  -----
          >>> From: "Cliff Wells" <cliff at develix.com>
          >>> To: "FreeSWITCH  Users Help" <freeswitch-users at lists.freeswitch.org>
          >>>  Sent: Thursday, October 27, 2011 12:26 PM
          >>> Subject: Re:  [Freeswitch-users] High load on database  server
          >>>
          >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
          >>>>> I means that depend the  quality of your script ;)
          >>>>
          >>>> Quality  isn't the issue here.  The script is too simple to be  incorrect
          >>>> (and it carefully releases the odbc  connection back to the pool in a
          >>>> hangup handler).   I have written poor-quality code plenty of times,  but
          >>>> 25 years as a programmer usually allows me the  luxury of knowing when
          >>>> I'm doing it,  thanks.
          >>>>
          >>>> In any case, Lua scripts  work fine.  Google will tell you that unixODBC
          >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies there,  but
          >>>> again, I emphasize, everything WORKS GREAT (been  using it for a couple
          >>>> of years now) until you get to  very high concurrency (about 330
          >>>> concurrent calls on a  single system).  If you don't expect to handle
          >>>>  more than 300 concurrent calls, then you do not need to worry about  it.
          >>>> At all.
          >>>>
          >>>>  Regards,
          >>>>  Cliff
          >>>>
          >>>>
          >>>>
          >>>>  FreeSWITCH-users mailing list
          >>>> FreeSWITCH-users at lists.freeswitch.org
          >>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >>>>
          >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >>>>  http://www.freeswitch.org
          >>
          >>  FreeSWITCH-users mailing list
          >> FreeSWITCH-users at lists.freeswitch.org
          >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >>  http://www.freeswitch.org
          >
          >
          >
          >  FreeSWITCH-users mailing list
          > FreeSWITCH-users at lists.freeswitch.org
          > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >  http://www.freeswitch.org
          >
          >
          >
          >  FreeSWITCH-users mailing list
          > FreeSWITCH-users at lists.freeswitch.org
          > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >  http://www.freeswitch.org


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



         

         



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



  FreeSWITCH-users mailing 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 Oct 27 22:08:30 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 13:08:30 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <D5E62885E535486BB92F94CB5CBBC1A1@e1705>
References: <CACF0505.211E1%krice@freeswitch.org>
	<D5E62885E535486BB92F94CB5CBBC1A1@e1705>
Message-ID: <CAKbxfG8Hm_a2CBK6S1upZm43rT3brbp8XDAFXuYcycNZneGB2Q@mail.gmail.com>

mysql does not work with transactions by default in defense of injection
attacks... read the thread its in the top.


On Thu, Oct 27, 2011 at 1:07 PM, Madovsky <infos at madovsky.org> wrote:

> **
> Thanks Ken. here the link of official SIPP website
> http://sipp.sourceforge.net/
>
> I didn't know it was a HP app ! :0)
>
> ----- Original Message -----
> *From:* Ken Rice <krice at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, October 27, 2011 1:54 PM
> *Subject:* Re: [Freeswitch-users] High load on database server
>
> SIPP works fine for this... However keep in mind some of their default
> scenario files arent exactly the best in the world and can in many
> situations leave a bit to be desired... Check the wiki I think there is more
> info on this on there
>
> K
>
>
> On 10/27/11 12:51 PM, "Madovsky" <infos at madovsky.org> wrote:
>
> on this subject,
> is SIPp can be used to test to reproduce hundreds calls ?
>
>
> ----- Original Message -----
>
> *From:*  Anthony Minessale <mailto:anthony.minessale at gmail.com<anthony.minessale at gmail.com>>
>
>
> *To:* FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
>
>
> *Sent:* Thursday, October 27, 2011 1:39  PM
>
> *Subject:* Re: [Freeswitch-users] High load  on database server
>
>
> BTW I know it's a contradiction to tell you to consider older  ODBC and
> newer FS but I wrote FS so I can attest to its stability especially  the
> ODBC code in the core.
>
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <
> anthony.minessale at gmail.com>  wrote:
>
>
>
> Let's recap
>
>
>
> You have 4 moving parts, you have various versions of all 4 to choose  from
> and even different implementations of 3 of the 4.
>
>
> ODBC LIB
> ODBC DRIVER
>
> DATABASE
>
> FS
>
>
>
> You must choose a stable combination of all 4 which may require  specific
> configuration of each component as well.
>
>
>
> My only issue is FUD, I do not want people to advertise that FS does  not
> work on ODBC under load when the whole story is FS does not work under  load
> when you use fooODBC with barDRIVER with basBASE database server  configured
> a particular way.
>
>
>
> It should never crash, if it does, its a bug in something.  I can  speak
> for FS that there is no bug there.  The other 3 are  variables.
>
> Remember this next time there is a discussion about using system libs  and
> why we build our own depends.  The one thing we use system libs  for, ODBC,
> is a huge PITA.........
>
>
>
> Things to remember:
>
>
>
> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem  and
> has been made the default in latest versions.
>
> 2) If you must use Mysql, enable transactions and make sure your  odbc.ini
> is referencing the threadsafe version of the lib
>
>      libmyodbc3_r.so <-- note _r
>
> 3) Try various combos of drivers and odbc libs, many time newer is not
>  better, stable versions lie in the past.
>
> 4) build the drivers and odbc yourself or get the debug symbols so you  can
> get a backtrace, you could be finding a bug for them.......
>
> 5) Find a test to reproduce your problem so you can try different  database
> engines and driver combos.
>
> 6) Make sure you are on latest FS git so you know you have a stable  copy.
>
>
>
>
>
> Anyway, this is a pain, that's why ppl pay you to do it.
>
> Discuss this all you want here, just minimize any FUD to scare away  people
> who want to use it.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com>
> wrote:
>
>
> Do not agree. Issues happening under extreme cases like  high load with
> some specific conditions are hard to reproduce, sharing  the
> information in this case is a lot more efficient, for all.
>
>
> Sent from my mobile device
>
>
>
>
> On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com>
> wrote:
>
> > My bologna  has a first name - it's O S C A R...
> >
> > This topic is getting  really old... I have to agree with Antm - get out
> of
> > the lazy  pants and do some research / hard work.
> >
> >
> >  -----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 Sergey
> > Okhapkin
> > Sent: Thursday, October 27, 2011 12:43  PM
> > To: FreeSWITCH Users Help
> > Subject: Re:  [Freeswitch-users] High load on database server
> >
> > What do  you mean "shared ODBC"?
> >
> > On Thursday 27 October 2011,  Madovsky wrote:
> >> but, is anyone experienced that with shared  ODBC ? (managing more than
> 330
> >> concurrent  calls)
> >>
> >> ----- Original Message -----
> >>  From: "Madovsky" <infos at madovsky.org>
> >> To: "FreeSWITCH Users  Help" <freeswitch-users at lists.freeswitch.org>
> >>  Sent: Thursday, October 27, 2011 12:33 PM
> >> Subject: Re:  [Freeswitch-users] High load on database  server
> >>
> >>> ha ok, good luck  so
> >>>
> >>> ----- Original Message  -----
> >>> From: "Cliff Wells" <cliff at develix.com>
> >>> To: "FreeSWITCH  Users Help" <freeswitch-users at lists.freeswitch.org>
> >>>  Sent: Thursday, October 27, 2011 12:26 PM
> >>> Subject: Re:  [Freeswitch-users] High load on database  server
> >>>
> >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
> >>>>> I means that depend the  quality of your script ;)
> >>>>
> >>>> Quality  isn't the issue here.  The script is too simple to be
>  incorrect
> >>>> (and it carefully releases the odbc  connection back to the pool in a
> >>>> hangup handler).   I have written poor-quality code plenty of times,
>  but
> >>>> 25 years as a programmer usually allows me the  luxury of knowing when
> >>>> I'm doing it,  thanks.
> >>>>
> >>>> In any case, Lua scripts  work fine.  Google will tell you that
> unixODBC
> >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies there,
>  but
> >>>> again, I emphasize, everything WORKS GREAT (been  using it for a
> couple
> >>>> of years now) until you get to  very high concurrency (about 330
> >>>> concurrent calls on a  single system).  If you don't expect to handle
> >>>>  more than 300 concurrent calls, then you do not need to worry about
>  it.
> >>>> At all.
> >>>>
> >>>>  Regards,
> >>>>  Cliff
> >>>>
> >>>>
> >>>>
> >>>>  FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>  http://www.freeswitch.org
> >>
> >>  FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>  http://www.freeswitch.org
> >
> >
> >
> >  FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
> >
> >
> >
> >  FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
>
>
> FreeSWITCH-users  mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
>
>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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 aakashviswam at gmail.com  Thu Oct 27 22:26:27 2011
From: aakashviswam at gmail.com (Aakash)
Date: Thu, 27 Oct 2011 11:26:27 -0700 (PDT)
Subject: [Freeswitch-users] caller tone breakage
Message-ID: <1319739987637-6937501.post@n2.nabble.com>

Hi,

I am using freeswitch latest git version.For testing i configured my new US
DID inbound numbers landing straight to an extension,i am able to speak two
way without any issues.But the call ringing tone is breaking when we try to
reach the new US DID number.

Could you please tell whether we have any setting to control the caller tone
breakage.


Thanks in advance

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/caller-tone-breakage-tp6937501p6937501.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From hynek.cihlar at gmail.com  Thu Oct 27 22:29:50 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Thu, 27 Oct 2011 20:29:50 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG8Hm_a2CBK6S1upZm43rT3brbp8XDAFXuYcycNZneGB2Q@mail.gmail.com>
References: <CACF0505.211E1%krice@freeswitch.org>
	<D5E62885E535486BB92F94CB5CBBC1A1@e1705>
	<CAKbxfG8Hm_a2CBK6S1upZm43rT3brbp8XDAFXuYcycNZneGB2Q@mail.gmail.com>
Message-ID: <3244411099634732796@unknownmsgid>

Transactions are by default supported by mysql, the transactions are driven
by sql statements themselves.

Your statement is either incorrect or I interpret it wrong.

Sent from my mobile device

On Oct 27, 2011, at 20:09, Anthony Minessale <anthony.minessale at gmail.com>
wrote:

mysql does not work with transactions by default in defense of injection
attacks... read the thread its in the top.


On Thu, Oct 27, 2011 at 1:07 PM, Madovsky <infos at madovsky.org> wrote:

> **
> Thanks Ken. here the link of official SIPP website
> http://sipp.sourceforge.net/
>
> I didn't know it was a HP app ! :0)
>
> ----- Original Message -----
> *From:* Ken Rice <krice at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, October 27, 2011 1:54 PM
> *Subject:* Re: [Freeswitch-users] High load on database server
>
> SIPP works fine for this... However keep in mind some of their default
> scenario files arent exactly the best in the world and can in many
> situations leave a bit to be desired... Check the wiki I think there is more
> info on this on there
>
> K
>
>
> On 10/27/11 12:51 PM, "Madovsky" <infos at madovsky.org> wrote:
>
> on this subject,
> is SIPp can be used to test to reproduce hundreds calls ?
>
>
> ----- Original Message -----
>
> *From:*  Anthony Minessale <mailto:anthony.minessale at gmail.com<anthony.minessale at gmail.com><anthony.minessale at gmail.com>>
>
>
> *To:* FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org><freeswitch-users at lists.freeswitch.org>>
>
>
> *Sent:* Thursday, October 27, 2011 1:39  PM
>
> *Subject:* Re: [Freeswitch-users] High load  on database server
>
>
> BTW I know it's a contradiction to tell you to consider older  ODBC and
> newer FS but I wrote FS so I can attest to its stability especially  the
> ODBC code in the core.
>
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <
> anthony.minessale at gmail.com>  wrote:
>
>
>
> Let's recap
>
>
>
> You have 4 moving parts, you have various versions of all 4 to choose  from
> and even different implementations of 3 of the 4.
>
>
> ODBC LIB
> ODBC DRIVER
>
> DATABASE
>
> FS
>
>
>
> You must choose a stable combination of all 4 which may require  specific
> configuration of each component as well.
>
>
>
> My only issue is FUD, I do not want people to advertise that FS does  not
> work on ODBC under load when the whole story is FS does not work under  load
> when you use fooODBC with barDRIVER with basBASE database server  configured
> a particular way.
>
>
>
> It should never crash, if it does, its a bug in something.  I can  speak
> for FS that there is no bug there.  The other 3 are  variables.
>
> Remember this next time there is a discussion about using system libs  and
> why we build our own depends.  The one thing we use system libs  for, ODBC,
> is a huge PITA.........
>
>
>
> Things to remember:
>
>
>
> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem  and
> has been made the default in latest versions.
>
> 2) If you must use Mysql, enable transactions and make sure your  odbc.ini
> is referencing the threadsafe version of the lib
>
>      libmyodbc3_r.so <-- note _r
>
> 3) Try various combos of drivers and odbc libs, many time newer is not
>  better, stable versions lie in the past.
>
> 4) build the drivers and odbc yourself or get the debug symbols so you  can
> get a backtrace, you could be finding a bug for them.......
>
> 5) Find a test to reproduce your problem so you can try different  database
> engines and driver combos.
>
> 6) Make sure you are on latest FS git so you know you have a stable  copy.
>
>
>
>
>
> Anyway, this is a pain, that's why ppl pay you to do it.
>
> Discuss this all you want here, just minimize any FUD to scare away  people
> who want to use it.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com>
> wrote:
>
>
> Do not agree. Issues happening under extreme cases like  high load with
> some specific conditions are hard to reproduce, sharing  the
> information in this case is a lot more efficient, for all.
>
>
> Sent from my mobile device
>
>
>
>
> On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com>
> wrote:
>
> > My bologna  has a first name - it's O S C A R...
> >
> > This topic is getting  really old... I have to agree with Antm - get out
> of
> > the lazy  pants and do some research / hard work.
> >
> >
> >  -----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><freeswitch-users-bounces at lists.freeswitch.org>]
> On Behalf  Of Sergey
> > Okhapkin
> > Sent: Thursday, October 27, 2011 12:43  PM
> > To: FreeSWITCH Users Help
> > Subject: Re:  [Freeswitch-users] High load on database server
> >
> > What do  you mean "shared ODBC"?
> >
> > On Thursday 27 October 2011,  Madovsky wrote:
> >> but, is anyone experienced that with shared  ODBC ? (managing more than
> 330
> >> concurrent  calls)
> >>
> >> ----- Original Message -----
> >>  From: "Madovsky" <infos at madovsky.org>
> >> To: "FreeSWITCH Users  Help" <freeswitch-users at lists.freeswitch.org>
> >>  Sent: Thursday, October 27, 2011 12:33 PM
> >> Subject: Re:  [Freeswitch-users] High load on database  server
> >>
> >>> ha ok, good luck  so
> >>>
> >>> ----- Original Message  -----
> >>> From: "Cliff Wells" <cliff at develix.com>
> >>> To: "FreeSWITCH  Users Help" <freeswitch-users at lists.freeswitch.org>
> >>>  Sent: Thursday, October 27, 2011 12:26 PM
> >>> Subject: Re:  [Freeswitch-users] High load on database  server
> >>>
> >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
> >>>>> I means that depend the  quality of your script ;)
> >>>>
> >>>> Quality  isn't the issue here.  The script is too simple to be
>  incorrect
> >>>> (and it carefully releases the odbc  connection back to the pool in a
> >>>> hangup handler).   I have written poor-quality code plenty of times,
>  but
> >>>> 25 years as a programmer usually allows me the  luxury of knowing when
> >>>> I'm doing it,  thanks.
> >>>>
> >>>> In any case, Lua scripts  work fine.  Google will tell you that
> unixODBC
> >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies there,
>  but
> >>>> again, I emphasize, everything WORKS GREAT (been  using it for a
> couple
> >>>> of years now) until you get to  very high concurrency (about 330
> >>>> concurrent calls on a  single system).  If you don't expect to handle
> >>>>  more than 300 concurrent calls, then you do not need to worry about
>  it.
> >>>> At all.
> >>>>
> >>>>  Regards,
> >>>>  Cliff
> >>>>
> >>>>
> >>>>
> >>>>  FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>  http://www.freeswitch.org
> >>
> >>  FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>  http://www.freeswitch.org
> >
> >
> >
> >  FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
> >
> >
> >
> >  FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
>
>
> FreeSWITCH-users  mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
>
>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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 <sip%3A888 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 cliff at develix.com  Thu Oct 27 23:07:54 2011
From: cliff at develix.com (Cliff Wells)
Date: Thu, 27 Oct 2011 12:07:54 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1759442497975160368@unknownmsgid>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<2934644557544732041@unknownmsgid>
	<17277C2D60E140D0B41A28F4223062CC@e1705>
	<1319732799.2234.34.camel@portable-evil>
	<1759442497975160368@unknownmsgid>
Message-ID: <1319742474.2774.7.camel@portable-evil>

On Thu, 2011-10-27 at 18:42 +0200, Hynek Cihlar wrote:
> Did you try to locate the source of the segmantation fault? At least
> the module.

Here's a sample:

freeswitch[30818]: segfault at 710 ip 00007f343516b491 sp 00007f33f17b7fc0 error 4 in psqlodbcw.so[7f3435126000+7d000]

I know it's pointing at the PostgreSQL ODBC driver (and this may, in
fact, be the case), but it's hard to know whether the driver itself is
the issue, or whether mishandling of the driver is at fault.

Also, in 2.3.0, unixODBC stopped shipping their own PostgreSQL driver
and now use the one shipped with PostgreSQL instead, so this may have an
impact on stability.  In any case, until I'm able to do more testing
with 2.3.0 I'm just speculating.

Cliff





From anthony.minessale at gmail.com  Thu Oct 27 23:08:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 14:08:21 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <3244411099634732796@unknownmsgid>
References: <CACF0505.211E1%krice@freeswitch.org>
	<D5E62885E535486BB92F94CB5CBBC1A1@e1705>
	<CAKbxfG8Hm_a2CBK6S1upZm43rT3brbp8XDAFXuYcycNZneGB2Q@mail.gmail.com>
	<3244411099634732796@unknownmsgid>
Message-ID: <CAKbxfG-bCvU_tMPse5egBHZUf7oYLzxLaHzieMneycZCfi9F5Q@mail.gmail.com>

Blah,

I said I don't like mysql, but ok I'll look it up for you.......

http://www.mail-archive.com/profox at leafe.com/msg33150.html

AND

http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-connection-parameters.html
search for 'FLAG_MULTI_STATEMENTS'

>From our own FS resources:

http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg19883.html

or

http://tinyurl.com/4xo26sv






On Thu, Oct 27, 2011 at 1:29 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> Transactions are by default supported by mysql, the transactions are driven
> by sql statements themselves.
>
> Your statement is either incorrect or I interpret it wrong.
>
>
> Sent from my mobile device
>
> On Oct 27, 2011, at 20:09, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> mysql does not work with transactions by default in defense of injection
> attacks... read the thread its in the top.
>
>
> On Thu, Oct 27, 2011 at 1:07 PM, Madovsky < <infos at madovsky.org>
> infos at madovsky.org> wrote:
>
>> **
>> Thanks Ken. here the link of official SIPP website
>>  <http://sipp.sourceforge.net/>http://sipp.sourceforge.net/
>>
>> I didn't know it was a HP app ! :0)
>>
>> ----- Original Message -----
>> *From:* Ken Rice <krice at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Thursday, October 27, 2011 1:54 PM
>> *Subject:* Re: [Freeswitch-users] High load on database server
>>
>> SIPP works fine for this... However keep in mind some of their default
>> scenario files arent exactly the best in the world and can in many
>> situations leave a bit to be desired... Check the wiki I think there is more
>> info on this on there
>>
>> K
>>
>>
>> On 10/27/11 12:51 PM, "Madovsky" < <infos at madovsky.org>infos at madovsky.org>
>> wrote:
>>
>> on this subject,
>> is SIPp can be used to test to reproduce hundreds calls ?
>>
>>
>> ----- Original Message -----
>>
>> *From:*  Anthony Minessale < <anthony.minessale at gmail.com>
>> mailto:anthony.minessale at gmail.com <anthony.minessale at gmail.com>>
>>
>> *To:* FreeSWITCH Users Help < <freeswitch-users at lists.freeswitch.org>
>> mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
>>
>>
>> *Sent:* Thursday, October 27, 2011 1:39  PM
>>
>> *Subject:* Re: [Freeswitch-users] High load  on database server
>>
>>
>> BTW I know it's a contradiction to tell you to consider older  ODBC and
>> newer FS but I wrote FS so I can attest to its stability especially  the
>> ODBC code in the core.
>>
>>
>>
>>
>>
>>
>>
>> On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <<http://anthony.minessale at gmail.com>
>> anthony.minessale at gmail.com>  wrote:
>>
>>
>>
>> Let's recap
>>
>>
>>
>> You have 4 moving parts, you have various versions of all 4 to choose
>>  from and even different implementations of 3 of the 4.
>>
>>
>> ODBC LIB
>> ODBC DRIVER
>>
>> DATABASE
>>
>> FS
>>
>>
>>
>> You must choose a stable combination of all 4 which may require  specific
>> configuration of each component as well.
>>
>>
>>
>> My only issue is FUD, I do not want people to advertise that FS does  not
>> work on ODBC under load when the whole story is FS does not work under  load
>> when you use fooODBC with barDRIVER with basBASE database server  configured
>> a particular way.
>>
>>
>>
>> It should never crash, if it does, its a bug in something.  I can  speak
>> for FS that there is no bug there.  The other 3 are  variables.
>>
>> Remember this next time there is a discussion about using system libs  and
>> why we build our own depends.  The one thing we use system libs  for, ODBC,
>> is a huge PITA.........
>>
>>
>>
>> Things to remember:
>>
>>
>>
>> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem  and
>> has been made the default in latest versions.
>>
>> 2) If you must use Mysql, enable transactions and make sure your  odbc.ini
>> is referencing the threadsafe version of the lib
>>
>>      libmyodbc3_r.so <-- note _r
>>
>> 3) Try various combos of drivers and odbc libs, many time newer is not
>>  better, stable versions lie in the past.
>>
>> 4) build the drivers and odbc yourself or get the debug symbols so you
>>  can get a backtrace, you could be finding a bug for them.......
>>
>> 5) Find a test to reproduce your problem so you can try different
>>  database engines and driver combos.
>>
>> 6) Make sure you are on latest FS git so you know you have a stable  copy.
>>
>>
>>
>>
>>
>> Anyway, this is a pain, that's why ppl pay you to do it.
>>
>> Discuss this all you want here, just minimize any FUD to scare away
>>  people who want to use it.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <<http://hynek.cihlar at gmail.com>
>> hynek.cihlar at gmail.com> wrote:
>>
>>
>> Do not agree. Issues happening under extreme cases like  high load with
>> some specific conditions are hard to reproduce, sharing  the
>> information in this case is a lot more efficient, for all.
>>
>>
>> Sent from my mobile device
>>
>>
>>
>>
>> On Oct 27, 2011, at 18:56, Robert Huddleston <<http://rhuddleston at gmail.com>
>> rhuddleston at gmail.com> wrote:
>>
>> > My bologna  has a first name - it's O S C A R...
>> >
>> > This topic is getting  really old... I have to agree with Antm - get out
>> of
>> > the lazy  pants and do some research / hard work.
>> >
>> >
>> >  -----Original Message-----
>> > From: <http://freeswitch-users-bounces at lists.freeswitch.org>
>> freeswitch-users-bounces at lists.freeswitch.org
>> >  [ <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 Sergey
>> > Okhapkin
>> > Sent: Thursday, October 27, 2011 12:43  PM
>> > To: FreeSWITCH Users Help
>> > Subject: Re:  [Freeswitch-users] High load on database server
>> >
>> > What do  you mean "shared ODBC"?
>> >
>> > On Thursday 27 October 2011,  Madovsky wrote:
>> >> but, is anyone experienced that with shared  ODBC ? (managing more than
>> 330
>> >> concurrent  calls)
>> >>
>> >> ----- Original Message -----
>> >>  From: "Madovsky" < <http://infos at madovsky.org>infos at madovsky.org>
>> >> To: "FreeSWITCH Users  Help" <<http://freeswitch-users at lists.freeswitch.org>
>> freeswitch-users at lists.freeswitch.org>
>> >>  Sent: Thursday, October 27, 2011 12:33 PM
>> >> Subject: Re:  [Freeswitch-users] High load on database  server
>> >>
>> >>> ha ok, good luck  so
>> >>>
>> >>> ----- Original Message  -----
>> >>> From: "Cliff Wells" < <http://cliff at develix.com>cliff at develix.com>
>> >>> To: "FreeSWITCH  Users Help" <<http://freeswitch-users at lists.freeswitch.org>
>> freeswitch-users at lists.freeswitch.org>
>> >>>  Sent: Thursday, October 27, 2011 12:26 PM
>> >>> Subject: Re:  [Freeswitch-users] High load on database  server
>> >>>
>> >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
>> >>>>> I means that depend the  quality of your script ;)
>> >>>>
>> >>>> Quality  isn't the issue here.  The script is too simple to be
>>  incorrect
>> >>>> (and it carefully releases the odbc  connection back to the pool in a
>> >>>> hangup handler).   I have written poor-quality code plenty of times,
>>  but
>> >>>> 25 years as a programmer usually allows me the  luxury of knowing
>> when
>> >>>> I'm doing it,  thanks.
>> >>>>
>> >>>> In any case, Lua scripts  work fine.  Google will tell you that
>> unixODBC
>> >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies there,
>>  but
>> >>>> again, I emphasize, everything WORKS GREAT (been  using it for a
>> couple
>> >>>> of years now) until you get to  very high concurrency (about 330
>> >>>> concurrent calls on a  single system).  If you don't expect to handle
>> >>>>  more than 300 concurrent calls, then you do not need to worry about
>>  it.
>> >>>> At all.
>> >>>>
>> >>>>  Regards,
>> >>>>  Cliff
>> >>>>
>> >>>>
>> >>>>
>> >>>>  FreeSWITCH-users mailing list
>> >>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>> 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
>> >>
>> >>  FreeSWITCH-users mailing list
>> >> <http://FreeSWITCH-users at lists.freeswitch.org>
>> 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
>> >
>> >
>> >
>> >  FreeSWITCH-users mailing list
>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>> 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
>> >
>> >
>> >
>> >  FreeSWITCH-users mailing list
>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>> 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
>>
>>
>> FreeSWITCH-users  mailing list
>> <http://FreeSWITCH-users at lists.freeswitch.org>
>> 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
>>
>>
>>
>>
>>
>>
>>
>>  ------------------------------
>>
>>
>> FreeSWITCH-users mailing list
>> <FreeSWITCH-users at lists.freeswitch.org>
>> 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
>>
>>
>>
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> 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
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH <http://www.freeswitch.org/>http://www.freeswitch.org/
> ClueCon <http://www.cluecon.com/>http://www.cluecon.com/
> Twitter: <http://twitter.com/FreeSWITCH_wire>
> http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: <http://irc.freenode.net>irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip: <sip%3A888 at conference.freeswitch.org>888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Oct 27 23:11:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 14:11:42 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1319742474.2774.7.camel@portable-evil>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<2934644557544732041@unknownmsgid>
	<17277C2D60E140D0B41A28F4223062CC@e1705>
	<1319732799.2234.34.camel@portable-evil>
	<1759442497975160368@unknownmsgid>
	<1319742474.2774.7.camel@portable-evil>
Message-ID: <CAKbxfG99gTcpqukR1-XycbWUtUkbDx=fAhY2mhXZjCtu6JVfwA@mail.gmail.com>

That's not useful, you need to collect the core and run gdb on it.


On Thu, Oct 27, 2011 at 2:07 PM, Cliff Wells <cliff at develix.com> wrote:

> On Thu, 2011-10-27 at 18:42 +0200, Hynek Cihlar wrote:
> > Did you try to locate the source of the segmantation fault? At least
> > the module.
>
> Here's a sample:
>
> freeswitch[30818]: segfault at 710 ip 00007f343516b491 sp 00007f33f17b7fc0
> error 4 in psqlodbcw.so[7f3435126000+7d000]
>
> I know it's pointing at the PostgreSQL ODBC driver (and this may, in
> fact, be the case), but it's hard to know whether the driver itself is
> the issue, or whether mishandling of the driver is at fault.
>
> Also, in 2.3.0, unixODBC stopped shipping their own PostgreSQL driver
> and now use the one shipped with PostgreSQL instead, so this may have an
> impact on stability.  In any case, until I'm able to do more testing
> with 2.3.0 I'm just speculating.
>
> Cliff
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 cliff at develix.com  Thu Oct 27 23:28:29 2011
From: cliff at develix.com (Cliff Wells)
Date: Thu, 27 Oct 2011 12:28:29 -0700
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG99gTcpqukR1-XycbWUtUkbDx=fAhY2mhXZjCtu6JVfwA@mail.gmail.com>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<2934644557544732041@unknownmsgid>
	<17277C2D60E140D0B41A28F4223062CC@e1705>
	<1319732799.2234.34.camel@portable-evil>
	<1759442497975160368@unknownmsgid>
	<1319742474.2774.7.camel@portable-evil>
	<CAKbxfG99gTcpqukR1-XycbWUtUkbDx=fAhY2mhXZjCtu6JVfwA@mail.gmail.com>
Message-ID: <1319743709.2774.9.camel@portable-evil>

On Thu, 2011-10-27 at 14:11 -0500, Anthony Minessale wrote:
> That's not useful, you need to collect the core and run gdb on it.

I also assume I'd need the debug versions of the related software
installed?

Cliff


> 
> On Thu, Oct 27, 2011 at 2:07 PM, Cliff Wells <cliff at develix.com>
> wrote:
>         On Thu, 2011-10-27 at 18:42 +0200, Hynek Cihlar wrote:
>         > Did you try to locate the source of the segmantation fault?
>         At least
>         > the module.
>         
>         
>         Here's a sample:
>         
>         freeswitch[30818]: segfault at 710 ip 00007f343516b491 sp
>         00007f33f17b7fc0 error 4 in psqlodbcw.so[7f3435126000+7d000]
>         
>         I know it's pointing at the PostgreSQL ODBC driver (and this
>         may, in
>         fact, be the case), but it's hard to know whether the driver
>         itself is
>         the issue, or whether mishandling of the driver is at fault.
>         
>         Also, in 2.3.0, unixODBC stopped shipping their own PostgreSQL
>         driver
>         and now use the one shipped with PostgreSQL instead, so this
>         may have an
>         impact on stability.  In any case, until I'm able to do more
>         testing
>         with 2.3.0 I'm just speculating.
>         
>         
>         Cliff
>         
>         
>         
>         
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Thu Oct 27 23:32:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 14:32:59 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <1319743709.2774.9.camel@portable-evil>
References: <A49B2950111D4F7CA5CBBC805B21B970@e1705>
	<2934644557544732041@unknownmsgid>
	<17277C2D60E140D0B41A28F4223062CC@e1705>
	<1319732799.2234.34.camel@portable-evil>
	<1759442497975160368@unknownmsgid>
	<1319742474.2774.7.camel@portable-evil>
	<CAKbxfG99gTcpqukR1-XycbWUtUkbDx=fAhY2mhXZjCtu6JVfwA@mail.gmail.com>
	<1319743709.2774.9.camel@portable-evil>
Message-ID: <CAKbxfG8Lz7xT1yZu9hSbFMd6417a7cY8vVCXCRVQYJmUS3pbOw@mail.gmail.com>

ya

On Thu, Oct 27, 2011 at 2:28 PM, Cliff Wells <cliff at develix.com> wrote:

> On Thu, 2011-10-27 at 14:11 -0500, Anthony Minessale wrote:
> > That's not useful, you need to collect the core and run gdb on it.
>
> I also assume I'd need the debug versions of the related software
> installed?
>
> Cliff
>
>
> >
> > On Thu, Oct 27, 2011 at 2:07 PM, Cliff Wells <cliff at develix.com>
> > wrote:
> >         On Thu, 2011-10-27 at 18:42 +0200, Hynek Cihlar wrote:
> >         > Did you try to locate the source of the segmantation fault?
> >         At least
> >         > the module.
> >
> >
> >         Here's a sample:
> >
> >         freeswitch[30818]: segfault at 710 ip 00007f343516b491 sp
> >         00007f33f17b7fc0 error 4 in psqlodbcw.so[7f3435126000+7d000]
> >
> >         I know it's pointing at the PostgreSQL ODBC driver (and this
> >         may, in
> >         fact, be the case), but it's hard to know whether the driver
> >         itself is
> >         the issue, or whether mishandling of the driver is at fault.
> >
> >         Also, in 2.3.0, unixODBC stopped shipping their own PostgreSQL
> >         driver
> >         and now use the one shipped with PostgreSQL instead, so this
> >         may have an
> >         impact on stability.  In any case, until I'm able to do more
> >         testing
> >         with 2.3.0 I'm just speculating.
> >
> >
> >         Cliff
> >
> >
> >
> >
> >
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From krice at freeswitch.org  Thu Oct 27 23:35:56 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Oct 2011 14:35:56 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG-bCvU_tMPse5egBHZUf7oYLzxLaHzieMneycZCfi9F5Q@mail.gmail.com>
Message-ID: <CACF1CCC.21213%krice@freeswitch.org>

Hey Anthony,

I think he?s being a little confused here...

What tony is refering to, is stacked statements   ?select * from foo; update
bar set foo=bar; some other statement;?

Allowing stacked statements in 1 call over ODBC is bad mojo that?s exactly
host most sql injections work in the first place

?select * from users where username = ?$USERNAME?? ... Replace $USERNAME
with the next line

?;  UPDATE users set password=NEWPASSWORD where username = ?admin?; --

Boom sql injection... Now there are other ways to protect against this like
properly escaping input from 3rd parties going into your sql statements but
it still happens all the time...

This is one of the main reasons I use prepared statements on Postgresql
cause it stops this sort of behavior cold in its tracks... You should still
properly escape inputs but the extra layer is worth it




On 10/27/11 2:08 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> Blah,
> 
> I said I don't like mysql, but ok I'll look it up for you.......
> 
> http://www.mail-archive.com/profox at leafe.com/msg33150.html
> 
> AND
> 
> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-connection
> -parameters.html
> search for 'FLAG_MULTI_STATEMENTS'?
> 
> From our own FS resources:
> 
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
> 
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg19883.htm>
l
> 
> or?
> 
> http://tinyurl.com/4xo26sv
> 
> 
> 
> 
> 
> 
> On Thu, Oct 27, 2011 at 1:29 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:
>> Transactions are by default supported by mysql, the transactions are driven
>> by sql statements themselves.?
>> 
>> Your statement is either incorrect or I interpret it wrong.?
>> 
>> 
>> Sent from my mobile device
>> 
>> On Oct 27, 2011, at 20:09, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>> 
>>> mysql does not work with transactions by default in defense of injection
>>> attacks... read the thread its in the top.
>>> 
>>> 
>>> On Thu, Oct 27, 2011 at 1:07 PM, Madovsky < <mailto:infos at madovsky.org>
>>> infos at madovsky.org> wrote:
>>>> Thanks Ken. here the link of official SIPP website
>>>>  <http://sipp.sourceforge.net/> http://sipp.sourceforge.net/
>>>> ?
>>>> I didn't know it was a HP app ! :0)
>>>>>  
>>>>> ----- Original Message -----
>>>>>  
>>>>> From:  Ken Rice <mailto:krice at freeswitch.org>
>>>>>  
>>>>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
>>>>>  
>>>>> Sent: Thursday, October 27, 2011 1:54  PM
>>>>>  
>>>>> Subject: Re: [Freeswitch-users] High load  on database server
>>>>>  
>>>>> 
>>>>> SIPP works fine for this... However keep in mind some  of their default
>>>>> scenario files arent exactly the best in the world and can in  many
>>>>> situations leave a bit to be desired... Check the wiki I think there is
>>>>> more info on this on there
>>>>> 
>>>>> K
>>>>> 
>>>>> 
>>>>> On 10/27/11 12:51 PM,  "Madovsky" < <mailto:infos at madovsky.org>
>>>>> infos at madovsky.org>  wrote:
>>>>> 
>>>>>  
>>>>>> on this subject,
>>>>>> is SIPp can be used to test to  reproduce hundreds calls ?
>>>>>>  
>>>>>>> 
>>>>>>> ----- Original Message -----
>>>>>>> ?
>>>>>>> From: ?Anthony Minessale < <mailto:anthony.minessale at gmail.com>
>>>>>>> mailto:anthony.minessale at gmail.com>  ?
>>>>>>> ?
>>>>>>> To: FreeSWITCH Users Help <
>>>>>>> <mailto:freeswitch-users at lists.freeswitch.org>
>>>>>>> mailto:freeswitch-users at lists.freeswitch.org>  ??
>>>>>>> ?
>>>>>>> Sent: Thursday, October 27, 2011 1:39  ?PM
>>>>>>> ?
>>>>>>> Subject: Re: [Freeswitch-users] High load  ?on database server
>>>>>>> ?
>>>>>>> 
>>>>>>> BTW I know it's a contradiction  to tell you to consider older ?ODBC and
>>>>>>> newer FS but I wrote FS so I  can attest to its stability especially
>>>>>>> ?the ODBC code in the core.
>>>>>>> 
>>>>>>> ?
>>>>>>> 
>>>>>>> ?
>>>>>>> 
>>>>>>> 
>>>>>>> ?
>>>>>>> On Thu, Oct 27, 2011 at  12:36 PM, Anthony Minessale ?<
>>>>>>> <http://anthony.minessale at gmail.com> anthony.minessale at gmail.com>
>>>>>>> ?wrote:
>>>>>>> ?
>>>>>>>  

Let's recap
?

?
You  have 4 moving parts, you have various versions of all 4 to choose
?from and even different implementations of 3 of the  4.
?

ODBC LIB 
ODBC  DRIVER
?
DATABASE
?
FS
?

?
You  must choose a stable combination of all 4 which may require  ?specific
configuration of each component as  well.
?

?
My only issue is FUD, I do not want  people to advertise that FS does ?not
work on ODBC under load when  the whole story is FS does not work under
?load when you use  fooODBC with barDRIVER with basBASE database server
?configured a  particular way.
?

?
It should never crash, if it  does, its a bug in something. ?I can ?speak
for FS that there  is no bug there. ?The other 3 are ?variables.
?
Remember this next time there is a discussion about using  system libs ?and
why we build our own depends. ?The one thing  we use system libs ?for, ODBC,
is a huge  PITA.........
?

?
Things to  remember:
?

?
1) try the Threading=0 in the  odbcinst.ini, this is a serious problem ?and
has been made the  default in latest versions.
?
2) If you must use Mysql,  enable transactions and make sure your ?odbc.ini
is referencing the  threadsafe version of the lib
?
?????libmyodbc3_r.so <-- note  _r
?
3) Try various combos of drivers and odbc libs, many  time newer is not
?better, stable versions lie in the  past.
?
4) build the drivers and odbc yourself or get the  debug symbols so you ?can
get a backtrace, you could be finding a  bug for them.......
?
5) Find a test to reproduce your  problem so you can try different ?database
engines and driver  combos.
?
6) Make sure you are on latest FS git so you know  you have a stable  ?copy.
?

?

?
Anyway, this is a  pain, that's why ppl pay you to do it.
?
Discuss this all you  want here, just minimize any FUD to scare away ?people
who want to  use  it.
?

?

?

?

?
?
?
?


?
On  Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <
<http://hynek.cihlar at gmail.com> hynek.cihlar at gmail.com>  wrote:
?
 
Do not agree. Issues happening under extreme  cases like ?high load with
some specific conditions are hard  to reproduce, sharing ?the
information in this case is a lot  more efficient, for all.
?

Sent from my mobile  device

?
?
?
On Oct 27, 2011, at 18:56,  Robert Huddleston <
<http://rhuddleston at gmail.com> rhuddleston at gmail.com>  wrote:

> My bologna ?has a first name - it's O S C A  R...
>
> This topic is getting ?really old... I have  to agree with Antm - get out of
> the lazy ?pants and do  some research / hard work.
>
>
> ?-----Original  Message-----
> From:  <http://freeswitch-users-bounces at lists.freeswitch.org>
freeswitch-users-bounces at lists.freeswitch.org
>  ?[ <mailto:freeswitch-users-bounces at lists.freeswitch.org>
mailto:freeswitch-users-bounces at lists.freeswitch.org]  On Behalf ?Of Sergey
> Okhapkin
> Sent: Thursday,  October 27, 2011 12:43 ?PM
> To: FreeSWITCH Users  Help
> Subject: Re: ?[Freeswitch-users] High load on  database server
>
> What do ?you mean "shared  ODBC"?
>
> On Thursday 27 October 2011, ?Madovsky  wrote:
>> but, is anyone experienced that with shared  ?ODBC ? (managing more than 330
>> concurrent  ?calls)
>>
>> ----- Original Message  -----
>> ?From: "Madovsky" < <http://infos at madovsky.org> infos at madovsky.org>
>> To:  "FreeSWITCH Users ?Help" <
<http://freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org>
>>  ?Sent: Thursday, October 27, 2011 12:33 PM
>> Subject:  Re: ?[Freeswitch-users] High load on database  ?server
>>
>>> ha ok, good luck  ?so
>>>
>>> ----- Original Message  ?-----
>>> From: "Cliff Wells" < <http://cliff at develix.com> cliff at develix.com>
>>> To:  "FreeSWITCH ?Users Help" <
<http://freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org>
>>>  ?Sent: Thursday, October 27, 2011 12:26 PM
>>>  Subject: Re: ?[Freeswitch-users] High load on database  ?server
>>>
>>>> On Thu, 2011-10-27 at  11:55 ?-0400, Madovsky wrote:
>>>>> I means  that depend the ?quality of your script  ;)
>>>>
>>>> Quality ?isn't the  issue here. ?The script is too simple to be  ?incorrect
>>>> (and it carefully releases the  odbc ?connection back to the pool in a
>>>> hangup  handler). ??I have written poor-quality code plenty of  times, ?but
>>>> 25 years as a programmer usually  allows me the ?luxury of knowing when
>>>> I'm  doing it, ?thanks.
>>>>
>>>> In any  case, Lua scripts ?work fine. ?Google will tell you that  unixODBC
>>>> ?before 2.3.0 was a bit of a mess,  so I expect the issue lies there, ?but
>>>> again,  I emphasize, everything WORKS GREAT (been ?using it for a  couple
>>>> of years now) until you get to ?very  high concurrency (about 330
>>>> concurrent calls on a  ?single system). ?If you don't expect to  handle
>>>> ?more than 300 concurrent calls, then  you do not need to worry about ?it.
>>>> At  all.
>>>>
>>>>  ?Regards,
>>>>  ?Cliff
>>>>
>>>>
>>>>
>>>>  ?FreeSWITCH-users mailing list
>>>>  <http://FreeSWITCH-users at lists.freeswitch.org>
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
>>
>>  ?FreeSWITCH-users mailing list
>>  <http://FreeSWITCH-users at lists.freeswitch.org>
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
>
>
>
>  ?FreeSWITCH-users mailing list
>  <http://FreeSWITCH-users at lists.freeswitch.org>
FreeSWITCH-users at lists.freeswitch.org
>   <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
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>  ?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
>
>
>
>  ?FreeSWITCH-users mailing list
>  <http://FreeSWITCH-users at lists.freeswitch.org>
FreeSWITCH-users at lists.freeswitch.org
>   <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
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>  ?UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
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>  ? <http://www.freeswitch.org> http://www.freeswitch.org


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

?
>>>>>  
>>>>>  
>>>>> 
>>>>> 
>>>>>  
>>>>> 
>>>>> 
>>>>> FreeSWITCH-users mailing  list
>>>>>  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>> 
>>>> 
>>>> 
>>>> FreeSWITCH-users mailing list
>>>>  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
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>>>> 
>>> 
>>> 

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From anthony.minessale at gmail.com  Thu Oct 27 23:39:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Oct 2011 14:39:37 -0500
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CACF1CCC.21213%krice@freeswitch.org>
References: <CAKbxfG-bCvU_tMPse5egBHZUf7oYLzxLaHzieMneycZCfi9F5Q@mail.gmail.com>
	<CACF1CCC.21213%krice@freeswitch.org>
Message-ID: <CAKbxfG8U07W_+j7wZZMejvM5AqN7ieSC3BJaWXL9AG6rJC3LYg@mail.gmail.com>

ya so furthermore,

The core and sofia has to be generic so it uses ANSI sql transactions with
BEGIN; <2000 sql stmts> COMMIT;
This falls under the category of multi-line stmts and breaks w/o that
enabled thus making the DB suck much more.


On Thu, Oct 27, 2011 at 2:35 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Hey Anthony,
>
> I think he?s being a little confused here...
>
> What tony is refering to, is stacked statements   ?select * from foo;
> update bar set foo=bar; some other statement;?
>
> Allowing stacked statements in 1 call over ODBC is bad mojo that?s exactly
> host most sql injections work in the first place
>
> ?select * from users where username = ?$USERNAME?? ... Replace $USERNAME
> with the next line
>
> ?;  UPDATE users set password=NEWPASSWORD where username = ?admin?; --
>
> Boom sql injection... Now there are other ways to protect against this like
> properly escaping input from 3rd parties going into your sql statements but
> it still happens all the time...
>
> This is one of the main reasons I use prepared statements on Postgresql
> cause it stops this sort of behavior cold in its tracks... You should still
> properly escape inputs but the extra layer is worth it
>
>
>
>
>
> On 10/27/11 2:08 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
> wrote:
>
> Blah,
>
> I said I don't like mysql, but ok I'll look it up for you.......
>
> http://www.mail-archive.com/profox at leafe.com/msg33150.html
>
> AND
>
>
> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-connection-parameters.html
> search for 'FLAG_MULTI_STATEMENTS'
>
> >From our own FS resources:
>
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg19883.html
>
> or
>
> http://tinyurl.com/4xo26sv
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at 1:29 PM, Hynek Cihlar <hynek.cihlar at gmail.com>
> wrote:
>
> Transactions are by default supported by mysql, the transactions are driven
> by sql statements themselves.
>
> Your statement is either incorrect or I interpret it wrong.
>
>
> Sent from my mobile device
>
> On Oct 27, 2011, at 20:09, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> mysql does not work with transactions by default in defense of injection
> attacks... read the thread its in the top.
>
>
> On Thu, Oct 27, 2011 at 1:07 PM, Madovsky < <mailto:infos at madovsky.org<infos at madovsky.org>>
> infos at madovsky.org> wrote:
>
> Thanks Ken. here the link of official SIPP website
>  <http://sipp.sourceforge.net/> http://sipp.sourceforge.net/
>
>
> I didn't know it was a HP app ! :0)
>
>
> ----- Original Message -----
>
> *From:*  Ken Rice <mailto:krice at freeswitch.org <krice at freeswitch.org>>
>
> *To:* FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
>
>
> *Sent:* Thursday, October 27, 2011 1:54  PM
>
> *Subject:* Re: [Freeswitch-users] High load  on database server
>
>
> SIPP works fine for this... However keep in mind some  of their default
> scenario files arent exactly the best in the world and can in  many
> situations leave a bit to be desired... Check the wiki I think there is
>  more info on this on there
>
> K
>
>
> On 10/27/11 12:51 PM,  "Madovsky" < <mailto:infos at madovsky.org<infos at madovsky.org>>
> infos at madovsky.org>  wrote:
>
>
>
> on this subject,
> is SIPp can be used to test to  reproduce hundreds calls ?
>
>
>
> ----- Original Message -----
>
> *From:*  Anthony Minessale < <mailto:anthony.minessale at gmail.com<anthony.minessale at gmail.com>>
> mailto:anthony.minessale at gmail.com <anthony.minessale at gmail.com>>
>
> *To:* FreeSWITCH Users Help < <
> mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
> mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
>
>
> *Sent:* Thursday, October 27, 2011 1:39   PM
>
> *Subject:* Re: [Freeswitch-users] High load   on database server
>
>
> BTW I know it's a contradiction  to tell you to consider older  ODBC and
> newer FS but I wrote FS so I  can attest to its stability especially  the
> ODBC code in the core.
>
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at  12:36 PM, Anthony Minessale  < <
> http://anthony.minessale at gmail.com> anthony.minessale at gmail.com>   wrote:
>
>
>
>
> Let's recap
>
>
>
> You  have 4 moving parts, you have various versions of all 4 to choose
>   from and even different implementations of 3 of the  4.
>
>
> ODBC LIB
> ODBC  DRIVER
>
> DATABASE
>
> FS
>
>
>
> You  must choose a stable combination of all 4 which may require   specific
> configuration of each component as  well.
>
>
>
> My only issue is FUD, I do not want  people to advertise that FS does  not
> work on ODBC under load when  the whole story is FS does not work under
>  load when you use  fooODBC with barDRIVER with basBASE database server
>  configured a  particular way.
>
>
>
> It should never crash, if it  does, its a bug in something.  I can  speak
> for FS that there  is no bug there.  The other 3 are  variables.
>
> Remember this next time there is a discussion about using  system libs  and
> why we build our own depends.  The one thing  we use system libs  for, ODBC,
> is a huge  PITA.........
>
>
>
> Things to  remember:
>
>
>
> 1) try the Threading=0 in the  odbcinst.ini, this is a serious problem  and
> has been made the  default in latest versions.
>
> 2) If you must use Mysql,  enable transactions and make sure your  odbc.ini
> is referencing the  threadsafe version of the lib
>
>      libmyodbc3_r.so <-- note  _r
>
> 3) Try various combos of drivers and odbc libs, many  time newer is not
>  better, stable versions lie in the  past.
>
> 4) build the drivers and odbc yourself or get the  debug symbols so you
>  can get a backtrace, you could be finding a  bug for them.......
>
> 5) Find a test to reproduce your  problem so you can try different
>  database engines and driver  combos.
>
> 6) Make sure you are on latest FS git so you know  you have a stable
>   copy.
>
>
>
>
>
> Anyway, this is a  pain, that's why ppl pay you to do it.
>
> Discuss this all you  want here, just minimize any FUD to scare away
>  people who want to  use  it.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On  Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar < <
> http://hynek.cihlar at gmail.com> hynek.cihlar at gmail.com>  wrote:
>
>
>
> Do not agree. Issues happening under extreme  cases like  high load with
> some specific conditions are hard  to reproduce, sharing  the
> information in this case is a lot  more efficient, for all.
>
>
> Sent from my mobile  device
>
>
>
>
> On Oct 27, 2011, at 18:56,  Robert Huddleston < <
> http://rhuddleston at gmail.com> rhuddleston at gmail.com>  wrote:
>
>
> > My bologna  has a first name - it's O S C A  R...
> >
> > This topic is getting  really old... I have  to agree with Antm - get out
> of
> > the lazy  pants and do  some research / hard work.
> >
> >
> >  -----Original  Message-----
> > From:  <http://freeswitch-users-bounces at lists.freeswitch.org>
> freeswitch-users-bounces at lists.freeswitch.org
> >   [ <mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>>
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
>  On Behalf  Of Sergey
>
> > Okhapkin
> > Sent: Thursday,  October 27, 2011 12:43  PM
> > To: FreeSWITCH Users  Help
> > Subject: Re:  [Freeswitch-users] High load on  database server
> >
> > What do  you mean "shared  ODBC"?
> >
> > On Thursday 27 October 2011,  Madovsky  wrote:
> >> but, is anyone experienced that with shared   ODBC ? (managing more than
> 330
> >> concurrent   calls)
> >>
> >> ----- Original Message  -----
> >>  From: "Madovsky" < <http://infos at madovsky.org> infos at madovsky.org>
> >> To:  "FreeSWITCH Users  Help" < <
> http://freeswitch-users at lists.freeswitch.org>
> freeswitch-users at lists.freeswitch.org>
> >>   Sent: Thursday, October 27, 2011 12:33 PM
> >> Subject:  Re:  [Freeswitch-users] High load on database   server
> >>
> >>> ha ok, good luck   so
> >>>
> >>> ----- Original Message   -----
> >>> From: "Cliff Wells" < <http://cliff at develix.com> cliff at develix.com>
> >>> To:  "FreeSWITCH  Users Help" < <
> http://freeswitch-users at lists.freeswitch.org>
> freeswitch-users at lists.freeswitch.org>
> >>>   Sent: Thursday, October 27, 2011 12:26 PM
> >>>  Subject: Re:  [Freeswitch-users] High load on database   server
> >>>
> >>>> On Thu, 2011-10-27 at  11:55  -0400, Madovsky wrote:
> >>>>> I means  that depend the  quality of your script  ;)
> >>>>
> >>>> Quality  isn't the  issue here.  The script is too simple to be
>   incorrect
> >>>> (and it carefully releases the  odbc  connection back to the pool in a
> >>>> hangup  handler).   I have written poor-quality code plenty of  times,
>  but
> >>>> 25 years as a programmer usually  allows me the  luxury of knowing
> when
> >>>> I'm  doing it,  thanks.
> >>>>
> >>>> In any  case, Lua scripts  work fine.  Google will tell you that
>  unixODBC
> >>>>  before 2.3.0 was a bit of a mess,  so I expect the issue lies there,
>  but
> >>>> again,  I emphasize, everything WORKS GREAT (been  using it for a
>  couple
> >>>> of years now) until you get to  very  high concurrency (about 330
> >>>> concurrent calls on a   single system).  If you don't expect to
>  handle
> >>>>  more than 300 concurrent calls, then  you do not need to worry about
>  it.
> >>>> At  all.
> >>>>
> >>>>   Regards,
> >>>>   Cliff
> >>>>
> >>>>
> >>>>
> >>>>   FreeSWITCH-users mailing list
> >>>>  <http://FreeSWITCH-users at lists.freeswitch.org>
> 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
> >>
> >>   FreeSWITCH-users mailing list
> >>  <http://FreeSWITCH-users at lists.freeswitch.org>
> 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
> >
> >
> >
> >   FreeSWITCH-users mailing list
> >  <http://FreeSWITCH-users at lists.freeswitch.org>
> 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
> >
> >
> >
> >   FreeSWITCH-users mailing list
> >  <http://FreeSWITCH-users at lists.freeswitch.org>
> 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
>
>
> FreeSWITCH-users   mailing list
>  <http://FreeSWITCH-users at lists.freeswitch.org>
> FreeSWITCH-users at lists.freeswitch.org
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>
>
>
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>
>
> ------------------------------
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>
> FreeSWITCH-users mailing  list
>  <mailto:FreeSWITCH-users at lists.freeswitch.org<FreeSWITCH-users at lists.freeswitch.org>>
> 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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>
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>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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From infos at madovsky.org  Fri Oct 28 00:33:46 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 16:33:46 -0400
Subject: [Freeswitch-users] originate_disposition
Message-ID: <D16AF8797FA347B3A2E1923AAE592321@e1705>

Afer hours of test I didn' succeed how
to catch the result of originate_disposition

--------

I have this extensions in my dialplan

         <extension name="internal_test">
                <condition field="${sip_to_uri}" expression="^(999\d{10})@$${domain}$">
                        <action inline="true" application="set" data="nibble_account="/>
                        <action inline="true" application="set" data="nibble_rate="/>
                        <action inline="true" application="set" data="dialed_extension=$1"/>
                </condition>
                <condition field="${from_admin_services}" expression="^true$" break="never">
                        <action application="set" data="ignore_early_media=true"/>
                        <action application="set" data="hangup_after_bridge=false"/>
                        <action application="set" data="continue_on_fail=true"/>
                        <action application="bridge" data="{sip_append_audio_sdp='a=rtpmap:0 PCMU/8000',sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
                </condition>
        </extension>
        <extension name="internal_test_tools">
                <condition field="${originate_disposition}" expression="^SUBSCRIBER_ABSENT$" break="on-true">
                        <action application="phrase" data="bad_ob_prepaid_dest"/>
                        <action application="transfer" data="ob_prepaid_call XML features"/>
                        <action application="info"/>
                </condition>
                <condition field="${cond(expr(${originate_disposition} != NORMAL_CLEARING) ? YES : NO)}" expression="^YES$"/>
                <condition field="${cond(expr(${originate_disposition} != SUBSCRIBER_ABSENT) ? YES : NO)}" expression="^YES$">
                        <action application="ring_ready"/>
                        <action application="sleep" data="2000"/>
                        <action application="answer"/>
                        <action application="sleep" data="1400"/>
                        <action application="voicemail" data="default $${domain} ${dialed_extension}"/>
               </condition>
        </extension>


-----------


I tried to merge it in one extension without success also.
where is the trick ? :)

Thanks

Franck
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From trever.adams at gmail.com  Fri Oct 28 00:44:49 2011
From: trever.adams at gmail.com (Trever L. Adams)
Date: Thu, 27 Oct 2011 14:44:49 -0600
Subject: [Freeswitch-users] trouble making conditions for OR
Message-ID: <4EA9C2C1.1020809@gmail.com>

Hello everyone,

The last few days I am beginning to explore freeswitch. I am having a
bit of trouble. I am trying to user the examples near
http://wiki.freeswitch.org/wiki/Dialplan_XML#Complex_Condition.2FAction_Rules
in the documentation. If I use '\' as it says, I get an error about
</extension> being unexpected. If I use '/' it doesn't work. Below is my
dialplan. It should trigger on empty number as well as the strings
mentioned. I am pretty sure the regex is accurate. However, instead of
running, I get:


2011-10-27 14:38:16.269266 [INFO] mod_dialplan_xml.c:336 Processing
private <private>-> in context Incoming-FXO
2011-10-27 14:38:16.269266 [INFO] switch_core_state_machine.c:177 No
Route, Aborting

The dialplan:

<include>
    <context name="Incoming-FXO">
        <extension name="screen">
            <condition field="caller_id_number" expression="^$"
break="on-true"/>
            <condition field="caller_id_number" expression=".nonymous.*"
break="on-true"/>
            <condition field="caller_id_number" expression=".nknown.*"
break="on-true"/>
            <condition field="caller_id_number" expression=".*rivate.*"
break="on-true"/>
            <condition field="caller_id_number"
expression=".estricted.*" break="on-true"/>
            <condition field="caller_id_number" expression="PTSN.*"
break="on-true"/>
            <condition field="caller_id_number" expression=".oll.*"
break="on-true"/>
            <condition field="caller_id_name" expression=".nonymous.*"
break="on-true"/>
            <condition field="caller_id_name" expression=".nknown.*"
break="on-true"/>
            <condition field="caller_id_name" expression=".*rivate.*"
break="on-true"/>
            <condition field="caller_id_name" expression=".estricted.*"
break="on-true"/>
            <condition field="caller_id_name" expression="PTSN.*"
break="on-true"/>
            <condition field="caller_id_name" expression=".oll.*"
break="on-true"/>
            <condition field="caller_id_name" expression=".--.*"
break="on-true"/>
            <condition field="caller_id_name" expression=",.*"
break="on-true"/>
            <condition field="caller_id_name" expression="000.*">
                <action application="set"
data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
                <action application="answer"/>
                <action application="sleep" data="1000"/>
                <action application="phrase" data="voicemail_record_name"/>
                <action application="playback"
data="tone_stream://%(500, 0, 640)"/>
                <action application="set"
data="playback_terminators=#*0123456789"/>
                <action application="record"
data="${call_screen_filename} 7 200 2"/>
                <action application="set" data="group_confirm_key=1"/>
                <action application="set"
data="fail_on_single_reject=true"/>
                <action application="set"
data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/>
                <action application="set" data="continue_on_fail=true"/>
<!--                <action application="bridge" data="user/$1"/> -->
                <action application="voicemail" data="default $${domain}
$1"/>
                <action application="hangup"/>
            </condition>
        </extension>
    </context>
</include>

Any help would be greatly appreciated.

Thank you,
Trever
-- 
"I know who I am. No one else knows who I am. If I was a giraffe, and
someone said I was a snake, I'd think, no, actually I'm a giraffe." --
Richard Gere


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From msc at freeswitch.org  Fri Oct 28 02:07:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Oct 2011 15:07:02 -0700
Subject: [Freeswitch-users] trouble making conditions for OR
In-Reply-To: <4EA9C2C1.1020809@gmail.com>
References: <4EA9C2C1.1020809@gmail.com>
Message-ID: <CAKzWOxV4JmzKuEBiQANWmh5MmfV3=rGz+LiZYxw-aoG6oD4iwg@mail.gmail.com>

Trever,

You're in one of those gray areas where xml_curl may actually be easier in
the long run unless you have just this one single set of goofy conditions to
handle. However, if I understand your question correctly you should be able
to do this in the dialplan.

I can't be exactly certain why your call is failing since the debug log
snippet you gave was a bit small, however my guess is that you don't
actually have any actions inside of a matching <condition> for the call that
was made. It looks like it was a "private" caller so this line would no
doubt match:

<condition field="caller_id_name" expression=".*rivate.*"
break="on-true"/>

When that condition matches, the dialplan processing will stop checking for
any more conditions inside of the extension named "screen" and simply move
on in the dialplan. (That behavior is caused by the break="on-true"
directive.) So, do you have any other extensions in this context? Whether
you do or not, it seems like none of your conditions that actually have
actions inside them is being matched, therefore you have no actions for this
call and get the no route error.

My recommendation is to pastebin the entire Incoming-FXO context as well as
debug level output for the incoming call. Paste the call log from start to
finish. That will help us see what's really going on.

-MC

On Thu, Oct 27, 2011 at 1:44 PM, Trever L. Adams <trever.adams at gmail.com>wrote:

> Hello everyone,
>
> The last few days I am beginning to explore freeswitch. I am having a
> bit of trouble. I am trying to user the examples near
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Complex_Condition.2FAction_Rules
> in the documentation. If I use '\' as it says, I get an error about
> </extension> being unexpected. If I use '/' it doesn't work. Below is my
> dialplan. It should trigger on empty number as well as the strings
> mentioned. I am pretty sure the regex is accurate. However, instead of
> running, I get:
>
>
> 2011-10-27 14:38:16.269266 [INFO] mod_dialplan_xml.c:336 Processing
> private <private>-> in context Incoming-FXO
> 2011-10-27 14:38:16.269266 [INFO] switch_core_state_machine.c:177 No
> Route, Aborting
>
> The dialplan:
>
> <include>
>    <context name="Incoming-FXO">
>        <extension name="screen">
>            <condition field="caller_id_number" expression="^$"
> break="on-true"/>
>            <condition field="caller_id_number" expression=".nonymous.*"
> break="on-true"/>
>            <condition field="caller_id_number" expression=".nknown.*"
> break="on-true"/>
>            <condition field="caller_id_number" expression=".*rivate.*"
> break="on-true"/>
>            <condition field="caller_id_number"
> expression=".estricted.*" break="on-true"/>
>            <condition field="caller_id_number" expression="PTSN.*"
> break="on-true"/>
>            <condition field="caller_id_number" expression=".oll.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression=".nonymous.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression=".nknown.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression=".*rivate.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression=".estricted.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression="PTSN.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression=".oll.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression=".--.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression=",.*"
> break="on-true"/>
>            <condition field="caller_id_name" expression="000.*">
>                <action application="set"
> data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
>                <action application="answer"/>
>                <action application="sleep" data="1000"/>
>                <action application="phrase" data="voicemail_record_name"/>
>                <action application="playback"
> data="tone_stream://%(500, 0, 640)"/>
>                <action application="set"
> data="playback_terminators=#*0123456789"/>
>                <action application="record"
> data="${call_screen_filename} 7 200 2"/>
>                <action application="set" data="group_confirm_key=1"/>
>                <action application="set"
> data="fail_on_single_reject=true"/>
>                <action application="set"
> data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/>
>                <action application="set" data="continue_on_fail=true"/>
> <!--                <action application="bridge" data="user/$1"/> -->
>                <action application="voicemail" data="default $${domain}
> $1"/>
>                <action application="hangup"/>
>            </condition>
>        </extension>
>    </context>
> </include>
>
> Any help would be greatly appreciated.
>
> Thank you,
> Trever
> --
> "I know who I am. No one else knows who I am. If I was a giraffe, and
> someone said I was a snake, I'd think, no, actually I'm a giraffe." --
> Richard Gere
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From xyangni at gmail.com  Fri Oct 28 02:17:08 2011
From: xyangni at gmail.com (Yihui Li)
Date: Fri, 28 Oct 2011 06:17:08 +0800
Subject: [Freeswitch-users] ./configure fails on vanilla checkout
In-Reply-To: <CALNZuwmBvrK0SF1RCAMVMJsOvv-V9-0StO4Fuz6+gNupJMYe8A@mail.gmail.com>
References: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>
	<1319731991828-6936975.post@n2.nabble.com>
	<CALNZuwmBvrK0SF1RCAMVMJsOvv-V9-0StO4Fuz6+gNupJMYe8A@mail.gmail.com>
Message-ID: <CAO1DH1_jAC387QQG4L0aoePw9T72SPTBZa05TDEbX2uOxzR5oA@mail.gmail.com>

I got the same problem on ubuntu 10.10 running on linode VPS

here is some of the outputs:

checking for library containing recv... none required
checking for getopt_long... yes
checking for getaddrinfo... yes
./configure: line 11073: syntax error near unexpected token `openssl,'
./configure: line 11073: `    PKG_CHECK_MODULES(openssl, openssl,'
configure: error: ./configure.gnu failed for libs/iksemel

however it does not occur on another newly installed ubuntu 10.04LTS.


On Fri, Oct 28, 2011 at 12:18 AM, Avi Marcus <avi at avimarcus.net> wrote:

> This sounds like something to be jira'd.
> -Avi
>
>
> On Thu, Oct 27, 2011 at 6:13 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>> You should provide somemore details here such as platform etc. I assume
>> this
>> does not happen for all platforms?
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/configure-fails-on-vanilla-checkout-tp6935612p6936975.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From stkn at freeswitch.org  Fri Oct 28 02:27:56 2011
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Fri, 28 Oct 2011 00:27:56 +0200
Subject: [Freeswitch-users] ./configure fails on vanilla checkout
In-Reply-To: <CAO1DH1_jAC387QQG4L0aoePw9T72SPTBZa05TDEbX2uOxzR5oA@mail.gmail.com>
References: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>
	<1319731991828-6936975.post@n2.nabble.com>
	<CALNZuwmBvrK0SF1RCAMVMJsOvv-V9-0StO4Fuz6+gNupJMYe8A@mail.gmail.com>
	<CAO1DH1_jAC387QQG4L0aoePw9T72SPTBZa05TDEbX2uOxzR5oA@mail.gmail.com>
Message-ID: <4EA9DAEC.1030507@freeswitch.org>

Please install pkg-config and ./bootstrap.sh again.

On 10/28/11 00:17, Yihui Li wrote:
> I got the same problem on ubuntu 10.10 running on linode VPS
> 
> here is some of the outputs:
> 
> checking for library containing recv... none required
> checking for getopt_long... yes
> checking for getaddrinfo... yes
> ./configure: line 11073: syntax error near unexpected token `openssl,'
> ./configure: line 11073: `    PKG_CHECK_MODULES(openssl, openssl,'
> configure: error: ./configure.gnu failed for libs/iksemel
> 
> however it does not occur on another newly installed ubuntu 10.04LTS.
> 
> 
> On Fri, Oct 28, 2011 at 12:18 AM, Avi Marcus <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
> 
>     This sounds like something to be jira'd.
>     -Avi
> 
> 
>     On Thu, Oct 27, 2011 at 6:13 PM, Jeff Lenk <jeff at jefflenk.com <mailto:jeff at jefflenk.com>> wrote:
> 
>         You should provide somemore details here such as platform etc. I assume this
>         does not happen for all platforms?
> 
>         --
>         View this message in context: http://freeswitch-users.2379917.n2.nabble.com/configure-fails-on-vanilla-checkout-tp6935612p6936975.html
>         Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> 
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> 
> 
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fcintrono at gmail.com  Fri Oct 28 03:15:55 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Thu, 27 Oct 2011 18:15:55 -0500
Subject: [Freeswitch-users] SPA921 only sends 10 digits to freeswtitch...
Message-ID: <CANtw_ZMLBq_8XxpXwh6BdLGUZdMiuij7aRPAnEHk5B6fv2dKGQ@mail.gmail.com>

Hello everybody from Guadalajara, Mexico.

I have 2 pots lines connected to a Cisco SPA8800 and with the help of fs
wiki site I was able to interface this device  to fs.

To make calls outside my office I created a file
in /usr/local/freeswitch/conf/dialplan/default

[root at pbx default]# pwd
/usr/local/freeswitch/conf/dialplan/default
[root at pbx default]# cat 00_spa8800_linea_1.xml
<include>
  <extension name="axtel1">
    <condition field="destination_number" expression="^1(\d{8,16})$">
      <action application="bridge" data="sofia/internal/$1 at 10.10.100.24:5061"
/>
    </condition>
  </extension>
</include>

As you see I have to dial "1" an then the phone number to be dialed.

With this dialplan entry I am able to make local calls from my linksys
SPA921 IP Phone without any problem.

Now I am trying to make some long distance calls. The phone I was trying to
call has 12 digits.

The problem is that when I dial this 12 digit phone number, log messages in
fs_cli console shows me that freeswtich only has 10 numbers to try to make
the long distance call.

If it helps when I dial this exact 12 digits phone in Xlite from my windows
rig, it works without any problem, I can make my long distance call.

This is Xlite log:
http://pastebin.freeswitch.org/17626

This is SPA921 log:
http://pastebin.freeswitch.org/17627


Thanks in advance for your kind help.
Have a nice day.
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From marhbere at hotmail.com  Fri Oct 28 03:23:13 2011
From: marhbere at hotmail.com (MarhBere)
Date: Thu, 27 Oct 2011 16:23:13 -0700 (PDT)
Subject: [Freeswitch-users] switch_rtp.c - Auto Changing Port
In-Reply-To: <09B4A6BE-F0D2-48E5-BC04-5104EF2EE8A3@freeswitch.org>
References: <77290C3C-B7E3-4FC7-892C-C2BEDBD0314F@ossiantelecom.co.uk>
	<09B4A6BE-F0D2-48E5-BC04-5104EF2EE8A3@freeswitch.org>
Message-ID: <1319757793079-6938401.post@n2.nabble.com>


I have same issue, with phone behind nat.

The RTP arrive to FS but don't have the auto change IP/Port, so Not media
can still  stablish. and FS have not sent any rtp packets.

In Asterisk after received 2 packets, it started to sending to rexternal
IP/port .

Is there any variable can setup for solve the problem?

Thanks,
Marhbere. 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/switch-rtp-c-Auto-Changing-Port-tp5503142p6938401.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter at onemetric.com  Fri Oct 28 02:28:13 2011
From: peter at onemetric.com (Peter Blackford)
Date: Fri, 28 Oct 2011 09:28:13 +1100
Subject: [Freeswitch-users] Docs Down?
Message-ID: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>

Is docs.freeswitch.org returning blank pages for everyone or is it just me?

Regards,
Peter
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From cliff at develix.com  Fri Oct 28 03:49:47 2011
From: cliff at develix.com (Cliff Wells)
Date: Thu, 27 Oct 2011 16:49:47 -0700
Subject: [Freeswitch-users] Docs Down?
In-Reply-To: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
References: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
Message-ID: <1319759387.2774.23.camel@portable-evil>

On Fri, 2011-10-28 at 09:28 +1100, Peter Blackford wrote:
> Is docs.freeswitch.org returning blank pages for everyone or is it
> just me?

Blank for me as well.

Cliff




From darknesslabs at gmail.com  Fri Oct 28 03:56:53 2011
From: darknesslabs at gmail.com (Karol)
Date: Thu, 27 Oct 2011 19:56:53 -0400
Subject: [Freeswitch-users] Docs Down?
In-Reply-To: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
References: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
Message-ID: <CAFZs+v8JhuvBhrNbSez42UWDBzVTpL+_oYeC2JDjLNiAYv3FZA@mail.gmail.com>

Same result.
On Oct 27, 2011 7:27 PM, "Peter Blackford" <peter at onemetric.com> wrote:

> Is docs.freeswitch.org returning blank pages for everyone or is it just
> me?
>
> Regards,
> Peter
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-list at puzzled.xs4all.nl  Fri Oct 28 04:04:37 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 28 Oct 2011 02:04:37 +0200
Subject: [Freeswitch-users] Docs Down?
In-Reply-To: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
References: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
Message-ID: <4EA9F195.4030106@puzzled.xs4all.nl>

On 10/28/2011 12:28 AM, Peter Blackford wrote:
> Is docs.freeswitch.org <http://docs.freeswitch.org> returning blank
> pages for everyone or is it just me?

Blank for me.

Regards,
Patrick


From 12ukwn at gmail.com  Fri Oct 28 04:24:08 2011
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 28 Oct 2011 02:24:08 +0200
Subject: [Freeswitch-users] Docs Down?
In-Reply-To: <4EA9F195.4030106@puzzled.xs4all.nl>
References: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
	<4EA9F195.4030106@puzzled.xs4all.nl>
Message-ID: <20111028022408.4e752ebc@anubis.defcon1>

On Fri, 28 Oct 2011 02:04:37 +0200
Patrick Lists <freeswitch-list at puzzled.xs4all.nl> wrote:

> Blank for me.

me2

-- 



From infos at madovsky.org  Fri Oct 28 04:40:48 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 20:40:48 -0400
Subject: [Freeswitch-users] originate_disposition
Message-ID: <4EE702AEC83348E0B4448B036AB68EB5@e1705>

To MC,

in this thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075412.html

if I understand what you explain does it mena that we can't have the b-leg bridge status
while a-leg is answered by FS itself ?

maybe that's my prob.. I'm trying to use bind_dgit_action to redirect to an Ivr with the option
"call a number" so yes the a-leg is already answered....

So todayno way to get the b-leg bridge status if a-leg did not hang up ?

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, October 27, 2011 4:33 PM
  Subject: originate_disposition


  Afer hours of test I didn' succeed how
  to catch the result of originate_disposition

  --------

  I have this extensions in my dialplan

           <extension name="internal_test">
                  <condition field="${sip_to_uri}" expression="^(999\d{10})@$${domain}$">
                          <action inline="true" application="set" data="nibble_account="/>
                          <action inline="true" application="set" data="nibble_rate="/>
                          <action inline="true" application="set" data="dialed_extension=$1"/>
                  </condition>
                  <condition field="${from_admin_services}" expression="^true$" break="never">
                          <action application="set" data="ignore_early_media=true"/>
                          <action application="set" data="hangup_after_bridge=false"/>
                          <action application="set" data="continue_on_fail=true"/>
                          <action application="bridge" data="{sip_append_audio_sdp='a=rtpmap:0 PCMU/8000',sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
                  </condition>
          </extension>
          <extension name="internal_test_tools">
                  <condition field="${originate_disposition}" expression="^SUBSCRIBER_ABSENT$" break="on-true">
                          <action application="phrase" data="bad_ob_prepaid_dest"/>
                          <action application="transfer" data="ob_prepaid_call XML features"/>
                          <action application="info"/>
                  </condition>
                  <condition field="${cond(expr(${originate_disposition} != NORMAL_CLEARING) ? YES : NO)}" expression="^YES$"/>
                  <condition field="${cond(expr(${originate_disposition} != SUBSCRIBER_ABSENT) ? YES : NO)}" expression="^YES$">
                          <action application="ring_ready"/>
                          <action application="sleep" data="2000"/>
                          <action application="answer"/>
                          <action application="sleep" data="1400"/>
                          <action application="voicemail" data="default $${domain} ${dialed_extension}"/>
                 </condition>
          </extension>


  -----------


  I tried to merge it in one extension without success also.
  where is the trick ? :)

  Thanks

  Franck
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From infos at madovsky.org  Fri Oct 28 04:47:19 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 27 Oct 2011 20:47:19 -0400
Subject: [Freeswitch-users] originate_disposition
Message-ID: <5F0F62EA80E446199D8EEC5BBA156348@e1705>

ha there's a nice explanation of Anthony later...

http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075433.html

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, October 27, 2011 8:40 PM
  Subject: Re: originate_disposition


  To MC,

  in this thread
  http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075412.html

  if I understand what you explain does it mena that we can't have the b-leg bridge status
  while a-leg is answered by FS itself ?

  maybe that's my prob.. I'm trying to use bind_dgit_action to redirect to an Ivr with the option
  "call a number" so yes the a-leg is already answered....

  So todayno way to get the b-leg bridge status if a-leg did not hang up ?

  Thanks
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, October 27, 2011 4:33 PM
    Subject: originate_disposition


    Afer hours of test I didn' succeed how
    to catch the result of originate_disposition

    --------

    I have this extensions in my dialplan

             <extension name="internal_test">
                    <condition field="${sip_to_uri}" expression="^(999\d{10})@$${domain}$">
                            <action inline="true" application="set" data="nibble_account="/>
                            <action inline="true" application="set" data="nibble_rate="/>
                            <action inline="true" application="set" data="dialed_extension=$1"/>
                    </condition>
                    <condition field="${from_admin_services}" expression="^true$" break="never">
                            <action application="set" data="ignore_early_media=true"/>
                            <action application="set" data="hangup_after_bridge=false"/>
                            <action application="set" data="continue_on_fail=true"/>
                            <action application="bridge" data="{sip_append_audio_sdp='a=rtpmap:0 PCMU/8000',sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
                    </condition>
            </extension>
            <extension name="internal_test_tools">
                    <condition field="${originate_disposition}" expression="^SUBSCRIBER_ABSENT$" break="on-true">
                            <action application="phrase" data="bad_ob_prepaid_dest"/>
                            <action application="transfer" data="ob_prepaid_call XML features"/>
                            <action application="info"/>
                    </condition>
                    <condition field="${cond(expr(${originate_disposition} != NORMAL_CLEARING) ? YES : NO)}" expression="^YES$"/>
                    <condition field="${cond(expr(${originate_disposition} != SUBSCRIBER_ABSENT) ? YES : NO)}" expression="^YES$">
                            <action application="ring_ready"/>
                            <action application="sleep" data="2000"/>
                            <action application="answer"/>
                            <action application="sleep" data="1400"/>
                            <action application="voicemail" data="default $${domain} ${dialed_extension}"/>
                   </condition>
            </extension>


    -----------


    I tried to merge it in one extension without success also.
    where is the trick ? :)

    Thanks

    Franck
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From trever.adams at gmail.com  Fri Oct 28 06:14:05 2011
From: trever.adams at gmail.com (Trever L. Adams)
Date: Thu, 27 Oct 2011 20:14:05 -0600
Subject: [Freeswitch-users] trouble making conditions for OR
In-Reply-To: <CAKzWOxV4JmzKuEBiQANWmh5MmfV3=rGz+LiZYxw-aoG6oD4iwg@mail.gmail.com>
References: <4EA9C2C1.1020809@gmail.com>
	<CAKzWOxV4JmzKuEBiQANWmh5MmfV3=rGz+LiZYxw-aoG6oD4iwg@mail.gmail.com>
Message-ID: <4EAA0FED.9000905@gmail.com>

On 10/27/2011 04:07 PM, Michael Collins wrote:
> Trever,
>
> You're in one of those gray areas where xml_curl may actually be
> easier in the long run unless you have just this one single set of
> goofy conditions to handle. However, if I understand your question
> correctly you should be able to do this in the dialplan.
>
> I can't be exactly certain why your call is failing since the debug
> log snippet you gave was a bit small, however my guess is that you
> don't actually have any actions inside of a matching <condition> for
> the call that was made. It looks like it was a "private" caller so
> this line would no doubt match:
>
> <condition field="caller_id_name" expression=".*rivate.*"
> break="on-true"/>
>
> When that condition matches, the dialplan processing will stop
> checking for any more conditions inside of the extension named
> "screen" and simply move on in the dialplan. (That behavior is caused
> by the break="on-true" directive.) So, do you have any other
> extensions in this context? Whether you do or not, it seems like none
> of your conditions that actually have actions inside them is being
> matched, therefore you have no actions for this call and get the no
> route error.
>
> My recommendation is to pastebin the entire Incoming-FXO context as
> well as debug level output for the incoming call. Paste the call log
> from start to finish. That will help us see what's really going on.
>
> -MC
>
MC,

Thank you, I will have to look into xml_curl. Apparently I misunderstood
the documentation. If ANY of these things match, the actions to screen
the call should be taken. Am I messing this up?

At the moment I have no other processing of incoming calls. Still
working on things.

> On Thu, Oct 27, 2011 at 1:44 PM, Trever L. Adams
> <trever.adams at gmail.com <mailto:trever.adams at gmail.com>> wrote:
>
>     The dialplan:
>
>     <include>
>        <context name="Incoming-FXO">
>            <extension name="screen">
>                <condition field="caller_id_number" expression="^$"
>     break="on-true"/>
>                <condition field="caller_id_number"
>     expression=".nonymous.*"
>     break="on-true"/>
>                <condition field="caller_id_number" expression=".nknown.*"
>     break="on-true"/>
>                <condition field="caller_id_number" expression=".*rivate.*"
>     break="on-true"/>
>                <condition field="caller_id_number"
>     expression=".estricted.*" break="on-true"/>
>                <condition field="caller_id_number" expression="PTSN.*"
>     break="on-true"/>
>                <condition field="caller_id_number" expression=".oll.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression=".nonymous.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression=".nknown.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression=".*rivate.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression=".estricted.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression="PTSN.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression=".oll.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression=".--.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression=",.*"
>     break="on-true"/>
>                <condition field="caller_id_name" expression="000.*">
>                    <action application="set"
>     data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
>                    <action application="answer"/>
>                    <action application="sleep" data="1000"/>
>                    <action application="phrase"
>     data="voicemail_record_name"/>
>                    <action application="playback"
>     data="tone_stream://%(500, 0, 640)"/>
>                    <action application="set"
>     data="playback_terminators=#*0123456789"/>
>                    <action application="record"
>     data="${call_screen_filename} 7 200 2"/>
>                    <action application="set" data="group_confirm_key=1"/>
>                    <action application="set"
>     data="fail_on_single_reject=true"/>
>                    <action application="set"
>     data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/>
>                    <action application="set"
>     data="continue_on_fail=true"/>
>     <!--                <action application="bridge" data="user/$1"/> -->
>                    <action application="voicemail" data="default
>     $${domain}
>     $1"/>
>                    <action application="hangup"/>
>                </condition>
>            </extension>
>        </context>
>     </include>
>

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From trever.adams at gmail.com  Fri Oct 28 06:16:44 2011
From: trever.adams at gmail.com (Trever L. Adams)
Date: Thu, 27 Oct 2011 20:16:44 -0600
Subject: [Freeswitch-users] FreeTDM and internal extensions and caller_id
	(user directory?)
Message-ID: <4EAA108C.1030907@gmail.com>

Hello all,

I have a small set of internal extensions working. I have an 8 port
DAHDI card. 6 FXS, 2 FXO. I have it so I can call between FXS ports
(extensions 1001-1005 right now are configured).

This works fine. The problem is that the caller ID information is wrong
(the default is FreeSwitch/0000000000). I need to be able to set this on
a per port basis. I see that conf/directory is the place to set this up
for sip. How/where do I do this for FreeTDM? I have not been able to
find any documentation for this. I am afraid I have spent a few hours
looking. If there is anything out there, I must be using the wrong
search terms.

Thanks for any help,
Trever

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From boris at tagnet.ru  Fri Oct 28 06:39:31 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 28 Oct 2011 08:39:31 +0600
Subject: [Freeswitch-users] SPA921 only sends 10 digits to freeswtitch...
In-Reply-To: <CANtw_ZMLBq_8XxpXwh6BdLGUZdMiuij7aRPAnEHk5B6fv2dKGQ@mail.gmail.com>
References: <CANtw_ZMLBq_8XxpXwh6BdLGUZdMiuij7aRPAnEHk5B6fv2dKGQ@mail.gmail.com>
Message-ID: <4EAA15E3.8010008@tagnet.ru>

Hello!

     Have not worked with SPA921 but SPA8000 and PAP-2T. Last two 
devices have "dialplan" feature to speedup digits processing. May be 
SPA921 has it too and You have misconfigured it?

> Hello everybody from Guadalajara, Mexico.
>
> I have 2 pots lines connected to a Cisco SPA8800 and with the help of 
> fs wiki site I was able to interface this device  to fs.
>
> To make calls outside my office I created a file 
> in /usr/local/freeswitch/conf/dialplan/default
>
> [root at pbx default]# pwd
> /usr/local/freeswitch/conf/dialplan/default
> [root at pbx default]# cat 00_spa8800_linea_1.xml
> <include>
> <extension name="axtel1">
> <condition field="destination_number" expression="^1(\d{8,16})$">
> <action application="bridge" data="sofia/internal/$1 at 10.10.100.24:5061 
> <http://1 at 10.10.100.24:5061>" />
> </condition>
> </extension>
> </include>
>
> As you see I have to dial "1" an then the phone number to be dialed.
>
> With this dialplan entry I am able to make local calls from my linksys 
> SPA921 IP Phone without any problem.
>
> Now I am trying to make some long distance calls. The phone I was 
> trying to call has 12 digits.
>
> The problem is that when I dial this 12 digit phone number, log 
> messages in fs_cli console shows me that freeswtich only has 10 
> numbers to try to make the long distance call.
>
> If it helps when I dial this exact 12 digits phone in Xlite from my 
> windows rig, it works without any problem, I can make my long distance 
> call.
>
> This is Xlite log:
> http://pastebin.freeswitch.org/17626
>
> This is SPA921 log:
> http://pastebin.freeswitch.org/17627
>
>
> Thanks in advance for your kind help.
> Have a nice day.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Boris


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From fcintrono at gmail.com  Fri Oct 28 07:43:32 2011
From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=)
Date: Thu, 27 Oct 2011 22:43:32 -0500
Subject: [Freeswitch-users] SPA921 only sends 10 digits to freeswtitch...
In-Reply-To: <4EAA15E3.8010008@tagnet.ru>
References: <CANtw_ZMLBq_8XxpXwh6BdLGUZdMiuij7aRPAnEHk5B6fv2dKGQ@mail.gmail.com>
	<4EAA15E3.8010008@tagnet.ru>
Message-ID: <CANtw_ZM4_YhHCqjvbc-z6nHPu1rDed_49oDFYe21C1+VreX-Mw@mail.gmail.com>

Thanks very much sir, I just disable this SPA921 dialplan feature.

Problem resolved. :)

Have a nice day.

On Thu, Oct 27, 2011 at 9:39 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     Have not worked with SPA921 but SPA8000 and PAP-2T. Last two devices
> have "dialplan" feature to speedup digits processing. May be SPA921 has it
> too and You have misconfigured it?
>
>  Hello everybody from Guadalajara, Mexico.
>
>  I have 2 pots lines connected to a Cisco SPA8800 and with the help of fs
> wiki site I was able to interface this device  to fs.
>
>  To make calls outside my office I created a file
> in /usr/local/freeswitch/conf/dialplan/default
>
>   [root at pbx default]# pwd
> /usr/local/freeswitch/conf/dialplan/default
> [root at pbx default]# cat 00_spa8800_linea_1.xml
> <include>
>   <extension name="axtel1">
>     <condition field="destination_number" expression="^1(\d{8,16})$">
>       <action application="bridge" data="sofia/internal/$
> 1 at 10.10.100.24:5061" />
>     </condition>
>   </extension>
> </include>
>
>  As you see I have to dial "1" an then the phone number to be dialed.
>
>  With this dialplan entry I am able to make local calls from my linksys
> SPA921 IP Phone without any problem.
>
>  Now I am trying to make some long distance calls. The phone I was trying
> to call has 12 digits.
>
>  The problem is that when I dial this 12 digit phone number, log messages
> in fs_cli console shows me that freeswtich only has 10 numbers to try to
> make the long distance call.
>
>  If it helps when I dial this exact 12 digits phone in Xlite from my
> windows rig, it works without any problem, I can make my long distance
> call.
>
>  This is Xlite log:
> http://pastebin.freeswitch.org/17626
>
>  This is SPA921 log:
> http://pastebin.freeswitch.org/17627
>
>
>  Thanks in advance for your kind help.
> Have a nice day.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> 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
>
>
>
> --
> Regards,
> Boris
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Fri Oct 28 08:20:20 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 28 Oct 2011 00:20:20 -0400
Subject: [Freeswitch-users] FreeTDM and internal extensions and
	caller_id (user directory?)
In-Reply-To: <4EAA108C.1030907@gmail.com>
References: <4EAA108C.1030907@gmail.com>
Message-ID: <3008.1319775620@ccs.covici.com>

What worked for me is in the [span zt <name>] you can set  number =>
 <caller id number> and name => <caller id name> .

Perhaps an ata would be better -- I have found freetdm to not be the
most stable of modules I have seen.

Trever L. Adams <trever.adams at gmail.com> wrote:

> Hello all,
> 
> I have a small set of internal extensions working. I have an 8 port
> DAHDI card. 6 FXS, 2 FXO. I have it so I can call between FXS ports
> (extensions 1001-1005 right now are configured).
> 
> This works fine. The problem is that the caller ID information is wrong
> (the default is FreeSwitch/0000000000). I need to be able to set this on
> a per port basis. I see that conf/directory is the place to set this up
> for sip. How/where do I do this for FreeTDM? I have not been able to
> find any documentation for this. I am afraid I have spent a few hours
> looking. If there is anything out there, I must be using the wrong
> search terms.
> 
> Thanks for any help,
> Trever
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 infos at madovsky.org  Fri Oct 28 09:01:48 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Oct 2011 01:01:48 -0400
Subject: [Freeswitch-users] High load on database server
Message-ID: <1C74EB6C3F0E44C4912454C3590B3E03@e1705>

Re: [Freeswitch-users] High load on database serveranother idea, maybe you have some hardware problem....

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Thursday, October 27, 2011 2:07 PM
  Subject: Re: [Freeswitch-users] High load on database server


  Thanks Ken. here the link of official SIPP website
  http://sipp.sourceforge.net/

  I didn't know it was a HP app ! :0)
    ----- Original Message ----- 
    From: Ken Rice 
    To: FreeSWITCH Users Help 
    Sent: Thursday, October 27, 2011 1:54 PM
    Subject: Re: [Freeswitch-users] High load on database server


    SIPP works fine for this... However keep in mind some of their default scenario files arent exactly the best in the world and can in many situations leave a bit to be desired... Check the wiki I think there is more info on this on there

    K


    On 10/27/11 12:51 PM, "Madovsky" <infos at madovsky.org> wrote:


      on this subject,
      is SIPp can be used to test to reproduce hundreds calls ?


        ----- Original Message ----- 
         
        From:  Anthony Minessale <mailto:anthony.minessale at gmail.com>  
         
        To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>   
         
        Sent: Thursday, October 27, 2011 1:39  PM
         
        Subject: Re: [Freeswitch-users] High load  on database server
         

        BTW I know it's a contradiction to tell you to consider older  ODBC and newer FS but I wrote FS so I can attest to its stability especially  the ODBC code in the core. 

         

         


         
        On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <anthony.minessale at gmail.com>  wrote:
         


          Let's recap
           

           
          You have 4 moving parts, you have various versions of all 4 to choose  from and even different implementations of 3 of the 4.
           

          ODBC LIB 
          ODBC DRIVER
           
          DATABASE
           
          FS
           

           
          You must choose a stable combination of all 4 which may require  specific configuration of each component as well.
           

           
          My only issue is FUD, I do not want people to advertise that FS does  not work on ODBC under load when the whole story is FS does not work under  load when you use fooODBC with barDRIVER with basBASE database server  configured a particular way.
           

           
          It should never crash, if it does, its a bug in something.  I can  speak for FS that there is no bug there.  The other 3 are  variables. 
           
          Remember this next time there is a discussion about using system libs  and why we build our own depends.  The one thing we use system libs  for, ODBC, is a huge PITA.........
           

           
          Things to remember:
           

           
          1) try the Threading=0 in the odbcinst.ini, this is a serious problem  and has been made the default in latest versions.
           
          2) If you must use Mysql, enable transactions and make sure your  odbc.ini is referencing the threadsafe version of the lib 
           
               libmyodbc3_r.so <-- note _r
           
          3) Try various combos of drivers and odbc libs, many time newer is not  better, stable versions lie in the past.
           
          4) build the drivers and odbc yourself or get the debug symbols so you  can get a backtrace, you could be finding a bug for them.......
           
          5) Find a test to reproduce your problem so you can try different  database engines and driver combos.
           
          6) Make sure you are on latest FS git so you know you have a stable  copy.
           

           

           
          Anyway, this is a pain, that's why ppl pay you to do it.
           
          Discuss this all you want here, just minimize any FUD to scare away  people who want to use it.
           

           

           

           

           
           
           
           


           
          On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:
           

            Do not agree. Issues happening under extreme cases like  high load with
            some specific conditions are hard to reproduce, sharing  the
            information in this case is a lot more efficient, for all.
             

            Sent from my mobile device

             
             
             
            On Oct 27, 2011, at 18:56, Robert Huddleston <rhuddleston at gmail.com> wrote:

            > My bologna  has a first name - it's O S C A R...
            >
            > This topic is getting  really old... I have to agree with Antm - get out of
            > the lazy  pants and do some research / hard work.
            >
            >
            >  -----Original Message-----
            > From: freeswitch-users-bounces at lists.freeswitch.org
            >  [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf  Of Sergey
            > Okhapkin
            > Sent: Thursday, October 27, 2011 12:43  PM
            > To: FreeSWITCH Users Help
            > Subject: Re:  [Freeswitch-users] High load on database server
            >
            > What do  you mean "shared ODBC"?
            >
            > On Thursday 27 October 2011,  Madovsky wrote:
            >> but, is anyone experienced that with shared  ODBC ? (managing more than 330
            >> concurrent  calls)
            >>
            >> ----- Original Message -----
            >>  From: "Madovsky" <infos at madovsky.org>
            >> To: "FreeSWITCH Users  Help" <freeswitch-users at lists.freeswitch.org>
            >>  Sent: Thursday, October 27, 2011 12:33 PM
            >> Subject: Re:  [Freeswitch-users] High load on database  server
            >>
            >>> ha ok, good luck  so
            >>>
            >>> ----- Original Message  -----
            >>> From: "Cliff Wells" <cliff at develix.com>
            >>> To: "FreeSWITCH  Users Help" <freeswitch-users at lists.freeswitch.org>
            >>>  Sent: Thursday, October 27, 2011 12:26 PM
            >>> Subject: Re:  [Freeswitch-users] High load on database  server
            >>>
            >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
            >>>>> I means that depend the  quality of your script ;)
            >>>>
            >>>> Quality  isn't the issue here.  The script is too simple to be  incorrect
            >>>> (and it carefully releases the odbc  connection back to the pool in a
            >>>> hangup handler).   I have written poor-quality code plenty of times,  but
            >>>> 25 years as a programmer usually allows me the  luxury of knowing when
            >>>> I'm doing it,  thanks.
            >>>>
            >>>> In any case, Lua scripts  work fine.  Google will tell you that unixODBC
            >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies there,  but
            >>>> again, I emphasize, everything WORKS GREAT (been  using it for a couple
            >>>> of years now) until you get to  very high concurrency (about 330
            >>>> concurrent calls on a  single system).  If you don't expect to handle
            >>>>  more than 300 concurrent calls, then you do not need to worry about  it.
            >>>> At all.
            >>>>
            >>>>  Regards,
            >>>>  Cliff
            >>>>
            >>>>
            >>>>
            >>>>  FreeSWITCH-users mailing list
            >>>> FreeSWITCH-users at lists.freeswitch.org
            >>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            >>>>
            >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            >>>>  http://www.freeswitch.org
            >>
            >>  FreeSWITCH-users mailing list
            >> FreeSWITCH-users at lists.freeswitch.org
            >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            >>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            >>  http://www.freeswitch.org
            >
            >
            >
            >  FreeSWITCH-users mailing list
            > FreeSWITCH-users at lists.freeswitch.org
            > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            >  http://www.freeswitch.org
            >
            >
            >
            >  FreeSWITCH-users mailing list
            > FreeSWITCH-users at lists.freeswitch.org
            > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            >  http://www.freeswitch.org


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



           

           



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From msc at freeswitch.org  Fri Oct 28 10:16:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Oct 2011 23:16:07 -0700
Subject: [Freeswitch-users] trouble making conditions for OR
In-Reply-To: <4EAA0FED.9000905@gmail.com>
References: <4EA9C2C1.1020809@gmail.com>
	<CAKzWOxV4JmzKuEBiQANWmh5MmfV3=rGz+LiZYxw-aoG6oD4iwg@mail.gmail.com>
	<4EAA0FED.9000905@gmail.com>
Message-ID: <CAKzWOxWLeP1ofJbQkdgmezk-4X8UWjObVpPuWyKRjhF88roogQ@mail.gmail.com>

On Thu, Oct 27, 2011 at 7:14 PM, Trever L. Adams <trever.adams at gmail.com>wrote:

>  On 10/27/2011 04:07 PM, Michael Collins wrote:
>
> Trever,
>
>  You're in one of those gray areas where xml_curl may actually be easier
> in the long run unless you have just this one single set of goofy conditions
> to handle. However, if I understand your question correctly you should be
> able to do this in the dialplan.
>
>  I can't be exactly certain why your call is failing since the debug log
> snippet you gave was a bit small, however my guess is that you don't
> actually have any actions inside of a matching <condition> for the call that
> was made. It looks like it was a "private" caller so this line would no
> doubt match:
>
>  <condition field="caller_id_name" expression=".*rivate.*"
> break="on-true"/>
>
>  When that condition matches, the dialplan processing will stop checking
> for any more conditions inside of the extension named "screen" and simply
> move on in the dialplan. (That behavior is caused by the break="on-true"
> directive.) So, do you have any other extensions in this context? Whether
> you do or not, it seems like none of your conditions that actually have
> actions inside them is being matched, therefore you have no actions for this
> call and get the no route error.
>
>  My recommendation is to pastebin the entire Incoming-FXO context as well
> as debug level output for the incoming call. Paste the call log from start
> to finish. That will help us see what's really going on.
>
>  -MC
>
>  MC,
>
> Thank you, I will have to look into xml_curl. Apparently I misunderstood
> the documentation. If ANY of these things match, the actions to screen the
> call should be taken. Am I messing this up?
>

To be frank, yes, you are messing it up. :)

The "proper" way to do an OR is by using the | (pipe) char in the regex.
However, you are checking two different fields and you want the same actions
if either field matches the values you have supplied. In this case I'd do
something like the following:

<extension name="screen">
  <condition field="caller_id_name" expression="^$|
nonymous|nknown|rivate|PSTN|oll|restricted|.--.*|,.*|000.*">
    <action application="set" data="private_caller=true" inline="true"/>
  </condition>

  <condition field="caller_id_number"
expression="^$|nonymous|nknown|rivate|PSTN|oll">
    <action application="set" data="private_caller=true" inline="true"/>
  </condition>

  <condition field="${private_caller}" expression="^true$">
    <action application="set"
data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
    <action application="answer"/>
    <action application="sleep" data="1000"/>
    <action application="phrase" data="voicemail_record_name"/>
    <action application="playback" data="tone_stream://%(500, 0, 640)"/>
    <action application="set" data="playback_terminators=#*0123456789"/>
    <action application="record" data="${call_screen_filename} 7 200 2"/>
    <action application="set" data="group_confirm_key=1"/>
    <action application="set" data="fail_on_single_reject=true"/>
    <action application="set"
data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/>
    <action application="set" data="continue_on_fail=true"/>
    <action application="bridge" data="user/${dialed_number}"/>
    <action application="voicemail" data="default
$${domain} ${dialed_number}"/>
    <action application="hangup"/>
  </condition>
</extension>

NOTE: I took some liberties with your regex patterns. You may need to tweak
the expressions depending on exactly what comes down the pipe.

Bonus question: why did I replace $1 with ${dialed_number}? :)  Be sure that
you route the call properly. I'm making an assumption about what the dialed
number is, but you won't have that luxury - you will need to make sure that
you have a user ID that is the same as the dialed number or you will need to
parse the dialed number and relate it somehow to one of your users.

-MC


> At the moment I have no other processing of incoming calls. Still working
> on things.
>
>  On Thu, Oct 27, 2011 at 1:44 PM, Trever L. Adams <trever.adams at gmail.com>wrote:
>
>> The dialplan:
>>
>> <include>
>>    <context name="Incoming-FXO">
>>        <extension name="screen">
>>            <condition field="caller_id_number" expression="^$"
>> break="on-true"/>
>>            <condition field="caller_id_number" expression=".nonymous.*"
>> break="on-true"/>
>>            <condition field="caller_id_number" expression=".nknown.*"
>> break="on-true"/>
>>            <condition field="caller_id_number" expression=".*rivate.*"
>> break="on-true"/>
>>            <condition field="caller_id_number"
>> expression=".estricted.*" break="on-true"/>
>>            <condition field="caller_id_number" expression="PTSN.*"
>> break="on-true"/>
>>            <condition field="caller_id_number" expression=".oll.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression=".nonymous.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression=".nknown.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression=".*rivate.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression=".estricted.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression="PTSN.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression=".oll.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression=".--.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression=",.*"
>> break="on-true"/>
>>            <condition field="caller_id_name" expression="000.*">
>>                <action application="set"
>> data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
>>                <action application="answer"/>
>>                <action application="sleep" data="1000"/>
>>                <action application="phrase" data="voicemail_record_name"/>
>>                <action application="playback"
>> data="tone_stream://%(500, 0, 640)"/>
>>                <action application="set"
>> data="playback_terminators=#*0123456789"/>
>>                <action application="record"
>> data="${call_screen_filename} 7 200 2"/>
>>                <action application="set" data="group_confirm_key=1"/>
>>                <action application="set"
>> data="fail_on_single_reject=true"/>
>>                <action application="set"
>> data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/>
>>                <action application="set" data="continue_on_fail=true"/>
>> <!--                <action application="bridge" data="user/$1"/> -->
>>                <action application="voicemail" data="default $${domain}
>> $1"/>
>>                <action application="hangup"/>
>>            </condition>
>>        </extension>
>>    </context>
>> </include>
>>
>
>
>
> FreeSWITCH-users mailing 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 Jacob_Hodges at data3.com.au  Fri Oct 28 05:31:28 2011
From: Jacob_Hodges at data3.com.au (Jacob Hodges)
Date: Fri, 28 Oct 2011 12:31:28 +1100
Subject: [Freeswitch-users] FreeSwitch as MOH for Lync
Message-ID: <3842AA5160166449B2C35289ECEFEBC803C58BC8@D3COLEXM01.data3.com.au>

I am looking at using FreeSwitch to provide Music On Hold for Lync. Just wondering if anyone has done this before or could point me in the right direction.

Jacob.

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From msc at freeswitch.org  Fri Oct 28 10:24:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Oct 2011 23:24:15 -0700
Subject: [Freeswitch-users] originate_disposition
In-Reply-To: <D16AF8797FA347B3A2E1923AAE592321@e1705>
References: <D16AF8797FA347B3A2E1923AAE592321@e1705>
Message-ID: <CAKzWOxXXks-WQ42VR12=a2wcsBaMCAp0ox2yW-dbrKc=qre0Rg@mail.gmail.com>

How about the trick that Jay Binks added for scenarios not unlike this one?
http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail

-MC

On Thu, Oct 27, 2011 at 1:33 PM, Madovsky <infos at madovsky.org> wrote:

> **
> Afer hours of test I didn' succeed how
> to catch the result of originate_disposition
>
> --------
>
> I have this extensions in my dialplan
>
>          <extension name="internal_test">
>                 <condition field="${sip_to_uri}"
> expression="^(999\d{10})@$${domain}$">
>                         <action inline="true" application="set"
> data="nibble_account="/>
>                         <action inline="true" application="set"
> data="nibble_rate="/>
>                         <action inline="true" application="set"
> data="dialed_extension=$1"/>
>                 </condition>
>                 <condition field="${from_admin_services}"
> expression="^true$" break="never">
>                         <action application="set"
> data="ignore_early_media=true"/>
>                         <action application="set"
> data="hangup_after_bridge=false"/>
>                         <action application="set"
> data="continue_on_fail=true"/>
>                         <action application="bridge"
> data="{sip_append_audio_sdp='a=rtpmap:0
> PCMU/8000',sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
>                 </condition>
>         </extension>
>         <extension name="internal_test_tools">
>                 <condition field="${originate_disposition}"
> expression="^SUBSCRIBER_ABSENT$" break="on-true">
>                         <action application="phrase"
> data="bad_ob_prepaid_dest"/>
>                         <action application="transfer"
> data="ob_prepaid_call XML features"/>
>                         <action application="info"/>
>                 </condition>
>                 <condition field="${cond(expr(${originate_disposition} !=
> NORMAL_CLEARING) ? YES : NO)}" expression="^YES$"/>
>                 <condition field="${cond(expr(${originate_disposition} !=
> SUBSCRIBER_ABSENT) ? YES : NO)}" expression="^YES$">
>                         <action application="ring_ready"/>
>                         <action application="sleep" data="2000"/>
>                         <action application="answer"/>
>                         <action application="sleep" data="1400"/>
>                         <action application="voicemail" data="default
> $${domain} ${dialed_extension}"/>
>                </condition>
>         </extension>
>
> -----------
>
>
> I tried to merge it in one extension without success also.
> where is the trick ? :)
>
> Thanks
>
> Franck
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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From andrewkt at aktzero.com  Fri Oct 28 10:28:02 2011
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Fri, 28 Oct 2011 02:28:02 -0400
Subject: [Freeswitch-users] FreeSwitch as MOH for Lync
In-Reply-To: <3842AA5160166449B2C35289ECEFEBC803C58BC8@D3COLEXM01.data3.com.au>
Message-ID: <mailman.1.1319783290.29746.freeswitch-users@lists.freeswitch.org>

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From infos at madovsky.org  Fri Oct 28 10:52:51 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 28 Oct 2011 02:52:51 -0400
Subject: [Freeswitch-users] originate_disposition
References: <D16AF8797FA347B3A2E1923AAE592321@e1705>
	<CAKzWOxXXks-WQ42VR12=a2wcsBaMCAp0ox2yW-dbrKc=qre0Rg@mail.gmail.com>
Message-ID: <C7E1545D96BE42FBB1F8FB344A39857B@e1705>

Great shot MC, thanks a lot.

is this variable can contain more than one cause ?

Thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, October 28, 2011 2:24 AM
  Subject: Re: [Freeswitch-users] originate_disposition


  How about the trick that Jay Binks added for scenarios not unlike this one?
  http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail


  -MC


  On Thu, Oct 27, 2011 at 1:33 PM, Madovsky <infos at madovsky.org> wrote:

    Afer hours of test I didn' succeed how
    to catch the result of originate_disposition

    --------

    I have this extensions in my dialplan

             <extension name="internal_test">
                    <condition field="${sip_to_uri}" expression="^(999\d{10})@$${domain}$">
                            <action inline="true" application="set" data="nibble_account="/>
                            <action inline="true" application="set" data="nibble_rate="/>
                            <action inline="true" application="set" data="dialed_extension=$1"/>
                    </condition>
                    <condition field="${from_admin_services}" expression="^true$" break="never">
                            <action application="set" data="ignore_early_media=true"/>
                            <action application="set" data="hangup_after_bridge=false"/>
                            <action application="set" data="continue_on_fail=true"/>
                            <action application="bridge" data="{sip_append_audio_sdp='a=rtpmap:0 PCMU/8000',sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
                    </condition>
            </extension>
            <extension name="internal_test_tools">
                    <condition field="${originate_disposition}" expression="^SUBSCRIBER_ABSENT$" break="on-true">
                            <action application="phrase" data="bad_ob_prepaid_dest"/>
                            <action application="transfer" data="ob_prepaid_call XML features"/>
                            <action application="info"/>
                    </condition>
                    <condition field="${cond(expr(${originate_disposition} != NORMAL_CLEARING) ? YES : NO)}" expression="^YES$"/>
                    <condition field="${cond(expr(${originate_disposition} != SUBSCRIBER_ABSENT) ? YES : NO)}" expression="^YES$">
                            <action application="ring_ready"/>
                            <action application="sleep" data="2000"/>
                            <action application="answer"/>
                            <action application="sleep" data="1400"/>
                            <action application="voicemail" data="default $${domain} ${dialed_extension}"/>
                   </condition>
            </extension>


    -----------


    I tried to merge it in one extension without success also.
    where is the trick ? :)

    Thanks

    Franck



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From boris at tagnet.ru  Fri Oct 28 11:07:17 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 28 Oct 2011 13:07:17 +0600
Subject: [Freeswitch-users] Strange echo problems
Message-ID: <4EAA54A5.1070607@tagnet.ru>

Hello!

     I have found a very strange echo problem with different devices. 
When then user starts speaking there is no echo, but after about 1 
minute of talking echo begins. If user stop talking echo disappears and 
again when user start talking there is echo again after about 1 minute 
talking. What is this? Codec is G711a, FreeSWITCH Version 1.0.head 
(git-abc5d7c 2011-10-18 20-48-34 -0400). Any help please.

-- 
Regards,
Boris




From pkelly at gmail.com  Fri Oct 28 11:13:50 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 28 Oct 2011 08:13:50 +0100
Subject: [Freeswitch-users] ./configure fails on vanilla checkout
In-Reply-To: <4EA9DAEC.1030507@freeswitch.org>
References: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>
	<1319731991828-6936975.post@n2.nabble.com>
	<CALNZuwmBvrK0SF1RCAMVMJsOvv-V9-0StO4Fuz6+gNupJMYe8A@mail.gmail.com>
	<CAO1DH1_jAC387QQG4L0aoePw9T72SPTBZa05TDEbX2uOxzR5oA@mail.gmail.com>
	<4EA9DAEC.1030507@freeswitch.org>
Message-ID: <CAEWH9M8BF_iUyFvFp=7O1Lyf2Xh_8DNw8=z6vi3Ez5hAyGAH5w@mail.gmail.com>

Hi,

It was a vanilla install of Squeeze 64 bit.

Is the solution to install pkg-config?

Pete

On 27 October 2011 23:27, Stefan Knoblich <stkn at freeswitch.org> wrote:

> Please install pkg-config and ./bootstrap.sh again.
>
> On 10/28/11 00:17, Yihui Li wrote:
> > I got the same problem on ubuntu 10.10 running on linode VPS
> >
> > here is some of the outputs:
> >
> > checking for library containing recv... none required
> > checking for getopt_long... yes
> > checking for getaddrinfo... yes
> > ./configure: line 11073: syntax error near unexpected token `openssl,'
> > ./configure: line 11073: `    PKG_CHECK_MODULES(openssl, openssl,'
> > configure: error: ./configure.gnu failed for libs/iksemel
> >
> > however it does not occur on another newly installed ubuntu 10.04LTS.
> >
> >
> > On Fri, Oct 28, 2011 at 12:18 AM, Avi Marcus <avi at avimarcus.net <mailto:
> avi at avimarcus.net>> wrote:
> >
> >     This sounds like something to be jira'd.
> >     -Avi
> >
> >
> >     On Thu, Oct 27, 2011 at 6:13 PM, Jeff Lenk <jeff at jefflenk.com<mailto:
> jeff at jefflenk.com>> wrote:
> >
> >         You should provide somemore details here such as platform etc. I
> assume this
> >         does not happen for all platforms?
> >
> >         --
> >         View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/configure-fails-on-vanilla-checkout-tp6935612p6936975.html
> >         Sent from the freeswitch-users mailing list archive at
> Nabble.com.
> >
> >
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org <mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org <mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From trever.adams at gmail.com  Fri Oct 28 11:40:20 2011
From: trever.adams at gmail.com (Trever L. Adams)
Date: Fri, 28 Oct 2011 01:40:20 -0600
Subject: [Freeswitch-users] FreeTDM and internal extensions and
 caller_id (user directory?)
In-Reply-To: <3008.1319775620@ccs.covici.com>
References: <4EAA108C.1030907@gmail.com> <3008.1319775620@ccs.covici.com>
Message-ID: <4EAA5C64.8010905@gmail.com>

On 10/27/2011 10:20 PM, covici at ccs.covici.com wrote:
> What worked for me is in the [span zt <name>] you can set  number =>
>  <caller id number> and name => <caller id name> .
>
> Perhaps an ata would be better -- I have found freetdm to not be the
> most stable of modules I have seen.
I am afraid that number and name are not complete. Was it
caller_id_number and caller_id_name?
Thank you,
Trever

-- 
"Well-taught doctrines and principles have a more powerful influence on
behavior than rules. " -- Elder Dallin H. Oaks - Ensign, Nov. 1999, 79

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From trever.adams at gmail.com  Fri Oct 28 11:39:44 2011
From: trever.adams at gmail.com (Trever L. Adams)
Date: Fri, 28 Oct 2011 01:39:44 -0600
Subject: [Freeswitch-users] trouble making conditions for OR
In-Reply-To: <CAKzWOxWLeP1ofJbQkdgmezk-4X8UWjObVpPuWyKRjhF88roogQ@mail.gmail.com>
References: <4EA9C2C1.1020809@gmail.com>
	<CAKzWOxV4JmzKuEBiQANWmh5MmfV3=rGz+LiZYxw-aoG6oD4iwg@mail.gmail.com>
	<4EAA0FED.9000905@gmail.com>
	<CAKzWOxWLeP1ofJbQkdgmezk-4X8UWjObVpPuWyKRjhF88roogQ@mail.gmail.com>
Message-ID: <4EAA5C40.6080604@gmail.com>

On 10/28/2011 12:16 AM, Michael Collins wrote:
>
> To be frank, yes, you are messing it up. :)
>
Glad to be set straight. Thank you! The method you suggested worked
perfectly. Unfortunately, the regex did need those prefix periods. Now
it works. Thank you so very much!

Trever
-- 
"If there is righteousness in the heart, there will be beauty in the
character. If there is beauty in the character, there will be harmony in
the home. If there is harmony in the home, there will be order in the
nation. If there is order in the nation, there will be peace in the
world. So let it be." -- Scottish Blessing

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From faisal.rehman22 at hotmail.com  Fri Oct 28 11:47:18 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Fri, 28 Oct 2011 12:47:18 +0500
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <CAKzWOxUCgJbP0XmH-+AXyKfcQX=TCf8cQeh0irPP1Fc__HLcvA@mail.gmail.com>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>,
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>,
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>,
	<CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>,
	<BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>,
	<CAKzWOxUCgJbP0XmH-+AXyKfcQX=TCf8cQeh0irPP1Fc__HLcvA@mail.gmail.com>
Message-ID: <BAY167-W1790CAFE7260E497A93155ECD30@phx.gbl>


Hi Micheal,
What would be the difference between $14 and $9, will you please a little explain?

Thanks
Faisal Rehman
Date: Thu, 27 Oct 2011 08:40:12 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring



On Thu, Oct 27, 2011 at 1:02 AM, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:






Hi David,
Thanks for sharing the script but will that work too for Asterisk's Master.csv if I needed the ASR & ACD records from that file?
Use $14 (instead of $9) for Asterisk's Master.csv file.
-MC 

Regards,
Faisal Rehman


From: david.villasmil.work at gmail.com
Date: Wed, 26 Oct 2011 19:48:57 +0200
To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring

Hello,
That's a faily simple bash script. 
"cat" the log file to an awk script like so;

cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{ FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT: "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR; if(ASR<100){ system("echo send email");} }' 



the "system" is where you put the mail command
also, bear in mind that the master can get VERY large if you don't rotate it.
david




On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:








Hi Waseem,
Thanks for your help, I saw the scripts that were on the link that you provided to me but actually the problem is that I don't want to involve the database though it is a bit difficult task but according to my requirement I just want to read the whole Master.csv file & if the ASR or ACD is low on any server, it should send me email, so my question is how can I skip the database from it to interact directly with the Master.csv file.




Thanks in Advance!!
Faisal Rehman

From: wasim at convergence.pk
Date: Tue, 25 Oct 2011 07:32:35 +0500



To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring



Faisal:
Load up the CDR into a db (MySQL or otherwise) ...
For ACD you'll want a SQL query akinSELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec, SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE disposition=1;




For ASR you'll need another querySELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;ASR is calls_answered/calls_totalThere is sample (read: poor) code in http://jira.freeswitch.org/browse/FS-1981




which a google for "freeswitch ACD ASR" would have helped you get to in the first place.
-wasim
On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com> wrote:










Hi Everyone,
I want to write a script that reads the Master.csv & generate an alert if the ASR & ACD is low from the desired value. Anybody has written such script, please help me write it or provide any sample.






Thanks

Faisal Rehman 		 	   		  



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-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | peace be upon you ...





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From trever.adams at gmail.com  Fri Oct 28 11:49:53 2011
From: trever.adams at gmail.com (Trever L. Adams)
Date: Fri, 28 Oct 2011 01:49:53 -0600
Subject: [Freeswitch-users] FreeTDM woes
Message-ID: <4EAA5EA1.2010008@gmail.com>

I have a span of 4 FXS modules. They are all configured to use context
Internal-FXS. Channel 5 will dial. All the others won't even stop
sending the dialtone when one attempts to dial.

Channel 5 can call channels 1-3 (there is no 4 at the moment, I have
tried it where 4 isn't used and 5 becomes 4, same problem), but if 5
hangs up, the called channel continues to ring.

zt.conf has echo_cancellation_level at 16. Yet, I constantly get
messages like: Disable echo cancel for 1:5

I get messages like this continually: [ERR] ftmod_zt.c:1188 [s1c2][1:2]
Failed to process event from channel

Finally, with the help of someone on this list I now have my call
screening setup to work for the appropriate caller id names and numbers
when it has been blocked. The problem is, the screening part completely
fails.

These are the actions from the file, which are only slightly modified
from the example at
http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_13:_Call_Screening
. It asks for your name, it records. But it doesn't give ANYTHING to the
dialed party.

            <action application="set"
data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
            <action application="answer"/>
            <action application="sleep" data="1000"/>
            <action application="phrase" data="voicemail_record_name"/>
            <action application="playback" data="tone_stream://%(500, 0,
640)"/>
            <action application="set"
data="playback_terminators=#*0123456789"/>
            <action application="record" data="${call_screen_filename} 7
200 2"/>
            <action application="set" data="group_confirm_key=1"/>
            <action application="set" data="fail_on_single_reject=true"/>
            <action application="set"
data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/>
            <action application="set" data="continue_on_fail=true"/>
             <action application="bridge"
data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6"/>
            <action application="voicemail" data="default $${domain}
${dialed_number}"/>
            <action application="hangup"/>

I don't think the problem is the phrase:screen_confirm as I have tried
removing it. I do set dialed_number to my phone number at the top of
this extension so it is set.

It should be noted that all calls to this FXO port are routed as the
action above shows. The non-blocked-caller_id route has the exact same
bridge line.

Am I just plain missing something about the FreeTDM module? Is it this
broken?

Any help would be greatly appreciated.

Trever
-- 
"Karl's version of Parkinson's Law: Work expands to exceed the time
alloted it."


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From nbhatti at gmail.com  Fri Oct 28 11:57:25 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Fri, 28 Oct 2011 10:57:25 +0300
Subject: [Freeswitch-users] ./configure fails on vanilla checkout
In-Reply-To: <CAEWH9M8BF_iUyFvFp=7O1Lyf2Xh_8DNw8=z6vi3Ez5hAyGAH5w@mail.gmail.com>
References: <CAEWH9M8YO0F=1aKzqis5GVMbvze14e6s4NidkknPk=nd4mL0pQ@mail.gmail.com>
	<1319731991828-6936975.post@n2.nabble.com>
	<CALNZuwmBvrK0SF1RCAMVMJsOvv-V9-0StO4Fuz6+gNupJMYe8A@mail.gmail.com>
	<CAO1DH1_jAC387QQG4L0aoePw9T72SPTBZa05TDEbX2uOxzR5oA@mail.gmail.com>
	<4EA9DAEC.1030507@freeswitch.org>
	<CAEWH9M8BF_iUyFvFp=7O1Lyf2Xh_8DNw8=z6vi3Ez5hAyGAH5w@mail.gmail.com>
Message-ID: <CACak_yy6Tnzr1X-zKSsRuv1m8iPFwwOwvX4DRJd98pU8ZrJz1A@mail.gmail.com>

pkg-config install, and then bootstrap again fixes the issue.

On Fri, Oct 28, 2011 at 10:13 AM, Pete Kelly <pkelly at gmail.com> wrote:
> Hi,
> It was a vanilla install of Squeeze 64 bit.
> Is the solution to install pkg-config?
> Pete
> On 27 October 2011 23:27, Stefan Knoblich <stkn at freeswitch.org> wrote:
>>
>> Please install pkg-config and ./bootstrap.sh again.
>>
>> On 10/28/11 00:17, Yihui Li wrote:
>> > I got the same problem on ubuntu 10.10 running on linode VPS
>> >
>> > here is some of the outputs:
>> >
>> > checking for library containing recv... none required
>> > checking for getopt_long... yes
>> > checking for getaddrinfo... yes
>> > ./configure: line 11073: syntax error near unexpected token `openssl,'
>> > ./configure: line 11073: ` ? ?PKG_CHECK_MODULES(openssl, openssl,'
>> > configure: error: ./configure.gnu failed for libs/iksemel
>> >
>> > however it does not occur on another newly installed ubuntu 10.04LTS.
>> >
>> >
>> > On Fri, Oct 28, 2011 at 12:18 AM, Avi Marcus <avi at avimarcus.net
>> > <mailto:avi at avimarcus.net>> wrote:
>> >
>> > ? ? This sounds like something to be jira'd.
>> > ? ? -Avi
>> >
>> >
>> > ? ? On Thu, Oct 27, 2011 at 6:13 PM, Jeff Lenk <jeff at jefflenk.com
>> > <mailto:jeff at jefflenk.com>> wrote:
>> >
>> > ? ? ? ? You should provide somemore details here such as platform etc. I
>> > assume this
>> > ? ? ? ? does not happen for all platforms?
>> >
>> > ? ? ? ? --
>> > ? ? ? ? View this message in context:
>> > http://freeswitch-users.2379917.n2.nabble.com/configure-fails-on-vanilla-checkout-tp6935612p6936975.html
>> > ? ? ? ? Sent from the freeswitch-users mailing list archive at
>> > Nabble.com.
>> >
>> >
>> > ? ? ? ? FreeSWITCH-users mailing list
>> > ? ? ? ? FreeSWITCH-users at lists.freeswitch.org
>> > <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > ? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > ? ? ? ? http://www.freeswitch.org
>> >
>> >
>> >
>> >
>> > ? ? FreeSWITCH-users mailing list
>> > ? ? FreeSWITCH-users at lists.freeswitch.org
>> > <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > ? ? http://www.freeswitch.org
>> >
>> >
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
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>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
>
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>


From miha at softnet.si  Fri Oct 28 12:35:46 2011
From: miha at softnet.si (Miha Zoubek)
Date: Fri, 28 Oct 2011 10:35:46 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <CAKzWOxWRM1xUeJHwYkd7dOwkBELY8jkzjvJaFZFS1gjaQ00Axg@mail.gmail.com>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
	<4EA92D49.5030909@softnet.si>
	<CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>
	<4EA9552C.2030205@softnet.si> <4EA96090.2090908@softnet.si>
	<CAKzWOxWRM1xUeJHwYkd7dOwkBELY8jkzjvJaFZFS1gjaQ00Axg@mail.gmail.com>
Message-ID: <4EAA6962.5000509@softnet.si>

Hi @Michael,

I have done as you said.

http://pastebin.freeswitch.org/17628 (usr miha-)

I have installed on new server (cenots) freeswitch (from git), as sofia 
global command was not recognised on version 1.0.6.

I configured it like on previous freeswitch.

Thank you!

BR,
Miha

On 10/27/2011 5:18 PM, Michael Collins wrote:
> Okay, a few things. First, let's use pastebin.freeswitch.org 
> <http://pastebin.freeswitch.org> since there is so much information. 
> Second, I recommend turning on sofia sip trace:
> sofia global siptrace on
>
> Make another test call and capture the output. Let's see what the 
> other side sends leading up to the no route message.
> -MC
>
> On Thu, Oct 27, 2011 at 6:45 AM, Miha Zoubek <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     This log is from cli:
>
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1164]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 136
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_thread
>     Log-Line: 1164
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.325859 [DEBUG] switch_core_session.c:1164
>     Session 8 (sofia/external/051357952) Locked, Waiting on external
>     entities
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [106]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [5]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_session.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_thread]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1182]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 106
>     Log-Level: 5
>     Text-Channel: 3
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_thread
>     Log-Line: 1182
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1182
>     Session 8 (sofia/external/051357952) Ended
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [115]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [5]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_session.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_thread]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1184]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 115
>     Log-Level: 5
>     Text-Channel: 0
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_thread
>     Log-Line: 1184
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1184
>     Close Channel sofia/external/051357952 [CS_DESTROY]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [126]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_destroy_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [428]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 126
>     Log-Level: 7
>     Text-Channel: 0
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_destroy_state
>     Log-Line: 428
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:428
>     (sofia/external/051357952) Running State Change CS_DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [108]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_destroy_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [439]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 108
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_destroy_state
>     Log-Line: 439
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:439
>     (sofia/external/051357952) State DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [90]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [mod_sofia.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [sofia_on_destroy]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [341]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 90
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: mod_sofia.c
>     Log-Func: sofia_on_destroy
>     Log-Line: 341
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.328931 [DEBUG] mod_sofia.c:341
>     sofia/external/051357952 SOFIA DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [108]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_standard_on_destroy]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [60]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 108
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_standard_on_destroy
>     Log-Line: 60
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:60
>     sofia/external/051357952 Standard DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [123]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_destroy_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [439]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 123
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_destroy_state
>     Log-Line: 439
>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>
>
>     2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:439
>     (sofia/external/051357952) State DESTROY going to sleep
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [82]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [mod_sofia.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [sofia_on_hangup]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [476]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 82
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: mod_sofia.c
>     Log-Func: sofia_on_hangup
>     Log-Line: 476
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.355813 [DEBUG] mod_sofia.c:476 Responding to
>     INVITE with: 404
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [144]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_standard_on_hangup]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [46]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 144
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_standard_on_hangup
>     Log-Line: 46
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:46
>     sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard HANGUP, cause:
>     NO_ROUTE_DESTINATION
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [130]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_hangup_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [499]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 130
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_hangup_state
>     Log-Line: 499
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:499
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP going to sleep
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [141]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_run]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [333]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 141
>     Log-Level: 7
>     Text-Channel: 0
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_run
>     Log-Line: 333
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:333
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change CS_HANGUP
>     -> CS_REPORTING
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [115]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_session.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_signal_state_change]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1021]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 115
>     Log-Level: 7
>     Text-Channel: 0
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_signal_state_change
>     Log-Line: 1021
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021 Send
>     signal sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [136]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_run]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [314]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 136
>     Log-Level: 7
>     Text-Channel: 0
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_run
>     Log-Line: 314
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:314
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>     CS_REPORTING
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [118]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_reporting_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [590]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 118
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_reporting_state
>     Log-Line: 590
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:590
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [147]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_standard_on_reporting]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [53]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 147
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_standard_on_reporting
>     Log-Line: 53
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:53
>     sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard REPORTING,
>     cause: NO_ROUTE_DESTINATION
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [133]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_reporting_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [590]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 133
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_reporting_state
>     Log-Line: 590
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:590
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING going
>     to sleep
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [142]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_run]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [327]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 142
>     Log-Level: 7
>     Text-Channel: 0
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_run
>     Log-Line: 327
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:327
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>     CS_REPORTING -> CS_DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [115]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_session.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_signal_state_change]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1021]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 115
>     Log-Level: 7
>     Text-Channel: 0
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_signal_state_change
>     Log-Line: 1021
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021 Send
>     signal sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [144]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_session.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_thread]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1164]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 144
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_thread
>     Log-Line: 1164
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1164
>     Session 7 (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) Locked, Waiting on
>     external entities
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [114]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [5]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_session.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_thread]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1182]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 114
>     Log-Level: 5
>     Text-Channel: 3
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_thread
>     Log-Line: 1182
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1182
>     Session 7 (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) Ended
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [123]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [5]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_session.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_thread]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [1184]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 123
>     Log-Level: 5
>     Text-Channel: 0
>     Log-File: switch_core_session.c
>     Log-Func: switch_core_session_thread
>     Log-Line: 1184
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1184
>     Close Channel sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_DESTROY]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [134]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [0]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_destroy_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [428]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 134
>     Log-Level: 7
>     Text-Channel: 0
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_destroy_state
>     Log-Line: 428
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:428
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>     CS_DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [116]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_destroy_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [439]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 116
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_destroy_state
>     Log-Line: 439
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:439
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [98]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [mod_sofia.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [sofia_on_destroy]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [341]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 98
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: mod_sofia.c
>     Log-Func: sofia_on_destroy
>     Log-Line: 341
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] mod_sofia.c:341
>     sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [116]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_standard_on_destroy]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [60]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 116
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_standard_on_destroy
>     Log-Line: 60
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:60
>     sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard DESTROY
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Type] = [log/data]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Content-Length] = [131]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Level] = [7]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Text-Channel] = [3]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-File] = [switch_core_state_machine.c]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Func] = [switch_core_session_destroy_state]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [Log-Line] = [439]
>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>     Event-Name: SOCKET_DATA
>     Content-Type: log/data
>     Content-Length: 131
>     Log-Level: 7
>     Text-Channel: 3
>     Log-File: switch_core_state_machine.c
>     Log-Func: switch_core_session_destroy_state
>     Log-Line: 439
>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>
>
>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:439
>     (sofia/internal/1030 at 172.31.1.192
>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY going to
>     sleep
>
>
>
>     On 10/27/2011 2:57 PM, Miha Zoubek wrote:
>>     Hi,
>>
>>     path of my dialplan is:
>>     /usr/local/freeswitch/conf/dialplan/default/test.xml
>>
>>
>>
>>     I rename IP800 for innovaphone.
>>
>>     Here is my debugging log:
>>
>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Regex (FAIL)
>>     [hold_music] destination_number(9051357952) =~ /^9999$/
>>     break=on-false
>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> parsing [default->Dial
>>     Out Custom Gateway] continue=false
>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Regex (PASS) [Dial Out
>>     Custom Gateway] destination_number(9051357952) =~ /^9(\d{9})$/
>>     break=on-false
>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Action
>>     bridge(sofia/gateway/innovaphone/051357952)
>>     2011-10-27 14:54:58.927692 [DEBUG]
>>     switch_core_state_machine.c:119 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>     CS_ROUTING -> CS_EXECUTE
>>     2011-10-27 14:54:58.927692 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>     2011-10-27 14:54:58.927692 [DEBUG]
>>     switch_core_state_machine.c:341 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State ROUTING going to
>>     sleep
>>     2011-10-27 14:54:58.927692 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>     CS_EXECUTE
>>     2011-10-27 14:54:58.927692 [DEBUG]
>>     switch_core_state_machine.c:348 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State EXECUTE
>>     2011-10-27 14:54:58.927692 [DEBUG] mod_sofia.c:226
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA EXECUTE
>>     2011-10-27 14:54:58.927692 [DEBUG]
>>     switch_core_state_machine.c:157 sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard EXECUTE
>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> set(open=true)
>>     2011-10-27 14:54:58.927692 [DEBUG] mod_dptools.c:816
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> SET [open]=[true]
>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>     hash(insert/172.31.1.192-spymap/1030/147c8578-500d-4d91-bad5-9391c4400934)
>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>     hash(insert/172.31.1.192-last_dial/1030/9051357952)
>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>     hash(insert/172.31.1.192-last_dial/global/147c8578-500d-4d91-bad5-9391c4400934)
>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>     bridge(sofia/gateway/innovaphone/051357952)
>>     2011-10-27 14:54:58.931685 [NOTICE] switch_channel.c:669 New
>>     Channel sofia/external/051357952
>>     [c4a81269-04e7-42db-aaf8-ecf528b74d68]
>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:3384
>>     (sofia/external/051357952) State Change CS_NEW -> CS_INIT
>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/external/051357952 [BREAK]
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>     Running State Change CS_INIT
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:338 (sofia/external/051357952) State INIT
>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:83
>>     sofia/external/051357952 SOFIA INIT
>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:117
>>     (sofia/external/051357952) State Change CS_INIT -> CS_ROUTING
>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/external/051357952 [BREAK]
>>     2011-10-27 14:54:58.933717 [DEBUG] sofia.c:4153 Channel
>>     sofia/external/051357952 entering state [calling][0]
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:338 (sofia/external/051357952) State
>>     INIT going to sleep
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>     Running State Change CS_ROUTING
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:341 (sofia/external/051357952) State
>>     ROUTING
>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:140
>>     sofia/external/051357952 SOFIA ROUTING
>>     2011-10-27 14:54:58.933717 [DEBUG] switch_ivr_originate.c:66
>>     (sofia/external/051357952) State Change CS_ROUTING ->
>>     CS_CONSUME_MEDIA
>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/external/051357952 [BREAK]
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:341 (sofia/external/051357952) State
>>     ROUTING going to sleep
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>     Running State Change CS_CONSUME_MEDIA
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:360 (sofia/external/051357952) State
>>     CONSUME_MEDIA
>>     2011-10-27 14:54:58.933717 [DEBUG]
>>     switch_core_state_machine.c:360 (sofia/external/051357952) State
>>     CONSUME_MEDIA going to sleep
>>     2011-10-27 14:54:58.937691 [DEBUG] sofia.c:4153 Channel
>>     sofia/external/051357952 entering state [calling][0]
>>     2011-10-27 14:54:59.062721 [DEBUG] sofia.c:4153 Channel
>>     sofia/external/051357952 entering state [terminated][404]
>>     2011-10-27 14:54:59.062721 [NOTICE] sofia.c:4789 Hangup
>>     sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>>     2011-10-27 14:54:59.062721 [DEBUG] switch_channel.c:2102 Send
>>     signal sofia/external/051357952 [KILL]
>>     2011-10-27 14:54:59.062721 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/external/051357952 [BREAK]
>>     2011-10-27 14:54:59.062721 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>     Running State Change CS_HANGUP
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_ivr_originate.c:3228
>>     Originate Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
>>     2011-10-27 14:54:59.064679 [INFO] mod_dptools.c:2355 Originate
>>     Failed.  Cause: NO_ROUTE_DESTINATION
>>     2011-10-27 14:54:59.064679 [NOTICE] mod_dptools.c:2418 Hangup
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_EXECUTE]
>>     [NO_ROUTE_DESTINATION]
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_channel.c:2102 Send
>>     signal sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [KILL]
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:348 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State EXECUTE going to
>>     sleep
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>     CS_HANGUP
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:499 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP
>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Overriding SIP cause
>>     404 with 404 from the other leg
>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> hanging up, cause:
>>     NO_ROUTE_DESTINATION
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:499 (sofia/external/051357952) State
>>     HANGUP
>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408
>>     sofia/external/051357952 Overriding SIP cause 404 with 404 from
>>     the other leg
>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
>>     sofia/external/051357952 hanging up, cause: NO_ROUTE_DESTINATION
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:46
>>     sofia/external/051357952 Standard HANGUP, cause: NO_ROUTE_DESTINATION
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:499 (sofia/external/051357952) State
>>     HANGUP going to sleep
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:333 (sofia/external/051357952) State
>>     Change CS_HANGUP -> CS_REPORTING
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/external/051357952 [BREAK]
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>     Running State Change CS_REPORTING
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:590 (sofia/external/051357952) State
>>     REPORTING
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_state_machine.c:53
>>     sofia/external/051357952 Standard REPORTING, cause:
>>     NO_ROUTE_DESTINATION
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:590 (sofia/external/051357952) State
>>     REPORTING going to sleep
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:327 (sofia/external/051357952) State
>>     Change CS_REPORTING -> CS_DESTROY
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/external/051357952 [BREAK]
>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1164
>>     Session 4 (sofia/external/051357952) Locked, Waiting on external
>>     entities
>>     2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1182
>>     Session 4 (sofia/external/051357952) Ended
>>     2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1184
>>     Close Channel sofia/external/051357952 [CS_DESTROY]
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:428 (sofia/external/051357952)
>>     Running State Change CS_DESTROY
>>     2011-10-27 14:54:59.064679 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>     DESTROY
>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:341
>>     sofia/external/051357952 SOFIA DESTROY
>>     2011-10-27 14:54:59.066681 [DEBUG] switch_core_state_machine.c:60
>>     sofia/external/051357952 Standard DESTROY
>>     2011-10-27 14:54:59.066681 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>     DESTROY going to sleep
>>     2011-10-27 14:54:59.108644 [DEBUG] mod_sofia.c:476 Responding to
>>     INVITE with: 404
>>     2011-10-27 14:54:59.108644 [DEBUG] switch_core_state_machine.c:46
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard HANGUP, cause:
>>     NO_ROUTE_DESTINATION
>>     2011-10-27 14:54:59.108644 [DEBUG]
>>     switch_core_state_machine.c:499 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP going to
>>     sleep
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:333 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change CS_HANGUP
>>     -> CS_REPORTING
>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>     CS_REPORTING
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:590 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING
>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:53
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard REPORTING,
>>     cause: NO_ROUTE_DESTINATION
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:590 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING going
>>     to sleep
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:327 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>     CS_REPORTING -> CS_DESTROY
>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1164
>>     Session 3 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Locked, Waiting on
>>     external entities
>>     2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1182
>>     Session 3 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Ended
>>     2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1184
>>     Close Channel sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_DESTROY]
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:428 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>     CS_DESTROY
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY
>>     2011-10-27 14:54:59.110650 [DEBUG] mod_sofia.c:341
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA DESTROY
>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_state_machine.c:60
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard DESTROY
>>     2011-10-27 14:54:59.110650 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY going to
>>     slee
>>
>>
>>     BR,
>>     Miha
>>
>>     On 10/27/2011 2:15 PM, Nandy Dagondon wrote:
>>>     what is the full path/filename of your dialplan?
>>>     your debug level is not detailed enough.  set the log level to 7.
>>>     $fs_cli -l 7
>>>
>>>
>>>     On Thu, Oct 27, 2011 at 6:07 PM, Miha Zoubek <miha at softnet.si
>>>     <mailto:miha at softnet.si>> wrote:
>>>
>>>         Hi,
>>>
>>>         I have dialed 9051357952 <tel:9051357952>. IP800 is our GW.
>>>         I created on it one SIP account and connect freeswitch with it.
>>>
>>>         BR,
>>>         Miha
>>>
>>>
>>>         On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
>>>>         hi,
>>>>
>>>>         what is IP800? is this a FXO gateway? or a SIP provider?
>>>>
>>>>         u hv to dial 9051357952 <tel:9051357952> to match the
>>>>         dialplan.
>>>>
>>>>         -nandy
>>>>
>>>>         On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek
>>>>         <miha at softnet.si <mailto:miha at softnet.si>> wrote:
>>>>
>>>>             Hi,
>>>>
>>>>             I am new Freeswitch user. I have read freeswitch book.
>>>>             I registered first few phones and make few inbound calls.
>>>>             I am trying to get outgoing calls working but having no
>>>>             luck.
>>>>             I have configured my gw and set up calling plan. When I
>>>>             try to dial number 051357952 I get this:
>>>>
>>>>             reeswitch at localhost.localdomain> 2011-10-26
>>>>             <tel:2011-10-26> 15:27:25.919562 <tel:25.919562>
>>>>             [NOTICE] switch_channel.c:669 New Channel
>>>>             sofia/internal/1030 at 172.31.1.157
>>>>             <mailto:1030 at 172.31.1.157>
>>>>             [f2132149-dc40-4601-bd34-c22e5b06b696]
>>>>             2011-10-26 <tel:2011-10-26> 15:27:25.955550
>>>>             <tel:25.955550> [INFO] mod_dialplan_xml.c:418
>>>>             Processing 1030->9051357952 <tel:9051357952> in context
>>>>             default
>>>>             2011-10-26 <tel:2011-10-26> 15:27:25.960548
>>>>             <tel:25.960548> [NOTICE] switch_channel.c:669 New
>>>>             Channel sofia/external/051357952
>>>>             [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.095518
>>>>             <tel:26.095518> [NOTICE] sofia.c:4789 Hangup
>>>>             sofia/external/051357952 [CS_CONSUME_MEDIA]
>>>>             [NO_ROUTE_DESTINATION]
>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>             <tel:26.098534> [INFO] mod_dptools.c:2355 Originate
>>>>             Failed.  Cause: NO_ROUTE_DESTINATION
>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>             <tel:26.098534> [NOTICE] mod_dptools.c:2418 Hangup
>>>>             sofia/internal/1030 at 172.31.1.157
>>>>             <mailto:1030 at 172.31.1.157> [CS_EXECUTE]
>>>>             [NO_ROUTE_DESTINATION]
>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>             <tel:26.098534> [NOTICE] switch_core_session.c:1182
>>>>             Session 49 (sofia/external/051357952) Ended
>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>             <tel:26.098534> [NOTICE] switch_core_session.c:1184
>>>>             Close Channel sofia/external/051357952 [CS_DESTROY]
>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.140506
>>>>             <tel:26.140506> [NOTICE] switch_core_session.c:1182
>>>>             Session 48 (sofia/internal/1030 at 172.31.1.157
>>>>             <mailto:1030 at 172.31.1.157>) Ended
>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.140506
>>>>             <tel:26.140506> [NOTICE] switch_core_session.c:1184
>>>>             Close Channel sofia/internal/1030 at 172.31.1.157
>>>>             <mailto:1030 at 172.31.1.157> [CS_DESTROY]
>>>>
>>>>             GW:
>>>>
>>>>             <include>
>>>>             <gateway name="IP800">
>>>>             <!-- Your username, which is not your DID -->
>>>>             <param name="username" value="Demo2"/>
>>>>             <param name="password" value="1234"/>
>>>>             <param name="register" value="true"/>
>>>>             <param name="realm" value="172.31.1.200"/>
>>>>             </gateway>
>>>>             </include>
>>>>
>>>>             This is my dial plan:
>>>>
>>>>             <include>
>>>>             <extension name="Dial Out Custom Gateway">
>>>>             <condition field="destination_number"
>>>>             expression="^9(\d{9})$">
>>>>             <action application="bridge"
>>>>             data="sofia/gateway/IP800/$1"/>
>>>>             </condition>
>>>>             </extension>
>>>>             </include>
>>>>
>>>>
>>>>             sofia status
>>>>
>>>>                                  Name         
>>>>             Type                                       Data      State
>>>>             =================================================================================================
>>>>                              external       profile
>>>>             sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>>>>                 external::example.com <http://example.com>      
>>>>             gateway sip:joeuser at example.com      NOREG
>>>>                       external::IP800       gateway
>>>>             sip:Demo2 at 172.31.1.200      REGED
>>>>                              internal       profile
>>>>             sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>>>>                          172.31.1.157        
>>>>             alias                                   internal     
>>>>             ALIASED
>>>>             =================================================================================================
>>>>             2 profiles 1 alias
>>>>
>>>>
>>>>             Please help me why I am getting not route to destination.
>>>>
>>>>             Thank you very much for all your help!
>>>>
>>>>             BR,
>>>>             Miha
>>>>
>>>>
>>>>             FreeSWITCH-users mailing list
>>>>             FreeSWITCH-users at lists.freeswitch.org
>>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>             http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org
>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From hynek.cihlar at gmail.com  Fri Oct 28 12:37:16 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 28 Oct 2011 10:37:16 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAKbxfG-bCvU_tMPse5egBHZUf7oYLzxLaHzieMneycZCfi9F5Q@mail.gmail.com>
References: <CACF0505.211E1%krice@freeswitch.org>
	<D5E62885E535486BB92F94CB5CBBC1A1@e1705>
	<CAKbxfG8Hm_a2CBK6S1upZm43rT3brbp8XDAFXuYcycNZneGB2Q@mail.gmail.com>
	<3244411099634732796@unknownmsgid>
	<CAKbxfG-bCvU_tMPse5egBHZUf7oYLzxLaHzieMneycZCfi9F5Q@mail.gmail.com>
Message-ID: <CAJh8XdWSWjOGM1Ys8RPqF_QMnTrFZm+8mwNHR+aPZMiw_i5Jsg@mail.gmail.com>

My god!

Multi statements and transactions are two completely different thinks! You
should read up on the topic.

Hynek



On Thu, Oct 27, 2011 at 9:08 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Blah,
>
> I said I don't like mysql, but ok I'll look it up for you.......
>
> http://www.mail-archive.com/profox at leafe.com/msg33150.html
>
> AND
>
>
> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-connection-parameters.html
> search for 'FLAG_MULTI_STATEMENTS'
>
> From our own FS resources:
>
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg19883.html
>
> or
>
> http://tinyurl.com/4xo26sv
>
>
>
>
>
>
> On Thu, Oct 27, 2011 at 1:29 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> Transactions are by default supported by mysql, the transactions are
>> driven by sql statements themselves.
>>
>> Your statement is either incorrect or I interpret it wrong.
>>
>>
>> Sent from my mobile device
>>
>> On Oct 27, 2011, at 20:09, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>>
>> mysql does not work with transactions by default in defense of injection
>> attacks... read the thread its in the top.
>>
>>
>> On Thu, Oct 27, 2011 at 1:07 PM, Madovsky < <infos at madovsky.org>
>> infos at madovsky.org> wrote:
>>
>>> **
>>> Thanks Ken. here the link of official SIPP website
>>>  <http://sipp.sourceforge.net/>http://sipp.sourceforge.net/
>>>
>>> I didn't know it was a HP app ! :0)
>>>
>>> ----- Original Message -----
>>> *From:* Ken Rice <krice at freeswitch.org>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Thursday, October 27, 2011 1:54 PM
>>> *Subject:* Re: [Freeswitch-users] High load on database server
>>>
>>> SIPP works fine for this... However keep in mind some of their default
>>> scenario files arent exactly the best in the world and can in many
>>> situations leave a bit to be desired... Check the wiki I think there is more
>>> info on this on there
>>>
>>> K
>>>
>>>
>>> On 10/27/11 12:51 PM, "Madovsky" < <infos at madovsky.org>
>>> infos at madovsky.org> wrote:
>>>
>>> on this subject,
>>> is SIPp can be used to test to reproduce hundreds calls ?
>>>
>>>
>>> ----- Original Message -----
>>>
>>> *From:*  Anthony Minessale < <anthony.minessale at gmail.com>
>>> mailto:anthony.minessale at gmail.com <anthony.minessale at gmail.com>>
>>>
>>> *To:* FreeSWITCH Users Help < <freeswitch-users at lists.freeswitch.org>
>>> mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
>>>
>>>
>>> *Sent:* Thursday, October 27, 2011 1:39  PM
>>>
>>> *Subject:* Re: [Freeswitch-users] High load  on database server
>>>
>>>
>>> BTW I know it's a contradiction to tell you to consider older  ODBC and
>>> newer FS but I wrote FS so I can attest to its stability especially  the
>>> ODBC code in the core.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <<http://anthony.minessale at gmail.com>
>>> anthony.minessale at gmail.com>  wrote:
>>>
>>>
>>>
>>> Let's recap
>>>
>>>
>>>
>>> You have 4 moving parts, you have various versions of all 4 to choose
>>>  from and even different implementations of 3 of the 4.
>>>
>>>
>>> ODBC LIB
>>> ODBC DRIVER
>>>
>>> DATABASE
>>>
>>> FS
>>>
>>>
>>>
>>> You must choose a stable combination of all 4 which may require  specific
>>> configuration of each component as well.
>>>
>>>
>>>
>>> My only issue is FUD, I do not want people to advertise that FS does  not
>>> work on ODBC under load when the whole story is FS does not work under  load
>>> when you use fooODBC with barDRIVER with basBASE database server  configured
>>> a particular way.
>>>
>>>
>>>
>>> It should never crash, if it does, its a bug in something.  I can  speak
>>> for FS that there is no bug there.  The other 3 are  variables.
>>>
>>> Remember this next time there is a discussion about using system libs
>>>  and why we build our own depends.  The one thing we use system libs  for,
>>> ODBC, is a huge PITA.........
>>>
>>>
>>>
>>> Things to remember:
>>>
>>>
>>>
>>> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem
>>>  and has been made the default in latest versions.
>>>
>>> 2) If you must use Mysql, enable transactions and make sure your
>>>  odbc.ini is referencing the threadsafe version of the lib
>>>
>>>      libmyodbc3_r.so <-- note _r
>>>
>>> 3) Try various combos of drivers and odbc libs, many time newer is not
>>>  better, stable versions lie in the past.
>>>
>>> 4) build the drivers and odbc yourself or get the debug symbols so you
>>>  can get a backtrace, you could be finding a bug for them.......
>>>
>>> 5) Find a test to reproduce your problem so you can try different
>>>  database engines and driver combos.
>>>
>>> 6) Make sure you are on latest FS git so you know you have a stable
>>>  copy.
>>>
>>>
>>>
>>>
>>>
>>> Anyway, this is a pain, that's why ppl pay you to do it.
>>>
>>> Discuss this all you want here, just minimize any FUD to scare away
>>>  people who want to use it.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <<http://hynek.cihlar at gmail.com>
>>> hynek.cihlar at gmail.com> wrote:
>>>
>>>
>>> Do not agree. Issues happening under extreme cases like  high load with
>>> some specific conditions are hard to reproduce, sharing  the
>>> information in this case is a lot more efficient, for all.
>>>
>>>
>>> Sent from my mobile device
>>>
>>>
>>>
>>>
>>> On Oct 27, 2011, at 18:56, Robert Huddleston <<http://rhuddleston at gmail.com>
>>> rhuddleston at gmail.com> wrote:
>>>
>>> > My bologna  has a first name - it's O S C A R...
>>> >
>>> > This topic is getting  really old... I have to agree with Antm - get
>>> out of
>>> > the lazy  pants and do some research / hard work.
>>> >
>>> >
>>> >  -----Original Message-----
>>> > From: <http://freeswitch-users-bounces at lists.freeswitch.org>
>>> freeswitch-users-bounces at lists.freeswitch.org
>>> >  [ <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 Sergey
>>> > Okhapkin
>>> > Sent: Thursday, October 27, 2011 12:43  PM
>>> > To: FreeSWITCH Users Help
>>> > Subject: Re:  [Freeswitch-users] High load on database server
>>> >
>>> > What do  you mean "shared ODBC"?
>>> >
>>> > On Thursday 27 October 2011,  Madovsky wrote:
>>> >> but, is anyone experienced that with shared  ODBC ? (managing more
>>> than 330
>>> >> concurrent  calls)
>>> >>
>>> >> ----- Original Message -----
>>> >>  From: "Madovsky" < <http://infos at madovsky.org>infos at madovsky.org>
>>> >> To: "FreeSWITCH Users  Help" <<http://freeswitch-users at lists.freeswitch.org>
>>> freeswitch-users at lists.freeswitch.org>
>>> >>  Sent: Thursday, October 27, 2011 12:33 PM
>>> >> Subject: Re:  [Freeswitch-users] High load on database  server
>>> >>
>>> >>> ha ok, good luck  so
>>> >>>
>>> >>> ----- Original Message  -----
>>> >>> From: "Cliff Wells" < <http://cliff at develix.com>cliff at develix.com>
>>> >>> To: "FreeSWITCH  Users Help" <<http://freeswitch-users at lists.freeswitch.org>
>>> freeswitch-users at lists.freeswitch.org>
>>> >>>  Sent: Thursday, October 27, 2011 12:26 PM
>>> >>> Subject: Re:  [Freeswitch-users] High load on database  server
>>> >>>
>>> >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
>>> >>>>> I means that depend the  quality of your script ;)
>>> >>>>
>>> >>>> Quality  isn't the issue here.  The script is too simple to be
>>>  incorrect
>>> >>>> (and it carefully releases the odbc  connection back to the pool in
>>> a
>>> >>>> hangup handler).   I have written poor-quality code plenty of times,
>>>  but
>>> >>>> 25 years as a programmer usually allows me the  luxury of knowing
>>> when
>>> >>>> I'm doing it,  thanks.
>>> >>>>
>>> >>>> In any case, Lua scripts  work fine.  Google will tell you that
>>> unixODBC
>>> >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies there,
>>>  but
>>> >>>> again, I emphasize, everything WORKS GREAT (been  using it for a
>>> couple
>>> >>>> of years now) until you get to  very high concurrency (about 330
>>> >>>> concurrent calls on a  single system).  If you don't expect to
>>> handle
>>> >>>>  more than 300 concurrent calls, then you do not need to worry about
>>>  it.
>>> >>>> At all.
>>> >>>>
>>> >>>>  Regards,
>>> >>>>  Cliff
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>  FreeSWITCH-users mailing list
>>> >>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>> >>
>>> >>  FreeSWITCH-users mailing list
>>> >> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>> >
>>> >
>>> >
>>> >  FreeSWITCH-users mailing list
>>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>> >
>>> >
>>> >
>>> >  FreeSWITCH-users mailing list
>>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>>
>>> FreeSWITCH-users  mailing list
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>  ------------------------------
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> 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
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH <http://www.freeswitch.org/>http://www.freeswitch.org/
>> ClueCon <http://www.cluecon.com/>http://www.cluecon.com/
>> Twitter: <http://twitter.com/FreeSWITCH_wire>
>> http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: <http://irc.freenode.net>irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip: <sip%3A888 at conference.freeswitch.org>888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Fri Oct 28 12:46:37 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Oct 2011 10:46:37 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CAJh8XdWSWjOGM1Ys8RPqF_QMnTrFZm+8mwNHR+aPZMiw_i5Jsg@mail.gmail.com>
References: <CACF0505.211E1%krice@freeswitch.org>
	<D5E62885E535486BB92F94CB5CBBC1A1@e1705>
	<CAKbxfG8Hm_a2CBK6S1upZm43rT3brbp8XDAFXuYcycNZneGB2Q@mail.gmail.com>
	<3244411099634732796@unknownmsgid>
	<CAKbxfG-bCvU_tMPse5egBHZUf7oYLzxLaHzieMneycZCfi9F5Q@mail.gmail.com>
	<CAJh8XdWSWjOGM1Ys8RPqF_QMnTrFZm+8mwNHR+aPZMiw_i5Jsg@mail.gmail.com>
Message-ID: <CALNZuwmTxNBW2OEhWc7PC7=vO9nGZGH_rKh-ySGUi53i14eFPg@mail.gmail.com>

He gave you the link to what he's talking about for a DB that he doesn't
even use.

Feel free to clean up the wiki page on this.

-Avi


On Fri, Oct 28, 2011 at 10:37 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> My god!
>
> Multi statements and transactions are two completely different thinks! You
> should read up on the topic.
>
> Hynek
>
>
>
> On Thu, Oct 27, 2011 at 9:08 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Blah,
>>
>> I said I don't like mysql, but ok I'll look it up for you.......
>>
>> http://www.mail-archive.com/profox at leafe.com/msg33150.html
>>
>> AND
>>
>>
>> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-connection-parameters.html
>> search for 'FLAG_MULTI_STATEMENTS'
>>
>> From our own FS resources:
>>
>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
>>
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg19883.html
>>
>> or
>>
>> http://tinyurl.com/4xo26sv
>>
>>
>>
>>
>>
>>
>> On Thu, Oct 27, 2011 at 1:29 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>>
>>> Transactions are by default supported by mysql, the transactions are
>>> driven by sql statements themselves.
>>>
>>> Your statement is either incorrect or I interpret it wrong.
>>>
>>>
>>> Sent from my mobile device
>>>
>>> On Oct 27, 2011, at 20:09, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>> mysql does not work with transactions by default in defense of injection
>>> attacks... read the thread its in the top.
>>>
>>>
>>> On Thu, Oct 27, 2011 at 1:07 PM, Madovsky < <infos at madovsky.org>
>>> infos at madovsky.org> wrote:
>>>
>>>> **
>>>> Thanks Ken. here the link of official SIPP website
>>>>  <http://sipp.sourceforge.net/>http://sipp.sourceforge.net/
>>>>
>>>> I didn't know it was a HP app ! :0)
>>>>
>>>> ----- Original Message -----
>>>> *From:* Ken Rice <krice at freeswitch.org>
>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> *Sent:* Thursday, October 27, 2011 1:54 PM
>>>> *Subject:* Re: [Freeswitch-users] High load on database server
>>>>
>>>> SIPP works fine for this... However keep in mind some of their default
>>>> scenario files arent exactly the best in the world and can in many
>>>> situations leave a bit to be desired... Check the wiki I think there is more
>>>> info on this on there
>>>>
>>>> K
>>>>
>>>>
>>>> On 10/27/11 12:51 PM, "Madovsky" < <infos at madovsky.org>
>>>> infos at madovsky.org> wrote:
>>>>
>>>> on this subject,
>>>> is SIPp can be used to test to reproduce hundreds calls ?
>>>>
>>>>
>>>> ----- Original Message -----
>>>>
>>>> *From:*  Anthony Minessale < <anthony.minessale at gmail.com>
>>>> mailto:anthony.minessale at gmail.com <anthony.minessale at gmail.com>>
>>>>
>>>> *To:* FreeSWITCH Users Help < <freeswitch-users at lists.freeswitch.org>
>>>> mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
>>>>
>>>>
>>>> *Sent:* Thursday, October 27, 2011 1:39  PM
>>>>
>>>> *Subject:* Re: [Freeswitch-users] High load  on database server
>>>>
>>>>
>>>> BTW I know it's a contradiction to tell you to consider older  ODBC and
>>>> newer FS but I wrote FS so I can attest to its stability especially  the
>>>> ODBC code in the core.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <<http://anthony.minessale at gmail.com>
>>>> anthony.minessale at gmail.com>  wrote:
>>>>
>>>>
>>>>
>>>> Let's recap
>>>>
>>>>
>>>>
>>>> You have 4 moving parts, you have various versions of all 4 to choose
>>>>  from and even different implementations of 3 of the 4.
>>>>
>>>>
>>>> ODBC LIB
>>>> ODBC DRIVER
>>>>
>>>> DATABASE
>>>>
>>>> FS
>>>>
>>>>
>>>>
>>>> You must choose a stable combination of all 4 which may require
>>>>  specific configuration of each component as well.
>>>>
>>>>
>>>>
>>>> My only issue is FUD, I do not want people to advertise that FS does
>>>>  not work on ODBC under load when the whole story is FS does not work under
>>>>  load when you use fooODBC with barDRIVER with basBASE database server
>>>>  configured a particular way.
>>>>
>>>>
>>>>
>>>> It should never crash, if it does, its a bug in something.  I can  speak
>>>> for FS that there is no bug there.  The other 3 are  variables.
>>>>
>>>> Remember this next time there is a discussion about using system libs
>>>>  and why we build our own depends.  The one thing we use system libs  for,
>>>> ODBC, is a huge PITA.........
>>>>
>>>>
>>>>
>>>> Things to remember:
>>>>
>>>>
>>>>
>>>> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem
>>>>  and has been made the default in latest versions.
>>>>
>>>> 2) If you must use Mysql, enable transactions and make sure your
>>>>  odbc.ini is referencing the threadsafe version of the lib
>>>>
>>>>      libmyodbc3_r.so <-- note _r
>>>>
>>>> 3) Try various combos of drivers and odbc libs, many time newer is not
>>>>  better, stable versions lie in the past.
>>>>
>>>> 4) build the drivers and odbc yourself or get the debug symbols so you
>>>>  can get a backtrace, you could be finding a bug for them.......
>>>>
>>>> 5) Find a test to reproduce your problem so you can try different
>>>>  database engines and driver combos.
>>>>
>>>> 6) Make sure you are on latest FS git so you know you have a stable
>>>>  copy.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Anyway, this is a pain, that's why ppl pay you to do it.
>>>>
>>>> Discuss this all you want here, just minimize any FUD to scare away
>>>>  people who want to use it.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <<http://hynek.cihlar at gmail.com>
>>>> hynek.cihlar at gmail.com> wrote:
>>>>
>>>>
>>>> Do not agree. Issues happening under extreme cases like  high load with
>>>> some specific conditions are hard to reproduce, sharing  the
>>>> information in this case is a lot more efficient, for all.
>>>>
>>>>
>>>> Sent from my mobile device
>>>>
>>>>
>>>>
>>>>
>>>> On Oct 27, 2011, at 18:56, Robert Huddleston <<http://rhuddleston at gmail.com>
>>>> rhuddleston at gmail.com> wrote:
>>>>
>>>> > My bologna  has a first name - it's O S C A R...
>>>> >
>>>> > This topic is getting  really old... I have to agree with Antm - get
>>>> out of
>>>> > the lazy  pants and do some research / hard work.
>>>> >
>>>> >
>>>> >  -----Original Message-----
>>>> > From: <http://freeswitch-users-bounces at lists.freeswitch.org>
>>>> freeswitch-users-bounces at lists.freeswitch.org
>>>> >  [ <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 Sergey
>>>> > Okhapkin
>>>> > Sent: Thursday, October 27, 2011 12:43  PM
>>>> > To: FreeSWITCH Users Help
>>>> > Subject: Re:  [Freeswitch-users] High load on database server
>>>> >
>>>> > What do  you mean "shared ODBC"?
>>>> >
>>>> > On Thursday 27 October 2011,  Madovsky wrote:
>>>> >> but, is anyone experienced that with shared  ODBC ? (managing more
>>>> than 330
>>>> >> concurrent  calls)
>>>> >>
>>>> >> ----- Original Message -----
>>>> >>  From: "Madovsky" < <http://infos at madovsky.org>infos at madovsky.org>
>>>> >> To: "FreeSWITCH Users  Help" <<http://freeswitch-users at lists.freeswitch.org>
>>>> freeswitch-users at lists.freeswitch.org>
>>>> >>  Sent: Thursday, October 27, 2011 12:33 PM
>>>> >> Subject: Re:  [Freeswitch-users] High load on database  server
>>>> >>
>>>> >>> ha ok, good luck  so
>>>> >>>
>>>> >>> ----- Original Message  -----
>>>> >>> From: "Cliff Wells" < <http://cliff at develix.com>cliff at develix.com>
>>>> >>> To: "FreeSWITCH  Users Help" <<http://freeswitch-users at lists.freeswitch.org>
>>>> freeswitch-users at lists.freeswitch.org>
>>>> >>>  Sent: Thursday, October 27, 2011 12:26 PM
>>>> >>> Subject: Re:  [Freeswitch-users] High load on database  server
>>>> >>>
>>>> >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
>>>> >>>>> I means that depend the  quality of your script ;)
>>>> >>>>
>>>> >>>> Quality  isn't the issue here.  The script is too simple to be
>>>>  incorrect
>>>> >>>> (and it carefully releases the odbc  connection back to the pool in
>>>> a
>>>> >>>> hangup handler).   I have written poor-quality code plenty of
>>>> times,  but
>>>> >>>> 25 years as a programmer usually allows me the  luxury of knowing
>>>> when
>>>> >>>> I'm doing it,  thanks.
>>>> >>>>
>>>> >>>> In any case, Lua scripts  work fine.  Google will tell you that
>>>> unixODBC
>>>> >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies
>>>> there,  but
>>>> >>>> again, I emphasize, everything WORKS GREAT (been  using it for a
>>>> couple
>>>> >>>> of years now) until you get to  very high concurrency (about 330
>>>> >>>> concurrent calls on a  single system).  If you don't expect to
>>>> handle
>>>> >>>>  more than 300 concurrent calls, then you do not need to worry
>>>> about  it.
>>>> >>>> At all.
>>>> >>>>
>>>> >>>>  Regards,
>>>> >>>>  Cliff
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>  FreeSWITCH-users mailing list
>>>> >>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>> >>
>>>> >>  FreeSWITCH-users mailing list
>>>> >> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>> >
>>>> >
>>>> >
>>>> >  FreeSWITCH-users mailing list
>>>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>> >
>>>> >
>>>> >
>>>> >  FreeSWITCH-users mailing list
>>>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>>
>>>>
>>>> FreeSWITCH-users  mailing list
>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>  ------------------------------
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> <FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> 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
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH <http://www.freeswitch.org/>http://www.freeswitch.org/
>>> ClueCon <http://www.cluecon.com/>http://www.cluecon.com/
>>> Twitter: <http://twitter.com/FreeSWITCH_wire>
>>> http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: <http://irc.freenode.net>irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip: <sip%3A888 at conference.freeswitch.org>888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fieldpeak at gmail.com  Fri Oct 28 12:49:59 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 28 Oct 2011 16:49:59 +0800
Subject: [Freeswitch-users] Log level of log file by fs_cli
Message-ID: <CAOTmbMwj8qnJZYGDFQeWQ83jyhwRXOQwj88g2UjGLDkKnr4Xdw@mail.gmail.com>

Hi friends,

i know we can set the loglevel inside switch.conf.xml to change the log
level of log files,

however, i would like to change the log level of log files thru fs_cli on
the fly(no need restart FS), is there any commands to support it, thanks.

-- 
Regards,
Charles
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From hynek.cihlar at gmail.com  Fri Oct 28 13:06:56 2011
From: hynek.cihlar at gmail.com (Hynek Cihlar)
Date: Fri, 28 Oct 2011 11:06:56 +0200
Subject: [Freeswitch-users] High load on database server
In-Reply-To: <CALNZuwmTxNBW2OEhWc7PC7=vO9nGZGH_rKh-ySGUi53i14eFPg@mail.gmail.com>
References: <CACF0505.211E1%krice@freeswitch.org>
	<D5E62885E535486BB92F94CB5CBBC1A1@e1705>
	<CAKbxfG8Hm_a2CBK6S1upZm43rT3brbp8XDAFXuYcycNZneGB2Q@mail.gmail.com>
	<3244411099634732796@unknownmsgid>
	<CAKbxfG-bCvU_tMPse5egBHZUf7oYLzxLaHzieMneycZCfi9F5Q@mail.gmail.com>
	<CAJh8XdWSWjOGM1Ys8RPqF_QMnTrFZm+8mwNHR+aPZMiw_i5Jsg@mail.gmail.com>
	<CALNZuwmTxNBW2OEhWc7PC7=vO9nGZGH_rKh-ySGUi53i14eFPg@mail.gmail.com>
Message-ID: <CAJh8XdUafT+7sSgcadSeiz0E2LBKbgYb0EStDW_nZauj8_g2-Q@mail.gmail.com>

Yes, I see the link - transaction statements are composed of multiple
statements, hence enabling multi statements... fine.

But what the heck does it have in common with "enabling transactions in
mysql"?? Transactions in mysql can't even be turned off in the first place!

Hynek



On Fri, Oct 28, 2011 at 10:46 AM, Avi Marcus <avi at avimarcus.net> wrote:

> He gave you the link to what he's talking about for a DB that he doesn't
> even use.
>
> Feel free to clean up the wiki page on this.
>
> -Avi
>
>
> On Fri, Oct 28, 2011 at 10:37 AM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>
>> My god!
>>
>> Multi statements and transactions are two completely different thinks! You
>> should read up on the topic.
>>
>> Hynek
>>
>>
>>
>> On Thu, Oct 27, 2011 at 9:08 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Blah,
>>>
>>> I said I don't like mysql, but ok I'll look it up for you.......
>>>
>>> http://www.mail-archive.com/profox at leafe.com/msg33150.html
>>>
>>> AND
>>>
>>>
>>> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-connection-parameters.html
>>> search for 'FLAG_MULTI_STATEMENTS'
>>>
>>> From our own FS resources:
>>>
>>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
>>>
>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg19883.html
>>>
>>> or
>>>
>>> http://tinyurl.com/4xo26sv
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Oct 27, 2011 at 1:29 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>>>
>>>> Transactions are by default supported by mysql, the transactions are
>>>> driven by sql statements themselves.
>>>>
>>>> Your statement is either incorrect or I interpret it wrong.
>>>>
>>>>
>>>> Sent from my mobile device
>>>>
>>>> On Oct 27, 2011, at 20:09, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>> mysql does not work with transactions by default in defense of injection
>>>> attacks... read the thread its in the top.
>>>>
>>>>
>>>> On Thu, Oct 27, 2011 at 1:07 PM, Madovsky < <infos at madovsky.org>
>>>> infos at madovsky.org> wrote:
>>>>
>>>>> **
>>>>> Thanks Ken. here the link of official SIPP website
>>>>>  <http://sipp.sourceforge.net/>http://sipp.sourceforge.net/
>>>>>
>>>>> I didn't know it was a HP app ! :0)
>>>>>
>>>>> ----- Original Message -----
>>>>> *From:* Ken Rice <krice at freeswitch.org>
>>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> *Sent:* Thursday, October 27, 2011 1:54 PM
>>>>> *Subject:* Re: [Freeswitch-users] High load on database server
>>>>>
>>>>> SIPP works fine for this... However keep in mind some of their default
>>>>> scenario files arent exactly the best in the world and can in many
>>>>> situations leave a bit to be desired... Check the wiki I think there is more
>>>>> info on this on there
>>>>>
>>>>> K
>>>>>
>>>>>
>>>>> On 10/27/11 12:51 PM, "Madovsky" < <infos at madovsky.org>
>>>>> infos at madovsky.org> wrote:
>>>>>
>>>>> on this subject,
>>>>> is SIPp can be used to test to reproduce hundreds calls ?
>>>>>
>>>>>
>>>>> ----- Original Message -----
>>>>>
>>>>> *From:*  Anthony Minessale < <anthony.minessale at gmail.com>
>>>>> mailto:anthony.minessale at gmail.com <anthony.minessale at gmail.com>>
>>>>>
>>>>> *To:* FreeSWITCH Users Help < <freeswitch-users at lists.freeswitch.org>
>>>>> mailto:freeswitch-users at lists.freeswitch.org<freeswitch-users at lists.freeswitch.org>>
>>>>>
>>>>>
>>>>> *Sent:* Thursday, October 27, 2011 1:39  PM
>>>>>
>>>>> *Subject:* Re: [Freeswitch-users] High load  on database server
>>>>>
>>>>>
>>>>> BTW I know it's a contradiction to tell you to consider older  ODBC and
>>>>> newer FS but I wrote FS so I can attest to its stability especially  the
>>>>> ODBC code in the core.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Oct 27, 2011 at 12:36 PM, Anthony Minessale  <<http://anthony.minessale at gmail.com>
>>>>> anthony.minessale at gmail.com>  wrote:
>>>>>
>>>>>
>>>>>
>>>>> Let's recap
>>>>>
>>>>>
>>>>>
>>>>> You have 4 moving parts, you have various versions of all 4 to choose
>>>>>  from and even different implementations of 3 of the 4.
>>>>>
>>>>>
>>>>> ODBC LIB
>>>>> ODBC DRIVER
>>>>>
>>>>> DATABASE
>>>>>
>>>>> FS
>>>>>
>>>>>
>>>>>
>>>>> You must choose a stable combination of all 4 which may require
>>>>>  specific configuration of each component as well.
>>>>>
>>>>>
>>>>>
>>>>> My only issue is FUD, I do not want people to advertise that FS does
>>>>>  not work on ODBC under load when the whole story is FS does not work under
>>>>>  load when you use fooODBC with barDRIVER with basBASE database server
>>>>>  configured a particular way.
>>>>>
>>>>>
>>>>>
>>>>> It should never crash, if it does, its a bug in something.  I can
>>>>>  speak for FS that there is no bug there.  The other 3 are  variables.
>>>>>
>>>>> Remember this next time there is a discussion about using system libs
>>>>>  and why we build our own depends.  The one thing we use system libs  for,
>>>>> ODBC, is a huge PITA.........
>>>>>
>>>>>
>>>>>
>>>>> Things to remember:
>>>>>
>>>>>
>>>>>
>>>>> 1) try the Threading=0 in the odbcinst.ini, this is a serious problem
>>>>>  and has been made the default in latest versions.
>>>>>
>>>>> 2) If you must use Mysql, enable transactions and make sure your
>>>>>  odbc.ini is referencing the threadsafe version of the lib
>>>>>
>>>>>      libmyodbc3_r.so <-- note _r
>>>>>
>>>>> 3) Try various combos of drivers and odbc libs, many time newer is not
>>>>>  better, stable versions lie in the past.
>>>>>
>>>>> 4) build the drivers and odbc yourself or get the debug symbols so you
>>>>>  can get a backtrace, you could be finding a bug for them.......
>>>>>
>>>>> 5) Find a test to reproduce your problem so you can try different
>>>>>  database engines and driver combos.
>>>>>
>>>>> 6) Make sure you are on latest FS git so you know you have a stable
>>>>>  copy.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Anyway, this is a pain, that's why ppl pay you to do it.
>>>>>
>>>>> Discuss this all you want here, just minimize any FUD to scare away
>>>>>  people who want to use it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Oct 27, 2011 at 12:24 PM, Hynek Cihlar <<http://hynek.cihlar at gmail.com>
>>>>> hynek.cihlar at gmail.com> wrote:
>>>>>
>>>>>
>>>>> Do not agree. Issues happening under extreme cases like  high load with
>>>>> some specific conditions are hard to reproduce, sharing  the
>>>>> information in this case is a lot more efficient, for all.
>>>>>
>>>>>
>>>>> Sent from my mobile device
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Oct 27, 2011, at 18:56, Robert Huddleston <<http://rhuddleston at gmail.com>
>>>>> rhuddleston at gmail.com> wrote:
>>>>>
>>>>> > My bologna  has a first name - it's O S C A R...
>>>>> >
>>>>> > This topic is getting  really old... I have to agree with Antm - get
>>>>> out of
>>>>> > the lazy  pants and do some research / hard work.
>>>>> >
>>>>> >
>>>>> >  -----Original Message-----
>>>>> > From: <http://freeswitch-users-bounces at lists.freeswitch.org>
>>>>> freeswitch-users-bounces at lists.freeswitch.org
>>>>> >  [ <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 Sergey
>>>>> > Okhapkin
>>>>> > Sent: Thursday, October 27, 2011 12:43  PM
>>>>> > To: FreeSWITCH Users Help
>>>>> > Subject: Re:  [Freeswitch-users] High load on database server
>>>>> >
>>>>> > What do  you mean "shared ODBC"?
>>>>> >
>>>>> > On Thursday 27 October 2011,  Madovsky wrote:
>>>>> >> but, is anyone experienced that with shared  ODBC ? (managing more
>>>>> than 330
>>>>> >> concurrent  calls)
>>>>> >>
>>>>> >> ----- Original Message -----
>>>>> >>  From: "Madovsky" < <http://infos at madovsky.org>infos at madovsky.org>
>>>>> >> To: "FreeSWITCH Users  Help" <<http://freeswitch-users at lists.freeswitch.org>
>>>>> freeswitch-users at lists.freeswitch.org>
>>>>> >>  Sent: Thursday, October 27, 2011 12:33 PM
>>>>> >> Subject: Re:  [Freeswitch-users] High load on database  server
>>>>> >>
>>>>> >>> ha ok, good luck  so
>>>>> >>>
>>>>> >>> ----- Original Message  -----
>>>>> >>> From: "Cliff Wells" < <http://cliff at develix.com>cliff at develix.com>
>>>>> >>> To: "FreeSWITCH  Users Help" <<http://freeswitch-users at lists.freeswitch.org>
>>>>> freeswitch-users at lists.freeswitch.org>
>>>>> >>>  Sent: Thursday, October 27, 2011 12:26 PM
>>>>> >>> Subject: Re:  [Freeswitch-users] High load on database  server
>>>>> >>>
>>>>> >>>> On Thu, 2011-10-27 at 11:55  -0400, Madovsky wrote:
>>>>> >>>>> I means that depend the  quality of your script ;)
>>>>> >>>>
>>>>> >>>> Quality  isn't the issue here.  The script is too simple to be
>>>>>  incorrect
>>>>> >>>> (and it carefully releases the odbc  connection back to the pool
>>>>> in a
>>>>> >>>> hangup handler).   I have written poor-quality code plenty of
>>>>> times,  but
>>>>> >>>> 25 years as a programmer usually allows me the  luxury of knowing
>>>>> when
>>>>> >>>> I'm doing it,  thanks.
>>>>> >>>>
>>>>> >>>> In any case, Lua scripts  work fine.  Google will tell you that
>>>>> unixODBC
>>>>> >>>>  before 2.3.0 was a bit of a mess, so I expect the issue lies
>>>>> there,  but
>>>>> >>>> again, I emphasize, everything WORKS GREAT (been  using it for a
>>>>> couple
>>>>> >>>> of years now) until you get to  very high concurrency (about 330
>>>>> >>>> concurrent calls on a  single system).  If you don't expect to
>>>>> handle
>>>>> >>>>  more than 300 concurrent calls, then you do not need to worry
>>>>> about  it.
>>>>> >>>> At all.
>>>>> >>>>
>>>>> >>>>  Regards,
>>>>> >>>>  Cliff
>>>>> >>>>
>>>>> >>>>
>>>>> >>>>
>>>>> >>>>  FreeSWITCH-users mailing list
>>>>> >>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>> >>
>>>>> >>  FreeSWITCH-users mailing list
>>>>> >> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>> >
>>>>> >
>>>>> >
>>>>> >  FreeSWITCH-users mailing list
>>>>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>> >
>>>>> >
>>>>> >
>>>>> >  FreeSWITCH-users mailing list
>>>>> > <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>>
>>>>>
>>>>> FreeSWITCH-users  mailing list
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>  ------------------------------
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH <http://www.freeswitch.org/>http://www.freeswitch.org/
>>>> ClueCon <http://www.cluecon.com/>http://www.cluecon.com/
>>>> Twitter: <http://twitter.com/FreeSWITCH_wire>
>>>> http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: <http://irc.freenode.net>irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip: <sip%3A888 at conference.freeswitch.org>888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 28 13:09:30 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 28 Oct 2011 11:09:30 +0200
Subject: [Freeswitch-users] Log level of log file by fs_cli
In-Reply-To: <CAOTmbMwj8qnJZYGDFQeWQ83jyhwRXOQwj88g2UjGLDkKnr4Xdw@mail.gmail.com>
References: <CAOTmbMwj8qnJZYGDFQeWQ83jyhwRXOQwj88g2UjGLDkKnr4Xdw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F2B77504@cooper>

Just enter "/log [loglevel]" inside fs_cli. I think there is a startup parameter as well.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
Skickat: den 28 oktober 2011 10:50
Till: FreeSWITCH-users
?mne: [Freeswitch-users] Log level of log file by fs_cli

Hi friends,

i know we can set the loglevel inside switch.conf.xml to change the log level of log files,

however, i would like to change the log level of log files thru fs_cli on the fly(no need restart FS), is there any commands to support it, thanks.

--
Regards,
Charles

!DSPAM:4eaa6d4932761466214554!
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From miha at softnet.si  Fri Oct 28 14:34:58 2011
From: miha at softnet.si (Miha Zoubek)
Date: Fri, 28 Oct 2011 12:34:58 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <4EAA6962.5000509@softnet.si>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
	<4EA92D49.5030909@softnet.si>
	<CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>
	<4EA9552C.2030205@softnet.si> <4EA96090.2090908@softnet.si>
	<CAKzWOxWRM1xUeJHwYkd7dOwkBELY8jkzjvJaFZFS1gjaQ00Axg@mail.gmail.com>
	<4EAA6962.5000509@softnet.si>
Message-ID: <4EAA8552.4070103@softnet.si>

Please look at the newest file.

Sorry for my mistake.

BR,
Miha

On 10/28/2011 10:35 AM, Miha Zoubek wrote:
> Hi @Michael,
>
> I have done as you said.
>
> http://pastebin.freeswitch.org/17628 (usr miha-)
>
> I have installed on new server (cenots) freeswitch (from git), as 
> sofia global command was not recognised on version 1.0.6.
>
> I configured it like on previous freeswitch.
>
> Thank you!
>
> BR,
> Miha
>
> On 10/27/2011 5:18 PM, Michael Collins wrote:
>> Okay, a few things. First, let's use pastebin.freeswitch.org 
>> <http://pastebin.freeswitch.org> since there is so much information. 
>> Second, I recommend turning on sofia sip trace:
>> sofia global siptrace on
>>
>> Make another test call and capture the output. Let's see what the 
>> other side sends leading up to the no route message.
>> -MC
>>
>> On Thu, Oct 27, 2011 at 6:45 AM, Miha Zoubek <miha at softnet.si 
>> <mailto:miha at softnet.si>> wrote:
>>
>>     This log is from cli:
>>
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1164]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 136
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_thread
>>     Log-Line: 1164
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.325859 [DEBUG] switch_core_session.c:1164
>>     Session 8 (sofia/external/051357952) Locked, Waiting on external
>>     entities
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [106]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [5]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_session.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_thread]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1182]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 106
>>     Log-Level: 5
>>     Text-Channel: 3
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_thread
>>     Log-Line: 1182
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1182
>>     Session 8 (sofia/external/051357952) Ended
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [115]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [5]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_session.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_thread]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1184]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 115
>>     Log-Level: 5
>>     Text-Channel: 0
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_thread
>>     Log-Line: 1184
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1184
>>     Close Channel sofia/external/051357952 [CS_DESTROY]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [126]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_destroy_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [428]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 126
>>     Log-Level: 7
>>     Text-Channel: 0
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_destroy_state
>>     Log-Line: 428
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.328931 [DEBUG]
>>     switch_core_state_machine.c:428 (sofia/external/051357952)
>>     Running State Change CS_DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [108]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_destroy_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [439]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 108
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_destroy_state
>>     Log-Line: 439
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.328931 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>     DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [90]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [mod_sofia.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [sofia_on_destroy]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [341]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 90
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: mod_sofia.c
>>     Log-Func: sofia_on_destroy
>>     Log-Line: 341
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.328931 [DEBUG] mod_sofia.c:341
>>     sofia/external/051357952 SOFIA DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [108]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_standard_on_destroy]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [60]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 108
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_standard_on_destroy
>>     Log-Line: 60
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.328931 [DEBUG] switch_core_state_machine.c:60
>>     sofia/external/051357952 Standard DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [123]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_destroy_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [439]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 123
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_destroy_state
>>     Log-Line: 439
>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>
>>
>>     2011-10-27 15:45:39.328931 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>     DESTROY going to sleep
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [82]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [mod_sofia.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [sofia_on_hangup]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [476]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 82
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: mod_sofia.c
>>     Log-Func: sofia_on_hangup
>>     Log-Line: 476
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.355813 [DEBUG] mod_sofia.c:476 Responding to
>>     INVITE with: 404
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [144]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_standard_on_hangup]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [46]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 144
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_standard_on_hangup
>>     Log-Line: 46
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:46
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard HANGUP, cause:
>>     NO_ROUTE_DESTINATION
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [130]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_hangup_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [499]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 130
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_hangup_state
>>     Log-Line: 499
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:499 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP going to
>>     sleep
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [141]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_run]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [333]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 141
>>     Log-Level: 7
>>     Text-Channel: 0
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_run
>>     Log-Line: 333
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:333 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change CS_HANGUP
>>     -> CS_REPORTING
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [115]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_session.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_signal_state_change]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1021]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 115
>>     Log-Level: 7
>>     Text-Channel: 0
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_signal_state_change
>>     Log-Line: 1021
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [136]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_run]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [314]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 136
>>     Log-Level: 7
>>     Text-Channel: 0
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_run
>>     Log-Line: 314
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:314 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>     CS_REPORTING
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [118]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_reporting_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [590]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 118
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_reporting_state
>>     Log-Line: 590
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:590 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [147]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_standard_on_reporting]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [53]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 147
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_standard_on_reporting
>>     Log-Line: 53
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:53
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard REPORTING,
>>     cause: NO_ROUTE_DESTINATION
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [133]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_reporting_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [590]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 133
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_reporting_state
>>     Log-Line: 590
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:590 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING going
>>     to sleep
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [142]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_run]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [327]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 142
>>     Log-Level: 7
>>     Text-Channel: 0
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_run
>>     Log-Line: 327
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:327 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>     CS_REPORTING -> CS_DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [115]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_session.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_signal_state_change]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1021]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 115
>>     Log-Level: 7
>>     Text-Channel: 0
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_signal_state_change
>>     Log-Line: 1021
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021
>>     Send signal sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [144]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_session.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_thread]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1164]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 144
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_thread
>>     Log-Line: 1164
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1164
>>     Session 7 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Locked, Waiting on
>>     external entities
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [114]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [5]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_session.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_thread]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1182]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 114
>>     Log-Level: 5
>>     Text-Channel: 3
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_thread
>>     Log-Line: 1182
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1182
>>     Session 7 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Ended
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [123]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [5]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_session.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_thread]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [1184]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 123
>>     Log-Level: 5
>>     Text-Channel: 0
>>     Log-File: switch_core_session.c
>>     Log-Func: switch_core_session_thread
>>     Log-Line: 1184
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1184
>>     Close Channel sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_DESTROY]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [134]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [0]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_destroy_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [428]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 134
>>     Log-Level: 7
>>     Text-Channel: 0
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_destroy_state
>>     Log-Line: 428
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:428 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>     CS_DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [116]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_destroy_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [439]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 116
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_destroy_state
>>     Log-Line: 439
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [98]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [mod_sofia.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [sofia_on_destroy]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [341]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 98
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: mod_sofia.c
>>     Log-Func: sofia_on_destroy
>>     Log-Line: 341
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG] mod_sofia.c:341
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [116]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_standard_on_destroy]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [60]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 116
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_standard_on_destroy
>>     Log-Line: 60
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_state_machine.c:60
>>     sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard DESTROY
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Type] = [log/data]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Content-Length] = [131]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Level] = [7]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Text-Channel] = [3]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-File] = [switch_core_state_machine.c]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Func] = [switch_core_session_destroy_state]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [Log-Line] = [439]
>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>     Event-Name: SOCKET_DATA
>>     Content-Type: log/data
>>     Content-Length: 131
>>     Log-Level: 7
>>     Text-Channel: 3
>>     Log-File: switch_core_state_machine.c
>>     Log-Func: switch_core_session_destroy_state
>>     Log-Line: 439
>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>
>>
>>     2011-10-27 15:45:39.357844 [DEBUG]
>>     switch_core_state_machine.c:439 (sofia/internal/1030 at 172.31.1.192
>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY going to
>>     sleep
>>
>>
>>
>>     On 10/27/2011 2:57 PM, Miha Zoubek wrote:
>>>     Hi,
>>>
>>>     path of my dialplan is:
>>>     /usr/local/freeswitch/conf/dialplan/default/test.xml
>>>
>>>
>>>
>>>     I rename IP800 for innovaphone.
>>>
>>>     Here is my debugging log:
>>>
>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Regex (FAIL)
>>>     [hold_music] destination_number(9051357952) =~ /^9999$/
>>>     break=on-false
>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> parsing [default->Dial
>>>     Out Custom Gateway] continue=false
>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Regex (PASS) [Dial Out
>>>     Custom Gateway] destination_number(9051357952) =~ /^9(\d{9})$/
>>>     break=on-false
>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Action
>>>     bridge(sofia/gateway/innovaphone/051357952)
>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>     switch_core_state_machine.c:119
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>     CS_ROUTING -> CS_EXECUTE
>>>     2011-10-27 14:54:58.927692 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>     switch_core_state_machine.c:341
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State ROUTING going
>>>     to sleep
>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>     switch_core_state_machine.c:314
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>     CS_EXECUTE
>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>     switch_core_state_machine.c:348
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State EXECUTE
>>>     2011-10-27 14:54:58.927692 [DEBUG] mod_sofia.c:226
>>>     sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA EXECUTE
>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>     switch_core_state_machine.c:157 sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard EXECUTE
>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> set(open=true)
>>>     2011-10-27 14:54:58.927692 [DEBUG] mod_dptools.c:816
>>>     sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> SET [open]=[true]
>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>     hash(insert/172.31.1.192-spymap/1030/147c8578-500d-4d91-bad5-9391c4400934)
>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>     hash(insert/172.31.1.192-last_dial/1030/9051357952)
>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>     hash(insert/172.31.1.192-last_dial/global/147c8578-500d-4d91-bad5-9391c4400934)
>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>     bridge(sofia/gateway/innovaphone/051357952)
>>>     2011-10-27 14:54:58.931685 [NOTICE] switch_channel.c:669 New
>>>     Channel sofia/external/051357952
>>>     [c4a81269-04e7-42db-aaf8-ecf528b74d68]
>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:3384
>>>     (sofia/external/051357952) State Change CS_NEW -> CS_INIT
>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/external/051357952 [BREAK]
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>     Running State Change CS_INIT
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:338 (sofia/external/051357952) State
>>>     INIT
>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:83
>>>     sofia/external/051357952 SOFIA INIT
>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:117
>>>     (sofia/external/051357952) State Change CS_INIT -> CS_ROUTING
>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/external/051357952 [BREAK]
>>>     2011-10-27 14:54:58.933717 [DEBUG] sofia.c:4153 Channel
>>>     sofia/external/051357952 entering state [calling][0]
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:338 (sofia/external/051357952) State
>>>     INIT going to sleep
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>     Running State Change CS_ROUTING
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:341 (sofia/external/051357952) State
>>>     ROUTING
>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:140
>>>     sofia/external/051357952 SOFIA ROUTING
>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_ivr_originate.c:66
>>>     (sofia/external/051357952) State Change CS_ROUTING ->
>>>     CS_CONSUME_MEDIA
>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/external/051357952 [BREAK]
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:341 (sofia/external/051357952) State
>>>     ROUTING going to sleep
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>     Running State Change CS_CONSUME_MEDIA
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:360 (sofia/external/051357952) State
>>>     CONSUME_MEDIA
>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>     switch_core_state_machine.c:360 (sofia/external/051357952) State
>>>     CONSUME_MEDIA going to sleep
>>>     2011-10-27 14:54:58.937691 [DEBUG] sofia.c:4153 Channel
>>>     sofia/external/051357952 entering state [calling][0]
>>>     2011-10-27 14:54:59.062721 [DEBUG] sofia.c:4153 Channel
>>>     sofia/external/051357952 entering state [terminated][404]
>>>     2011-10-27 14:54:59.062721 [NOTICE] sofia.c:4789 Hangup
>>>     sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>>>     2011-10-27 14:54:59.062721 [DEBUG] switch_channel.c:2102 Send
>>>     signal sofia/external/051357952 [KILL]
>>>     2011-10-27 14:54:59.062721 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/external/051357952 [BREAK]
>>>     2011-10-27 14:54:59.062721 [DEBUG]
>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>     Running State Change CS_HANGUP
>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_ivr_originate.c:3228
>>>     Originate Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
>>>     2011-10-27 14:54:59.064679 [INFO] mod_dptools.c:2355 Originate
>>>     Failed.  Cause: NO_ROUTE_DESTINATION
>>>     2011-10-27 14:54:59.064679 [NOTICE] mod_dptools.c:2418 Hangup
>>>     sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_EXECUTE]
>>>     [NO_ROUTE_DESTINATION]
>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_channel.c:2102 Send
>>>     signal sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [KILL]
>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:348
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State EXECUTE going
>>>     to sleep
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:314
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>     CS_HANGUP
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:499
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP
>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408
>>>     sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Overriding SIP cause
>>>     404 with 404 from the other leg
>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
>>>     sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> hanging up, cause:
>>>     NO_ROUTE_DESTINATION
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:499 (sofia/external/051357952) State
>>>     HANGUP
>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408
>>>     sofia/external/051357952 Overriding SIP cause 404 with 404 from
>>>     the other leg
>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
>>>     sofia/external/051357952 hanging up, cause: NO_ROUTE_DESTINATION
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:46 sofia/external/051357952 Standard
>>>     HANGUP, cause: NO_ROUTE_DESTINATION
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:499 (sofia/external/051357952) State
>>>     HANGUP going to sleep
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:333 (sofia/external/051357952) State
>>>     Change CS_HANGUP -> CS_REPORTING
>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/external/051357952 [BREAK]
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>     Running State Change CS_REPORTING
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:590 (sofia/external/051357952) State
>>>     REPORTING
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:53 sofia/external/051357952 Standard
>>>     REPORTING, cause: NO_ROUTE_DESTINATION
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:590 (sofia/external/051357952) State
>>>     REPORTING going to sleep
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:327 (sofia/external/051357952) State
>>>     Change CS_REPORTING -> CS_DESTROY
>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/external/051357952 [BREAK]
>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1164
>>>     Session 4 (sofia/external/051357952) Locked, Waiting on external
>>>     entities
>>>     2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1182
>>>     Session 4 (sofia/external/051357952) Ended
>>>     2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1184
>>>     Close Channel sofia/external/051357952 [CS_DESTROY]
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:428 (sofia/external/051357952)
>>>     Running State Change CS_DESTROY
>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>>     DESTROY
>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:341
>>>     sofia/external/051357952 SOFIA DESTROY
>>>     2011-10-27 14:54:59.066681 [DEBUG]
>>>     switch_core_state_machine.c:60 sofia/external/051357952 Standard
>>>     DESTROY
>>>     2011-10-27 14:54:59.066681 [DEBUG]
>>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>>     DESTROY going to sleep
>>>     2011-10-27 14:54:59.108644 [DEBUG] mod_sofia.c:476 Responding to
>>>     INVITE with: 404
>>>     2011-10-27 14:54:59.108644 [DEBUG]
>>>     switch_core_state_machine.c:46 sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard HANGUP,
>>>     cause: NO_ROUTE_DESTINATION
>>>     2011-10-27 14:54:59.108644 [DEBUG]
>>>     switch_core_state_machine.c:499
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP going to
>>>     sleep
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:333
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>     CS_HANGUP -> CS_REPORTING
>>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:314
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>     CS_REPORTING
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:590
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:53 sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard REPORTING,
>>>     cause: NO_ROUTE_DESTINATION
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:590
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING going
>>>     to sleep
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:327
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>     CS_REPORTING -> CS_DESTROY
>>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1164
>>>     Session 3 (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Locked, Waiting on
>>>     external entities
>>>     2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1182
>>>     Session 3 (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Ended
>>>     2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1184
>>>     Close Channel sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_DESTROY]
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:428
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>     CS_DESTROY
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:439
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY
>>>     2011-10-27 14:54:59.110650 [DEBUG] mod_sofia.c:341
>>>     sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA DESTROY
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:60 sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard DESTROY
>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>     switch_core_state_machine.c:439
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY going
>>>     to slee
>>>
>>>
>>>     BR,
>>>     Miha
>>>
>>>     On 10/27/2011 2:15 PM, Nandy Dagondon wrote:
>>>>     what is the full path/filename of your dialplan?
>>>>     your debug level is not detailed enough.  set the log level to 7.
>>>>     $fs_cli -l 7
>>>>
>>>>
>>>>     On Thu, Oct 27, 2011 at 6:07 PM, Miha Zoubek <miha at softnet.si
>>>>     <mailto:miha at softnet.si>> wrote:
>>>>
>>>>         Hi,
>>>>
>>>>         I have dialed 9051357952 <tel:9051357952>. IP800 is our GW.
>>>>         I created on it one SIP account and connect freeswitch with it.
>>>>
>>>>         BR,
>>>>         Miha
>>>>
>>>>
>>>>         On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
>>>>>         hi,
>>>>>
>>>>>         what is IP800? is this a FXO gateway? or a SIP provider?
>>>>>
>>>>>         u hv to dial 9051357952 <tel:9051357952> to match the
>>>>>         dialplan.
>>>>>
>>>>>         -nandy
>>>>>
>>>>>         On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek
>>>>>         <miha at softnet.si <mailto:miha at softnet.si>> wrote:
>>>>>
>>>>>             Hi,
>>>>>
>>>>>             I am new Freeswitch user. I have read freeswitch book.
>>>>>             I registered first few phones and make few inbound calls.
>>>>>             I am trying to get outgoing calls working but having
>>>>>             no luck.
>>>>>             I have configured my gw and set up calling plan. When
>>>>>             I try to dial number 051357952 I get this:
>>>>>
>>>>>             reeswitch at localhost.localdomain> 2011-10-26
>>>>>             <tel:2011-10-26> 15:27:25.919562 <tel:25.919562>
>>>>>             [NOTICE] switch_channel.c:669 New Channel
>>>>>             sofia/internal/1030 at 172.31.1.157
>>>>>             <mailto:1030 at 172.31.1.157>
>>>>>             [f2132149-dc40-4601-bd34-c22e5b06b696]
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:25.955550
>>>>>             <tel:25.955550> [INFO] mod_dialplan_xml.c:418
>>>>>             Processing 1030->9051357952 <tel:9051357952> in
>>>>>             context default
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:25.960548
>>>>>             <tel:25.960548> [NOTICE] switch_channel.c:669 New
>>>>>             Channel sofia/external/051357952
>>>>>             [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.095518
>>>>>             <tel:26.095518> [NOTICE] sofia.c:4789 Hangup
>>>>>             sofia/external/051357952 [CS_CONSUME_MEDIA]
>>>>>             [NO_ROUTE_DESTINATION]
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>             <tel:26.098534> [INFO] mod_dptools.c:2355 Originate
>>>>>             Failed.  Cause: NO_ROUTE_DESTINATION
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>             <tel:26.098534> [NOTICE] mod_dptools.c:2418 Hangup
>>>>>             sofia/internal/1030 at 172.31.1.157
>>>>>             <mailto:1030 at 172.31.1.157> [CS_EXECUTE]
>>>>>             [NO_ROUTE_DESTINATION]
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>             <tel:26.098534> [NOTICE] switch_core_session.c:1182
>>>>>             Session 49 (sofia/external/051357952) Ended
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>             <tel:26.098534> [NOTICE] switch_core_session.c:1184
>>>>>             Close Channel sofia/external/051357952 [CS_DESTROY]
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.140506
>>>>>             <tel:26.140506> [NOTICE] switch_core_session.c:1182
>>>>>             Session 48 (sofia/internal/1030 at 172.31.1.157
>>>>>             <mailto:1030 at 172.31.1.157>) Ended
>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.140506
>>>>>             <tel:26.140506> [NOTICE] switch_core_session.c:1184
>>>>>             Close Channel sofia/internal/1030 at 172.31.1.157
>>>>>             <mailto:1030 at 172.31.1.157> [CS_DESTROY]
>>>>>
>>>>>             GW:
>>>>>
>>>>>             <include>
>>>>>             <gateway name="IP800">
>>>>>             <!-- Your username, which is not your DID -->
>>>>>             <param name="username" value="Demo2"/>
>>>>>             <param name="password" value="1234"/>
>>>>>             <param name="register" value="true"/>
>>>>>             <param name="realm" value="172.31.1.200"/>
>>>>>             </gateway>
>>>>>             </include>
>>>>>
>>>>>             This is my dial plan:
>>>>>
>>>>>             <include>
>>>>>             <extension name="Dial Out Custom Gateway">
>>>>>             <condition field="destination_number"
>>>>>             expression="^9(\d{9})$">
>>>>>             <action application="bridge"
>>>>>             data="sofia/gateway/IP800/$1"/>
>>>>>             </condition>
>>>>>             </extension>
>>>>>             </include>
>>>>>
>>>>>
>>>>>             sofia status
>>>>>
>>>>>                                  Name         
>>>>>             Type                                       Data      State
>>>>>             =================================================================================================
>>>>>                              external       profile
>>>>>             sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>>>>>                 external::example.com <http://example.com>      
>>>>>             gateway sip:joeuser at example.com      NOREG
>>>>>                       external::IP800       gateway
>>>>>             sip:Demo2 at 172.31.1.200      REGED
>>>>>                              internal       profile
>>>>>             sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>>>>>                          172.31.1.157        
>>>>>             alias                                   internal     
>>>>>             ALIASED
>>>>>             =================================================================================================
>>>>>             2 profiles 1 alias
>>>>>
>>>>>
>>>>>             Please help me why I am getting not route to destination.
>>>>>
>>>>>             Thank you very much for all your help!
>>>>>
>>>>>             BR,
>>>>>             Miha
>>>>>
>>>>>
>>>>>             FreeSWITCH-users mailing list
>>>>>             FreeSWITCH-users at lists.freeswitch.org
>>>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>             http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>         FreeSWITCH-users mailing list
>>>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>         http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>
>>>
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bnaylor at sirran.com  Fri Oct 28 15:34:45 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Fri, 28 Oct 2011 12:34:45 +0100
Subject: [Freeswitch-users] RTP auto adjust
Message-ID: <024701cc9565$989f8b50$c9dea1f0$@sirran.com>

Hi Freeswitch community

 

I was wondering if someone could help me with the above?

 

The scenario is that I have one sip client connected to Freeswitch over 3G,
and one connected via Wifi.

 

When I make a call from the wifi to 3G, I can see in the console debug
messages this -

 

2011-10-28 12:14:42.777087 [INFO] switch_rtp.c:3164 Auto Changing port from
192.168.16.18:60416 to 193.xxx.xxx.123:60416

 

Success, the audio flows both ways.

 

However, when I then make a call from 3G to Wifi, I do not get this line in
the debug messages for the wifi sip client.  Why is this?  Does the auto
change only apply to the originating sip client?

 

The 3G client is passing the correct IP in the SDP details due to there
being a SIP ALG router in place to change the details.

 

Any help is greatly appreciated, as always.

 

Kind regards

 

Ben

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From miha at softnet.si  Fri Oct 28 15:46:14 2011
From: miha at softnet.si (Miha Zoubek)
Date: Fri, 28 Oct 2011 13:46:14 +0200
Subject: [Freeswitch-users] Getting NO_ROUTE_DESTINATION
In-Reply-To: <4EAA8552.4070103@softnet.si>
References: <4EA80C5F.6020501@softnet.si>
	<CAHKqMb65kLgsYZ3cL4K3S4JEkAeMzy9=PFQc3t3AyeZcvNiA6Q@mail.gmail.com>
	<4EA92D49.5030909@softnet.si>
	<CAHKqMb7b1V-iLC-ic4-Abppf94Zzzi_D17fERqYOMXnJoLzKJA@mail.gmail.com>
	<4EA9552C.2030205@softnet.si> <4EA96090.2090908@softnet.si>
	<CAKzWOxWRM1xUeJHwYkd7dOwkBELY8jkzjvJaFZFS1gjaQ00Axg@mail.gmail.com>
	<4EAA6962.5000509@softnet.si> <4EAA8552.4070103@softnet.si>
Message-ID: <4EAA9606.5030604@softnet.si>

Hi @Michael,

sorry for spaming:) I figure it out.

Thanks a lot.

New verison from git helped:)

BR,
Miha

On 10/28/2011 12:34 PM, Miha Zoubek wrote:
> Please look at the newest file.
>
> Sorry for my mistake.
>
> BR,
> Miha
>
> On 10/28/2011 10:35 AM, Miha Zoubek wrote:
>> Hi @Michael,
>>
>> I have done as you said.
>>
>> http://pastebin.freeswitch.org/17628 (usr miha-)
>>
>> I have installed on new server (cenots) freeswitch (from git), as 
>> sofia global command was not recognised on version 1.0.6.
>>
>> I configured it like on previous freeswitch.
>>
>> Thank you!
>>
>> BR,
>> Miha
>>
>> On 10/27/2011 5:18 PM, Michael Collins wrote:
>>> Okay, a few things. First, let's use pastebin.freeswitch.org 
>>> <http://pastebin.freeswitch.org> since there is so much information. 
>>> Second, I recommend turning on sofia sip trace:
>>> sofia global siptrace on
>>>
>>> Make another test call and capture the output. Let's see what the 
>>> other side sends leading up to the no route message.
>>> -MC
>>>
>>> On Thu, Oct 27, 2011 at 6:45 AM, Miha Zoubek <miha at softnet.si 
>>> <mailto:miha at softnet.si>> wrote:
>>>
>>>     This log is from cli:
>>>
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1164]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 136
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_thread
>>>     Log-Line: 1164
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.325859 [DEBUG] switch_core_session.c:1164
>>>     Session 8 (sofia/external/051357952) Locked, Waiting on external
>>>     entities
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [106]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [5]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_session.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_thread]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1182]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 106
>>>     Log-Level: 5
>>>     Text-Channel: 3
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_thread
>>>     Log-Line: 1182
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1182
>>>     Session 8 (sofia/external/051357952) Ended
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [115]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [5]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_session.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_thread]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1184]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 115
>>>     Log-Level: 5
>>>     Text-Channel: 0
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_thread
>>>     Log-Line: 1184
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.325859 [NOTICE] switch_core_session.c:1184
>>>     Close Channel sofia/external/051357952 [CS_DESTROY]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [126]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_destroy_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [428]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 126
>>>     Log-Level: 7
>>>     Text-Channel: 0
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_destroy_state
>>>     Log-Line: 428
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.328931 [DEBUG]
>>>     switch_core_state_machine.c:428 (sofia/external/051357952)
>>>     Running State Change CS_DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [108]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_destroy_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [439]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 108
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_destroy_state
>>>     Log-Line: 439
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.328931 [DEBUG]
>>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>>     DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [90]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [mod_sofia.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [sofia_on_destroy]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [341]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 90
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: mod_sofia.c
>>>     Log-Func: sofia_on_destroy
>>>     Log-Line: 341
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.328931 [DEBUG] mod_sofia.c:341
>>>     sofia/external/051357952 SOFIA DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [108]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_standard_on_destroy]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [60]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 108
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_standard_on_destroy
>>>     Log-Line: 60
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.328931 [DEBUG]
>>>     switch_core_state_machine.c:60 sofia/external/051357952 Standard
>>>     DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [123]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_destroy_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [439]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [9240aaac-7466-4a4d-aded-e74998dfeb01]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 123
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_destroy_state
>>>     Log-Line: 439
>>>     User-Data: 9240aaac-7466-4a4d-aded-e74998dfeb01
>>>
>>>
>>>     2011-10-27 15:45:39.328931 [DEBUG]
>>>     switch_core_state_machine.c:439 (sofia/external/051357952) State
>>>     DESTROY going to sleep
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [82]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [mod_sofia.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [sofia_on_hangup]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [476]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 82
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: mod_sofia.c
>>>     Log-Func: sofia_on_hangup
>>>     Log-Line: 476
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.355813 [DEBUG] mod_sofia.c:476 Responding to
>>>     INVITE with: 404
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [144]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_standard_on_hangup]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [46]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 144
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_standard_on_hangup
>>>     Log-Line: 46
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:46 sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard HANGUP,
>>>     cause: NO_ROUTE_DESTINATION
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [130]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_hangup_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [499]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 130
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_hangup_state
>>>     Log-Line: 499
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:499
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP going to
>>>     sleep
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [141]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_run]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [333]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 141
>>>     Log-Level: 7
>>>     Text-Channel: 0
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_run
>>>     Log-Line: 333
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:333
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>     CS_HANGUP -> CS_REPORTING
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [115]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_session.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_signal_state_change]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1021]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 115
>>>     Log-Level: 7
>>>     Text-Channel: 0
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_signal_state_change
>>>     Log-Line: 1021
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [136]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_run]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [314]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 136
>>>     Log-Level: 7
>>>     Text-Channel: 0
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_run
>>>     Log-Line: 314
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:314
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>     CS_REPORTING
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [118]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_reporting_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [590]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 118
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_reporting_state
>>>     Log-Line: 590
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:590
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [147]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_standard_on_reporting]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [53]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 147
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_standard_on_reporting
>>>     Log-Line: 53
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:53 sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard REPORTING,
>>>     cause: NO_ROUTE_DESTINATION
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [133]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_reporting_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [590]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 133
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_reporting_state
>>>     Log-Line: 590
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:590
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING going
>>>     to sleep
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [142]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_run]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [327]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 142
>>>     Log-Level: 7
>>>     Text-Channel: 0
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_run
>>>     Log-Line: 327
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:327
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>     CS_REPORTING -> CS_DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [115]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_session.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_signal_state_change]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1021]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 115
>>>     Log-Level: 7
>>>     Text-Channel: 0
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_signal_state_change
>>>     Log-Line: 1021
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1021
>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [144]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_session.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_thread]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1164]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 144
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_thread
>>>     Log-Line: 1164
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG] switch_core_session.c:1164
>>>     Session 7 (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Locked, Waiting on
>>>     external entities
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [114]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [5]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_session.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_thread]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1182]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 114
>>>     Log-Level: 5
>>>     Text-Channel: 3
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_thread
>>>     Log-Line: 1182
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1182
>>>     Session 7 (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Ended
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [123]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [5]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_session.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_thread]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [1184]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 123
>>>     Log-Level: 5
>>>     Text-Channel: 0
>>>     Log-File: switch_core_session.c
>>>     Log-Func: switch_core_session_thread
>>>     Log-Line: 1184
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [NOTICE] switch_core_session.c:1184
>>>     Close Channel sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_DESTROY]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [134]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [0]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_destroy_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [428]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 134
>>>     Log-Level: 7
>>>     Text-Channel: 0
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_destroy_state
>>>     Log-Line: 428
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:428
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>     CS_DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [116]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_destroy_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [439]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 116
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_destroy_state
>>>     Log-Line: 439
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:439
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [98]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [mod_sofia.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [sofia_on_destroy]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [341]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 98
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: mod_sofia.c
>>>     Log-Func: sofia_on_destroy
>>>     Log-Line: 341
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG] mod_sofia.c:341
>>>     sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [116]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_standard_on_destroy]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [60]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 116
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_standard_on_destroy
>>>     Log-Line: 60
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:60 sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard DESTROY
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Type] = [log/data]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Content-Length] = [131]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Level] = [7]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Text-Channel] = [3]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-File] = [switch_core_state_machine.c]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Func] = [switch_core_session_destroy_state]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [Log-Line] = [439]
>>>     [DEBUG] libs/esl/src/esl.c:918 esl_recv_event() RECV HEADER
>>>     [User-Data] = [962db612-d472-4f8b-88f9-5349f0034660]
>>>     [DEBUG] libs/esl/src/esl.c:1037 esl_recv_event() RECV MESSAGE
>>>     Event-Name: SOCKET_DATA
>>>     Content-Type: log/data
>>>     Content-Length: 131
>>>     Log-Level: 7
>>>     Text-Channel: 3
>>>     Log-File: switch_core_state_machine.c
>>>     Log-Func: switch_core_session_destroy_state
>>>     Log-Line: 439
>>>     User-Data: 962db612-d472-4f8b-88f9-5349f0034660
>>>
>>>
>>>     2011-10-27 15:45:39.357844 [DEBUG]
>>>     switch_core_state_machine.c:439
>>>     (sofia/internal/1030 at 172.31.1.192
>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY going
>>>     to sleep
>>>
>>>
>>>
>>>     On 10/27/2011 2:57 PM, Miha Zoubek wrote:
>>>>     Hi,
>>>>
>>>>     path of my dialplan is:
>>>>     /usr/local/freeswitch/conf/dialplan/default/test.xml
>>>>
>>>>
>>>>
>>>>     I rename IP800 for innovaphone.
>>>>
>>>>     Here is my debugging log:
>>>>
>>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Regex (FAIL)
>>>>     [hold_music] destination_number(9051357952) =~ /^9999$/
>>>>     break=on-false
>>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> parsing
>>>>     [default->Dial Out Custom Gateway] continue=false
>>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Regex (PASS) [Dial
>>>>     Out Custom Gateway] destination_number(9051357952) =~
>>>>     /^9(\d{9})$/ break=on-false
>>>>     Dialplan: sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Action
>>>>     bridge(sofia/gateway/innovaphone/051357952)
>>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>>     switch_core_state_machine.c:119
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>>     CS_ROUTING -> CS_EXECUTE
>>>>     2011-10-27 14:54:58.927692 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>>     switch_core_state_machine.c:341
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State ROUTING going
>>>>     to sleep
>>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>>     switch_core_state_machine.c:314
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>>     CS_EXECUTE
>>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>>     switch_core_state_machine.c:348
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State EXECUTE
>>>>     2011-10-27 14:54:58.927692 [DEBUG] mod_sofia.c:226
>>>>     sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA EXECUTE
>>>>     2011-10-27 14:54:58.927692 [DEBUG]
>>>>     switch_core_state_machine.c:157
>>>>     sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard EXECUTE
>>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> set(open=true)
>>>>     2011-10-27 14:54:58.927692 [DEBUG] mod_dptools.c:816
>>>>     sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> SET [open]=[true]
>>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>>     hash(insert/172.31.1.192-spymap/1030/147c8578-500d-4d91-bad5-9391c4400934)
>>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>>     hash(insert/172.31.1.192-last_dial/1030/9051357952)
>>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>>     hash(insert/172.31.1.192-last_dial/global/147c8578-500d-4d91-bad5-9391c4400934)
>>>>     EXECUTE sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>
>>>>     bridge(sofia/gateway/innovaphone/051357952)
>>>>     2011-10-27 14:54:58.931685 [NOTICE] switch_channel.c:669 New
>>>>     Channel sofia/external/051357952
>>>>     [c4a81269-04e7-42db-aaf8-ecf528b74d68]
>>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:3384
>>>>     (sofia/external/051357952) State Change CS_NEW -> CS_INIT
>>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/external/051357952 [BREAK]
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>>     Running State Change CS_INIT
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:338 (sofia/external/051357952)
>>>>     State INIT
>>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:83
>>>>     sofia/external/051357952 SOFIA INIT
>>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:117
>>>>     (sofia/external/051357952) State Change CS_INIT -> CS_ROUTING
>>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/external/051357952 [BREAK]
>>>>     2011-10-27 14:54:58.933717 [DEBUG] sofia.c:4153 Channel
>>>>     sofia/external/051357952 entering state [calling][0]
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:338 (sofia/external/051357952)
>>>>     State INIT going to sleep
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>>     Running State Change CS_ROUTING
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:341 (sofia/external/051357952)
>>>>     State ROUTING
>>>>     2011-10-27 14:54:58.933717 [DEBUG] mod_sofia.c:140
>>>>     sofia/external/051357952 SOFIA ROUTING
>>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_ivr_originate.c:66
>>>>     (sofia/external/051357952) State Change CS_ROUTING ->
>>>>     CS_CONSUME_MEDIA
>>>>     2011-10-27 14:54:58.933717 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/external/051357952 [BREAK]
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:341 (sofia/external/051357952)
>>>>     State ROUTING going to sleep
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>>     Running State Change CS_CONSUME_MEDIA
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:360 (sofia/external/051357952)
>>>>     State CONSUME_MEDIA
>>>>     2011-10-27 14:54:58.933717 [DEBUG]
>>>>     switch_core_state_machine.c:360 (sofia/external/051357952)
>>>>     State CONSUME_MEDIA going to sleep
>>>>     2011-10-27 14:54:58.937691 [DEBUG] sofia.c:4153 Channel
>>>>     sofia/external/051357952 entering state [calling][0]
>>>>     2011-10-27 14:54:59.062721 [DEBUG] sofia.c:4153 Channel
>>>>     sofia/external/051357952 entering state [terminated][404]
>>>>     2011-10-27 14:54:59.062721 [NOTICE] sofia.c:4789 Hangup
>>>>     sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>>>>     2011-10-27 14:54:59.062721 [DEBUG] switch_channel.c:2102 Send
>>>>     signal sofia/external/051357952 [KILL]
>>>>     2011-10-27 14:54:59.062721 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/external/051357952 [BREAK]
>>>>     2011-10-27 14:54:59.062721 [DEBUG]
>>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>>     Running State Change CS_HANGUP
>>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_ivr_originate.c:3228
>>>>     Originate Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
>>>>     2011-10-27 14:54:59.064679 [INFO] mod_dptools.c:2355 Originate
>>>>     Failed.  Cause: NO_ROUTE_DESTINATION
>>>>     2011-10-27 14:54:59.064679 [NOTICE] mod_dptools.c:2418 Hangup
>>>>     sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_EXECUTE]
>>>>     [NO_ROUTE_DESTINATION]
>>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_channel.c:2102 Send
>>>>     signal sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [KILL]
>>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:348
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State EXECUTE going
>>>>     to sleep
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:314
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>>     CS_HANGUP
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:499
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP
>>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408
>>>>     sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Overriding SIP cause
>>>>     404 with 404 from the other leg
>>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
>>>>     sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> hanging up, cause:
>>>>     NO_ROUTE_DESTINATION
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:499 (sofia/external/051357952)
>>>>     State HANGUP
>>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:408
>>>>     sofia/external/051357952 Overriding SIP cause 404 with 404 from
>>>>     the other leg
>>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:414 Channel
>>>>     sofia/external/051357952 hanging up, cause: NO_ROUTE_DESTINATION
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:46 sofia/external/051357952
>>>>     Standard HANGUP, cause: NO_ROUTE_DESTINATION
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:499 (sofia/external/051357952)
>>>>     State HANGUP going to sleep
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:333 (sofia/external/051357952)
>>>>     State Change CS_HANGUP -> CS_REPORTING
>>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/external/051357952 [BREAK]
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:314 (sofia/external/051357952)
>>>>     Running State Change CS_REPORTING
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:590 (sofia/external/051357952)
>>>>     State REPORTING
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:53 sofia/external/051357952
>>>>     Standard REPORTING, cause: NO_ROUTE_DESTINATION
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:590 (sofia/external/051357952)
>>>>     State REPORTING going to sleep
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:327 (sofia/external/051357952)
>>>>     State Change CS_REPORTING -> CS_DESTROY
>>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/external/051357952 [BREAK]
>>>>     2011-10-27 14:54:59.064679 [DEBUG] switch_core_session.c:1164
>>>>     Session 4 (sofia/external/051357952) Locked, Waiting on
>>>>     external entities
>>>>     2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1182
>>>>     Session 4 (sofia/external/051357952) Ended
>>>>     2011-10-27 14:54:59.064679 [NOTICE] switch_core_session.c:1184
>>>>     Close Channel sofia/external/051357952 [CS_DESTROY]
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:428 (sofia/external/051357952)
>>>>     Running State Change CS_DESTROY
>>>>     2011-10-27 14:54:59.064679 [DEBUG]
>>>>     switch_core_state_machine.c:439 (sofia/external/051357952)
>>>>     State DESTROY
>>>>     2011-10-27 14:54:59.064679 [DEBUG] mod_sofia.c:341
>>>>     sofia/external/051357952 SOFIA DESTROY
>>>>     2011-10-27 14:54:59.066681 [DEBUG]
>>>>     switch_core_state_machine.c:60 sofia/external/051357952
>>>>     Standard DESTROY
>>>>     2011-10-27 14:54:59.066681 [DEBUG]
>>>>     switch_core_state_machine.c:439 (sofia/external/051357952)
>>>>     State DESTROY going to sleep
>>>>     2011-10-27 14:54:59.108644 [DEBUG] mod_sofia.c:476 Responding
>>>>     to INVITE with: 404
>>>>     2011-10-27 14:54:59.108644 [DEBUG]
>>>>     switch_core_state_machine.c:46 sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard HANGUP,
>>>>     cause: NO_ROUTE_DESTINATION
>>>>     2011-10-27 14:54:59.108644 [DEBUG]
>>>>     switch_core_state_machine.c:499
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State HANGUP going
>>>>     to sleep
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:333
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>>     CS_HANGUP -> CS_REPORTING
>>>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:314
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>>     CS_REPORTING
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:590
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:53 sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard REPORTING,
>>>>     cause: NO_ROUTE_DESTINATION
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:590
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State REPORTING
>>>>     going to sleep
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:327
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State Change
>>>>     CS_REPORTING -> CS_DESTROY
>>>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1021
>>>>     Send signal sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [BREAK]
>>>>     2011-10-27 14:54:59.110650 [DEBUG] switch_core_session.c:1164
>>>>     Session 3 (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Locked, Waiting on
>>>>     external entities
>>>>     2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1182
>>>>     Session 3 (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Ended
>>>>     2011-10-27 14:54:59.110650 [NOTICE] switch_core_session.c:1184
>>>>     Close Channel sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> [CS_DESTROY]
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:428
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) Running State Change
>>>>     CS_DESTROY
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:439
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY
>>>>     2011-10-27 14:54:59.110650 [DEBUG] mod_sofia.c:341
>>>>     sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> SOFIA DESTROY
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:60 sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192> Standard DESTROY
>>>>     2011-10-27 14:54:59.110650 [DEBUG]
>>>>     switch_core_state_machine.c:439
>>>>     (sofia/internal/1030 at 172.31.1.192
>>>>     <mailto:sofia/internal/1030 at 172.31.1.192>) State DESTROY going
>>>>     to slee
>>>>
>>>>
>>>>     BR,
>>>>     Miha
>>>>
>>>>     On 10/27/2011 2:15 PM, Nandy Dagondon wrote:
>>>>>     what is the full path/filename of your dialplan?
>>>>>     your debug level is not detailed enough.  set the log level to 7.
>>>>>     $fs_cli -l 7
>>>>>
>>>>>
>>>>>     On Thu, Oct 27, 2011 at 6:07 PM, Miha Zoubek <miha at softnet.si
>>>>>     <mailto:miha at softnet.si>> wrote:
>>>>>
>>>>>         Hi,
>>>>>
>>>>>         I have dialed 9051357952 <tel:9051357952>. IP800 is our
>>>>>         GW. I created on it one SIP account and connect freeswitch
>>>>>         with it.
>>>>>
>>>>>         BR,
>>>>>         Miha
>>>>>
>>>>>
>>>>>         On 10/27/2011 11:19 AM, Nandy Dagondon wrote:
>>>>>>         hi,
>>>>>>
>>>>>>         what is IP800? is this a FXO gateway? or a SIP provider?
>>>>>>
>>>>>>         u hv to dial 9051357952 <tel:9051357952> to match the
>>>>>>         dialplan.
>>>>>>
>>>>>>         -nandy
>>>>>>
>>>>>>         On Wed, Oct 26, 2011 at 9:34 PM, Miha Zoubek
>>>>>>         <miha at softnet.si <mailto:miha at softnet.si>> wrote:
>>>>>>
>>>>>>             Hi,
>>>>>>
>>>>>>             I am new Freeswitch user. I have read freeswitch
>>>>>>             book. I registered first few phones and make few
>>>>>>             inbound calls.
>>>>>>             I am trying to get outgoing calls working but having
>>>>>>             no luck.
>>>>>>             I have configured my gw and set up calling plan. When
>>>>>>             I try to dial number 051357952 I get this:
>>>>>>
>>>>>>             reeswitch at localhost.localdomain> 2011-10-26
>>>>>>             <tel:2011-10-26> 15:27:25.919562 <tel:25.919562>
>>>>>>             [NOTICE] switch_channel.c:669 New Channel
>>>>>>             sofia/internal/1030 at 172.31.1.157
>>>>>>             <mailto:1030 at 172.31.1.157>
>>>>>>             [f2132149-dc40-4601-bd34-c22e5b06b696]
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:25.955550
>>>>>>             <tel:25.955550> [INFO] mod_dialplan_xml.c:418
>>>>>>             Processing 1030->9051357952 <tel:9051357952> in
>>>>>>             context default
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:25.960548
>>>>>>             <tel:25.960548> [NOTICE] switch_channel.c:669 New
>>>>>>             Channel sofia/external/051357952
>>>>>>             [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4]
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.095518
>>>>>>             <tel:26.095518> [NOTICE] sofia.c:4789 Hangup
>>>>>>             sofia/external/051357952 [CS_CONSUME_MEDIA]
>>>>>>             [NO_ROUTE_DESTINATION]
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>>             <tel:26.098534> [INFO] mod_dptools.c:2355 Originate
>>>>>>             Failed.  Cause: NO_ROUTE_DESTINATION
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>>             <tel:26.098534> [NOTICE] mod_dptools.c:2418 Hangup
>>>>>>             sofia/internal/1030 at 172.31.1.157
>>>>>>             <mailto:1030 at 172.31.1.157> [CS_EXECUTE]
>>>>>>             [NO_ROUTE_DESTINATION]
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>>             <tel:26.098534> [NOTICE] switch_core_session.c:1182
>>>>>>             Session 49 (sofia/external/051357952) Ended
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.098534
>>>>>>             <tel:26.098534> [NOTICE] switch_core_session.c:1184
>>>>>>             Close Channel sofia/external/051357952 [CS_DESTROY]
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.140506
>>>>>>             <tel:26.140506> [NOTICE] switch_core_session.c:1182
>>>>>>             Session 48 (sofia/internal/1030 at 172.31.1.157
>>>>>>             <mailto:1030 at 172.31.1.157>) Ended
>>>>>>             2011-10-26 <tel:2011-10-26> 15:27:26.140506
>>>>>>             <tel:26.140506> [NOTICE] switch_core_session.c:1184
>>>>>>             Close Channel sofia/internal/1030 at 172.31.1.157
>>>>>>             <mailto:1030 at 172.31.1.157> [CS_DESTROY]
>>>>>>
>>>>>>             GW:
>>>>>>
>>>>>>             <include>
>>>>>>             <gateway name="IP800">
>>>>>>             <!-- Your username, which is not your DID -->
>>>>>>             <param name="username" value="Demo2"/>
>>>>>>             <param name="password" value="1234"/>
>>>>>>             <param name="register" value="true"/>
>>>>>>             <param name="realm" value="172.31.1.200"/>
>>>>>>             </gateway>
>>>>>>             </include>
>>>>>>
>>>>>>             This is my dial plan:
>>>>>>
>>>>>>             <include>
>>>>>>             <extension name="Dial Out Custom Gateway">
>>>>>>             <condition field="destination_number"
>>>>>>             expression="^9(\d{9})$">
>>>>>>             <action application="bridge"
>>>>>>             data="sofia/gateway/IP800/$1"/>
>>>>>>             </condition>
>>>>>>             </extension>
>>>>>>             </include>
>>>>>>
>>>>>>
>>>>>>             sofia status
>>>>>>
>>>>>>                                  Name         
>>>>>>             Type                                       Data     
>>>>>>             State
>>>>>>             =================================================================================================
>>>>>>                              external       profile
>>>>>>             sip:mod_sofia at 172.31.1.157:5080      RUNNING (0)
>>>>>>                 external::example.com <http://example.com>      
>>>>>>             gateway sip:joeuser at example.com      NOREG
>>>>>>                       external::IP800       gateway
>>>>>>             sip:Demo2 at 172.31.1.200      REGED
>>>>>>                              internal       profile
>>>>>>             sip:mod_sofia at 172.31.1.157:5060      RUNNING (0)
>>>>>>                          172.31.1.157        
>>>>>>             alias                                   internal     
>>>>>>             ALIASED
>>>>>>             =================================================================================================
>>>>>>             2 profiles 1 alias
>>>>>>
>>>>>>
>>>>>>             Please help me why I am getting not route to destination.
>>>>>>
>>>>>>             Thank you very much for all your help!
>>>>>>
>>>>>>             BR,
>>>>>>             Miha
>>>>>>
>>>>>>
>>>>>>             FreeSWITCH-users mailing list
>>>>>>             FreeSWITCH-users at lists.freeswitch.org
>>>>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>             http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>         FreeSWITCH-users mailing list
>>>>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>         http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>         FreeSWITCH-users mailing list
>>>>>         FreeSWITCH-users at lists.freeswitch.org
>>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>         http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>     FreeSWITCH-users mailing list
>>>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>
>>>
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 28 15:53:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Oct 2011 13:53:31 +0200
Subject: [Freeswitch-users] Log level of log file by fs_cli
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F2B77504@cooper>
References: <CAOTmbMwj8qnJZYGDFQeWQ83jyhwRXOQwj88g2UjGLDkKnr4Xdw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2B77504@cooper>
Message-ID: <76208DFD-604B-4C02-AD0B-4A6C36425134@gmail.com>

Peter, that affects only the logging level of your fs_cli session - not the log file.

Fran, use the fsctl command from the cli. It's in mod_commands and documented on the wiki.

Eg:
fsctl loglevel
- shows current level
fsctl loglevel debug
- set to debug

Steve on iPhone

On 28 Oct 2011, at 11:09, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Just enter ?/log [loglevel]? inside fs_cli. I think there is a startup parameter as well.
>  
> /Peter
>  
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
> Skickat: den 28 oktober 2011 10:50
> Till: FreeSWITCH-users
> ?mne: [Freeswitch-users] Log level of log file by fs_cli
>  
> Hi friends,
> 
> i know we can set the loglevel inside switch.conf.xml to change the log level of log files, 
> 
> however, i would like to change the log level of log files thru fs_cli on the fly(no need restart FS), is there any commands to support it, thanks.
> 
> -- 
> Regards,
> Charles
> 
> !DSPAM:4eaa6d4932761466214554!
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From beppe.grillo at gmail.com  Fri Oct 28 16:13:36 2011
From: beppe.grillo at gmail.com (Beppe Grillo)
Date: Fri, 28 Oct 2011 14:13:36 +0200
Subject: [Freeswitch-users] Docs Down?
In-Reply-To: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
References: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
Message-ID: <CA+zSTk2uf4Ug-3uKUAX0eRQcuJ19LQTfH2mMyro4Vme4R07VJQ@mail.gmail.com>

No, i see
FreeSWITCH API Documentation.
Regards,
Giuseppe.

2011/10/28 Peter Blackford <peter at onemetric.com>

> Is docs.freeswitch.org returning blank pages for everyone or is it just
> me?
>
> Regards,
> Peter
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter at onemetric.com  Fri Oct 28 16:16:11 2011
From: peter at onemetric.com (Peter Blackford)
Date: Fri, 28 Oct 2011 23:16:11 +1100
Subject: [Freeswitch-users] Docs Down?
In-Reply-To: <CA+zSTk2uf4Ug-3uKUAX0eRQcuJ19LQTfH2mMyro4Vme4R07VJQ@mail.gmail.com>
References: <CAG1aRi_0xDX6c3eJWpw=Lzy3tufGp1n+NqN=kgXOg1Bq2aTytA@mail.gmail.com>
	<CA+zSTk2uf4Ug-3uKUAX0eRQcuJ19LQTfH2mMyro4Vme4R07VJQ@mail.gmail.com>
Message-ID: <CAG1aRi-dxT_mFU4uK3qxvHpRV-a1n4O+uxBf8e9WYtJv8T4FAw@mail.gmail.com>

It looks like it has been fixed

Thanks to whoever it was
On Oct 28, 2011 11:15 PM, "Beppe Grillo" <beppe.grillo at gmail.com> wrote:

> No, i see
> FreeSWITCH API Documentation.
> Regards,
> Giuseppe.
>
> 2011/10/28 Peter Blackford <peter at onemetric.com>
>
>> Is docs.freeswitch.org returning blank pages for everyone or is it just
>> me?
>>
>> Regards,
>> Peter
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dipkumar.mehta at gmail.com  Fri Oct 28 12:28:29 2011
From: dipkumar.mehta at gmail.com (Dip Mehta)
Date: Fri, 28 Oct 2011 13:58:29 +0530
Subject: [Freeswitch-users] Dialing issues
Message-ID: <CAGL3AgFKfDxi6qHmimhA9DjgahB21fA53WmRbsqJgespBPWFjw@mail.gmail.com>

Hi,

I am trying to integrate Newfies Dialer with Freeswitch.

This is my sofia status

freeswitch at 127.0.0.1@internal> sofia status
                     Name          Type
  Data      State
=================================================================================================
          *:XX.XX.XX.XX*         alias
internal      ALIASED
                 external       profile
sip:mod_sofia@*:XX.XX.XX.XX*:5080
     RUNNING (0)
    external::example.com       gateway
sip:joeuser at example.com      NOREG
 * external::XX.XX.XX.XX       gateway         sip:password at XX.XX.XX.XX
 REGED*
                 internal       profile
sip:mod_sofia at 199.167.145.102:5060      RUNNING (0)
            internal-ipv6       profile
sip:mod_sofia@[::1]:5060
     RUNNING (0)
=================================================================================================
3 profiles 1 alias


Looks like the Service provider is now registered with the Freeswitch. I am
asuming as it says REGED in the status.

However, when i start my campaign through the dialer, I am unable to make
the outgoing calls which results in these errors.

2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:1884 Parsing
global variables
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_request_uuid]=[48df3dbe-013e-11e1-8b92-bc305bec2901]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_hangup_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/hangupcall/]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[origination_caller_id_number]=[8888888888]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[bridge_early_media]=[true]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[hangup_after_bridge]=[true]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_app]=[true]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[absolute_codec_string]=[PCMA]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[originate_timeout]=[1800]
2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
[ignore_early_media]=[true]
2011-10-28 04:24:41.992674 [ERR] switch_core_session.c:426 *Could not locate
channel type* *XX.XX.XX.XX Customers contact number*
2011-10-28 04:24:41.992674 [NOTICE] switch_ivr_originate.c:2459 Cannot
create outgoing channel of type [*XX.XX.XX.XX Customers contact number*]
cause: [CHAN_NOT_IMPLEMENTED]
2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:3348 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:1884 Parsing
global variables
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_request_uuid]=[53a1c460-013e-11e1-8b92-bc305bec2901]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_hangup_url]=[http://*XX.XX.XX.XX*2:9080/api/dialer_cdr/hangupcall/]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[origination_caller_id_number]=[8888888888]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[bridge_early_media]=[true]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[hangup_after_bridge]=[true]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[plivo_app]=[true]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[absolute_codec_string]=[PCMA]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[originate_timeout]=[1800]
2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
[ignore_early_media]=[true]
2011-10-28 04:25:00.052673 [ERR] switch_core_session.c:426 Could not locate
channel type *XX.XX.XX.XX Customers contact number*
2011-10-28 04:25:00.052673 [NOTICE] switch_ivr_originate.c:2459 Cannot
create outgoing channel of type [*XX.XX.XX.XX Customers contact number*]
cause: [CHAN_NOT_IMPLEMENTED]
2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:3348 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]


Please advice.
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From marhbere at hotmail.com  Fri Oct 28 17:52:20 2011
From: marhbere at hotmail.com (MarhBere)
Date: Fri, 28 Oct 2011 06:52:20 -0700 (PDT)
Subject: [Freeswitch-users] switch_rtp.c - Auto Changing Port
Message-ID: <1319809940595-6940220.post@n2.nabble.com>

Hi, I have an issue with extension (eyebeam) behind nat.

Scenario: The call Incoming From ITSP  and is bridged with the a user
registered; The RTP arrive to FS from register user, but don't have the auto
adjust IP/Port. So only had silence in both points. IF the phone Hold the
call and Unhold the call. FS now yet change the RTP to correct private
IP/Port and call is now with bidirectional audio.

I red that is hapened in old release :
http://freeswitch-users.2379917.n2.nabble.com/No-RTP-destination-auto-correction-tt3906730.html#none
But I running the last Head trunk. "FreeSWITCH Version 1.0.head (git-1bf97fa
2011-10-11 15-13-23 -0500)" SO: CentOs 5. 64x

This happens to me a long time and I can not solve it.

Someone know what is wrong?
PD: Sorry for my traduction.

Thanks,
Marcelo Bereterbide. 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/switch-rtp-c-Auto-Changing-Port-tp6940220p6940220.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Oct 28 19:33:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Oct 2011 08:33:19 -0700
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <BAY167-W1790CAFE7260E497A93155ECD30@phx.gbl>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
	<CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>
	<BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>
	<CAKzWOxUCgJbP0XmH-+AXyKfcQX=TCf8cQeh0irPP1Fc__HLcvA@mail.gmail.com>
	<BAY167-W1790CAFE7260E497A93155ECD30@phx.gbl>
Message-ID: <CAKzWOxXD8Qgo0zuxrfPGiGpKwryfx=TZJ+pKXcD4xGPp3t_-qQ@mail.gmail.com>

Basic awk scripting. The $9 means "ninth field" and $14 means "14th" field.
Those are the respective locations of the "billsec" field in the FS and
Asterisk Master.csv files.

-MC

On Fri, Oct 28, 2011 at 12:47 AM, Faisal Rehman <faisal.rehman22 at hotmail.com
> wrote:

>  Hi Micheal,
>
> What would be the difference between $14 and $9, will you please a little
> explain?
>
>
> Thanks
>
> Faisal Rehman
>
> ------------------------------
> Date: Thu, 27 Oct 2011 08:40:12 -0700
> From: msc at freeswitch.org
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
>
>
> On Thu, Oct 27, 2011 at 1:02 AM, Faisal Rehman <
> faisal.rehman22 at hotmail.com> wrote:
>
>  Hi David,
>
> Thanks for sharing the script but will that work too for Asterisk's
> Master.csv if I needed the ASR & ACD records from that file?
>
>
> Use $14 (instead of $9) for Asterisk's Master.csv file.
> -MC
>
>
>
>
> Regards,
>
> Faisal Rehman
>
> ------------------------------
> From: david.villasmil.work at gmail.com
> Date: Wed, 26 Oct 2011 19:48:57 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
> Hello,
>
> That's a faily simple bash script.
>
> "cat" the log file to an awk script like so;
>
> cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{
> FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT:
> "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR;
> if(ASR<100){ system("echo send email");} }'
>
> the "system" is where you put the mail command
>
> also, bear in mind that the master can get VERY large if you don't rotate
> it.
>
> david
>
> On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman <
> faisal.rehman22 at hotmail.com> wrote:
>
>  *Hi Waseem,*
>
> Thanks for your help, I saw the scripts that were on the link that you
> provided to me but actually the problem is that I don't want to involve the
> database though it is a bit difficult task but according to my requirement I
> just want to read the whole Master.csv file & if the ASR or ACD is low on
> any server, it should send me email, so my question is how can I skip the
> database from it to interact directly with the Master.csv file.
>
>
> Thanks in Advance!!
>
> *Faisal Rehman
> *
> ------------------------------
> From: wasim at convergence.pk
> Date: Tue, 25 Oct 2011 07:32:35 +0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>
>
> Faisal:
>
> Load up the CDR into a db (MySQL or otherwise) ...
>
> For ACD you'll want a SQL query akin
>
>    - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
>    SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
>    disposition=1;
>
> For ASR you'll need another query
>
>    - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
>    - ASR is calls_answered/calls_total
>
> There is sample (read: poor) code in
> http://jira.freeswitch.org/browse/FS-1981
> which a google for "freeswitch ACD ASR" would have helped you get to in the
> first place.
>
> -wasim
>
> On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com>wrote:
>
>  Hi Everyone,
>
> I want to write a script that reads the Master.csv & generate an alert if
> the ASR & ACD is low from the desired value. Anybody has written such
> script, please help me write it or provide any sample.
>
>
> Thanks
>
>
> Faisal Rehman
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 28 19:34:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Oct 2011 08:34:28 -0700
Subject: [Freeswitch-users] originate_disposition
In-Reply-To: <C7E1545D96BE42FBB1F8FB344A39857B@e1705>
References: <D16AF8797FA347B3A2E1923AAE592321@e1705>
	<CAKzWOxXXks-WQ42VR12=a2wcsBaMCAp0ox2yW-dbrKc=qre0Rg@mail.gmail.com>
	<C7E1545D96BE42FBB1F8FB344A39857B@e1705>
Message-ID: <CAKzWOxV6pAO+gqF0hNRi_NjPoC-tBOJzfMmsYMvjvt+W4vbhBQ@mail.gmail.com>

I believe so. I've never tried it. Ask Jay Binks (IRC: jaybinks) if you are
not able to figure it out.
-MC

On Thu, Oct 27, 2011 at 11:52 PM, Madovsky <infos at madovsky.org> wrote:

> **
> Great shot MC, thanks a lot.
>
> is this variable can contain more than one cause ?
>
> Thanks
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, October 28, 2011 2:24 AM
> *Subject:* Re: [Freeswitch-users] originate_disposition
>
> How about the trick that Jay Binks added for scenarios not unlike this one?
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
>
> -MC
>
> On Thu, Oct 27, 2011 at 1:33 PM, Madovsky <infos at madovsky.org> wrote:
>
>> **
>> Afer hours of test I didn' succeed how
>> to catch the result of originate_disposition
>>
>> --------
>>
>> I have this extensions in my dialplan
>>
>>          <extension name="internal_test">
>>                 <condition field="${sip_to_uri}"
>> expression="^(999\d{10})@$${domain}$">
>>                         <action inline="true" application="set"
>> data="nibble_account="/>
>>                         <action inline="true" application="set"
>> data="nibble_rate="/>
>>                         <action inline="true" application="set"
>> data="dialed_extension=$1"/>
>>                 </condition>
>>                 <condition field="${from_admin_services}"
>> expression="^true$" break="never">
>>                         <action application="set"
>> data="ignore_early_media=true"/>
>>                         <action application="set"
>> data="hangup_after_bridge=false"/>
>>                         <action application="set"
>> data="continue_on_fail=true"/>
>>                         <action application="bridge"
>> data="{sip_append_audio_sdp='a=rtpmap:0
>> PCMU/8000',sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
>>                 </condition>
>>         </extension>
>>         <extension name="internal_test_tools">
>>                 <condition field="${originate_disposition}"
>> expression="^SUBSCRIBER_ABSENT$" break="on-true">
>>                         <action application="phrase"
>> data="bad_ob_prepaid_dest"/>
>>                         <action application="transfer"
>> data="ob_prepaid_call XML features"/>
>>                         <action application="info"/>
>>                 </condition>
>>                 <condition field="${cond(expr(${originate_disposition} !=
>> NORMAL_CLEARING) ? YES : NO)}" expression="^YES$"/>
>>                 <condition field="${cond(expr(${originate_disposition} !=
>> SUBSCRIBER_ABSENT) ? YES : NO)}" expression="^YES$">
>>                         <action application="ring_ready"/>
>>                         <action application="sleep" data="2000"/>
>>                         <action application="answer"/>
>>                         <action application="sleep" data="1400"/>
>>                         <action application="voicemail" data="default
>> $${domain} ${dialed_extension}"/>
>>                </condition>
>>         </extension>
>>
>> -----------
>>
>>
>> I tried to merge it in one extension without success also.
>> where is the trick ? :)
>>
>> Thanks
>>
>> Franck
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From basit.engg at gmail.com  Fri Oct 28 20:14:03 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Fri, 28 Oct 2011 21:14:03 +0500
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <CAKzWOxXD8Qgo0zuxrfPGiGpKwryfx=TZJ+pKXcD4xGPp3t_-qQ@mail.gmail.com>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
	<CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>
	<BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>
	<CAKzWOxUCgJbP0XmH-+AXyKfcQX=TCf8cQeh0irPP1Fc__HLcvA@mail.gmail.com>
	<BAY167-W1790CAFE7260E497A93155ECD30@phx.gbl>
	<CAKzWOxXD8Qgo0zuxrfPGiGpKwryfx=TZJ+pKXcD4xGPp3t_-qQ@mail.gmail.com>
Message-ID: <CAGQG5Ma=xTw3XQiCwURpbVOtpcF_iTxB6bYDzMZFG4HjjuUgkA@mail.gmail.com>

For asterisk, ASR and ACD is as follows: (just addition) billsec is 14th
field

cat /var/log/asterisk/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{ FS=",";
}{ if($14>0){ CONN=CONN+1; BSEC=BSEC+$14} TOTAL=TOTAL+1; }END{ print
"CONNECT: "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR:
"ASR; print "BILLSEC: "BSEC; print "ACD: "BSEC/CONN; if(ASR<100){
system("echo send email");} }'

here is the result

CONNECT: 1573542
TOTAL: 1584174
ASR: 99.3289
BILLSEC: 448223564
ACD: 284.85
send email


-- 
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445

On Fri, Oct 28, 2011 at 8:33 PM, Michael Collins <msc at freeswitch.org> wrote:

> Basic awk scripting. The $9 means "ninth field" and $14 means "14th" field.
> Those are the respective locations of the "billsec" field in the FS and
> Asterisk Master.csv files.
>
> -MC
>
>
> On Fri, Oct 28, 2011 at 12:47 AM, Faisal Rehman <
> faisal.rehman22 at hotmail.com> wrote:
>
>>  Hi Micheal,
>>
>> What would be the difference between $14 and $9, will you please a little
>> explain?
>>
>>
>> Thanks
>>
>> Faisal Rehman
>>
>> ------------------------------
>> Date: Thu, 27 Oct 2011 08:40:12 -0700
>> From: msc at freeswitch.org
>>
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>>
>>
>>
>> On Thu, Oct 27, 2011 at 1:02 AM, Faisal Rehman <
>> faisal.rehman22 at hotmail.com> wrote:
>>
>>  Hi David,
>>
>> Thanks for sharing the script but will that work too for Asterisk's
>> Master.csv if I needed the ASR & ACD records from that file?
>>
>>
>> Use $14 (instead of $9) for Asterisk's Master.csv file.
>> -MC
>>
>>
>>
>>
>> Regards,
>>
>> Faisal Rehman
>>
>> ------------------------------
>> From: david.villasmil.work at gmail.com
>> Date: Wed, 26 Oct 2011 19:48:57 +0200
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>>
>> Hello,
>>
>> That's a faily simple bash script.
>>
>> "cat" the log file to an awk script like so;
>>
>> cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk
>> '{ FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT:
>> "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR;
>> if(ASR<100){ system("echo send email");} }'
>>
>> the "system" is where you put the mail command
>>
>> also, bear in mind that the master can get VERY large if you don't rotate
>> it.
>>
>> david
>>
>> On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman <
>> faisal.rehman22 at hotmail.com> wrote:
>>
>>  *Hi Waseem,*
>>
>> Thanks for your help, I saw the scripts that were on the link that you
>> provided to me but actually the problem is that I don't want to involve the
>> database though it is a bit difficult task but according to my requirement I
>> just want to read the whole Master.csv file & if the ASR or ACD is low on
>> any server, it should send me email, so my question is how can I skip the
>> database from it to interact directly with the Master.csv file.
>>
>>
>> Thanks in Advance!!
>>
>> *Faisal Rehman
>> *
>> ------------------------------
>> From: wasim at convergence.pk
>> Date: Tue, 25 Oct 2011 07:32:35 +0500
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>>
>>
>> Faisal:
>>
>> Load up the CDR into a db (MySQL or otherwise) ...
>>
>> For ACD you'll want a SQL query akin
>>
>>    - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
>>    SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
>>    disposition=1;
>>
>> For ASR you'll need another query
>>
>>    - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
>>    - ASR is calls_answered/calls_total
>>
>> There is sample (read: poor) code in
>> http://jira.freeswitch.org/browse/FS-1981
>> which a google for "freeswitch ACD ASR" would have helped you get to in
>> the first place.
>>
>> -wasim
>>
>> On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <faisal.rehman22 at hotmail.com
>> > wrote:
>>
>>  Hi Everyone,
>>
>> I want to write a script that reads the Master.csv & generate an alert if
>> the ASR & ACD is low from the desired value. Anybody has written such
>> script, please help me write it or provide any sample.
>>
>>
>> Thanks
>>
>>
>> Faisal Rehman
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070| peace be upon you ...
>>
>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing 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 jmoran at secureachsystems.com  Fri Oct 28 20:19:43 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Fri, 28 Oct 2011 12:19:43 -0400
Subject: [Freeswitch-users] Hangup Causes
Message-ID: <361E98F99D3CC3439EED59BC1924ED69547635@SERVER2003.SecuReachSystems.local>

I have some questions about the types of statuses we are getting (Yes,
I'm already familiar with http://wiki.freeswitch.org/wiki/Hangup_causes
)

 

We just deployed FS to production a couple of days ago and here is a
sample of our Hangup Causes:

CALL_REJECTED                1

NO_ANSWER     738

NO_USER_RESPONSE    175

NORMAL_CLEARING      1667 (Expected that this is only Early Hangups)

NORMAL_TEMPORARY_FAILURE             4

NORMAL_TEMPORARY_FAILURE (Rang)               37 (Rang means this status
happened more than 5 seconds after the session was started)

Complete            224  (Complete is our own status instead of
NORMAL_CLEARING, meaning WE hung up because we finished the entire
message).

ORIGINATOR_CANCEL (Rang)    3

RECOVERY_ON_TIMER_EXPIRE (Rang)   9 

UNALLOCATED_NUMBER            2

USER_BUSY        35

 

My questions:

1.       NORMAL_TEMPORARY_FAILURE happens a lot, and often returns 30 or
more seconds after starting. Could this be a "silent call" (etc)? We
have gotten 3-4 complaints of silent calls and I'm hoping to find the
cause.

2.       Is there anything here (or ideas elsewhere?) to explain why a
call will be silent once in a while?

3.       Is this spread of causes normal or does it seem strange

4.       We've had a few people note that the message on their voicemail
is over before they can hear it - even though we play the message twice
(to make short messages longer). Does anybody know of a solution for
this?

 

Thanks! -Jason

 

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From msc at freeswitch.org  Fri Oct 28 22:10:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Oct 2011 11:10:21 -0700
Subject: [Freeswitch-users] Hangup Causes
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED69547635@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED69547635@SERVER2003.SecuReachSystems.local>
Message-ID: <CAKzWOxU7ewV3WWq=nV59By0a_NNEGrUFfrqHd4PVMEOp=SZDGg@mail.gmail.com>

Although not easy on a production system, your best bet is to catch the
behavior in the act. Usually this means that you have to rely upon humans to
report the symptoms to you in a timely and accurate manner.

One way you can help track these down is to use something like pcapsipdump
and do a ring buffer that covers at least several days. Then, when you hear
about a symptom you can go back and look at the corresponding pcap file
which will have both signaling and audio. From there you can hopefully trace
the root of the problem.

-MC

http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump

On Fri, Oct 28, 2011 at 9:19 AM, Jason Moran <jmoran at secureachsystems.com>wrote:

>  I have some questions about the types of statuses we are getting (Yes,
> I?m already familiar with http://wiki.freeswitch.org/wiki/Hangup_causes )*
> ***
>
> ** **
>
> We just deployed FS to production a couple of days ago and here is a sample
> of our Hangup Causes:****
>
> CALL_REJECTED                1****
>
> NO_ANSWER     738****
>
> NO_USER_RESPONSE    175****
>
> NORMAL_CLEARING      1667 (Expected that this is only Early Hangups)****
>
> NORMAL_TEMPORARY_FAILURE             4****
>
> NORMAL_TEMPORARY_FAILURE (Rang)               37 (Rang means this status
> happened more than 5 seconds after the session was started)****
>
> Complete            224  (Complete is our own status instead of
> NORMAL_CLEARING, meaning WE hung up because we finished the entire message).
> ****
>
> ORIGINATOR_CANCEL (Rang)    3****
>
> RECOVERY_ON_TIMER_EXPIRE (Rang)   9 ****
>
> UNALLOCATED_NUMBER            2****
>
> USER_BUSY        35****
>
> ** **
>
> My questions:****
>
> **1.       **NORMAL_TEMPORARY_FAILURE happens a lot, and often returns 30
> or more seconds after starting. Could this be a ?silent call? (etc)? We have
> gotten 3-4 complaints of silent calls and I?m hoping to find the cause.***
> *
>
> **2.       **Is there anything here (or ideas elsewhere?) to explain why a
> call will be silent once in a while?****
>
> **3.       **Is this spread of causes normal or does it seem strange****
>
> **4.       **We?ve had a few people note that the message on their
> voicemail is over before they can hear it - even though we play the message
> twice (to make short messages longer). Does anybody know of a solution for
> this?****
>
> ** **
>
> Thanks! -Jason****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 28 22:15:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Oct 2011 11:15:13 -0700
Subject: [Freeswitch-users] Dialing issues
In-Reply-To: <CAGL3AgFKfDxi6qHmimhA9DjgahB21fA53WmRbsqJgespBPWFjw@mail.gmail.com>
References: <CAGL3AgFKfDxi6qHmimhA9DjgahB21fA53WmRbsqJgespBPWFjw@mail.gmail.com>
Message-ID: <CAKzWOxVZ3Sikyh6NFTBrf0U62EVmWuUkd7mRHFb7z06gJHgAow@mail.gmail.com>

This is a newfies configuration issue. I've only looked briefly at the
newfies dialer (I don't want to get anywhere near it because it's AGPL -
yuck) and I do know that it requires you to specify your gateway in some
manner. It does not look like you have done that. I'd review this
information from the newfies-dialer.org documentation site:

Newfies-Dialer?s Gateway

Log into the Newfies-Dialer interface using your root username and password
previously created in these instructions.

Click Admin, and locate and enter the ?Dialer Gateways? panel on the
dashboard, then add a dialer gateway. Only the fields in bold need to be
filled in. That is to say, the name of gateway, for identification, and the
Gateway?s field, e.g.

sofia/gateway/myprovider/

?myprovider? is the name of the gateway which you can take from the gateway
name when you typed ?sofia status? when configuring the Freeswitch Gateway.
Finally, check the status is active, and save. The trunk should now appear
in the gateway list.
-MC

On Fri, Oct 28, 2011 at 1:28 AM, Dip Mehta <dipkumar.mehta at gmail.com> wrote:

> Hi,
>
> I am trying to integrate Newfies Dialer with Freeswitch.
>
> This is my sofia status
>
> freeswitch at 127.0.0.1@internal> sofia status
>                      Name          Type
>   Data      State
>
> =================================================================================================
>           *:XX.XX.XX.XX*         alias
> internal      ALIASED
>                  external       profile         sip:mod_sofia@*
> :XX.XX.XX.XX*:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com      NOREG
>  * external::XX.XX.XX.XX       gateway         sip:password at XX.XX.XX.XX
>    REGED*
>                  internal       profile
> sip:mod_sofia at 199.167.145.102:5060      RUNNING (0)
>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060
>      RUNNING (0)
>
> =================================================================================================
> 3 profiles 1 alias
>
>
> Looks like the Service provider is now registered with the Freeswitch. I am
> asuming as it says REGED in the status.
>
> However, when i start my campaign through the dialer, I am unable to make
> the outgoing calls which results in these errors.
>
> 2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_request_uuid]=[48df3dbe-013e-11e1-8b92-bc305bec2901]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_hangup_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/hangupcall/]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [origination_caller_id_number]=[8888888888]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [bridge_early_media]=[true]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [hangup_after_bridge]=[true]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_app]=[true]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [absolute_codec_string]=[PCMA]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [originate_timeout]=[1800]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
> [ignore_early_media]=[true]
> 2011-10-28 04:24:41.992674 [ERR] switch_core_session.c:426 *Could not
> locate channel type* *XX.XX.XX.XX Customers contact number*
> 2011-10-28 04:24:41.992674 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [*XX.XX.XX.XX Customers contact number*]
> cause: [CHAN_NOT_IMPLEMENTED]
> 2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:3348 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_request_uuid]=[53a1c460-013e-11e1-8b92-bc305bec2901]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_hangup_url]=[http://*XX.XX.XX.XX*2:9080/api/dialer_cdr/hangupcall/]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [origination_caller_id_number]=[8888888888]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [bridge_early_media]=[true]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [hangup_after_bridge]=[true]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [plivo_app]=[true]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [absolute_codec_string]=[PCMA]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [originate_timeout]=[1800]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
> [ignore_early_media]=[true]
> 2011-10-28 04:25:00.052673 [ERR] switch_core_session.c:426 Could not locate
> channel type *XX.XX.XX.XX Customers contact number*
> 2011-10-28 04:25:00.052673 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [*XX.XX.XX.XX Customers contact number*]
> cause: [CHAN_NOT_IMPLEMENTED]
> 2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:3348 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>
>
> Please advice.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From john at mobex.biz  Fri Oct 28 22:36:45 2011
From: john at mobex.biz (john at mobex.biz)
Date: Fri, 28 Oct 2011 11:36:45 -0700
Subject: [Freeswitch-users] High load on database server
Message-ID: <20111028113645.c4df8f4d94a0e5285fca477124561e40.12560a1171.wbe@email17.secureserver.net>

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From cvogel at lyonl.com  Fri Oct 28 22:55:46 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 28 Oct 2011 18:55:46 +0000
Subject: [Freeswitch-users] Feedback
Message-ID: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

  *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
  *   FS now runs inside its own namespace
  *   We replace FS_CLI with a PowerShell shell app
  *   Supports SQL Server support via the Native SQL Server Client API
  *   Support for windows clustering (up to 32 node active/active cluster)
  *   Ties more directly into the Win32 API and has less reliance on open source Libs
  *   Replaces OpenSSL sockets with Windows encrypted sockets
  *   Added windows performance monitoring
  *   No longer need to use mod_managed for managed modules
  *   Memory management relies on .Net garbage collection
  *   Added support for Microsoft Speech
  *   VoiceXML 3.0 and SCXML support
  *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
  *   Fixed RTP clock timing issues in virtualized environments
  *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)
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From krice at freeswitch.org  Fri Oct 28 23:19:05 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 28 Oct 2011 14:19:05 -0500
Subject: [Freeswitch-users] Feedback
In-Reply-To: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
Message-ID: <CAD06A59.212EE%krice@freeswitch.org>

So you are forking the project and then asking the rest of us to help you
with that fork?


On 10/28/11 1:55 PM, "Chad Vogel" <cvogel at lyonl.com> wrote:

> Hello,
> 
> I wanted to reach out and get some feedback from everyone because my company
> started a project using FS as base several weeks ago, our goal was to optimize
> and integrate FS more directly into Windows environments. Here is my question,
> we would like to share our changes and where should we host our project?
> Because we understand our build breaks compatibility with the UNIX
> environments and fails to meet the objectives cross platform compatibility,
> however it could contribute greatly and be beneficial to other windows users.
> 
> Here are some of the changes we have made:
> * We converted FS core to C++ and having it compiling to .NET 4.0 CLI
> * FS now runs inside its own namespace
> * We replace FS_CLI with a PowerShell shell app
> * Supports SQL Server support via the Native SQL Server Client API
> * Support for windows clustering (up to 32 node active/active cluster)
> * Ties more directly into the Win32 API and has less reliance on open source
> Libs 
> * Replaces OpenSSL sockets with Windows encrypted sockets
> * Added windows performance monitoring
> * No longer need to use mod_managed for managed modules
> * Memory management relies on .Net garbage collection
> * Added support for Microsoft Speech
> * VoiceXML 3.0 and SCXML support
> * No longer supports JS, LUA, PHP development - Modules can be only developed
> in .NET, C or C++( we feel support for other languages can be added back in
> but falls outside the scope of our objectives at this time, support for SCXML
> should make limit the need for other scripting languages)
> * Fixed RTP clock timing issues in virtualized environments
> * New configuration file format using .Net App Configuration files (kind of
> looks like an IIS config file)
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dipkumar.mehta at gmail.com  Fri Oct 28 22:51:54 2011
From: dipkumar.mehta at gmail.com (Dip Mehta)
Date: Sat, 29 Oct 2011 00:21:54 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 64, Issue 174
In-Reply-To: <mailman.8403.1319827037.29749.freeswitch-users@lists.freeswitch.org>
References: <mailman.8403.1319827037.29749.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAGL3AgG_FfgyaG=+oLxiRp9UZSYVNnOh0anfBfqxn9k2DikGjQ@mail.gmail.com>

Thanks MC. You got it right.

Appreciate your reply.

Cheers !
Dip Mehta

On Sat, Oct 29, 2011 at 12:07 AM, <
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. Re: Hangup Causes (Michael Collins)
>   2. Re: Dialing issues (Michael Collins)
>   3. Re: High load on database server (john at mobex.biz)
>
>
> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 Oct 2011 11:10:21 -0700
> Subject: Re: [Freeswitch-users] Hangup Causes
> Although not easy on a production system, your best bet is to catch the
> behavior in the act. Usually this means that you have to rely upon humans to
> report the symptoms to you in a timely and accurate manner.
>
> One way you can help track these down is to use something like pcapsipdump
> and do a ring buffer that covers at least several days. Then, when you hear
> about a symptom you can go back and look at the corresponding pcap file
> which will have both signaling and audio. From there you can hopefully trace
> the root of the problem.
>
> -MC
>
> http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump
>
> On Fri, Oct 28, 2011 at 9:19 AM, Jason Moran <jmoran at secureachsystems.com>wrote:
>
>>  I have some questions about the types of statuses we are getting (Yes,
>> I?m already familiar with http://wiki.freeswitch.org/wiki/Hangup_causes )
>> ****
>>
>> ** **
>>
>> We just deployed FS to production a couple of days ago and here is a
>> sample of our Hangup Causes:****
>>
>> CALL_REJECTED                1****
>>
>> NO_ANSWER     738****
>>
>> NO_USER_RESPONSE    175****
>>
>> NORMAL_CLEARING      1667 (Expected that this is only Early Hangups)****
>>
>> NORMAL_TEMPORARY_FAILURE             4****
>>
>> NORMAL_TEMPORARY_FAILURE (Rang)               37 (Rang means this status
>> happened more than 5 seconds after the session was started)****
>>
>> Complete            224  (Complete is our own status instead of
>> NORMAL_CLEARING, meaning WE hung up because we finished the entire message).
>> ****
>>
>> ORIGINATOR_CANCEL (Rang)    3****
>>
>> RECOVERY_ON_TIMER_EXPIRE (Rang)   9 ****
>>
>> UNALLOCATED_NUMBER            2****
>>
>> USER_BUSY        35****
>>
>> ** **
>>
>> My questions:****
>>
>> **1.       **NORMAL_TEMPORARY_FAILURE happens a lot, and often returns 30
>> or more seconds after starting. Could this be a ?silent call? (etc)? We have
>> gotten 3-4 complaints of silent calls and I?m hoping to find the cause.**
>> **
>>
>> **2.       **Is there anything here (or ideas elsewhere?) to explain why
>> a call will be silent once in a while?****
>>
>> **3.       **Is this spread of causes normal or does it seem strange****
>>
>> **4.       **We?ve had a few people note that the message on their
>> voicemail is over before they can hear it - even though we play the message
>> twice (to make short messages longer). Does anybody know of a solution for
>> this?****
>>
>> ** **
>>
>> Thanks! -Jason****
>>
>> ** **
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 Oct 2011 11:15:13 -0700
> Subject: Re: [Freeswitch-users] Dialing issues
> This is a newfies configuration issue. I've only looked briefly at the
> newfies dialer (I don't want to get anywhere near it because it's AGPL -
> yuck) and I do know that it requires you to specify your gateway in some
> manner. It does not look like you have done that. I'd review this
> information from the newfies-dialer.org documentation site:
>
> Newfies-Dialer?s Gateway
>
> Log into the Newfies-Dialer interface using your root username and password
> previously created in these instructions.
>
> Click Admin, and locate and enter the ?Dialer Gateways? panel on the
> dashboard, then add a dialer gateway. Only the fields in bold need to be
> filled in. That is to say, the name of gateway, for identification, and the
> Gateway?s field, e.g.
>
> sofia/gateway/myprovider/
>
> ?myprovider? is the name of the gateway which you can take from the gateway
> name when you typed ?sofia status? when configuring the Freeswitch Gateway.
> Finally, check the status is active, and save. The trunk should now appear
> in the gateway list.
> -MC
>
> On Fri, Oct 28, 2011 at 1:28 AM, Dip Mehta <dipkumar.mehta at gmail.com>wrote:
>
>> Hi,
>>
>> I am trying to integrate Newfies Dialer with Freeswitch.
>>
>> This is my sofia status
>>
>> freeswitch at 127.0.0.1@internal> sofia status
>>                      Name          Type
>>     Data      State
>>
>> =================================================================================================
>>           *:XX.XX.XX.XX*         alias
>> internal      ALIASED
>>                  external       profile         sip:mod_sofia@*
>> :XX.XX.XX.XX*:5080      RUNNING (0)
>>     external::example.com       gateway
>> sip:joeuser at example.com      NOREG
>>  * external::XX.XX.XX.XX       gateway         sip:password at XX.XX.XX.XX
>>    REGED*
>>                  internal       profile
>> sip:mod_sofia at 199.167.145.102:5060      RUNNING (0)
>>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060
>>      RUNNING (0)
>>
>> =================================================================================================
>> 3 profiles 1 alias
>>
>>
>> Looks like the Service provider is now registered with the Freeswitch. I
>> am asuming as it says REGED in the status.
>>
>> However, when i start my campaign through the dialer, I am unable to make
>> the outgoing calls which results in these errors.
>>
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:1884 Parsing
>> global variables
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_request_uuid]=[48df3dbe-013e-11e1-8b92-bc305bec2901]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_hangup_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/hangupcall/]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [origination_caller_id_number]=[8888888888]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [bridge_early_media]=[true]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [hangup_after_bridge]=[true]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_app]=[true]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [absolute_codec_string]=[PCMA]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [originate_timeout]=[1800]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable
>> [ignore_early_media]=[true]
>> 2011-10-28 04:24:41.992674 [ERR] switch_core_session.c:426 *Could not
>> locate channel type* *XX.XX.XX.XX Customers contact number*
>> 2011-10-28 04:24:41.992674 [NOTICE] switch_ivr_originate.c:2459 Cannot
>> create outgoing channel of type [*XX.XX.XX.XX Customers contact number*]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:3348 Originate
>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:1884 Parsing
>> global variables
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_request_uuid]=[53a1c460-013e-11e1-8b92-bc305bec2901]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_hangup_url]=[http://*XX.XX.XX.XX*
>> 2:9080/api/dialer_cdr/hangupcall/]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [origination_caller_id_number]=[8888888888]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [bridge_early_media]=[true]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [hangup_after_bridge]=[true]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [plivo_app]=[true]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [absolute_codec_string]=[PCMA]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [originate_timeout]=[1800]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable
>> [ignore_early_media]=[true]
>> 2011-10-28 04:25:00.052673 [ERR] switch_core_session.c:426 Could not
>> locate channel type *XX.XX.XX.XX Customers contact number*
>> 2011-10-28 04:25:00.052673 [NOTICE] switch_ivr_originate.c:2459 Cannot
>> create outgoing channel of type [*XX.XX.XX.XX Customers contact number*]
>> cause: [CHAN_NOT_IMPLEMENTED]
>> 2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:3348 Originate
>> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>>
>>
>> Please advice.
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: <john at mobex.biz>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 Oct 2011 11:36:45 -0700
> Subject: Re: [Freeswitch-users] High load on database server
> I'm probably in over my head, but i could use help.
>
> it appears my Freeswitch went down and came back up (possible from volume)
> when it did a large volume accoutn is getting bolcked 407 messages:
>
> Im getting the following message log:
>
> 2011-10-28 14:24:54.778973 [DEBUG] sofia.c:7099 IP 66.42.61.24 Rejected by
> acl "domains". Falling back to Digest auth.
> 2011-10-28 14:24:54.778973 [WARNING] sofia_reg.c:1400 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [18125362643 at 216.134.222.93] from
> ip 66.42.61.24
>
> That IP address is set up as 66.42.61.24/32 in the cidr.
>
> What do i do to get them back up?
>
> Thanks
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mgg at giagnocavo.net  Fri Oct 28 23:30:28 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 28 Oct 2011 15:30:28 -0400
Subject: [Freeswitch-users] Feedback
Message-ID: <y4nm9j4ia1ey3ad3hvqarbwo.1319830230240@email.android.com>

Wow that looks very impressive. I'd love to see it. Have you considered putting it up on GitHub? That gives you a nice UI, bug tracking, and wiki.


-------- Original message --------
Subject: [Freeswitch-users] Feedback
From: Chad Vogel <cvogel at lyonl.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
CC:



Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

 *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
 *   FS now runs inside its own namespace
 *   We replace FS_CLI with a PowerShell shell app
 *   Supports SQL Server support via the Native SQL Server Client API
 *   Support for windows clustering (up to 32 node active/active cluster)
 *   Ties more directly into the Win32 API and has less reliance on open source Libs
 *   Replaces OpenSSL sockets with Windows encrypted sockets
 *   Added windows performance monitoring
 *   No longer need to use mod_managed for managed modules
 *   Memory management relies on .Net garbage collection
 *   Added support for Microsoft Speech
 *   VoiceXML 3.0 and SCXML support
 *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
 *   Fixed RTP clock timing issues in virtualized environments
 *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)
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From brian at freeswitch.org  Fri Oct 28 23:32:38 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Oct 2011 14:32:38 -0500
Subject: [Freeswitch-users] Feedback
In-Reply-To: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
Message-ID: <54A55543-6B5C-459F-B601-DDDBC02718F9@freeswitch.org>

I would start by posting all your changes back on Jira and let us see what all is done and make an assessment of how to integrate it all... chances are you didn't break as much as you think..... 

/b

On Oct 28, 2011, at 1:55 PM, Chad Vogel wrote:

> Hello,
> 
> I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.
> 
> Here are some of the changes we have made:
> 
>  *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>  *   FS now runs inside its own namespace
>  *   We replace FS_CLI with a PowerShell shell app
>  *   Supports SQL Server support via the Native SQL Server Client API
>  *   Support for windows clustering (up to 32 node active/active cluster)
>  *   Ties more directly into the Win32 API and has less reliance on open source Libs
>  *   Replaces OpenSSL sockets with Windows encrypted sockets
>  *   Added windows performance monitoring
>  *   No longer need to use mod_managed for managed modules
>  *   Memory management relies on .Net garbage collection
>  *   Added support for Microsoft Speech
>  *   VoiceXML 3.0 and SCXML support
>  *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
>  *   Fixed RTP clock timing issues in virtualized environments
>  *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

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From david.villasmil.work at gmail.com  Sat Oct 29 00:05:19 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 28 Oct 2011 22:05:19 +0200
Subject: [Freeswitch-users] Script for ASR & ACD Monitoring
In-Reply-To: <CAGQG5Ma=xTw3XQiCwURpbVOtpcF_iTxB6bYDzMZFG4HjjuUgkA@mail.gmail.com>
References: <BAY167-W96A4783B1202A89F5B6037ECEF0@phx.gbl>
	<CABHdwCotkab_+FhKxtVJfrTaH7XFoNOgz2MBWqxh7K5JiuqMWA@mail.gmail.com>
	<BAY167-W1390B9CD855BF66E30F2F36ECED0@phx.gbl>
	<CAFGRPVpkaRUoaMd7Qn-6awOYFtY1z3=L_ZUEaLXMyUrku-2uGw@mail.gmail.com>
	<BAY167-W616A06C7A176F55C8936AEECD20@phx.gbl>
	<CAKzWOxUCgJbP0XmH-+AXyKfcQX=TCf8cQeh0irPP1Fc__HLcvA@mail.gmail.com>
	<BAY167-W1790CAFE7260E497A93155ECD30@phx.gbl>
	<CAKzWOxXD8Qgo0zuxrfPGiGpKwryfx=TZJ+pKXcD4xGPp3t_-qQ@mail.gmail.com>
	<CAGQG5Ma=xTw3XQiCwURpbVOtpcF_iTxB6bYDzMZFG4HjjuUgkA@mail.gmail.com>
Message-ID: <CAFGRPVpAe+N2ECEniH_LvHEJ3Ndb22YuJuRYD9WOK1k0QEuk7g@mail.gmail.com>

Faisal,

What you should actually do is implement xml_curl to a web server and parse
the cdrs there. The just create a page where you can see all the date and
run a cron job periodically to calculate whether you need to send an alert
or not. This way you will have a historic asr and acd for your traffic.

maybe you can have a look at
http://wiki.freeswitch.org/wiki/Billing#FreeSWITCH_Viking where it points
you the git and you can take and use only what you need to have your
reports...

Hope it helps

David

On Fri, Oct 28, 2011 at 6:14 PM, Abdul Basit <basit.engg at gmail.com> wrote:

> For asterisk, ASR and ACD is as follows: (just addition) billsec is 14th
> field
>
> cat /var/log/asterisk/cdr-csv/Master.csv | sed "s/\\\"//g" | awk '{ FS=",";
> }{ if($14>0){ CONN=CONN+1; BSEC=BSEC+$14} TOTAL=TOTAL+1; }END{ print
> "CONNECT: "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR:
> "ASR; print "BILLSEC: "BSEC; print "ACD: "BSEC/CONN; if(ASR<100){
> system("echo send email");} }'
>
> here is the result
>
> CONNECT: 1573542
> TOTAL: 1584174
> ASR: 99.3289
> BILLSEC: 448223564
> ACD: 284.85
> send email
>
>
> --
> Regards,
>
> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
>
> On Fri, Oct 28, 2011 at 8:33 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Basic awk scripting. The $9 means "ninth field" and $14 means "14th"
>> field. Those are the respective locations of the "billsec" field in the FS
>> and Asterisk Master.csv files.
>>
>> -MC
>>
>>
>> On Fri, Oct 28, 2011 at 12:47 AM, Faisal Rehman <
>> faisal.rehman22 at hotmail.com> wrote:
>>
>>>  Hi Micheal,
>>>
>>> What would be the difference between $14 and $9, will you please a little
>>> explain?
>>>
>>>
>>> Thanks
>>>
>>> Faisal Rehman
>>>
>>> ------------------------------
>>> Date: Thu, 27 Oct 2011 08:40:12 -0700
>>> From: msc at freeswitch.org
>>>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>>>
>>>
>>>
>>> On Thu, Oct 27, 2011 at 1:02 AM, Faisal Rehman <
>>> faisal.rehman22 at hotmail.com> wrote:
>>>
>>>  Hi David,
>>>
>>> Thanks for sharing the script but will that work too for Asterisk's
>>> Master.csv if I needed the ASR & ACD records from that file?
>>>
>>>
>>> Use $14 (instead of $9) for Asterisk's Master.csv file.
>>> -MC
>>>
>>>
>>>
>>>
>>> Regards,
>>>
>>> Faisal Rehman
>>>
>>> ------------------------------
>>> From: david.villasmil.work at gmail.com
>>> Date: Wed, 26 Oct 2011 19:48:57 +0200
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>>>
>>> Hello,
>>>
>>> That's a faily simple bash script.
>>>
>>> "cat" the log file to an awk script like so;
>>>
>>> cat /usr/local/freeswitch/log/cdr-csv/Master.csv | sed "s/\\\"//g" | awk
>>> '{ FS=","; }{ if($9>0){ CONN=CONN+1; } TOTAL=TOTAL+1; }END{ print "CONNECT:
>>> "CONN; print "TOTAL: " TOTAL; ASR=(CONN/TOTAL)*100; print "ASR: "ASR;
>>> if(ASR<100){ system("echo send email");} }'
>>>
>>> the "system" is where you put the mail command
>>>
>>> also, bear in mind that the master can get VERY large if you don't rotate
>>> it.
>>>
>>> david
>>>
>>> On Wed, Oct 26, 2011 at 1:59 PM, Faisal Rehman <
>>> faisal.rehman22 at hotmail.com> wrote:
>>>
>>>  *Hi Waseem,*
>>>
>>> Thanks for your help, I saw the scripts that were on the link that you
>>> provided to me but actually the problem is that I don't want to involve the
>>> database though it is a bit difficult task but according to my requirement I
>>> just want to read the whole Master.csv file & if the ASR or ACD is low on
>>> any server, it should send me email, so my question is how can I skip the
>>> database from it to interact directly with the Master.csv file.
>>>
>>>
>>> Thanks in Advance!!
>>>
>>> *Faisal Rehman
>>> *
>>> ------------------------------
>>> From: wasim at convergence.pk
>>> Date: Tue, 25 Oct 2011 07:32:35 +0500
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Script for ASR & ACD Monitoring
>>>
>>>
>>> Faisal:
>>>
>>> Load up the CDR into a db (MySQL or otherwise) ...
>>>
>>> For ACD you'll want a SQL query akin
>>>
>>>    - SELECT COUNT(*) AS calls_answered, SUM(billusec) AS billusec,
>>>    SUM(billusec)/COUNT(*) AS acd FROM freeswitchdb.freeswitchcdr WHERE
>>>    disposition=1;
>>>
>>> For ASR you'll need another query
>>>
>>>    - SELECT COUNT(*) AS calls_total FROM freeswitchdb.freeswitchcdr;
>>>    - ASR is calls_answered/calls_total
>>>
>>> There is sample (read: poor) code in
>>> http://jira.freeswitch.org/browse/FS-1981
>>> which a google for "freeswitch ACD ASR" would have helped you get to in
>>> the first place.
>>>
>>> -wasim
>>>
>>> On Mon, Oct 24, 2011 at 15:39, Faisal Rehman <
>>> faisal.rehman22 at hotmail.com> wrote:
>>>
>>>  Hi Everyone,
>>>
>>> I want to write a script that reads the Master.csv & generate an alert if
>>> the ASR & ACD is low from the desired value. Anybody has written such
>>> script, please help me write it or provide any sample.
>>>
>>>
>>> Thanks
>>>
>>>
>>> Faisal Rehman
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070| peace be upon you ...
>>>
>>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Sat Oct 29 01:21:17 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 28 Oct 2011 21:21:17 +0000
Subject: [Freeswitch-users] Feedback
In-Reply-To: <CAD06A59.212EE%krice@freeswitch.org>
References: <CAD06A59.212EE%krice@freeswitch.org>
Message-ID: <BA39ED0F-D07C-41A0-AF8F-D263ABA83D40@lyonl.com>

No, we would support the project and help users with issues; we simply would like to share our work back to the FS community as continue development because some of our new core code could be useful in the FS mainline. The best way to think of our branch is it's kind of like trixbox - our goal is to provide an alternative to Microsoft Lync for enterprises. We will be offering for sale a commercial client, features and support services for our distribution while maintaining public and free open source core, basically we are trying to give back to the FS project.

On Oct 28, 2011, at 2:19 PM, Ken Rice wrote:

So you are forking the project and then asking the rest of us to help you with that fork?


On 10/28/11 1:55 PM, "Chad Vogel" <cvogel at lyonl.com<x-msg://119/cvogel at lyonl.com>> wrote:

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

  *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
  *   FS now runs inside its own namespace
  *   We replace FS_CLI with a PowerShell shell app
  *   Supports SQL Server support via the Native SQL Server Client API
  *   Support for windows clustering (up to 32 node active/active cluster)
  *   Ties more directly into the Win32 API and has less reliance on open source Libs
  *   Replaces OpenSSL sockets with Windows encrypted sockets
  *   Added windows performance monitoring
  *   No longer need to use mod_managed for managed modules
  *   Memory management relies on .Net garbage collection
  *   Added support for Microsoft Speech
  *   VoiceXML 3.0 and SCXML support
  *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
  *   Fixed RTP clock timing issues in virtualized environments
  *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

________________________________

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From msc at freeswitch.org  Sat Oct 29 01:28:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Oct 2011 14:28:30 -0700
Subject: [Freeswitch-users] Feedback
In-Reply-To: <BA39ED0F-D07C-41A0-AF8F-D263ABA83D40@lyonl.com>
References: <CAD06A59.212EE%krice@freeswitch.org>
	<BA39ED0F-D07C-41A0-AF8F-D263ABA83D40@lyonl.com>
Message-ID: <CAKzWOxUZOr=LCjrLk8n0b8ZJf7_-nj7B=QprS4K7vqYS_nYh8Q@mail.gmail.com>

Thank you for your willingness to give back your changes. We appreciate you
wanting to share.

Our resident Windows gurus no doubt have some technical questions for you.
Perhaps you could get with them at some point and discuss your project. I
think the biggest question they had was: Why? In other words, by doing all
of this work to make FS play more nicely with Windows, what exactly do you
gain?

Thanks,
Michael

On Fri, Oct 28, 2011 at 2:21 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  No, we would support the project and help users with issues; we simply
> would like to share our work back to the FS community as continue
> development because some of our new core code could be useful in the FS
> mainline. The best way to think of our branch is it's kind of like trixbox -
> our goal is to provide an alternative to Microsoft Lync for enterprises. We
> will be offering for sale a commercial client, features and support services
> for our distribution while maintaining public and free open source core,
> basically we are trying to give back to the FS project.
>
>  On Oct 28, 2011, at 2:19 PM, Ken Rice wrote:
>
>  So you are forking the project and then asking the rest of us to help you
> with that fork?
>
>
> On 10/28/11 1:55 PM, "Chad Vogel" <cvogel at lyonl.com> wrote:
>
>  Hello,
>
> I wanted to reach out and get some feedback from everyone because my
> company started a project using FS as base several weeks ago, our goal was
> to optimize and integrate FS more directly into Windows environments. Here
> is my question, we would like to share our changes and where should we host
> our project? Because we understand our build breaks compatibility with the
> UNIX environments and fails to meet the objectives cross platform
> compatibility, however it could contribute greatly and be beneficial to
> other windows users.
>
> Here are some of the changes we have made:
>
>    - We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>    - FS now runs inside its own namespace
>    - We replace FS_CLI with a PowerShell shell app
>    - Supports SQL Server support via the Native SQL Server Client API
>    - Support for windows clustering (up to 32 node active/active cluster)
>    - Ties more directly into the Win32 API and has less reliance on open
>    source Libs
>    - Replaces OpenSSL sockets with Windows encrypted sockets
>    - Added windows performance monitoring
>    - No longer need to use mod_managed for managed modules
>    - Memory management relies on .Net garbage collection
>    - Added support for Microsoft Speech
>    - VoiceXML 3.0 and SCXML support
>    - No longer supports JS, LUA, PHP development - Modules can be only
>    developed in .NET, C or C++( we feel support for other languages can be
>    added back in but falls outside the scope of our objectives at this time,
>    support for SCXML should make limit the need for other scripting languages)
>    - Fixed RTP clock timing issues in virtualized environments
>    - New configuration file format using .Net App Configuration files
>    (kind of looks like an IIS config file)
>
>
> ------------------------------
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From b_ball_henry at hotmail.com  Sat Oct 29 02:02:38 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 29 Oct 2011 06:02:38 +0800
Subject: [Freeswitch-users] Feedback
In-Reply-To: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
Message-ID: <CAP_G396+VJ7+yD=001xujd_YLKbWmuP+uKuELqABw4KHnzy-MQ@mail.gmail.com>

Only started a few weeks and you have so many things hacked(in good terms)
does make me a little skeptical. And I have the same question as Michael,
why make it work with windows? Isn't the sole purpose is just have a nice
.Net interface? Don't see other advantages making it run production on
windows.

Henry

On Sat, Oct 29, 2011 at 2:55 AM, Chad Vogel <cvogel at lyonl.com> wrote:

>  Hello,
>
> I wanted to reach out and get some feedback from everyone because my
> company started a project using FS as base several weeks ago, our goal was
> to optimize and integrate FS more directly into Windows environments. Here
> is my question, we would like to share our changes and where should we host
> our project? Because we understand our build breaks compatibility with the
> UNIX environments and fails to meet the objectives cross platform
> compatibility, however it could contribute greatly and be beneficial to
> other windows users.
>
> Here are some of the changes we have made:
>
>    - We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>    - FS now runs inside its own namespace
>    - We replace FS_CLI with a PowerShell shell app
>    - Supports SQL Server support via the Native SQL Server Client API
>    - Support for windows clustering (up to 32 node active/active cluster)
>    - Ties more directly into the Win32 API and has less reliance on open
>    source Libs
>    - Replaces OpenSSL sockets with Windows encrypted sockets
>    - Added windows performance monitoring
>    - No longer need to use mod_managed for managed modules
>    - Memory management relies on .Net garbage collection
>    - Added support for Microsoft Speech
>    - VoiceXML 3.0 and SCXML support
>    - No longer supports JS, LUA, PHP development - Modules can be only
>    developed in .NET, C or C++( we feel support for other languages can be
>    added back in but falls outside the scope of our objectives at this time,
>    support for SCXML should make limit the need for other scripting languages)
>    - Fixed RTP clock timing issues in virtualized environments
>    - New configuration file format using .Net App Configuration files
>    (kind of looks like an IIS config file)
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Oct 29 02:43:33 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 28 Oct 2011 17:43:33 -0500
Subject: [Freeswitch-users] Feedback
In-Reply-To: <BA39ED0F-D07C-41A0-AF8F-D263ABA83D40@lyonl.com>
Message-ID: <CAD09A45.213D6%krice@freeswitch.org>

Thanks for clarifying that... And sorry for the such off hand dismissal...
There are so many people that want to go the other way and just take
everything that tony and the team have worked on and not even bother giving
back...

Ken


On 10/28/11 4:21 PM, "Chad Vogel" <cvogel at lyonl.com> wrote:

> No, we would support the project and help users with issues; we simply would
> like to share our work back to the FS community as continue development
> because some of our new core code could be useful in the FS mainline. The best
> way to think of our branch is it's kind of like trixbox - our goal is to
> provide an alternative to Microsoft Lync for enterprises. We will be offering
> for sale a commercial client, features and support services for our
> distribution while maintaining public and free open source core, basically we
> are trying to give back to the FS project.
> 
> On Oct 28, 2011, at 2:19 PM, Ken Rice wrote:
> 
>> Re: [Freeswitch-users] Feedback
>> So you are forking the project and then asking the rest of us to help you
>> with that fork?
>> 
>> 
>> On 10/28/11 1:55 PM, "Chad Vogel" <cvogel at lyonl.com
>> <x-msg://119/cvogel at lyonl.com> > wrote:
>> 
>>> Hello,
>>> 
>>> I wanted to reach out and get some feedback from everyone because my company
>>> started a project using FS as base several weeks ago, our goal was to
>>> optimize and integrate FS more directly into Windows environments. Here is
>>> my question, we would like to share our changes and where should we host our
>>> project? Because we understand our build breaks compatibility with the UNIX
>>> environments and fails to meet the objectives cross platform compatibility,
>>> however it could contribute greatly and be beneficial to other windows
>>> users.
>>> 
>>> Here are some of the changes we have made:
>>> * We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>>> * FS now runs inside its own namespace
>>> * We replace FS_CLI with a PowerShell shell app
>>> * Supports SQL Server support via the Native SQL Server Client API
>>> * Support for windows clustering (up to 32 node active/active cluster)
>>> * Ties more directly into the Win32 API and has less reliance on open source
>>> Libs 
>>> * Replaces OpenSSL sockets with Windows encrypted sockets
>>> * Added windows performance monitoring
>>> * No longer need to use mod_managed for managed modules
>>> * Memory management relies on .Net garbage collection
>>> * Added support for Microsoft Speech
>>> * VoiceXML 3.0 and SCXML support
>>> * No longer supports JS, LUA, PHP development - Modules can be only
>>> developed in .NET, C or C++( we feel support for other languages can be
>>> added back in but falls outside the scope of our objectives at this time,
>>> support for SCXML should make limit the need for other scripting languages)
>>> * Fixed RTP clock timing issues in virtualized environments
>>> * New configuration file format using .Net App Configuration files (kind of
>>> looks like an IIS config file)
>>> 
>>> 
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <x-msg://119/FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From cvogel at lyonl.com  Sat Oct 29 02:58:37 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 28 Oct 2011 22:58:37 +0000
Subject: [Freeswitch-users] Feedback
In-Reply-To: <54A55543-6B5C-459F-B601-DDDBC02718F9@freeswitch.org>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<54A55543-6B5C-459F-B601-DDDBC02718F9@freeswitch.org>
Message-ID: <2D29343D-12CA-4512-89D6-B1A7C81A60AC@lyonl.com>

Brian, I will submit up in the next few days after we fix a small issue we found tonight, hopefully it will be in the Jira first part of next week. BTW - I know when we enabled the CLR and converting to managed code we needed change pointers, replace new with gcnew and we also went header less. The other big code changes are related to not using libs, we are now using .Net regex, XML and SqlClient. Also in our next phase we will be profiling the code, seeing if there are any optimization we can do and adding XML documentation.


On Oct 28, 2011, at 2:32 PM, Brian West wrote:

I would start by posting all your changes back on Jira and let us see what all is done and make an assessment of how to integrate it all... chances are you didn't break as much as you think.....

/b

On Oct 28, 2011, at 1:55 PM, Chad Vogel wrote:

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

 *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
 *   FS now runs inside its own namespace
 *   We replace FS_CLI with a PowerShell shell app
 *   Supports SQL Server support via the Native SQL Server Client API
 *   Support for windows clustering (up to 32 node active/active cluster)
 *   Ties more directly into the Win32 API and has less reliance on open source Libs
 *   Replaces OpenSSL sockets with Windows encrypted sockets
 *   Added windows performance monitoring
 *   No longer need to use mod_managed for managed modules
 *   Memory management relies on .Net garbage collection
 *   Added support for Microsoft Speech
 *   VoiceXML 3.0 and SCXML support
 *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
 *   Fixed RTP clock timing issues in virtualized environments
 *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)


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 cvogel at lyonl.com  Sat Oct 29 05:44:12 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Sat, 29 Oct 2011 01:44:12 +0000
Subject: [Freeswitch-users] Feedback
In-Reply-To: <CAP_G396+VJ7+yD=001xujd_YLKbWmuP+uKuELqABw4KHnzy-MQ@mail.gmail.com>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAP_G396+VJ7+yD=001xujd_YLKbWmuP+uKuELqABw4KHnzy-MQ@mail.gmail.com>
Message-ID: <674FF6D2-3344-4076-850C-7233B2ACFB4D@lyonl.com>

I had a couple of my developers work full time of converting FS for the last several weeks however on the VoiceXML / SCXML took a few months to develop and was part of another project we developed but it was in C++/CLR ? this is really why we converted the FS core. Now being in the CLR it?s really natural to take advantage of the existing namespaces for: regex, xml, configuration and SQLClient, this allows us to shrink the code that needs to be upgraded, a cleaner interface for module development, advantages of garbage collection and being managed code means having a vaster library of development options. Also we didn?t bother converting many modules with our build; we only have mod_sofia, mod_loopback, mod_spandsp and mod_dialplan_xml. We will be developing our proprietary modules and client - licensing them to enterprises.

On Oct 28, 2011, at 5:02 PM, Henry Huang wrote:

Only started a few weeks and you have so many things hacked(in good terms) does make me a little skeptical. And I have the same question as Michael, why make it work with windows? Isn't the sole purpose is just have a nice .Net interface? Don't see other advantages making it run production on windows.

Henry

On Sat, Oct 29, 2011 at 2:55 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

  *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
  *   FS now runs inside its own namespace
  *   We replace FS_CLI with a PowerShell shell app
  *   Supports SQL Server support via the Native SQL Server Client API
  *   Support for windows clustering (up to 32 node active/active cluster)
  *   Ties more directly into the Win32 API and has less reliance on open source Libs
  *   Replaces OpenSSL sockets with Windows encrypted sockets
  *   Added windows performance monitoring
  *   No longer need to use mod_managed for managed modules
  *   Memory management relies on .Net garbage collection
  *   Added support for Microsoft Speech
  *   VoiceXML 3.0 and SCXML support
  *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
  *   Fixed RTP clock timing issues in virtualized environments
  *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)


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 mgg at giagnocavo.net  Sat Oct 29 06:13:14 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 28 Oct 2011 22:13:14 -0400
Subject: [Freeswitch-users] Feedback
In-Reply-To: <674FF6D2-3344-4076-850C-7233B2ACFB4D@lyonl.com>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAP_G396+VJ7+yD=001xujd_YLKbWmuP+uKuELqABw4KHnzy-MQ@mail.gmail.com>
	<674FF6D2-3344-4076-850C-7233B2ACFB4D@lyonl.com>
Message-ID: <03351FCC6082174C8534AB714B8258A5E4D7DCC4@mse17be1.mse17.exchange.ms>

What's the plan for making sure critical (security) changes get patched into this version?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Vogel
Sent: Friday, October 28, 2011 7:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Feedback

I had a couple of my developers work full time of converting FS for the last several weeks however on the VoiceXML / SCXML took a few months to develop and was part of another project we developed but it was in C++/CLR - this is really why we converted the FS core. Now being in the CLR it's really natural to take advantage of the existing namespaces for: regex, xml, configuration and SQLClient, this allows us to shrink the code that needs to be upgraded, a cleaner interface for module development, advantages of garbage collection and being managed code means having a vaster library of development options. Also we didn't bother converting many modules with our build; we only have mod_sofia, mod_loopback, mod_spandsp and mod_dialplan_xml. We will be developing our proprietary modules and client - licensing them to enterprises.

On Oct 28, 2011, at 5:02 PM, Henry Huang wrote:


Only started a few weeks and you have so many things hacked(in good terms) does make me a little skeptical. And I have the same question as Michael, why make it work with windows? Isn't the sole purpose is just have a nice .Net interface? Don't see other advantages making it run production on windows.

Henry
On Sat, Oct 29, 2011 at 2:55 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

 *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
 *   FS now runs inside its own namespace
 *   We replace FS_CLI with a PowerShell shell app
 *   Supports SQL Server support via the Native SQL Server Client API
 *   Support for windows clustering (up to 32 node active/active cluster)
 *   Ties more directly into the Win32 API and has less reliance on open source Libs
 *   Replaces OpenSSL sockets with Windows encrypted sockets
 *   Added windows performance monitoring
 *   No longer need to use mod_managed for managed modules
 *   Memory management relies on .Net garbage collection
 *   Added support for Microsoft Speech
 *   VoiceXML 3.0 and SCXML support
 *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
 *   Fixed RTP clock timing issues in virtualized environments
 *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)


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 fieldpeak at gmail.com  Sat Oct 29 06:51:21 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 29 Oct 2011 10:51:21 +0800
Subject: [Freeswitch-users] Log level of log file by fs_cli
In-Reply-To: <76208DFD-604B-4C02-AD0B-4A6C36425134@gmail.com>
References: <CAOTmbMwj8qnJZYGDFQeWQ83jyhwRXOQwj88g2UjGLDkKnr4Xdw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2B77504@cooper>
	<76208DFD-604B-4C02-AD0B-4A6C36425134@gmail.com>
Message-ID: <CAOTmbMwQYK5z4CZqZ-srEy979+gyXpO47X8PwR+FBTDekMU5jA@mail.gmail.com>

Hi Steven,

i tried below cmd, and then make a call, inside the the log file,
there is still 'DEBUG' level message logged. very strange...

freeswitch at internal> fsctl loglevel alert
+OK log level: ALERT [1]

freeswitch at internal>

inside freeswitch.log
...

2011-10-29 10:44:02.496776 [DEBUG] switch_core_state_machine.c:464
(sofia/internal/13580358068 at 124.193.106.104) Callstate Change HANGUP
-> DOWN



2011/10/28, Steven Ayre <steveayre at gmail.com>:
> Peter, that affects only the logging level of your fs_cli session - not the
> log file.
>
> Fran, use the fsctl command from the cli. It's in mod_commands and
> documented on the wiki.
>
> Eg:
> fsctl loglevel
> - shows current level
> fsctl loglevel debug
> - set to debug
>
> Steve on iPhone
>
> On 28 Oct 2011, at 11:09, Peter Olsson <peter.olsson at visionutveckling.se>
> wrote:
>
>> Just enter ?/log [loglevel]? inside fs_cli. I think there is a startup
>> parameter as well.
>>
>> /Peter
>>
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
>> Skickat: den 28 oktober 2011 10:50
>> Till: FreeSWITCH-users
>> ?mne: [Freeswitch-users] Log level of log file by fs_cli
>>
>> Hi friends,
>>
>> i know we can set the loglevel inside switch.conf.xml to change the log
>> level of log files,
>>
>> however, i would like to change the log level of log files thru fs_cli on
>> the fly(no need restart FS), is there any commands to support it, thanks.
>>
>> --
>> Regards,
>> Charles
>>
>> !DSPAM:4eaa6d4932761466214554!
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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,
Charles


From anthony.minessale at gmail.com  Sat Oct 29 07:19:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Oct 2011 22:19:09 -0500
Subject: [Freeswitch-users] Feedback
In-Reply-To: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
Message-ID: <CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>

One thing that is disturbing is that after all the time and effort I put
into making FreeSWITCH cross platform, the very idea of such an initiative
wipes all that out in favor of one platform.  So even if you are willing to
share it, and it's not B.S. its bothersome to me that:

1) I spent 6 years making it work everywhere and you aim to put all your
extended efforts into steering it to a separate version
2) Making it use less open source libs is not really a good thing since its
an open source project.
3) You did not propose anything to us first, that's a bit disrespectful.

I think about everybody who uses FreeSWITCH when I make decisions, and this
just feels selfish to me, (probably not for windows users)  but hey....

On the other hand, Henry has a great point, that looks like a pretty big
list of changes to do in a few weeks considering you were only asking
trivial newbie questions as recently as sept 18th and your website is the
generic IIS document root.   I would be pretty concerned about vishing if I
were a reader of this post.

I guess seeing is believing and if you are legit our windows-friendly
developers will let me know.







On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  Hello,
>
> I wanted to reach out and get some feedback from everyone because my
> company started a project using FS as base several weeks ago, our goal was
> to optimize and integrate FS more directly into Windows environments. Here
> is my question, we would like to share our changes and where should we host
> our project? Because we understand our build breaks compatibility with the
> UNIX environments and fails to meet the objectives cross platform
> compatibility, however it could contribute greatly and be beneficial to
> other windows users.
>
> Here are some of the changes we have made:
>
>    - We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>    - FS now runs inside its own namespace
>    - We replace FS_CLI with a PowerShell shell app
>    - Supports SQL Server support via the Native SQL Server Client API
>    - Support for windows clustering (up to 32 node active/active cluster)
>    - Ties more directly into the Win32 API and has less reliance on open
>    source Libs
>    - Replaces OpenSSL sockets with Windows encrypted sockets
>    - Added windows performance monitoring
>    - No longer need to use mod_managed for managed modules
>    - Memory management relies on .Net garbage collection
>    - Added support for Microsoft Speech
>    - VoiceXML 3.0 and SCXML support
>    - No longer supports JS, LUA, PHP development - Modules can be only
>    developed in .NET, C or C++( we feel support for other languages can be
>    added back in but falls outside the scope of our objectives at this time,
>    support for SCXML should make limit the need for other scripting languages)
>    - Fixed RTP clock timing issues in virtualized environments
>    - New configuration file format using .Net App Configuration files
>    (kind of looks like an IIS config file)
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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 trever.adams at gmail.com  Sat Oct 29 10:39:08 2011
From: trever.adams at gmail.com (Trever L. Adams)
Date: Sat, 29 Oct 2011 00:39:08 -0600
Subject: [Freeswitch-users] FreeTDM woes
In-Reply-To: <4EAA5EA1.2010008@gmail.com>
References: <4EAA5EA1.2010008@gmail.com>
Message-ID: <4EAB9F8C.9050902@gmail.com>

Hello everyone. I know, poor form to reply to my own message, but some
of the problems I have solved.

On 10/28/2011 01:49 AM, Trever L. Adams wrote:
> I have a span of 4 FXS modules. They are all configured to use context
> Internal-FXS. Channel 5 will dial. All the others won't even stop
> sending the dialtone when one attempts to dial.
>
> Channel 5 can call channels 1-3 (there is no 4 at the moment, I have
> tried it where 4 isn't used and 5 becomes 4, same problem), but if 5
> hangs up, the called channel continues to ring.
>
> zt.conf has echo_cancellation_level at 16. Yet, I constantly get
> messages like: Disable echo cancel for 1:5
This was caused by me not having tones.conf in the right place. All of
this is fixed now. Except that the called channel still continues to
ring. It often looks like this:
2011-10-28 16:42:03.627167 [INFO] mod_dialplan_xml.c:336 Processing 
<0000000000>->1005 in context Internal-FXS
2011-10-28 16:42:03.627167 [INFO] ftmod_zt.c:655 Setting echo cancel to
32 taps for 1:5
2011-10-28 16:42:03.627167 [NOTICE] switch_channel.c:915 New Channel
FreeTDM/1:5/ [0e792586-01b6-11e1-b8e9-d3b08c8655be]
2011-10-28 16:42:03.627167 [INFO] ftmod_zt.c:655 Setting echo cancel to
32 taps for 1:5
2011-10-28 16:42:03.627167 [NOTICE] mod_freetdm.c:2079 Ring-Ready
FreeTDM/1:5/!
2011-10-28 16:42:04.147140 [NOTICE] mod_freetdm.c:1069 Ring-Ready
FreeTDM/1:3/1005!
2011-10-28 16:42:04.147140 [NOTICE] switch_ivr_originate.c:483 Ring
Ready FreeTDM/1:3/1005!
2011-10-28 16:42:05.627153 [ERR] ftmod_zt.c:1188 [s1c5][1:5] Failed to
process event from channel
2011-10-28 16:42:09.627161 [ERR] ftmod_zt.c:1188 [s1c5][1:5] Failed to
process event from channel
2011-10-28 16:42:11.627143 [ERR] ftmod_zt.c:1188 [s1c5][1:5] Failed to
process event from channel
2011-10-28 16:42:15.627170 [ERR] ftmod_zt.c:1188 [s1c5][1:5] Failed to
process event from channel
2011-10-28 16:42:17.627157 [ERR] ftmod_zt.c:1188 [s1c5][1:5] Failed to
process event from channel
2011-10-28 16:42:20.687138 [NOTICE] mod_freetdm.c:2157 Hangup
FreeTDM/1:3/1005 [CS_EXECUTE] [NORMAL_CLEARING]
2011-10-28 16:42:20.687138 [NOTICE] switch_ivr_originate.c:3167 Hangup
FreeTDM/1:5/ [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]

Freeswitch seems to think it is hanging up both legs, but the one
continues to ring. The dahdi module I am using is opvxa1200. This is an
8 port analog card from OpenVox. 2 ports are FXO 6 are FXS. The dahdi
package I am using is from openvox:
openvox_dahdi-linux-complete-2.5.0.1+2.5.0.1.tar.gz. I am not sure if
the bug is in FreeSwitch or the openvox/dahdi stuff.
>
> I get messages like this continually: [ERR] ftmod_zt.c:1188 [s1c2][1:2]
> Failed to process event from channel
I am still getting too many of the above.
>
> Finally, with the help of someone on this list I now have my call
> screening setup to work for the appropriate caller id names and numbers
> when it has been blocked. The problem is, the screening part completely
> fails.
>
> These are the actions from the file, which are only slightly modified
> from the example at
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_13:_Call_Screening
> . It asks for your name, it records. But it doesn't give ANYTHING to the
> dialed party.
>
>             <action application="set"
> data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
>             <action application="answer"/>
>             <action application="sleep" data="1000"/>
>             <action application="phrase" data="voicemail_record_name"/>
>             <action application="playback" data="tone_stream://%(500, 0,
> 640)"/>
>             <action application="set"
> data="playback_terminators=#*0123456789"/>
>             <action application="record" data="${call_screen_filename} 7
> 200 2"/>
>             <action application="set" data="group_confirm_key=1"/>
>             <action application="set" data="fail_on_single_reject=true"/>
>             <action application="set"
> data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/>
>             <action application="set" data="continue_on_fail=true"/>
>              <action application="bridge"
> data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6"/>
>             <action application="voicemail" data="default $${domain}
> ${dialed_number}"/>
>             <action application="hangup"/>
>
> I don't think the problem is the phrase:screen_confirm as I have tried
> removing it. I do set dialed_number to my phone number at the top of
> this extension so it is set.
>
> It should be noted that all calls to this FXO port are routed as the
> action above shows. The non-blocked-caller_id route has the exact same
> bridge line.
>
> Am I just plain missing something about the FreeTDM module? Is it this
> broken?
>
> Any help would be greatly appreciated.
>
> Trever
I haven't yet tried the call screening now that things are working MUCH
better. I will be trying that in the next 72 hours. If anyone can see
any obvious mistakes, I would love to know before hand.

Thank you,
Trever

P.S. FreeSwitch is really neat. I am loving it. I just have these few
problems.
-- 
"God is real, unless declared integer." -- Unknown

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From peter.olsson at visionutveckling.se  Sat Oct 29 11:57:25 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 29 Oct 2011 09:57:25 +0200
Subject: [Freeswitch-users] Feedback
In-Reply-To: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF199@cooper>

I'm one of those "Windows developers" out there, and I'm especially interested in how you solved "Fixed RTP clock timing issues in virtualized environments".

I guess we'll know more when you start sharing your code changes..

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Chad Vogel [cvogel at lyonl.com]
Skickat: den 28 oktober 2011 20:55
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Feedback

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

 *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
 *   FS now runs inside its own namespace
 *   We replace FS_CLI with a PowerShell shell app
 *   Supports SQL Server support via the Native SQL Server Client API
 *   Support for windows clustering (up to 32 node active/active cluster)
 *   Ties more directly into the Win32 API and has less reliance on open source Libs
 *   Replaces OpenSSL sockets with Windows encrypted sockets
 *   Added windows performance monitoring
 *   No longer need to use mod_managed for managed modules
 *   Memory management relies on .Net garbage collection
 *   Added support for Microsoft Speech
 *   VoiceXML 3.0 and SCXML support
 *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
 *   Fixed RTP clock timing issues in virtualized environments
 *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

!DSPAM:4eaafaaa32765825219108!


From infos at madovsky.org  Sat Oct 29 12:12:26 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Oct 2011 04:12:26 -0400
Subject: [Freeswitch-users] FreeTDM woes
References: <4EAA5EA1.2010008@gmail.com> <4EAB9F8C.9050902@gmail.com>
Message-ID: <E0847CA807A549A088052E3F6201B132@e1705>

Hi Trever,

post it to jira.freeswitch.org

thanks
----- Original Message ----- 
From: "Trever L. Adams" <trever.adams at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 29, 2011 2:39 AM
Subject: Re: [Freeswitch-users] FreeTDM woes


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


From infos at madovsky.org  Sat Oct 29 12:41:13 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Oct 2011 04:41:13 -0400
Subject: [Freeswitch-users] transfer_ringback and ivr
Message-ID: <163ABD92D7C3445A8E5674DC781B5D80@e1705>

I didnt' succeed to make transfer_ringback work
after an ivr and play_and_get digits and before transfer app.
must I set something else or is it the wrong variables to create
a ringback if a-leg answered already ?

Thanks

Franck
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From cvogel at lyonl.com  Sat Oct 29 21:58:44 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Sat, 29 Oct 2011 17:58:44 +0000
Subject: [Freeswitch-users] Feedback
In-Reply-To: <CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
Message-ID: <710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>

I'm thinking people are missing we are willing to share some of our code that we are developing back to the community that we developed for ourselves. We are not trying to change FS to a windows only platform. From the comments being posted I can see that my plans of sharing the code that my developers worked on isn't being well received and freaky I don't understand why - The community is always asking for people to share their code and I'm willing to share our code so others can use the VoiceXML/SCXML code to integrate into the main line also there may be some befits to our other changes for the main project.

As for the comment it being disrespectful developing our own branch for our business, there has been countless other companies that have done the same thing on other open source projects - for example Facebook has contributed code to other project in the same way and they did what is the best interest for their own needs - even this has led to benefits to open source projects. Also we are a for profit company and we did the changes that are best for our overall business and shrinking over all foot print of project was a befit to us so we didn't need to continue managing/converting code that is native to .Net and for converting to the CRL it business decision to decrees code of our future development - also in the current market it's easier to find good .Net developers then a good C++ developers, in about weeks' time we can have a C# developer writing C++/CLR.

Understand I'm the owner of the company and I spent my money on developing this code, for our use. I invested/paid for over 480 development hours on the rewrite FS to CLR and integration of our VoiceXML/SCXML code, my intentions are to make an offering to compete with Microsoft Lync that is open source but not open to contributions, it will patched and supported by my team - like trixbox. Our core will remain FREE but our modules will be licensed!  With this said I feel we have obligation to offer some of our source back to this project and help with integration if parts are chosen to be included into the FS base. Also to get a better understanding on what my people have been working on and to make our plans for our future I tried to install and learn about FS without help of my developers. Frankly I personally didn't do the development of our FS CLR rewrite, I only helped developed the base code that we used to add VoiceXML/SCXML to FS - the VoiceXML/SCXML code was from our own switch application but I discontinued the project in favor to using FS as a base.

I originally asked where should we post our project so we could make convenient for the community access our changes and I only posted the details so others could see that our changes aren't appropriate for the main line and see they may have some benefit. I'm starting to feel it maybe be the best option to release the code from our site like trixbox is doing with their asterisk code. Again, I'm not trying to impact FS in a negative way, I'm try to give back some of what we developed.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Friday, October 28, 2011 10:19 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Feedback

One thing that is disturbing is that after all the time and effort I put into making FreeSWITCH cross platform, the very idea of such an initiative wipes all that out in favor of one platform.  So even if you are willing to share it, and it's not B.S. its bothersome to me that:

1) I spent 6 years making it work everywhere and you aim to put all your extended efforts into steering it to a separate version
2) Making it use less open source libs is not really a good thing since its an open source project.
3) You did not propose anything to us first, that's a bit disrespectful.

I think about everybody who uses FreeSWITCH when I make decisions, and this just feels selfish to me, (probably not for windows users)  but hey....

On the other hand, Henry has a great point, that looks like a pretty big list of changes to do in a few weeks considering you were only asking trivial newbie questions as recently as sept 18th and your website is the generic IIS document root.   I would be pretty concerned about vishing if I were a reader of this post.

I guess seeing is believing and if you are legit our windows-friendly developers will let me know.







On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

  *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
  *   FS now runs inside its own namespace
  *   We replace FS_CLI with a PowerShell shell app
  *   Supports SQL Server support via the Native SQL Server Client API
  *   Support for windows clustering (up to 32 node active/active cluster)
  *   Ties more directly into the Win32 API and has less reliance on open source Libs
  *   Replaces OpenSSL sockets with Windows encrypted sockets
  *   Added windows performance monitoring
  *   No longer need to use mod_managed for managed modules
  *   Memory management relies on .Net garbage collection
  *   Added support for Microsoft Speech
  *   VoiceXML 3.0 and SCXML support
  *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
  *   Fixed RTP clock timing issues in virtualized environments
  *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)


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<tel:%2B19193869900>
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From avi at avimarcus.net  Sat Oct 29 22:25:48 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 29 Oct 2011 20:25:48 +0200
Subject: [Freeswitch-users] Feedback
In-Reply-To: <710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
Message-ID: <CALNZuwm-twgyRJmg2rKQV93JmgVsa5LVPg-u0uJGaO+YACS=PA@mail.gmail.com>

I think Anthony Minessale's main point - since he is the creator of FS - is
that he spent a lot of effort making FS cross-platform, and as such, it
would seem the bulk of your effort was directly counter to this intention.
(The same for using open source libs.)

It seems most of the community runs on *nix of some sort, but the folks
that do run on windows do report some very high performance.. I don't know
C, C++, C#/.net to really understand why you made the decision you did to
make these changes, but I believe that was the point of ESL and all the
mod_managed, mod_python, mod_lua, etc were made - to let you use nearly any
language to control what FS is doing, while enabling the core of FS to run
in native C to benefit from a code-base that be the same for any platform.
That's why Michael asked "What did you gain by doing this work?"

The community welcomes.. no, begs, for contributions but the the scale of
pushing into native windows code seems counter to the project's goals of
being universal (while maintaining performance and program-ability), hence
Anthony feeling bothered..

-Avi


On Sat, Oct 29, 2011 at 7:58 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  I?m thinking people are missing we are willing to share some of our code
> that we are developing back to the community that we developed for
> ourselves. We are not trying to change FS to a windows only platform. From
> the comments being posted I can see that my plans of sharing the code that
> my developers worked on isn?t being well received and freaky I don?t
> understand why ? The community is always asking for people to share their
> code and I?m willing to share our code so others can use the VoiceXML/SCXML
> code to integrate into the main line also there may be some befits to our
> other changes for the main project.****
>
> ** **
>
> As for the comment it being disrespectful developing our own branch for
> our business, there has been countless other companies that have done the
> same thing on other open source projects ? for example Facebook has
> contributed code to other project in the same way and they did what is the
> best interest for their own needs ? even this has led to benefits to open
> source projects. Also we are a for profit company and we did the changes
> that are best for our overall business and shrinking over all foot print of
> project was a befit to us so we didn?t need to continue managing/converting
> code that is native to .Net and for converting to the CRL it business
> decision to decrees code of our future development ? also in the current
> market it?s easier to find good .Net developers then a good C++ developers,
> in about weeks? time we can have a C# developer writing C++/CLR.****
>
> ** **
>
> Understand I?m the owner of the company and I spent my money on developing
> this code, for our use. I invested/paid for over 480 development hours on
> the rewrite FS to CLR and integration of our VoiceXML/SCXML code, my
> intentions are to make an offering to compete with Microsoft Lync that is
> open source but not open to contributions, it will patched and supported by
> my team ? like trixbox. Our core will remain FREE but our modules will be
> licensed!  With this said I feel we have obligation to offer some of our
> source back to this project and help with integration if parts are chosen
> to be included into the FS base. Also to get a better understanding on what
> my people have been working on and to make our plans for our future I tried
> to install and learn about FS without help of my developers. Frankly I
> personally didn?t do the development of our FS CLR rewrite, I only helped
> developed the base code that we used to add VoiceXML/SCXML to FS ? the
> VoiceXML/SCXML code was from our own switch application but I discontinued
> the project in favor to using FS as a base.****
>
> ** **
>
> I originally asked where should we post our project so we could make
> convenient for the community access our changes and I only posted the
> details so others could see that our changes aren?t appropriate for the
> main line and see they may have some benefit. I?m starting to feel it maybe
> be the best option to release the code from our site like trixbox is doing
> with their asterisk code. Again, I?m not trying to impact FS in a negative
> way, I?m try to give back some of what we developed.****
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Friday, October 28, 2011 10:19 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Feedback****
>
> ** **
>
> One thing that is disturbing is that after all the time and effort I put
> into making FreeSWITCH cross platform, the very idea of such an initiative
> wipes all that out in favor of one platform.  So even if you are willing to
> share it, and it's not B.S. its bothersome to me that:****
>
> ** **
>
> 1) I spent 6 years making it work everywhere and you aim to put all your
> extended efforts into steering it to a separate version ****
>
> 2) Making it use less open source libs is not really a good thing since
> its an open source project.****
>
> 3) You did not propose anything to us first, that's a bit disrespectful.**
> **
>
> ** **
>
> I think about everybody who uses FreeSWITCH when I make decisions, and
> this just feels selfish to me, (probably not for windows users)  but hey....
> ****
>
> ** **
>
> On the other hand, Henry has a great point, that looks like a pretty big
> list of changes to do in a few weeks considering you were only asking
> trivial newbie questions as recently as sept 18th and your website is the
> generic IIS document root.   I would be pretty concerned about vishing if I
> were a reader of this post.****
>
> ** **
>
> I guess seeing is believing and if you are legit our windows-friendly
> developers will let me know. ****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel <cvogel at lyonl.com> wrote:****
>
> Hello,****
>
> I wanted to reach out and get some feedback from everyone because my
> company started a project using FS as base several weeks ago, our goal was
> to optimize and integrate FS more directly into Windows environments. Here
> is my question, we would like to share our changes and where should we host
> our project? Because we understand our build breaks compatibility with the
> UNIX environments and fails to meet the objectives cross platform
> compatibility, however it could contribute greatly and be beneficial to
> other windows users.****
>
> Here are some of the changes we have made:****
>
>    - We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>    ****
>    - FS now runs inside its own namespace ****
>    - We replace FS_CLI with a PowerShell shell app ****
>    - Supports SQL Server support via the Native SQL Server Client API   **
>    **
>    - Support for windows clustering (up to 32 node active/active cluster)
>    ****
>    - Ties more directly into the Win32 API and has less reliance on open
>    source Libs ****
>    - Replaces OpenSSL sockets with Windows encrypted sockets ****
>    - Added windows performance monitoring ****
>    - No longer need to use mod_managed for managed modules ****
>    - Memory management relies on .Net garbage collection ****
>    - Added support for Microsoft Speech ****
>    - VoiceXML 3.0 and SCXML support****
>    - No longer supports JS, LUA, PHP development - Modules can be only
>    developed in .NET, C or C++( we feel support for other languages can be
>    added back in but falls outside the scope of our objectives at this time,
>    support for SCXML should make limit the need for other scripting languages)
>    ****
>    - Fixed RTP clock timing issues in virtualized environments ****
>    - New configuration file format using .Net App Configuration files
>    (kind of looks like an IIS config file) ****
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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 infos at madovsky.org  Sat Oct 29 22:27:06 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Oct 2011 14:27:06 -0400
Subject: [Freeswitch-users] transfer_ringback and ivr
Message-ID: <14D5AB918F454A02A5865410A88728A0@e1705>

I found a way to replace transfer_ringback
and produce a non blocking tone in case of a-leg answered already
(like after an ivr option)

  <action application="displace_session" data="$${sounds_dir}/extras/ringback.wav m"/>
  <action application="set" data="send_silence_when_idle=1400"/>
  if you  wish to use sleep before answer or bridge you can use it, slience_when_idle is done to send RTP packets
and let the a-leg to continue to hear the tone
  <action application="sleep" data="2000"/>

unfortunately displace_session doesn't have any stop option, so it needs to use uuid_displace
<action application="set" data="api_res=${uuid_displace ${uuid} stop $${sounds_dir}/extras/ringback.wav}"/>

or use it with execute_on_answer for example if you don't when the b-leg will answer...

Hope it helps

Franck
  


  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, October 29, 2011 4:41 AM
  Subject: transfer_ringback and ivr


  I didnt' succeed to make transfer_ringback work
  after an ivr and play_and_get digits and before transfer app.
  must I set something else or is it the wrong variables to create
  a ringback if a-leg answered already ?

  Thanks

  Franck
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From infos at madovsky.org  Sat Oct 29 22:36:41 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Oct 2011 14:36:41 -0400
Subject: [Freeswitch-users] Feedback
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM><CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com><710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
	<CALNZuwm-twgyRJmg2rKQV93JmgVsa5LVPg-u0uJGaO+YACS=PA@mail.gmail.com>
Message-ID: <A4EF10E82B304B4C90D0737C82A28BB8@e1705>

My opinion is, with this kind of email, it even doesn't need to answer,
if this guy was a bit serious he will contact directly FS team, not a FS user list.
it can considered as spam...
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Saturday, October 29, 2011 2:25 PM
  Subject: Re: [Freeswitch-users] Feedback


  I think Anthony Minessale's main point - since he is the creator of FS - is that he spent a lot of effort making FS cross-platform, and as such, it would seem the bulk of your effort was directly counter to this intention. (The same for using open source libs.)


  It seems most of the community runs on *nix of some sort, but the folks that do run on windows do report some very high performance.. I don't know C, C++, C#/.net to really understand why you made the decision you did to make these changes, but I believe that was the point of ESL and all the mod_managed, mod_python, mod_lua, etc were made - to let you use nearly any language to control what FS is doing, while enabling the core of FS to run in native C to benefit from a code-base that be the same for any platform.
  That's why Michael asked "What did you gain by doing this work?"


  The community welcomes.. no, begs, for contributions but the the scale of pushing into native windows code seems counter to the project's goals of being universal (while maintaining performance and program-ability), hence Anthony feeling bothered..


  -Avi



  On Sat, Oct 29, 2011 at 7:58 PM, Chad Vogel <cvogel at lyonl.com> wrote:

    I?m thinking people are missing we are willing to share some of our code that we are developing back to the community that we developed for ourselves. We are not trying to change FS to a windows only platform. From the comments being posted I can see that my plans of sharing the code that my developers worked on isn?t being well received and freaky I don?t understand why ? The community is always asking for people to share their code and I?m willing to share our code so others can use the VoiceXML/SCXML code to integrate into the main line also there may be some befits to our other changes for the main project.



    As for the comment it being disrespectful developing our own branch for our business, there has been countless other companies that have done the same thing on other open source projects ? for example Facebook has contributed code to other project in the same way and they did what is the best interest for their own needs ? even this has led to benefits to open source projects. Also we are a for profit company and we did the changes that are best for our overall business and shrinking over all foot print of project was a befit to us so we didn?t need to continue managing/converting code that is native to .Net and for converting to the CRL it business decision to decrees code of our future development ? also in the current market it?s easier to find good .Net developers then a good C++ developers, in about weeks? time we can have a C# developer writing C++/CLR.



    Understand I?m the owner of the company and I spent my money on developing this code, for our use. I invested/paid for over 480 development hours on the rewrite FS to CLR and integration of our VoiceXML/SCXML code, my intentions are to make an offering to compete with Microsoft Lync that is open source but not open to contributions, it will patched and supported by my team ? like trixbox. Our core will remain FREE but our modules will be licensed!  With this said I feel we have obligation to offer some of our source back to this project and help with integration if parts are chosen to be included into the FS base. Also to get a better understanding on what my people have been working on and to make our plans for our future I tried to install and learn about FS without help of my developers. Frankly I personally didn?t do the development of our FS CLR rewrite, I only helped developed the base code that we used to add VoiceXML/SCXML to FS ? the VoiceXML/SCXML code was from our own switch application but I discontinued the project in favor to using FS as a base.



    I originally asked where should we post our project so we could make convenient for the community access our changes and I only posted the details so others could see that our changes aren?t appropriate for the main line and see they may have some benefit. I?m starting to feel it maybe be the best option to release the code from our site like trixbox is doing with their asterisk code. Again, I?m not trying to impact FS in a negative way, I?m try to give back some of what we developed.





    From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
    Sent: Friday, October 28, 2011 10:19 PM
    To: FreeSWITCH Users Help
    Subject: Re: [Freeswitch-users] Feedback



    One thing that is disturbing is that after all the time and effort I put into making FreeSWITCH cross platform, the very idea of such an initiative wipes all that out in favor of one platform.  So even if you are willing to share it, and it's not B.S. its bothersome to me that:



    1) I spent 6 years making it work everywhere and you aim to put all your extended efforts into steering it to a separate version 

    2) Making it use less open source libs is not really a good thing since its an open source project.

    3) You did not propose anything to us first, that's a bit disrespectful.



    I think about everybody who uses FreeSWITCH when I make decisions, and this just feels selfish to me, (probably not for windows users)  but hey....



    On the other hand, Henry has a great point, that looks like a pretty big list of changes to do in a few weeks considering you were only asking trivial newbie questions as recently as sept 18th and your website is the generic IIS document root.   I would be pretty concerned about vishing if I were a reader of this post.



    I guess seeing is believing and if you are legit our windows-friendly developers will let me know. 















    On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel <cvogel at lyonl.com> wrote:

    Hello,

    I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

    Here are some of the changes we have made:

      a.. We converted FS core to C++ and having it compiling to .NET 4.0 CLI  
      b.. FS now runs inside its own namespace 
      c.. We replace FS_CLI with a PowerShell shell app 
      d.. Supports SQL Server support via the Native SQL Server Client API   
      e.. Support for windows clustering (up to 32 node active/active cluster) 
      f.. Ties more directly into the Win32 API and has less reliance on open source Libs 
      g.. Replaces OpenSSL sockets with Windows encrypted sockets 
      h.. Added windows performance monitoring 
      i.. No longer need to use mod_managed for managed modules 
      j.. Memory management relies on .Net garbage collection 
      k.. Added support for Microsoft Speech 
      l.. VoiceXML 3.0 and SCXML support
      m.. No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
      n.. Fixed RTP clock timing issues in virtualized environments 
      o.. New configuration file format using .Net App Configuration files (kind of looks like an IIS config file) 


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







    -- 
    Anthony Minessale II

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

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

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



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






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



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From infos at madovsky.org  Sat Oct 29 22:40:09 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 29 Oct 2011 14:40:09 -0400
Subject: [Freeswitch-users] Feedback
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM><CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
Message-ID: <9B8CDB9859AE4134AD0C686B64A31666@e1705>

>As for the comment it being disrespectful developing our own branch for our business, there has been countless other companies that have done the same thing on other >open source projects - for example Facebook has contributed code to other project in the same way and they did 

Facebook wants to play the open source game because they need a big community of developpers to maintain their website.
in reality, it's a big big private business that makes money with traffic at wall street, nothing else
  ----- Original Message ----- 
  From: Chad Vogel 
  To: FreeSWITCH Users Help 
  Sent: Saturday, October 29, 2011 1:58 PM
  Subject: Re: [Freeswitch-users] Feedback


  I'm thinking people are missing we are willing to share some of our code that we are developing back to the community that we developed for ourselves. We are not trying to change FS to a windows only platform. >From the comments being posted I can see that my plans of sharing the code that my developers worked on isn't being well received and freaky I don't understand why - The community is always asking for people to share their code and I'm willing to share our code so others can use the VoiceXML/SCXML code to integrate into the main line also there may be some befits to our other changes for the main project.

   

  As for the comment it being disrespectful developing our own branch for our business, there has been countless other companies that have done the same thing on other open source projects - for example Facebook has contributed code to other project in the same way and they did what is the best interest for their own needs - even this has led to benefits to open source projects. Also we are a for profit company and we did the changes that are best for our overall business and shrinking over all foot print of project was a befit to us so we didn't need to continue managing/converting code that is native to .Net and for converting to the CRL it business decision to decrees code of our future development - also in the current market it's easier to find good .Net developers then a good C++ developers, in about weeks' time we can have a C# developer writing C++/CLR.

   

  Understand I'm the owner of the company and I spent my money on developing this code, for our use. I invested/paid for over 480 development hours on the rewrite FS to CLR and integration of our VoiceXML/SCXML code, my intentions are to make an offering to compete with Microsoft Lync that is open source but not open to contributions, it will patched and supported by my team - like trixbox. Our core will remain FREE but our modules will be licensed!  With this said I feel we have obligation to offer some of our source back to this project and help with integration if parts are chosen to be included into the FS base. Also to get a better understanding on what my people have been working on and to make our plans for our future I tried to install and learn about FS without help of my developers. Frankly I personally didn't do the development of our FS CLR rewrite, I only helped developed the base code that we used to add VoiceXML/SCXML to FS - the VoiceXML/SCXML code was from our own switch application but I discontinued the project in favor to using FS as a base.

   

  I originally asked where should we post our project so we could make convenient for the community access our changes and I only posted the details so others could see that our changes aren't appropriate for the main line and see they may have some benefit. I'm starting to feel it maybe be the best option to release the code from our site like trixbox is doing with their asterisk code. Again, I'm not trying to impact FS in a negative way, I'm try to give back some of what we developed.

   

   

  From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
  Sent: Friday, October 28, 2011 10:19 PM
  To: FreeSWITCH Users Help
  Subject: Re: [Freeswitch-users] Feedback

   

  One thing that is disturbing is that after all the time and effort I put into making FreeSWITCH cross platform, the very idea of such an initiative wipes all that out in favor of one platform.  So even if you are willing to share it, and it's not B.S. its bothersome to me that:

   

  1) I spent 6 years making it work everywhere and you aim to put all your extended efforts into steering it to a separate version 

  2) Making it use less open source libs is not really a good thing since its an open source project.

  3) You did not propose anything to us first, that's a bit disrespectful.

   

  I think about everybody who uses FreeSWITCH when I make decisions, and this just feels selfish to me, (probably not for windows users)  but hey....

   

  On the other hand, Henry has a great point, that looks like a pretty big list of changes to do in a few weeks considering you were only asking trivial newbie questions as recently as sept 18th and your website is the generic IIS document root.   I would be pretty concerned about vishing if I were a reader of this post.

   

  I guess seeing is believing and if you are legit our windows-friendly developers will let me know. 

   

   

   

   

   

   

   

  On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel <cvogel at lyonl.com> wrote:

  Hello,

  I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

  Here are some of the changes we have made:

    a.. We converted FS core to C++ and having it compiling to .NET 4.0 CLI  
    b.. FS now runs inside its own namespace 
    c.. We replace FS_CLI with a PowerShell shell app 
    d.. Supports SQL Server support via the Native SQL Server Client API   
    e.. Support for windows clustering (up to 32 node active/active cluster) 
    f.. Ties more directly into the Win32 API and has less reliance on open source Libs 
    g.. Replaces OpenSSL sockets with Windows encrypted sockets 
    h.. Added windows performance monitoring 
    i.. No longer need to use mod_managed for managed modules 
    j.. Memory management relies on .Net garbage collection 
    k.. Added support for Microsoft Speech 
    l.. VoiceXML 3.0 and SCXML support
    m.. No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
    n.. Fixed RTP clock timing issues in virtualized environments 
    o.. New configuration file format using .Net App Configuration files (kind of looks like an IIS config file) 


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





   

  -- 
  Anthony Minessale II

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

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

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



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



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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From 12ukwn at gmail.com  Sun Oct 30 00:10:46 2011
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 29 Oct 2011 22:10:46 +0200
Subject: [Freeswitch-users] Feedback
In-Reply-To: <CALNZuwm-twgyRJmg2rKQV93JmgVsa5LVPg-u0uJGaO+YACS=PA@mail.gmail.com>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
	<CALNZuwm-twgyRJmg2rKQV93JmgVsa5LVPg-u0uJGaO+YACS=PA@mail.gmail.com>
Message-ID: <20111029221046.392e503c@anubis.defcon1>

On Sat, 29 Oct 2011 20:25:48 +0200
Avi Marcus <avi at avimarcus.net> wrote:

...
> It seems most of the community runs on *nix of some sort, but the folks
> that do run on windows do report some very high performance.. I don't know
> C, C++, C#/.net to really understand why you made the decision you did to
> make these changes, but I believe that was the point of ESL and all the
> mod_managed, mod_python, mod_lua, etc were made - to let you use nearly any
> language to control what FS is doing, while enabling the core of FS to run
> in native C to benefit from a code-base that be the same for any platform.
> That's why Michael asked "What did you gain by doing this work?"
...
I planned to rewrite it 6502 assembly language, but stepped back because
I feared an Apple patent infringing :p

-- 
BOFH excuse #151:
Some one needed the powerstrip, so they pulled the switch plug.


From freeswitch at earthspike.net  Sun Oct 30 01:14:41 2011
From: freeswitch at earthspike.net (John)
Date: Sat, 29 Oct 2011 22:14:41 +0100
Subject: [Freeswitch-users] Feedback
In-Reply-To: <20111029221046.392e503c@anubis.defcon1>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
	<CALNZuwm-twgyRJmg2rKQV93JmgVsa5LVPg-u0uJGaO+YACS=PA@mail.gmail.com>
	<20111029221046.392e503c@anubis.defcon1>
Message-ID: <4EAC6CC1.9070107@earthspike.net>

On 29/10/11 21:10, Jean-Yves F. Barbier wrote:
> On Sat, 29 Oct 2011 20:25:48 +0200
> Avi Marcus<avi at avimarcus.net>  wrote:
>
> ...
>> It seems most of the community runs on *nix of some sort, but the folks
>> that do run on windows do report some very high performance.. I don't know
>> C, C++, C#/.net to really understand why you made the decision you did to
>> make these changes, but I believe that was the point of ESL and all the
>> mod_managed, mod_python, mod_lua, etc were made - to let you use nearly any
>> language to control what FS is doing, while enabling the core of FS to run
>> in native C to benefit from a code-base that be the same for any platform.
>> That's why Michael asked "What did you gain by doing this work?"
> ...
> I planned to rewrite it 6502 assembly language, but stepped back because
> I feared an Apple patent infringing :p
>
I've had my 6809 port running for a month now.  First SIP session nearly 
set up!  6502 is likely to see some performance issues. ;)

John


From curriegrad2004 at gmail.com  Sun Oct 30 05:02:54 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 29 Oct 2011 19:02:54 -0700
Subject: [Freeswitch-users] Feedback
In-Reply-To: <4EAC6CC1.9070107@earthspike.net>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
	<CALNZuwm-twgyRJmg2rKQV93JmgVsa5LVPg-u0uJGaO+YACS=PA@mail.gmail.com>
	<20111029221046.392e503c@anubis.defcon1>
	<4EAC6CC1.9070107@earthspike.net>
Message-ID: <CAMWiMy-uQNvNzXvf76wpL2xQXRzneVqQwxzK5UEDijd+Vrj0KQ@mail.gmail.com>

Okay, okay let's tone down on the ridiculing on somebody's work right
now. If Chad is legitimate about his "fork" then he'd show us what
he's done. A code snippet or a sample of what his company has done
will be enough to prove if he's done what he has said to have did to
the FreeSWITCH code.

On Sat, Oct 29, 2011 at 2:14 PM, John <freeswitch at earthspike.net> wrote:
> On 29/10/11 21:10, Jean-Yves F. Barbier wrote:
>> On Sat, 29 Oct 2011 20:25:48 +0200
>> Avi Marcus<avi at avimarcus.net> ?wrote:
>>
>> ...
>>> It seems most of the community runs on *nix of some sort, but the folks
>>> that do run on windows do report some very high performance.. I don't know
>>> C, C++, C#/.net to really understand why you made the decision you did to
>>> make these changes, but I believe that was the point of ESL and all the
>>> mod_managed, mod_python, mod_lua, etc were made - to let you use nearly any
>>> language to control what FS is doing, while enabling the core of FS to run
>>> in native C to benefit from a code-base that be the same for any platform.
>>> That's why Michael asked "What did you gain by doing this work?"
>> ...
>> I planned to rewrite it 6502 assembly language, but stepped back because
>> I feared an Apple patent infringing :p
>>
> I've had my 6809 port running for a month now. ?First SIP session nearly
> set up! ?6502 is likely to see some performance issues. ;)
>
> John
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fieldpeak at gmail.com  Sun Oct 30 18:45:25 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 30 Oct 2011 23:45:25 +0800
Subject: [Freeswitch-users] b-leg pending
Message-ID: <CAOTmbMx9BQ7mbjEe5toH71QKtROy0fgST9koaf+_XxthppAFwQ@mail.gmail.com>

Hi friends,

i found when i make a call to pstn from a registed user via FS, after the
call hangup (no matter caller hangup or callee hangup, and no matter the
callee answer or not), the a-leg always was not released.
however, if i use two registed users call for each other, there is no
problem. can anyone help advise how to fix it, thanks!

below is my dialplan,

<extension name="CallToPSTN" >
   <condition field="destination_number" expression="^(0\d+)$" >
          <action application="bridge" data="sofia/internal/$
1 at 213.102.56.206" />
   </condition>
</extension>




-- 
Regards,
Charles
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From fieldpeak at gmail.com  Sun Oct 30 18:48:31 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 30 Oct 2011 23:48:31 +0800
Subject: [Freeswitch-users] b-leg pending
In-Reply-To: <CAOTmbMx9BQ7mbjEe5toH71QKtROy0fgST9koaf+_XxthppAFwQ@mail.gmail.com>
References: <CAOTmbMx9BQ7mbjEe5toH71QKtROy0fgST9koaf+_XxthppAFwQ@mail.gmail.com>
Message-ID: <CAOTmbMyd8oOyHhXanM9ixaNqA1WVjqh1_TTGXh4ZOR8C2MBeEw@mail.gmail.com>

sorry, the title should be a-leg pending...not b-leg

2011/10/30 fieldpeak <fieldpeak at gmail.com>

> Hi friends,
>
> i found when i make a call to pstn from a registed user via FS, after the
> call hangup (no matter caller hangup or callee hangup, and no matter the
> callee answer or not), the a-leg always was not released.
> however, if i use two registed users call for each other, there is no
> problem. can anyone help advise how to fix it, thanks!
>
> below is my dialplan,
>
> <extension name="CallToPSTN" >
>    <condition field="destination_number" expression="^(0\d+)$" >
>           <action application="bridge" data="sofia/internal/$
> 1 at 213.102.56.206" />
>    </condition>
> </extension>
>
>
>
>
> --
> Regards,
> Charles
>
>


-- 
Regards,
Charles
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From mgg at giagnocavo.net  Mon Oct 31 03:28:45 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sun, 30 Oct 2011 20:28:45 -0400
Subject: [Freeswitch-users] Feedback
In-Reply-To: <A4EF10E82B304B4C90D0737C82A28BB8@e1705>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM><CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com><710D094EC9AA8A458657A821CD66911201C80EF4@CH1PRD0502MB100.namprd05.prod.outlook.com>
	<CALNZuwm-twgyRJmg2rKQV93JmgVsa5LVPg-u0uJGaO+YACS=PA@mail.gmail.com>
	<A4EF10E82B304B4C90D0737C82A28BB8@e1705>
Message-ID: <03351FCC6082174C8534AB714B8258A5E4D7DD1B@mse17be1.mse17.exchange.ms>

So, because he asked about how the community would like him to host his code, then defended the attacks on him for offering to share code, he's a spammer?

I cannot fathom the benefit to the FS community to being rude to a potential contributor. At worst, I'd expect a "Well, that sounds pretty unbelievable, and it doesn't sound like it lines up with the FS project goals, but why not share your code and motivations and we'll go from there." No need to shoot someone down without evaluating properly.

So far, he's done nothing of the sort to deserve such treatment.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Madovsky
Sent: Saturday, October 29, 2011 12:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Feedback

My opinion is, with this kind of email, it even doesn't need to answer,
if this guy was a bit serious he will contact directly FS team, not a FS user list.
it can considered as spam...
----- Original Message -----
From: Avi Marcus<mailto:avi at avimarcus.net>
To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 29, 2011 2:25 PM
Subject: Re: [Freeswitch-users] Feedback

I think Anthony Minessale's main point - since he is the creator of FS - is that he spent a lot of effort making FS cross-platform, and as such, it would seem the bulk of your effort was directly counter to this intention. (The same for using open source libs.)

It seems most of the community runs on *nix of some sort, but the folks that do run on windows do report some very high performance.. I don't know C, C++, C#/.net to really understand why you made the decision you did to make these changes, but I believe that was the point of ESL and all the mod_managed, mod_python, mod_lua, etc were made - to let you use nearly any language to control what FS is doing, while enabling the core of FS to run in native C to benefit from a code-base that be the same for any platform.
That's why Michael asked "What did you gain by doing this work?"

The community welcomes.. no, begs, for contributions but the the scale of pushing into native windows code seems counter to the project's goals of being universal (while maintaining performance and program-ability), hence Anthony feeling bothered..

-Avi

On Sat, Oct 29, 2011 at 7:58 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
I'm thinking people are missing we are willing to share some of our code that we are developing back to the community that we developed for ourselves. We are not trying to change FS to a windows only platform. From the comments being posted I can see that my plans of sharing the code that my developers worked on isn't being well received and freaky I don't understand why - The community is always asking for people to share their code and I'm willing to share our code so others can use the VoiceXML/SCXML code to integrate into the main line also there may be some befits to our other changes for the main project.

As for the comment it being disrespectful developing our own branch for our business, there has been countless other companies that have done the same thing on other open source projects - for example Facebook has contributed code to other project in the same way and they did what is the best interest for their own needs - even this has led to benefits to open source projects. Also we are a for profit company and we did the changes that are best for our overall business and shrinking over all foot print of project was a befit to us so we didn't need to continue managing/converting code that is native to .Net and for converting to the CRL it business decision to decrees code of our future development - also in the current market it's easier to find good .Net developers then a good C++ developers, in about weeks' time we can have a C# developer writing C++/CLR.

Understand I'm the owner of the company and I spent my money on developing this code, for our use. I invested/paid for over 480 development hours on the rewrite FS to CLR and integration of our VoiceXML/SCXML code, my intentions are to make an offering to compete with Microsoft Lync that is open source but not open to contributions, it will patched and supported by my team - like trixbox. Our core will remain FREE but our modules will be licensed!  With this said I feel we have obligation to offer some of our source back to this project and help with integration if parts are chosen to be included into the FS base. Also to get a better understanding on what my people have been working on and to make our plans for our future I tried to install and learn about FS without help of my developers. Frankly I personally didn't do the development of our FS CLR rewrite, I only helped developed the base code that we used to add VoiceXML/SCXML to FS - the VoiceXML/SCXML code was from our own switch application but I discontinued the project in favor to using FS as a base.

I originally asked where should we post our project so we could make convenient for the community access our changes and I only posted the details so others could see that our changes aren't appropriate for the main line and see they may have some benefit. I'm starting to feel it maybe be the best option to release the code from our site like trixbox is doing with their asterisk code. Again, I'm not trying to impact FS in a negative way, I'm try to give back some of what we developed.


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 Anthony Minessale
Sent: Friday, October 28, 2011 10:19 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Feedback

One thing that is disturbing is that after all the time and effort I put into making FreeSWITCH cross platform, the very idea of such an initiative wipes all that out in favor of one platform.  So even if you are willing to share it, and it's not B.S. its bothersome to me that:

1) I spent 6 years making it work everywhere and you aim to put all your extended efforts into steering it to a separate version
2) Making it use less open source libs is not really a good thing since its an open source project.
3) You did not propose anything to us first, that's a bit disrespectful.

I think about everybody who uses FreeSWITCH when I make decisions, and this just feels selfish to me, (probably not for windows users)  but hey....

On the other hand, Henry has a great point, that looks like a pretty big list of changes to do in a few weeks considering you were only asking trivial newbie questions as recently as sept 18th and your website is the generic IIS document root.   I would be pretty concerned about vishing if I were a reader of this post.

I guess seeing is believing and if you are legit our windows-friendly developers will let me know.







On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:

Hello,

I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users.

Here are some of the changes we have made:

 *   We converted FS core to C++ and having it compiling to .NET 4.0 CLI
 *   FS now runs inside its own namespace
 *   We replace FS_CLI with a PowerShell shell app
 *   Supports SQL Server support via the Native SQL Server Client API
 *   Support for windows clustering (up to 32 node active/active cluster)
 *   Ties more directly into the Win32 API and has less reliance on open source Libs
 *   Replaces OpenSSL sockets with Windows encrypted sockets
 *   Added windows performance monitoring
 *   No longer need to use mod_managed for managed modules
 *   Memory management relies on .Net garbage collection
 *   Added support for Microsoft Speech
 *   VoiceXML 3.0 and SCXML support
 *   No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages)
 *   Fixed RTP clock timing issues in virtualized environments
 *   New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)


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<tel:%2B19193869900>


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

________________________________

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From Nabble_01394 at slickdeals.endjunk.com  Mon Oct 31 05:26:57 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sun, 30 Oct 2011 19:26:57 -0700 (PDT)
Subject: [Freeswitch-users] Feedback
In-Reply-To: <CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
Message-ID: <1320028017350-6946803.post@n2.nabble.com>

Anthony,

I am sure everyone really appreciates what you have done to contribute such
an excellent program to VoIP community. People like Linux Torvalds, and
including you, are what I considered a hero with a big heart to support
open-source community. Without them, the world will be different today. It
would probably be an M$ dominated world.

I guess what I tried to say here is it really is very disturbing to see your
comments below. The least you could have done to show our community was a
positive acceptance to embrace someone's effort to legitimately port FS to
other platforms in other programming languages. If Chad wanted to hi-jack
FS, I don't suppose he would even dare to represent himself here. BTw, can
you imagine if Linus Torvalds behaves like this?

I hope you don't take my comments in a negative direction. However, like
anyone who reads your comments below, I won't be surprised if you decide to
head in a negative direction to flame my comments. Regardless, I sincerely
wish you the best to come, especially with the contribution from Chad.


Anthony Minessale wrote:
> 
> One thing that is disturbing is that after all the time and effort I put
> into making FreeSWITCH cross platform, the very idea of such an initiative
> wipes all that out in favor of one platform.  So even if you are willing
> to
> share it, and it's not B.S. its bothersome to me that:
> 
> 1) I spent 6 years making it work everywhere and you aim to put all your
> extended efforts into steering it to a separate version
> 2) Making it use less open source libs is not really a good thing since
> its
> an open source project.
> 3) You did not propose anything to us first, that's a bit disrespectful.
> 
> I think about everybody who uses FreeSWITCH when I make decisions, and
> this
> just feels selfish to me, (probably not for windows users)  but hey....
> 
> On the other hand, Henry has a great point, that looks like a pretty big
> list of changes to do in a few weeks considering you were only asking
> trivial newbie questions as recently as sept 18th and your website is the
> generic IIS document root.   I would be pretty concerned about vishing if
> I
> were a reader of this post.
> 
> I guess seeing is believing and if you are legit our windows-friendly
> developers will let me know.
> 
> 
> 
> 
> 
> 
> 
> On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel &lt;cvogel@&gt; wrote:
> 
>>  Hello,
>>
>> I wanted to reach out and get some feedback from everyone because my
>> company started a project using FS as base several weeks ago, our goal
>> was
>> to optimize and integrate FS more directly into Windows environments.
>> Here
>> is my question, we would like to share our changes and where should we
>> host
>> our project? Because we understand our build breaks compatibility with
>> the
>> UNIX environments and fails to meet the objectives cross platform
>> compatibility, however it could contribute greatly and be beneficial to
>> other windows users.
>>
>> Here are some of the changes we have made:
>>
>>    - We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>>    - FS now runs inside its own namespace
>>    - We replace FS_CLI with a PowerShell shell app
>>    - Supports SQL Server support via the Native SQL Server Client API
>>    - Support for windows clustering (up to 32 node active/active cluster)
>>    - Ties more directly into the Win32 API and has less reliance on open
>>    source Libs
>>    - Replaces OpenSSL sockets with Windows encrypted sockets
>>    - Added windows performance monitoring
>>    - No longer need to use mod_managed for managed modules
>>    - Memory management relies on .Net garbage collection
>>    - Added support for Microsoft Speech
>>    - VoiceXML 3.0 and SCXML support
>>    - No longer supports JS, LUA, PHP development - Modules can be only
>>    developed in .NET, C or C++( we feel support for other languages can
>> be
>>    added back in but falls outside the scope of our objectives at this
>> time,
>>    support for SCXML should make limit the need for other scripting
>> languages)
>>    - Fixed RTP clock timing issues in virtualized environments
>>    - New configuration file format using .Net App Configuration files
>>    (kind of looks like an IIS config file)
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at .freeswitch
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale@
> GTALK/JABBER/PAYPAL:anthony.minessale@
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at .freeswitch
> googletalk:conf+888 at .freeswitch
> pstn:+19193869900
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at .freeswitch
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p6946803.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Mon Oct 31 05:38:08 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 30 Oct 2011 22:38:08 -0400
Subject: [Freeswitch-users] Feedback
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM><CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<1320028017350-6946803.post@n2.nabble.com>
Message-ID: <E3DD08C8FFE14F638B476540614656FF@e1705>

Chad, you are welcome to share your code
welcome to FS community ! :)

----- Original Message ----- 
From: "mazilo" <Nabble_01394 at slickdeals.endjunk.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, October 30, 2011 10:26 PM
Subject: Re: [Freeswitch-users] Feedback


> Anthony,
>
> I am sure everyone really appreciates what you have done to contribute 
> such
> an excellent program to VoIP community. People like Linux Torvalds, and
> including you, are what I considered a hero with a big heart to support
> open-source community. Without them, the world will be different today. It
> would probably be an M$ dominated world.
>
> I guess what I tried to say here is it really is very disturbing to see 
> your
> comments below. The least you could have done to show our community was a
> positive acceptance to embrace someone's effort to legitimately port FS to
> other platforms in other programming languages. If Chad wanted to hi-jack
> FS, I don't suppose he would even dare to represent himself here. BTw, can
> you imagine if Linus Torvalds behaves like this?
>
> I hope you don't take my comments in a negative direction. However, like
> anyone who reads your comments below, I won't be surprised if you decide 
> to
> head in a negative direction to flame my comments. Regardless, I sincerely
> wish you the best to come, especially with the contribution from Chad.
>
>
> Anthony Minessale wrote:
>>
>> One thing that is disturbing is that after all the time and effort I put
>> into making FreeSWITCH cross platform, the very idea of such an 
>> initiative
>> wipes all that out in favor of one platform.  So even if you are willing
>> to
>> share it, and it's not B.S. its bothersome to me that:
>>
>> 1) I spent 6 years making it work everywhere and you aim to put all your
>> extended efforts into steering it to a separate version
>> 2) Making it use less open source libs is not really a good thing since
>> its
>> an open source project.
>> 3) You did not propose anything to us first, that's a bit disrespectful.
>>
>> I think about everybody who uses FreeSWITCH when I make decisions, and
>> this
>> just feels selfish to me, (probably not for windows users)  but hey....
>>
>> On the other hand, Henry has a great point, that looks like a pretty big
>> list of changes to do in a few weeks considering you were only asking
>> trivial newbie questions as recently as sept 18th and your website is the
>> generic IIS document root.   I would be pretty concerned about vishing if
>> I
>> were a reader of this post.
>>
>> I guess seeing is believing and if you are legit our windows-friendly
>> developers will let me know.
>>
>>
>>
>>
>>
>>
>>
>> On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel &lt;cvogel@&gt; wrote:
>>
>>>  Hello,
>>>
>>> I wanted to reach out and get some feedback from everyone because my
>>> company started a project using FS as base several weeks ago, our goal
>>> was
>>> to optimize and integrate FS more directly into Windows environments.
>>> Here
>>> is my question, we would like to share our changes and where should we
>>> host
>>> our project? Because we understand our build breaks compatibility with
>>> the
>>> UNIX environments and fails to meet the objectives cross platform
>>> compatibility, however it could contribute greatly and be beneficial to
>>> other windows users.
>>>
>>> Here are some of the changes we have made:
>>>
>>>    - We converted FS core to C++ and having it compiling to .NET 4.0 CLI
>>>    - FS now runs inside its own namespace
>>>    - We replace FS_CLI with a PowerShell shell app
>>>    - Supports SQL Server support via the Native SQL Server Client API
>>>    - Support for windows clustering (up to 32 node active/active 
>>> cluster)
>>>    - Ties more directly into the Win32 API and has less reliance on open
>>>    source Libs
>>>    - Replaces OpenSSL sockets with Windows encrypted sockets
>>>    - Added windows performance monitoring
>>>    - No longer need to use mod_managed for managed modules
>>>    - Memory management relies on .Net garbage collection
>>>    - Added support for Microsoft Speech
>>>    - VoiceXML 3.0 and SCXML support
>>>    - No longer supports JS, LUA, PHP development - Modules can be only
>>>    developed in .NET, C or C++( we feel support for other languages can
>>> be
>>>    added back in but falls outside the scope of our objectives at this
>>> time,
>>>    support for SCXML should make limit the need for other scripting
>>> languages)
>>>    - Fixed RTP clock timing issues in virtualized environments
>>>    - New configuration file format using .Net App Configuration files
>>>    (kind of looks like an IIS config file)
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at .freeswitch
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale@
>> GTALK/JABBER/PAYPAL:anthony.minessale@
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at .freeswitch
>> googletalk:conf+888 at .freeswitch
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at .freeswitch
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 
> Watts of electricity.
> --
> View this message in context: 
> http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p6946803.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From spencer at 5ninesolutions.com  Mon Oct 31 05:44:00 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 30 Oct 2011 19:44:00 -0700
Subject: [Freeswitch-users] Using Presence with Valet Parking
Message-ID: <E572C020-3CF0-40A9-83A1-3F8EC4EEA639@5ninesolutions.com>

Hello,
I noticed there have been some recent improvements to mod_valet_parking regarding presence.  I was wondering if anyone had used this already and could shed some light on how to subscribe to the status of a particular slot.  I'm testing this with a Cisco SPA-509G and can't seem to get it working. My dialplan looks like this:

    <extension name="parking_slots">
        <condition field="destination_number" expression="^(7[1-4])$" break="on-false">
          <action application="answer"/>
          <action application="valet_park" data="parking_slots $1"/>
        </condition>
    </extension>

The phone is sending a SUBSCRIBE to park+71@<DOMAIN> and gets a 202 Accepted response.  

I do have presence configured and working correctly for user extensions.  When a call is transferred to 71, the parking works as expected but a NOTIFY is never sent to the phone.  What am I missing?

Thanks for any help!

Spencer


 

From adminjew at gmail.com  Mon Oct 31 07:46:22 2011
From: adminjew at gmail.com (Yitzchok)
Date: Mon, 31 Oct 2011 00:46:22 -0400
Subject: [Freeswitch-users] Feedback
In-Reply-To: <E3DD08C8FFE14F638B476540614656FF@e1705>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<1320028017350-6946803.post@n2.nabble.com>
	<E3DD08C8FFE14F638B476540614656FF@e1705>
Message-ID: <CAOQmPKpAJ+2aUX5c0WO962ivPd4eFx0VmPp8YQfU3oXkfex7mA@mail.gmail.com>

Will it work on linux using mono http://www.mono-project.com/CPlusPlus (I
don't think so)


Yitzchok

FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Mon Oct 31 07:55:42 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 Oct 2011 00:55:42 -0400
Subject: [Freeswitch-users] nibblebill SQL update
Message-ID: <AC7091ACA2B5480CB361A856EFAE4428@e1705>

Hi all,

little suggestion about cash update with nibblebill:
make a check of if nibble_rate is set to 0, avoid any update
and reduce write SQL request...

Franck
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From Nabble_01394 at slickdeals.endjunk.com  Mon Oct 31 14:02:36 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Mon, 31 Oct 2011 04:02:36 -0700 (PDT)
Subject: [Freeswitch-users] Feedback
In-Reply-To: <CAOQmPKpAJ+2aUX5c0WO962ivPd4eFx0VmPp8YQfU3oXkfex7mA@mail.gmail.com>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<1320028017350-6946803.post@n2.nabble.com>
	<E3DD08C8FFE14F638B476540614656FF@e1705>
	<CAOQmPKpAJ+2aUX5c0WO962ivPd4eFx0VmPp8YQfU3oXkfex7mA@mail.gmail.com>
Message-ID: <1320058956564-6947727.post@n2.nabble.com>

I have a doubt the it will. However, I won't be surprised if it does work and
hope it will.


Yitzchok wrote:
> 
> Will it work on linux using mono http://www.mono-project.com/CPlusPlus (I
> don't think so)


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p6947727.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From michaelt at callall.co.za  Mon Oct 31 14:07:28 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Mon, 31 Oct 2011 13:07:28 +0200
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <CAFmdODgcPWv2T5LmaYtxHRDMeHsQzTdfCmNOW4exZQ1LMyaihw@mail.gmail.com>
References: <20111027072637.26979.qmail@community22.interfree.it>
	<CAFmdODgcPWv2T5LmaYtxHRDMeHsQzTdfCmNOW4exZQ1LMyaihw@mail.gmail.com>
Message-ID: <CAFmdODhkFaCdk7JRpA1tm76_j8LoCp2FO_dgoaDZMFFT05qX8g@mail.gmail.com>

Hi,

 Is there someone in the list interested in some consulting work to assist
me to get to the bottom of this problem?

 Happy to pay for assistance here.

Thanks,

Michael

On Thu, Oct 27, 2011 at 11:34 AM, Michael Toop <michaelt at callall.co.za>wrote:

> Hi,
>
>  Start & stop RTP ports are commented out in switch.conf so must be the
> defaults, 16384 & 32768.
>
>  I have also tried the NDLB-allow-bad-iananame suggestion & have the same
> results, see SIP trace here: http://pastebin.com/vdztr7cf
>
>  What is very strange about this issue it only happens on certain routes
> i.e. the SDP of the B leg of the call is somehow breaking things.  Really
> at a loss here.
>
> Cheers,
>
> Michael
>
>
>
> On Thu, Oct 27, 2011 at 9:26 AM, <chrisbware at interfree.it> wrote:
>
>> Even payload type 96 for G729 sounds strange to me.
>> How did you set  *rtp-start-portand ***
>> ***rtp-end-port ?*
>> **
>> ***
>> *
>> **
>> ***
>> *
>> -----Messaggio originale-----
>> Da: Michael Toop <michaelt at callall.co.za>
>> Inviato il: 26 Ott 2011 - 05:00
>> A: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>
>>
>> Hi,
>>  When initiating a call from Freeswitch to a provider, Freeswitch is
>> putting '0'/ zero in the media port.  This looks like a bug to me.:
>>    m=audio 0 RTP/AVP 96 101
>> Media Description, name and address (m): audio 0 RTP/AVP 96 101
>>  Txt based SIP trace attached (not sure how to scrub pcap files of IPs).
>> Thanks,
>> Michael
>>
>>
>> -------------------------------------------------------------------------------
>> Valore legale alle tue mail
>> InterfreePEC - la tua Posta Elettronica Certificata
>> http://pec.interfree.it
>>
>> -------------------------------------------------------------------------------
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From spautz2 at telefaks.biz  Mon Oct 31 14:19:23 2011
From: spautz2 at telefaks.biz (David)
Date: Mon, 31 Oct 2011 12:19:23 +0100
Subject: [Freeswitch-users]
	CHANNEL_CALLSTATE=>switch_channel_perform_set_callstate
Message-ID: <4EAE843B.8090404@telefaks.biz>

Hi,

how can i push variables to the 
CHANNEL_CALLSTATE=>switch_channel_perform_set_callstate EVENT for a 
conference outbound channel?


Greez

David


From brian at freeswitch.org  Mon Oct 31 17:34:30 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 31 Oct 2011 09:34:30 -0500
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <CAFmdODhkFaCdk7JRpA1tm76_j8LoCp2FO_dgoaDZMFFT05qX8g@mail.gmail.com>
References: <20111027072637.26979.qmail@community22.interfree.it>
	<CAFmdODgcPWv2T5LmaYtxHRDMeHsQzTdfCmNOW4exZQ1LMyaihw@mail.gmail.com>
	<CAFmdODhkFaCdk7JRpA1tm76_j8LoCp2FO_dgoaDZMFFT05qX8g@mail.gmail.com>
Message-ID: <6D16158B-07AB-4FF6-8817-4166F855DC62@freeswitch.org>

Its not a bug.  zero is an indication that we rejected the media.  
/b

On Oct 31, 2011, at 6:07 AM, Michael Toop wrote:

> Hi,
> 
> Is there someone in the list interested in some consulting work to assist
> me to get to the bottom of this problem?
> 
> Happy to pay for assistance here.
> 
> Thanks,
> 
> Michael

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From boris at tagnet.ru  Mon Oct 31 17:58:04 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 31 Oct 2011 20:58:04 +0600
Subject: [Freeswitch-users] Linksys PAP2-T and echo
Message-ID: <4EAEB77C.9060901@tagnet.ru>

Hello!


     I'm wondering if it is possible to solve echo with Linksys PAP2-T 
devices. Does anybody here uses it? I read many articles and still can't 
understand how to eleminate echo :( I read 
http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found 
that Asterisk has its own EC. Does Freeswitch has it too? I found this 
article http://docs.freeswitch.org/echo_can_page.html but can't 
understand - has Freeswitch EC or not? If yes - how to turn it on and off?

-- 
Regards,
Boris



From anthony.minessale at gmail.com  Mon Oct 31 18:03:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Oct 2011 10:03:16 -0500
Subject: [Freeswitch-users] Using Presence with Valet Parking
In-Reply-To: <E572C020-3CF0-40A9-83A1-3F8EC4EEA639@5ninesolutions.com>
References: <E572C020-3CF0-40A9-83A1-3F8EC4EEA639@5ninesolutions.com>
Message-ID: <CAKbxfG9sLMRea7ADb5HrxxgxndrFjkjUfTZjnSnXjZzZBh9xAA@mail.gmail.com>

make sure DOMAIN is the default domain set in vars.xml otherwise name the
lot with @domain.com right in it.
Also make sure its the most recent rev of the mod.


On Sun, Oct 30, 2011 at 9:44 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello,
> I noticed there have been some recent improvements to mod_valet_parking
> regarding presence.  I was wondering if anyone had used this already and
> could shed some light on how to subscribe to the status of a particular
> slot.  I'm testing this with a Cisco SPA-509G and can't seem to get it
> working. My dialplan looks like this:
>
>    <extension name="parking_slots">
>        <condition field="destination_number" expression="^(7[1-4])$"
> break="on-false">
>          <action application="answer"/>
>          <action application="valet_park" data="parking_slots $1"/>
>        </condition>
>    </extension>
>
> The phone is sending a SUBSCRIBE to park+71@<DOMAIN> and gets a 202
> Accepted response.
>
> I do have presence configured and working correctly for user extensions.
>  When a call is transferred to 71, the parking works as expected but a
> NOTIFY is never sent to the phone.  What am I missing?
>
> Thanks for any help!
>
> Spencer
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 steveu at coppice.org  Mon Oct 31 18:11:51 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 31 Oct 2011 23:11:51 +0800
Subject: [Freeswitch-users] Linksys PAP2-T and echo
In-Reply-To: <4EAEB77C.9060901@tagnet.ru>
References: <4EAEB77C.9060901@tagnet.ru>
Message-ID: <4EAEBAB7.1030405@coppice.org>

On 10/31/2011 10:58 PM, Boris Kovalenko wrote:
> Hello!
>
>
>       I'm wondering if it is possible to solve echo with Linksys PAP2-T
> devices. Does anybody here uses it? I read many articles and still can't
> understand how to eleminate echo :( I read
> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found
> that Asterisk has its own EC. Does Freeswitch has it too? I found this
> article http://docs.freeswitch.org/echo_can_page.html but can't
> understand - has Freeswitch EC or not? If yes - how to turn it on and off?
Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo 
cancellation over IP is very problematic, and hardly ever attempted. The 
PAP2T should be echo cancelling for itself, and they usually do a fairly 
good job of this. I don't think it is configurable. Its always on.

Steve



From msc at freeswitch.org  Mon Oct 31 18:22:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Oct 2011 08:22:59 -0700
Subject: [Freeswitch-users] Log level of log file by fs_cli
In-Reply-To: <CAOTmbMwQYK5z4CZqZ-srEy979+gyXpO47X8PwR+FBTDekMU5jA@mail.gmail.com>
References: <CAOTmbMwj8qnJZYGDFQeWQ83jyhwRXOQwj88g2UjGLDkKnr4Xdw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F2B77504@cooper>
	<76208DFD-604B-4C02-AD0B-4A6C36425134@gmail.com>
	<CAOTmbMwQYK5z4CZqZ-srEy979+gyXpO47X8PwR+FBTDekMU5jA@mail.gmail.com>
Message-ID: <CAKzWOxV8MVk5VNb594-Awa3PJnE6Qz4U3FoxoK=H26Cq=rpuXA@mail.gmail.com>

Fieldpeak,

Looks like you just confirmed what Steven A. said. Try his suggestion re:
"fsctl loglevel xxx" from within fs_cli and see what you get.

-MC

On Fri, Oct 28, 2011 at 7:51 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Steven,
>
> i tried below cmd, and then make a call, inside the the log file,
> there is still 'DEBUG' level message logged. very strange...
>
> freeswitch at internal> fsctl loglevel alert
> +OK log level: ALERT [1]
>
> freeswitch at internal>
>
> inside freeswitch.log
> ...
>
> 2011-10-29 10:44:02.496776 [DEBUG] switch_core_state_machine.c:464
> (sofia/internal/13580358068 at 124.193.106.104) Callstate Change HANGUP
> -> DOWN
>
>
>
> 2011/10/28, Steven Ayre <steveayre at gmail.com>:
> > Peter, that affects only the logging level of your fs_cli session - not
> the
> > log file.
> >
> > Fran, use the fsctl command from the cli. It's in mod_commands and
> > documented on the wiki.
> >
> > Eg:
> > fsctl loglevel
> > - shows current level
> > fsctl loglevel debug
> > - set to debug
> >
> > Steve on iPhone
> >
> > On 28 Oct 2011, at 11:09, Peter Olsson <peter.olsson at visionutveckling.se
> >
> > wrote:
> >
> >> Just enter ?/log [loglevel]? inside fs_cli. I think there is a startup
> >> parameter as well.
> >>
> >> /Peter
> >>
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r fieldpeak
> >> Skickat: den 28 oktober 2011 10:50
> >> Till: FreeSWITCH-users
> >> ?mne: [Freeswitch-users] Log level of log file by fs_cli
> >>
> >> Hi friends,
> >>
> >> i know we can set the loglevel inside switch.conf.xml to change the log
> >> level of log files,
> >>
> >> however, i would like to change the log level of log files thru fs_cli
> on
> >> the fly(no need restart FS), is there any commands to support it,
> thanks.
> >>
> >> --
> >> Regards,
> >> Charles
> >>
> >> !DSPAM:4eaa6d4932761466214554!
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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,
> Charles
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 primrose.ws  Mon Oct 31 18:26:22 2011
From: darcy at primrose.ws (Darcy)
Date: Mon, 31 Oct 2011 11:26:22 -0400
Subject: [Freeswitch-users] Linksys PAP2-T and echo
In-Reply-To: <4EAEBAB7.1030405@coppice.org>
References: <4EAEB77C.9060901@tagnet.ru> <4EAEBAB7.1030405@coppice.org>
Message-ID: <3B1EABFB29D34F61BC96746169E2E2C4@DWP>

on your pap2t web interface, go to voice, advanced, select the line, the 
settings below will allow you to play with echo cancellation.  However, you 
also need to go the the regional tab in voice and play with the fxs input 
and output gains, the voice gain on the pap2t or any linksys gateway, will 
impact the echo on the line.  This is most likely where your problem is. 
You might not be able to cure the echo on every call, but you should be able 
to make it acceptable.

Darcy


Audio Configuration
Preferred Codec:        Second Preferred Codec:
Third Preferred Codec:        Use Pref Codec Only:
Silence Supp Enable:        Silence Threshold:
G729a Enable:        Echo Canc Enable:
G723 Enable:        Echo Canc Adapt Enable:
G726-16 Enable:        Echo Supp Enable:
G726-24 Enable:        FAX CED Detect Enable:
G726-32 Enable:        FAX CNG Detect Enable:
G726-40 Enable:        FAX Passthru Codec:
DTMF Process INFO:        FAX Codec Symmetric:
DTMF Process AVT:        FAX Passthru Method:
DTMF Tx Method:        DTMF Tx Mode:
DTMF Tx Strict Hold Off Time:        FAX Process NSE:
Hook Flash Tx Method:        FAX Disable ECAN:
Release Unused Codec:        FAX Enable T38:
FAX T38 Redundancy:        FAX Tone Detect Mode:

-----Original Message----- 
From: Steve Underwood
Sent: Monday, October 31, 2011 11:11 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo

On 10/31/2011 10:58 PM, Boris Kovalenko wrote:
> Hello!
>
>
>       I'm wondering if it is possible to solve echo with Linksys PAP2-T
> devices. Does anybody here uses it? I read many articles and still can't
> understand how to eleminate echo :( I read
> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found
> that Asterisk has its own EC. Does Freeswitch has it too? I found this
> article http://docs.freeswitch.org/echo_can_page.html but can't
> understand - has Freeswitch EC or not? If yes - how to turn it on and off?
Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo
cancellation over IP is very problematic, and hardly ever attempted. The
PAP2T should be echo cancelling for itself, and they usually do a fairly
good job of this. I don't think it is configurable. Its always on.

Steve



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




From anthony.minessale at gmail.com  Mon Oct 31 18:46:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 31 Oct 2011 10:46:57 -0500
Subject: [Freeswitch-users] Feedback
In-Reply-To: <E3DD08C8FFE14F638B476540614656FF@e1705>
References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM>
	<CAKbxfG9dqK77P0xpaLEzdkzqBJFYJstSkixwQHZ2pw7V0OqHgg@mail.gmail.com>
	<1320028017350-6946803.post@n2.nabble.com>
	<E3DD08C8FFE14F638B476540614656FF@e1705>
Message-ID: <CAKbxfG9MnTFaEFCJmfRTFYL6sBJFk8wjj4x5Jrwhtqk7qEWdDQ@mail.gmail.com>

Well the initial design of FreeSWITCH provides a way to make custom modules
in a variety of languages including C++.


>>    - We converted FS core to C++ and having it compiling to .NET 4.0 CLI

The entire FS core is C specifically designed to be completely compatible
with C++.
I am not sure what other changes are necessary but FreeSWITCH was not
designed to have people even touching the core.  They should be using
modules.


>>    - FS now runs inside its own namespace
Ummm ok,

>>    - We replace FS_CLI with a PowerShell shell app
Not really required to be a replacement when it's a standalone app
anyway.....


>>    - Supports SQL Server support via the Native SQL Server Client API
There was a core abstraction for db access, addition of this support could
have been worked into the existing framework without
causing proprietary changes....

>>    - Support for windows clustering (up to 32 node active/active cluster)
Not sure what this entails.

>>    - Ties more directly into the Win32 API and has less reliance on open
>>    source Libs

In my experience, the windows specific implementations of some of the
function suppled by other POSIX libs are very poor and do not provide the
same type of performance as the originals.  The later version of windows
now have become more and more truly POSIX compliant and all the suffering
man hours quoted above would have been better spent making all platforms
equally take advantage of newer stuff.

>>    - Replaces OpenSSL sockets with Windows encrypted sockets
Not sure but does this mean modifications to sofia etc because we don't
really use SSL direct in the core.


>>    - Added windows performance monitoring
I'm sure that could have been put into the mainstream very easily.

>>    - No longer need to use mod_managed for managed modules
Not sure how but it's most likely a trivial bypass of the abstraction
layers which is mostly just a greedy shortcut.

>>    - Memory management relies on .Net garbage collection
So you wrote a drop in replacement for the Apache Portable Runtime that
offers the dozens of pool based programming tools.  This alone seems like a
many man-year undertaking.

>>    - Added support for Microsoft Speech
This hopefully was done in a module that could easily be part of mainstream.

>>    - VoiceXML 3.0 and SCXML support
Again, I would hope in a module using the core speech module api.


>>    - No longer supports JS, LUA, PHP development - Modules can be only
>>    developed in .NET, C or C++( we feel support for other languages can
>> be
>>    added back in but falls outside the scope of our objectives at this
>> time,
Its easy to simply not load these modules, you never could write actual
modules in any of these languages to begin with, this comment suggest a
lack of understanding in how the core works that makes me even more
skeptical.


>>    support for SCXML should make limit the need for other scripting
>> languages)
Again something that should be done as a module.


>>    - Fixed RTP clock timing issues in virtualized environments
That must be a windows specific thing that could be applied in an ifdef
into the core.

>>    - New configuration file format using .Net App Configuration files
>>    (kind of looks like an IIS config file)

This change would render most of the existing functionality unusable and
would require countless man hours to deal with.  You cited 480 man hours
which is how much work I put in to this project every month for half a
decade straight.........


Anyway, FreeSWITCH was designed for businesses to share the core and make
their own custom modules for their own business use.   Any modifications to
the core or existing components require 100% of changes to be submitted to
the respective author on a per-file basis as stipulated by the MPL.  I may
or may not choose to do what I see fit with those changes but I would for
sure like to see them especially since you want us to host this branch
anyway.

In reality I suggest you make an effort to connect with the other windows
developers in the community and work toward a collaborative effort to make
your desired functionality work inside the scope of the projects actual
goals.  There are many of such developers and there has already been talk
of making some of the changes you boast only not in a way that destroys any
previous work done on the project or alienates any existing developers.













On Sun, Oct 30, 2011 at 9:38 PM, Madovsky <infos at madovsky.org> wrote:

> Chad, you are welcome to share your code
> welcome to FS community ! :)
>
> ----- Original Message -----
> From: "mazilo" <Nabble_01394 at slickdeals.endjunk.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, October 30, 2011 10:26 PM
> Subject: Re: [Freeswitch-users] Feedback
>
>
> > Anthony,
> >
> > I am sure everyone really appreciates what you have done to contribute
> > such
> > an excellent program to VoIP community. People like Linux Torvalds, and
> > including you, are what I considered a hero with a big heart to support
> > open-source community. Without them, the world will be different today.
> It
> > would probably be an M$ dominated world.
> >
> > I guess what I tried to say here is it really is very disturbing to see
> > your
> > comments below. The least you could have done to show our community was a
> > positive acceptance to embrace someone's effort to legitimately port FS
> to
> > other platforms in other programming languages. If Chad wanted to hi-jack
> > FS, I don't suppose he would even dare to represent himself here. BTw,
> can
> > you imagine if Linus Torvalds behaves like this?
> >
> > I hope you don't take my comments in a negative direction. However, like
> > anyone who reads your comments below, I won't be surprised if you decide
> > to
> > head in a negative direction to flame my comments. Regardless, I
> sincerely
> > wish you the best to come, especially with the contribution from Chad.
> >
> >
> > Anthony Minessale wrote:
> >>
> >> One thing that is disturbing is that after all the time and effort I put
> >> into making FreeSWITCH cross platform, the very idea of such an
> >> initiative
> >> wipes all that out in favor of one platform.  So even if you are willing
> >> to
> >> share it, and it's not B.S. its bothersome to me that:
> >>
> >> 1) I spent 6 years making it work everywhere and you aim to put all your
> >> extended efforts into steering it to a separate version
> >> 2) Making it use less open source libs is not really a good thing since
> >> its
> >> an open source project.
> >> 3) You did not propose anything to us first, that's a bit disrespectful.
> >>
> >> I think about everybody who uses FreeSWITCH when I make decisions, and
> >> this
> >> just feels selfish to me, (probably not for windows users)  but hey....
> >>
> >> On the other hand, Henry has a great point, that looks like a pretty big
> >> list of changes to do in a few weeks considering you were only asking
> >> trivial newbie questions as recently as sept 18th and your website is
> the
> >> generic IIS document root.   I would be pretty concerned about vishing
> if
> >> I
> >> were a reader of this post.
> >>
> >> I guess seeing is believing and if you are legit our windows-friendly
> >> developers will let me know.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> On Fri, Oct 28, 2011 at 1:55 PM, Chad Vogel &lt;cvogel@&gt; wrote:
> >>
> >>>  Hello,
> >>>
> >>> I wanted to reach out and get some feedback from everyone because my
> >>> company started a project using FS as base several weeks ago, our goal
> >>> was
> >>> to optimize and integrate FS more directly into Windows environments.
> >>> Here
> >>> is my question, we would like to share our changes and where should we
> >>> host
> >>> our project? Because we understand our build breaks compatibility with
> >>> the
> >>> UNIX environments and fails to meet the objectives cross platform
> >>> compatibility, however it could contribute greatly and be beneficial to
> >>> other windows users.
> >>>
> >>> Here are some of the changes we have made:
> >>>
> >>>    - We converted FS core to C++ and having it compiling to .NET 4.0
> CLI
> >>>    - FS now runs inside its own namespace
> >>>    - We replace FS_CLI with a PowerShell shell app
> >>>    - Supports SQL Server support via the Native SQL Server Client API
> >>>    - Support for windows clustering (up to 32 node active/active
> >>> cluster)
> >>>    - Ties more directly into the Win32 API and has less reliance on
> open
> >>>    source Libs
> >>>    - Replaces OpenSSL sockets with Windows encrypted sockets
> >>>    - Added windows performance monitoring
> >>>    - No longer need to use mod_managed for managed modules
> >>>    - Memory management relies on .Net garbage collection
> >>>    - Added support for Microsoft Speech
> >>>    - VoiceXML 3.0 and SCXML support
> >>>    - No longer supports JS, LUA, PHP development - Modules can be only
> >>>    developed in .NET, C or C++( we feel support for other languages can
> >>> be
> >>>    added back in but falls outside the scope of our objectives at this
> >>> time,
> >>>    support for SCXML should make limit the need for other scripting
> >>> languages)
> >>>    - Fixed RTP clock timing issues in virtualized environments
> >>>    - New configuration file format using .Net App Configuration files
> >>>    (kind of looks like an IIS config file)
> >>>
> >>>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at .freeswitch
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale@
> >> GTALK/JABBER/PAYPAL:anthony.minessale@
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at .freeswitch
> >> googletalk:conf+888 at .freeswitch
> >> pstn:+19193869900
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at .freeswitch
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > -----
> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> > Watts of electricity.
> > --
> > View this message in context:
> >
> http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p6946803.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 Hector.Geraldino at ip-soft.net  Mon Oct 31 18:47:42 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Mon, 31 Oct 2011 11:47:42 -0400
Subject: [Freeswitch-users] hangup/disconnect on transfer
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02249DBB7A@NY1-EXMB-01.ip-soft.net>

Hi,

I'm facing a problem that I'm not even sure that belongs to the FS boundaries but I'm asking anyway: I need to know when the legB of the call transfers the legA to another extension. Here's the scenario:

PSTN -> Cisco Unified CM -> Sip Trunk (FreeSwitch) -> ESL application

An incoming call is effectively bounded by a Cisco Call Manager platform to a SIP trunk managed by FreeSwitch, and controlled by a Java application. I can answer the call, play prompts and do some more stuff. After that, I'm originating a call to another extension (managed by the CUCM, not freeswitch) to be answered by an agent, and bridging both legs using uuid_bridge. So what I've is a classic call center scenario where I connect incoming calls to agents, and I calculate the call handling time based on the time between the call was bridged to an operator and the time the call was dropped because one of the two legs hanged up. Also after receiving the disconnection notice I perform some cleanup tasks. Great. So far so good.

But, there's a problem that begins when the agent (legB) transfers the call to another extension (or even an external phone number) using the transfer function on their IP Phone. As this phone is not 'controlled' by FreeSwitch there's no event registered for this call transfer. So what is happening is that both legs are still connected to FS even when the agent (legB) has transferred the call. This off course is an undesired behavior, because I'm saying that 'agentB spent 30 minutes handling call X' when he actually transferred this call to someone else after maybe just 20 seconds.

So what I need is an event or something that helps me to detect when a call has been transferred, not using the 'transfer' tool on freeswitch but transferred outside FreeSwitch. The only notification that FreeSwitch receives after a call transfer is performed is this SIP message (with its response):


recv 805 bytes from udp/[192.168.23.104]:5060 at 14:40:04.945886:
   ------------------------------------------------------------------------
   UPDATE sip:mod_sofia at 192.168.5.72:5060 SIP/2.0
   Via: SIP/2.0/UDP 192.168.23.104:5060;branch=z9hG4bK78836e9096b
   From: <sip:5562 at 192.168.23.104>;tag=63092ed1-42e0-4489-9bb9-bd2b77b7fcf3-28767966
   To: "" <sip:5512 at 192.168.5.72>;tag=cFQ5y9jHy0a0p
   Date: Mon, 31 Oct 2011 14:39:41 GMT
   Call-ID: e062f678-7e70-122f-f890-000c298202db
   User-Agent: Cisco-CUCM8.0
   Max-Forwards: 70
   Supported: timer,resource-priority,replaces
   Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
   CSeq: 101 UPDATE
   Contact: <sip:5562 at 192.168.23.104:5060>
   Supported: X-cisco-srtp-fallback
   Supported: Geolocation
   P-Asserted-Identity: "Leona C" <sip:5655 at 192.168.23.104>
   Remote-Party-ID: "Leona C" <sip:5655 at 192.168.23.104>;party=calling;screen=yes;privacy=off
   Content-Length: 0

   ------------------------------------------------------------------------
send 624 bytes to udp/[192.168.23.104]:5060 at 14:40:04.946088:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.23.104:5060;branch=z9hG4bK78836e9096b
   From: <sip:5562 at 192.168.23.104>;tag=63092ed1-42e0-4489-9bb9-bd2b77b7fcf3-28767966
   To: "" <sip:5512 at 192.168.5.72>;tag=cFQ5y9jHy0a0p
   Call-ID: e062f678-7e70-122f-f890-000c298202db
   CSeq: 101 UPDATE
   Contact: <sip:mod_sofia at 192.168.5.72:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Session-Expires: 1800;refresher=uas
   Min-SE: 120
   Content-Length: 0

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


Is there any way to react to this incoming message and generate an event or action? A channel variable or something that I may be missing? Any thoughts?

Thanks for your kindly attention,
Hector


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From boris at tagnet.ru  Mon Oct 31 19:42:37 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 31 Oct 2011 22:42:37 +0600
Subject: [Freeswitch-users] Linksys PAP2-T and echo
In-Reply-To: <3B1EABFB29D34F61BC96746169E2E2C4@DWP>
References: <4EAEB77C.9060901@tagnet.ru> <4EAEBAB7.1030405@coppice.org>
	<3B1EABFB29D34F61BC96746169E2E2C4@DWP>
Message-ID: <4EAECFFD.2090502@tagnet.ru>

Hello!

Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo 
acceptable but I still can hear it. So there is no way to eleminate echo 
with Linksys devices? Only to buy more professional devices like Addpac?
> on your pap2t web interface, go to voice, advanced, select the line, the
> settings below will allow you to play with echo cancellation.  However, you
> also need to go the the regional tab in voice and play with the fxs input
> and output gains, the voice gain on the pap2t or any linksys gateway, will
> impact the echo on the line.  This is most likely where your problem is.
> You might not be able to cure the echo on every call, but you should be able
> to make it acceptable.
>
> Darcy
>
>
> Audio Configuration
> Preferred Codec:        Second Preferred Codec:
> Third Preferred Codec:        Use Pref Codec Only:
> Silence Supp Enable:        Silence Threshold:
> G729a Enable:        Echo Canc Enable:
> G723 Enable:        Echo Canc Adapt Enable:
> G726-16 Enable:        Echo Supp Enable:
> G726-24 Enable:        FAX CED Detect Enable:
> G726-32 Enable:        FAX CNG Detect Enable:
> G726-40 Enable:        FAX Passthru Codec:
> DTMF Process INFO:        FAX Codec Symmetric:
> DTMF Process AVT:        FAX Passthru Method:
> DTMF Tx Method:        DTMF Tx Mode:
> DTMF Tx Strict Hold Off Time:        FAX Process NSE:
> Hook Flash Tx Method:        FAX Disable ECAN:
> Release Unused Codec:        FAX Enable T38:
> FAX T38 Redundancy:        FAX Tone Detect Mode:
>
> -----Original Message-----
> From: Steve Underwood
> Sent: Monday, October 31, 2011 11:11 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo
>
> On 10/31/2011 10:58 PM, Boris Kovalenko wrote:
>> Hello!
>>
>>
>>        I'm wondering if it is possible to solve echo with Linksys PAP2-T
>> devices. Does anybody here uses it? I read many articles and still can't
>> understand how to eleminate echo :( I read
>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found
>> that Asterisk has its own EC. Does Freeswitch has it too? I found this
>> article http://docs.freeswitch.org/echo_can_page.html but can't
>> understand - has Freeswitch EC or not? If yes - how to turn it on and off?
> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo
> cancellation over IP is very problematic, and hardly ever attempted. The
> PAP2T should be echo cancelling for itself, and they usually do a fairly
> good job of this. I don't think it is configurable. Its always on.
>
> 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From darcy at primrose.ws  Mon Oct 31 20:16:41 2011
From: darcy at primrose.ws (Darcy)
Date: Mon, 31 Oct 2011 13:16:41 -0400
Subject: [Freeswitch-users] Linksys PAP2-T and echo
In-Reply-To: <4EAECFFD.2090502@tagnet.ru>
References: <4EAEB77C.9060901@tagnet.ru>
	<4EAEBAB7.1030405@coppice.org><3B1EABFB29D34F61BC96746169E2E2C4@DWP>
	<4EAECFFD.2090502@tagnet.ru>
Message-ID: <FC2974685DF942C2A4A4369C4D4734BB@DWP>

Here are the settings from a gateway in the field, we have about 200 of them 
installed with no echo compliants

fax port gain, in and out   -2
set
echo canc enable yes
echo canc adapt enable yes
echo supp eanble yes

Also, set silence supp enable to yes
   and silence threshold to medium.

Give that a try, I find it works quite well.  These adapters are service by 
a freeswitch with a 729 license.


-----Original Message----- 
From: Boris Kovalenko
Sent: Monday, October 31, 2011 12:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo

Hello!

Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo
acceptable but I still can hear it. So there is no way to eleminate echo
with Linksys devices? Only to buy more professional devices like Addpac?
> on your pap2t web interface, go to voice, advanced, select the line, the
> settings below will allow you to play with echo cancellation.  However, 
> you
> also need to go the the regional tab in voice and play with the fxs input
> and output gains, the voice gain on the pap2t or any linksys gateway, will
> impact the echo on the line.  This is most likely where your problem is.
> You might not be able to cure the echo on every call, but you should be 
> able
> to make it acceptable.
>
> Darcy
>
>
> Audio Configuration
> Preferred Codec:        Second Preferred Codec:
> Third Preferred Codec:        Use Pref Codec Only:
> Silence Supp Enable:        Silence Threshold:
> G729a Enable:        Echo Canc Enable:
> G723 Enable:        Echo Canc Adapt Enable:
> G726-16 Enable:        Echo Supp Enable:
> G726-24 Enable:        FAX CED Detect Enable:
> G726-32 Enable:        FAX CNG Detect Enable:
> G726-40 Enable:        FAX Passthru Codec:
> DTMF Process INFO:        FAX Codec Symmetric:
> DTMF Process AVT:        FAX Passthru Method:
> DTMF Tx Method:        DTMF Tx Mode:
> DTMF Tx Strict Hold Off Time:        FAX Process NSE:
> Hook Flash Tx Method:        FAX Disable ECAN:
> Release Unused Codec:        FAX Enable T38:
> FAX T38 Redundancy:        FAX Tone Detect Mode:
>
> -----Original Message-----
> From: Steve Underwood
> Sent: Monday, October 31, 2011 11:11 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo
>
> On 10/31/2011 10:58 PM, Boris Kovalenko wrote:
>> Hello!
>>
>>
>>        I'm wondering if it is possible to solve echo with Linksys PAP2-T
>> devices. Does anybody here uses it? I read many articles and still can't
>> understand how to eleminate echo :( I read
>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found
>> that Asterisk has its own EC. Does Freeswitch has it too? I found this
>> article http://docs.freeswitch.org/echo_can_page.html but can't
>> understand - has Freeswitch EC or not? If yes - how to turn it on and 
>> off?
> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo
> cancellation over IP is very problematic, and hardly ever attempted. The
> PAP2T should be echo cancelling for itself, and they usually do a fairly
> good job of this. I don't think it is configurable. Its always on.
>
> 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



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




From steveu at coppice.org  Mon Oct 31 20:41:30 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 01 Nov 2011 01:41:30 +0800
Subject: [Freeswitch-users] Linksys PAP2-T and echo
In-Reply-To: <4EAECFFD.2090502@tagnet.ru>
References: <4EAEB77C.9060901@tagnet.ru>
	<4EAEBAB7.1030405@coppice.org>	<3B1EABFB29D34F61BC96746169E2E2C4@DWP>
	<4EAECFFD.2090502@tagnet.ru>
Message-ID: <4EAEDDCA.4070907@coppice.org>

Hi,

The echo canceller in the SPA3102 is a disaster, but the ones in the 
PAP2T and SPA2102 shouldn't give you problems. If you use the default 
gains you shouldn't really notice echo. If you set some wild gains you 
might bee poor results. Are you sure the echo originates from the PAP2T, 
and not some other part of the signal chain?

Steve


On 11/01/2011 12:42 AM, Boris Kovalenko wrote:
> Hello!
>
> Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo
> acceptable but I still can hear it. So there is no way to eleminate echo
> with Linksys devices? Only to buy more professional devices like Addpac?
>> on your pap2t web interface, go to voice, advanced, select the line, the
>> settings below will allow you to play with echo cancellation.  However, you
>> also need to go the the regional tab in voice and play with the fxs input
>> and output gains, the voice gain on the pap2t or any linksys gateway, will
>> impact the echo on the line.  This is most likely where your problem is.
>> You might not be able to cure the echo on every call, but you should be able
>> to make it acceptable.
>>
>> Darcy
>>
>>
>> Audio Configuration
>> Preferred Codec:        Second Preferred Codec:
>> Third Preferred Codec:        Use Pref Codec Only:
>> Silence Supp Enable:        Silence Threshold:
>> G729a Enable:        Echo Canc Enable:
>> G723 Enable:        Echo Canc Adapt Enable:
>> G726-16 Enable:        Echo Supp Enable:
>> G726-24 Enable:        FAX CED Detect Enable:
>> G726-32 Enable:        FAX CNG Detect Enable:
>> G726-40 Enable:        FAX Passthru Codec:
>> DTMF Process INFO:        FAX Codec Symmetric:
>> DTMF Process AVT:        FAX Passthru Method:
>> DTMF Tx Method:        DTMF Tx Mode:
>> DTMF Tx Strict Hold Off Time:        FAX Process NSE:
>> Hook Flash Tx Method:        FAX Disable ECAN:
>> Release Unused Codec:        FAX Enable T38:
>> FAX T38 Redundancy:        FAX Tone Detect Mode:
>>
>> -----Original Message-----
>> From: Steve Underwood
>> Sent: Monday, October 31, 2011 11:11 AM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo
>>
>> On 10/31/2011 10:58 PM, Boris Kovalenko wrote:
>>> Hello!
>>>
>>>
>>>         I'm wondering if it is possible to solve echo with Linksys PAP2-T
>>> devices. Does anybody here uses it? I read many articles and still can't
>>> understand how to eleminate echo :( I read
>>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found
>>> that Asterisk has its own EC. Does Freeswitch has it too? I found this
>>> article http://docs.freeswitch.org/echo_can_page.html but can't
>>> understand - has Freeswitch EC or not? If yes - how to turn it on and off?
>> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo
>> cancellation over IP is very problematic, and hardly ever attempted. The
>> PAP2T should be echo cancelling for itself, and they usually do a fairly
>> good job of this. I don't think it is configurable. Its always on.
>>
>> 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 jchavanton at gmail.com  Mon Oct 31 21:44:00 2011
From: jchavanton at gmail.com (Julien Chavanton)
Date: Mon, 31 Oct 2011 14:44:00 -0400
Subject: [Freeswitch-users] bridging calls from different sofia profiles
In-Reply-To: <CAKmcL2nrF0gyzOUWO+kWBQe3Q0jM=ucw6Z3mw6=gqn6G-eSpGg@mail.gmail.com>
References: <CAKmcL2nrF0gyzOUWO+kWBQe3Q0jM=ucw6Z3mw6=gqn6G-eSpGg@mail.gmail.com>
Message-ID: <CAKmcL2n+e_RFPKawq2tPVD8b-kvsQEnb=av++DSKJ6d7mFWx7w@mail.gmail.com>

Hi, I have audio problems when bridging calls from different sofia profiles.
I use 2 profiles in order to connect softphone through a VPN to avoid NAT
problem
One profile is sending/listening on a private IP (for softphone) and the
other profile is using a public IP (for external PSTN gateways)

I have tested with X-lite and Ekiga and I face different one way audio with
both:
With X-lite the audio received from the private IP is not forwarded
While with Ekiga the audio from the public IP profile is not forwarded

I have traced SIP and RTP and the signaling looks good. RTP is received by
FS on the correct socket but not forwarded.

I understand I do not provide enough information to diagnostic, but maybe I
could get input on how to  diagnostic RTP socket etc. in this scenario ?
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From vellayappan.n at mobax.com  Mon Oct 31 09:35:55 2011
From: vellayappan.n at mobax.com (Vellayappan.N)
Date: Mon, 31 Oct 2011 12:05:55 +0530
Subject: [Freeswitch-users] No caller id & DTMF detection fail issue
Message-ID: <CAH4nPoyEk+41ASh4yfgQTxqcKP-ufi9dqR8OquS4Lc+aVuhJeg@mail.gmail.com>

Hi,

I am  using  Digium X100P SE single port fxo analog card  for making pstn
calls.  I can able to ring to my SIP number from my mobile/land line
number. But I am not getting the caller id number who is actually calling,
instead I am getting only "0000000000"  always.  My service provider is
Airtel and I am in India. Freeswitch is detectling only few (two or three)
dtmf digits.   Can you please help me to get the incoming caller-id number
while calling from  PSTN number.

-- 
Regards
Vellayappan
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From vellayappan.n at mobax.com  Mon Oct 31 09:45:38 2011
From: vellayappan.n at mobax.com (Vellayappan.N)
Date: Mon, 31 Oct 2011 12:15:38 +0530
Subject: [Freeswitch-users] Phantom Calls
In-Reply-To: <754CEFADFC65654BAC484896D31C2C030566AC0D76@exchangesrv2007.ONEFONE.MSDOMAIN>
References: <754CEFADFC65654BAC484896D31C2C030566AC0D76@exchangesrv2007.ONEFONE.MSDOMAIN>
Message-ID: <CAH4nPoy0+2z8_G62pPg0y5XFpoNyZizCTRavCNS0wM=p_gnjug@mail.gmail.com>

Hi

I am also having the same issue. In my case all the calls have the same
issue. When I calling from my mobile to SIP phone in freeswtich,  it is
ringing to the registered sip number.  But  it continues to ring the sip
number even when i disconnect call in my mobile number.  Also the
registered SIP is automatically start gets ringing with in regular
interval  until the freeswitch is restarted.

Please let me know if you have addressed this issue.  Thanks in advance

.



On Wed, Oct 26, 2011 at 8:00 PM, Nir Cohen <nirc at fring.com> wrote:

> Hello, ****
>
> ** **
>
> I'm new to freeswitch. Even though I've been using it for quite a while, I
> hadn't had the chance to really dig into the config/monitoring/debugging.*
> ***
>
> ** **
>
> I have a freeswitch which runs flawlessly with a large number of calls but
> a small percentage of them do not disconnect at all until the freeswitch
> service is restarted.****
>
> I was wondering if there's some timeout which I need to define or if
> there's a way to see a direct reason for this to happen.****
>
> ** **
>
> **-          **I know they are stuck because I did "show calls" and saw
> calls from different times during the time it was up (see below)****
>
> **-          **I removed the freeswitch from the DNS to make sure it's
> not receiving anymore calls.****
>
> **-          **I opened wireshark and saw that the same amount of calls
> are referred to as "session in progress" ? that is, my fs sends a "Status:
> 183 session progress?" and the app server returns: (ICMP) "destination
> unreachable".****
>
> ** **
>
> Thank you.****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
>
> call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid
> ****
>
> 2011-10-09
> 15:10:08,1318165808,switch_ivr_signal_bridge,1001,1001,r0018095215036,sofia/external/
> 1001 at 172.16.100.155
> ,edac000a-bc5a-8642-aba8-ed3be16cd762,1001,fring,18095215036,sofia/external/18095215036,487fbf53-4d6c-134e-93bc-31cc3ee1baf8
> ****
>
> 2011-10-09
> 23:34:57,1318196097,switch_ivr_signal_bridge,1001,1001,r006498287601,sofia/external/
> 1001 at 172.16.100.155
> ,5d4b4b67-def4-2c44-addc-73f3366d6f83,1001,fring,6498287601,sofia/external/6498287601,6087cfe3-e9b6-de44-b38e-bc86d83ff95a
> ****
>
> 2011-10-09
> 23:38:30,1318196310,switch_ivr_signal_bridge,1001,1001,r0020472321728,sofia/external/
> 1001 at 172.16.100.155
> ,9913ff13-33ac-9d4b-b601-d761e49b81b6,1001,fring,20472321728,sofia/external/20472321728,954fff0a-4b7c-8444-8821-306dc6729e42
> ****
>
> 2011-10-09
> 23:41:09,1318196469,switch_ivr_signal_bridge,1001,1001,r0020108866522,sofia/external/
> 1001 at 172.16.100.155
> ,f6590d4f-bda0-2146-a309-f2e485ff231c,1001,fring,20108866522,sofia/external/20108866522,ba1f72d8-49fc-5145-8376-87e9f4a3b3b4
> ****
>
> 2011-10-10
> 02:16:36,1318205796,switch_ivr_signal_bridge,1001,1001,r0050687102330,sofia/external/
> 1001 at 172.16.100.155
> ,c48e48c1-590d-f84a-a34a-e2da40574d59,1001,fring,50687102330,sofia/external/50687102330,95c4c053-c925-5c42-9ac8-87cace55694b
> ****
>
> 2011-10-10
> 06:09:18,1318219758,switch_ivr_signal_bridge,1001,1001,r00919744973603,sofia/external/
> 1001 at 172.16.100.155
> ,04127975-c3f9-b949-907e-f915197bf4fc,1001,fring,919744973603,sofia/external/919744973603,f86302d2-e7ab-0940-b927-b673544c7241
> ****
>
> 2011-10-10
> 06:21:24,1318220484,switch_ivr_signal_bridge,1001,1001,r00917204674056,sofia/external/
> 1001 at 172.16.100.155
> ,c835ad85-9e04-b84d-993e-426261686cc7,1001,fring,917204674056,sofia/external/917204674056,1304eae6-4def-c743-9214-89991e6f3350
> ****
>
> 2011-10-10
> 06:53:04,1318222384,switch_ivr_signal_bridge,1001,1001,r00919715155054,sofia/external/
> 1001 at 172.16.100.155
> ,c4082f14-7cae-e846-9b05-65a519320619,1001,fring,919715155054,sofia/external/919715155054,16fa4e97-eb15-d24b-907d-f39a7741b493
> ****
>
> 2011-10-10
> 07:26:37,1318224397,switch_ivr_signal_bridge,1001,1001,r00919215900646,sofia/external/
> 1001 at 172.16.100.155
> ,40a79a56-cf31-824f-bd47-0006f4abb59b,1001,fring,919215900646,sofia/external/919215900646,f2a02bce-d25a-f448-af39-fa60f3509f51
> ****
>
> 2011-10-10
> 09:38:45,1318232325,switch_ivr_signal_bridge,1001,1001,r00919779210235,sofia/external/
> 1001 at 172.16.100.155
> ,3249aa91-d99f-8947-bcd4-749e199ecfbb,1001,fring,919779210235,sofia/external/919779210235,6b55ec96-6fe2-d047-a349-efd44df00c67
> ****
>
> 2011-10-10
> 13:22:42,1318245762,switch_ivr_signal_bridge,1001,1001,r00919312238801,sofia/external/
> 1001 at 172.16.100.155
> ,1226ffc9-f338-8b42-be8a-93868dc1ed0a,1001,fring,919312238801,sofia/external/919312238801,f63b997f-2371-2441-b0f3-cfcf0330c87a
> ****
>
> 2011-10-10
> 13:37:31,1318246651,switch_ivr_signal_bridge,1001,1001,r0091978999666,sofia/external/
> 1001 at 172.16.100.155
> ,3bf75841-ef52-8244-9c18-5ff783ad5bd4,1001,fring,91978999666,sofia/external/91978999666,8de43ba6-ff36-bf40-941f-a71d3574a95d
> ****
>
> 2011-10-10
> 14:15:44,1318248944,switch_ivr_signal_bridge,1001,1001,r00919855283870,sofia/external/
> 1001 at 172.16.100.155
> ,a1844ff2-c9f7-eb4b-a9ac-4e31d5c5a591,1001,fring,919855283870,sofia/external/919855283870,5c73749d-3f35-aa46-a13e-9c1b5d78c592
> ****
>
> 2011-10-10
> 14:56:30,1318251390,switch_ivr_signal_bridge,1001,1001,r00919891882161,sofia/external/
> 1001 at 172.16.100.155
> ,9b087d32-2a87-db4f-ac66-3bb91d7afdeb,1001,fring,919891882161,sofia/external/919891882161,0972c900-805d-f949-b47a-fb1398a5739d
> ****
>
> 2011-10-10
> 16:31:56,1318257116,switch_ivr_signal_bridge,1001,1001,r00919810399899,sofia/external/
> 1001 at 172.16.100.155
> ,c8b382a4-81ce-6740-bb5d-ebc9ab37d388,1001,fring,919810399899,sofia/external/919810399899,84313caf-fe39-5f45-bf4a-d1cc5e575eff
> ****
>
> 2011-10-10
> 16:47:27,1318258047,switch_ivr_signal_bridge,1001,1001,r0096597284481,sofia/external/
> 1001 at 172.16.100.155
> ,b8d93c17-97ee-074e-84c9-103e5259ddb1,1001,fring,96597284481,sofia/external/96597284481,1dee8a27-b3f1-e246-99f6-baca4cea0dd6
> ****
>
> 2011-10-10
> 20:11:30,1318270290,switch_ivr_signal_bridge,1001,1001,r00967771771821,sofia/external/
> 1001 at 172.16.100.155
> ,16462169-cb2f-6b42-bf01-64e43a8e4311,1001,fring,967771771821,sofia/external/967771771821,0a98e975-6bd1-4948-8a99-2b682935e893
> ****
>
> 2011-10-11
> 01:37:56,1318289876,switch_ivr_signal_bridge,1001,1001,r00584120208712,sofia/external/
> 1001 at 172.16.100.155
> ,7566612f-e7f5-a74a-848b-fa52db667bc1,1001,fring,584120208712,sofia/external/584120208712,bec24956-f8cc-9e46-a1b1-e5f75f827306
> ****
>
> 2011-10-11
> 02:23:53,1318292633,switch_ivr_signal_bridge,1001,1001,r00919815288834,sofia/external/
> 1001 at 172.16.100.155
> ,0c7c07a8-ed17-3e43-9d00-5928b998ac48,1001,fring,919815288834,sofia/external/919815288834,11c3f91e-4e84-7d4c-93d9-fbadbc0e0d69
> ****
>
> 2011-10-11
> 04:30:11,1318300211,switch_ivr_signal_bridge,1001,1001,r00919924392230,sofia/external/
> 1001 at 172.16.100.155
> ,63514e60-511f-5748-b1ed-6ebcc41d5878,1001,fring,919924392230,sofia/external/919924392230,b465f0f0-f0f5-fb4c-bb0f-93b23b59a4cf
> ****
>
> 2011-10-11
> 05:05:39,1318302339,switch_ivr_signal_bridge,1001,1001,r00919703570642,sofia/external/
> 1001 at 172.16.100.155
> ,d6181187-6d71-844a-a4cd-b8add2393c0b,1001,fring,919703570642,sofia/external/919703570642,ed78294e-33f5-7d4d-af4b-40e0efbc9d17
> ****
>
> 2011-10-11
> 07:24:57,1318310697,switch_ivr_signal_bridge,1001,1001,r00919845045625,sofia/external/
> 1001 at 172.16.100.155
> ,781823e0-1496-2f46-b6bc-a98cee121887,1001,fring,919845045625,sofia/external/919845045625,bd19f919-4b3c-3342-a23f-53088baedc8f
> ****
>
> 2011-10-11
> 07:55:46,1318312546,switch_ivr_signal_bridge,1001,1001,r0084090,sofia/external/
> 1001 at 172.16.100.155
> ,8edd5195-a573-4042-adf7-2c7d093b8178,1001,fring,84090,sofia/external/84090,7e83eb35-d155-384a-ab22-4545e0bb62bb
> ****
>
> 2011-10-11
> 10:11:50,1318320710,switch_ivr_signal_bridge,1001,1001,r009197845827258,sofia/external/
> 1001 at 172.16.100.155
> ,0dbff49d-e8cd-f44d-ab68-f0d5babd61a8,1001,fring,9197845827258,sofia/external/9197845827258,d11b3020-6def-2849-aac0-7454c25d5077
> ****
>
> 2011-10-11
> 11:28:26,1318325306,switch_ivr_signal_bridge,1001,1001,r00996550675305,sofia/external/
> 1001 at 172.16.100.155
> ,9354bc8e-100d-3e4e-972a-5814fde9ce4c,1001,fring,996550675305,sofia/external/996550675305,cf2662b9-3be1-a440-91f5-7d3088d574bb
> ****
>
> 2011-10-11
> 16:43:55,1318344235,switch_ivr_signal_bridge,1001,1001,r00919895419251,sofia/external/
> 1001 at 172.16.100.155
> ,69dab1ec-9d20-4c4d-a8b8-036cc2be4382,1001,fring,919895419251,sofia/external/919895419251,db22ba2d-dff2-1544-a67e-578886921592
> ****
>
> 2011-10-11
> 17:48:39,1318348119,switch_ivr_signal_bridge,1001,1001,r00919773212368,sofia/external/
> 1001 at 172.16.100.155
> ,7a35e17e-9604-4743-b131-d339e08245d0,1001,fring,919773212368,sofia/external/919773212368,0cce04b4-3776-4c4f-adf8-e42a1555d33a
> ****
>
> 2011-10-12
> 01:57:34,1318377454,switch_ivr_signal_bridge,1001,1001,r0023794311238,sofia/external/
> 1001 at 172.16.100.155
> ,4a09bce4-5342-3943-9b36-a9bd7c51d8bc,1001,fring,23794311238,sofia/external/23794311238,ede10bc2-45b2-1240-a20c-9b3e65868823
> ****
>
> 2011-10-12
> 05:39:54,1318390794,switch_ivr_signal_bridge,1001,1001,r00918121275024,sofia/external/
> 1001 at 172.16.100.155
> ,4c991cad-f71d-e644-9946-8fff485c9a40,1001,fring,918121275024,sofia/external/918121275024,4b3e9c9f-bb26-814e-89e5-0685c05692df
> ****
>
> 2011-10-12
> 06:39:05,1318394345,switch_ivr_signal_bridge,1001,1001,r00919947389305,sofia/external/
> 1001 at 172.16.100.155
> ,d2a0d36e-182e-bc47-8fe2-83032467eff3,1001,fring,919947389305,sofia/external/919947389305,e5d52ab2-2c36-1843-bea8-0cfdce391e15
> ****
>
> 2011-10-12
> 06:59:34,1318395574,switch_ivr_signal_bridge,1001,1001,r00865395760139,sofia/external/
> 1001 at 172.16.100.155
> ,03a603a0-6418-7a46-a1d4-5d4d2357bf23,1001,fring,865395760139,sofia/external/865395760139,ef325309-716c-1246-8f01-513c7e9a93bb
> ****
>
> 2011-10-12
> 08:49:22,1318402162,switch_ivr_signal_bridge,1001,1001,r00919944731940,sofia/external/
> 1001 at 172.16.100.155
> ,024d66fc-6582-b04b-a962-56c4db8cb013,1001,fring,919944731940,sofia/external/919944731940,1de9cb6b-4eed-3d4d-89bc-6b533b07849a
> ****
>
> 2011-10-12
> 09:37:03,1318405023,switch_ivr_signal_bridge,1001,1001,r00919916780545,sofia/external/
> 1001 at 172.16.100.155
> ,379b8c6e-a998-2649-a577-5f8f1b6b3c4e,1001,fring,919916780545,sofia/external/919916780545,097c1a46-1e65-6749-aa14-592b1cae6fd5
> ****
>
> 2011-10-12
> 10:17:08,1318407428,switch_ivr_signal_bridge,1001,1001,r00914562224442,sofia/external/
> 1001 at 172.16.100.155
> ,f91ec6f2-e8a1-b549-86b5-cbb1d17039b3,1001,fring,914562224442,sofia/external/914562224442,3c2ced4c-7b98-4948-8520-8bc357e4a437
> ****
>
> 2011-10-12
> 13:54:07,1318420447,switch_ivr_signal_bridge,1001,1001,r00919746980086,sofia/external/
> 1001 at 172.16.100.155
> ,6f1a25be-891e-ee4f-8c70-4ae79694a651,1001,fring,919746980086,sofia/external/919746980086,d0fdad10-f3a2-5347-ab1f-95d4383089b7
> ****
>
> 2011-10-12
> 21:04:21,1318446261,switch_ivr_signal_bridge,1001,1001,r0096597280025,sofia/external/
> 1001 at 172.16.100.155
> ,3349eb67-9f4b-f64f-98af-854dd9a07c23,1001,fring,96597280025,sofia/external/96597280025,14d7da2f-d2e6-084c-b5d3-375377400832
> ****
>
> 2011-10-13
> 08:38:46,1318487926,switch_ivr_signal_bridge,1001,1001,r00639214286751,sofia/external/
> 1001 at 172.16.100.155
> ,9b909f83-2d31-a145-82bb-1c0dd1f46695,1001,fring,639214286751,sofia/external/639214286751,6bd8bfbf-4391-1e48-ae3d-16c7f81c74a9
> ****
>
> 2011-10-13
> 09:28:42,1318490922,switch_ivr_signal_bridge,1001,1001,r00919855874737,sofia/external/
> 1001 at 172.16.100.155
> ,ff74debc-509c-2040-bf50-42313aa6629d,1001,fring,919855874737,sofia/external/919855874737,579ecfbf-d00c-7947-9c2f-6029dbf9b507
> ****
>
> 2011-10-13
> 10:44:23,1318495463,switch_ivr_signal_bridge,1001,1001,r00919966842195,sofia/external/
> 1001 at 172.16.100.155
> ,16e5bb3f-5ff1-0743-b787-78fbfb032583,1001,fring,919966842195,sofia/external/919966842195,77d980c9-4ccb-1c40-a901-3e6fd6960a2d
> ****
>
> 2011-10-13
> 11:54:10,1318499650,switch_ivr_signal_bridge,1001,1001,r00919751864788,sofia/external/
> 1001 at 172.16.100.155
> ,84725b33-97ab-d64f-8a36-b7f10a2b8723,1001,fring,919751864788,sofia/external/919751864788,f7e86569-01a2-5540-9c9b-3e300f1a0797
> ****
>
> 2011-10-13
> 14:26:04,1318508764,switch_ivr_signal_bridge,1001,1001,r00919922440249,sofia/external/
> 1001 at 172.16.100.155
> ,ad0c00ab-17c2-4949-880b-9ff164941f1b,1001,fring,919922440249,sofia/external/919922440249,1d139c8d-82eb-9248-8e81-e9d6315ba50d
> ****
>
> 2011-10-13
> 16:10:48,1318515048,switch_ivr_signal_bridge,1001,1001,r0020197596666,sofia/external/
> 1001 at 172.16.100.155
> ,ad64dcd2-232c-d543-8d01-2a513c3dc315,1001,fring,20197596666,sofia/external/20197596666,7f9b2f6c-41db-3840-ab78-e0052c56a885
> ****
>
> 2011-10-13
> 20:09:02,1318529342,switch_ivr_signal_bridge,1001,1001,r00551158958193,sofia/external/
> 1001 at 172.16.100.155
> ,0730806a-137f-924e-ace6-9852c72bb185,1001,fring,551158958193,sofia/external/551158958193,de18fc0d-d46f-014c-92f1-ff5ac3d69e47
> ****
>
> 2011-10-13
> 21:11:25,1318533085,switch_ivr_signal_bridge,1001,1001,r0020123946125,sofia/external/
> 1001 at 172.16.100.155
> ,74c80972-2dd4-c443-b831-d76003497612,1001,fring,20123946125,sofia/external/20123946125,be927ca2-99cc-4840-aa86-8c9a85e32f80
> ****
>
> 2011-10-13
> 22:43:24,1318538604,switch_ivr_signal_bridge,1001,1001,r00919769404307,sofia/external/
> 1001 at 172.16.100.155
> ,16b1cdf4-c7de-2642-a3d1-a901c7168829,1001,fring,919769404307,sofia/external/919769404307,74860ee3-d592-8441-a738-3b82103033a1
> ****
>
> 2011-10-14
> 03:50:54,1318557054,switch_ivr_signal_bridge,1001,1001,r00919988274760,sofia/external/
> 1001 at 172.16.100.155
> ,378a4ecc-7646-6147-9086-381a2ffc5570,1001,fring,919988274760,sofia/external/919988274760,49650a00-03dc-054a-ae56-2debe46897a4
> ****
>
> 2011-10-14
> 04:13:23,1318558403,switch_ivr_signal_bridge,1001,1001,r00919944010360,sofia/external/
> 1001 at 172.16.100.155
> ,5aec4e1e-ed85-ef41-b9fe-7f7e9666b0da,1001,fring,919944010360,sofia/external/919944010360,39aa0999-6817-1148-98a2-dc8b1cf768c8
> ****
>
> 2011-10-14
> 05:28:12,1318562892,switch_ivr_signal_bridge,1001,1001,r00919944191129,sofia/external/
> 1001 at 172.16.100.155
> ,f4e6f3dc-543a-0a4e-b81d-58c8e5434b0a,1001,fring,919944191129,sofia/external/919944191129,b702994a-2237-a442-89c3-1026966a1feb
> ****
>
> 2011-10-14
> 06:14:54,1318565694,switch_ivr_signal_bridge,1001,1001,r0060148144168,sofia/external/
> 1001 at 172.16.100.155
> ,95f21383-70db-2848-a1a4-bf1e279970cb,1001,fring,60148144168,sofia/external/60148144168,f9c3747a-4c39-aa4d-a655-7cea1222ada9
> ****
>
> 2011-10-14
> 06:40:42,1318567242,switch_ivr_signal_bridge,1001,1001,r008615952583367,sofia/external/
> 1001 at 172.16.100.155
> ,d4976a2b-d218-a14a-ab73-ed5252d23223,1001,fring,8615952583367,sofia/external/8615952583367,d514508a-e4c8-f14f-82fc-43b71560f565
> ****
>
> 2011-10-14
> 13:30:22,1318591822,switch_ivr_signal_bridge,1001,1001,r00919902730542,sofia/external/
> 1001 at 172.16.100.155
> ,535a6926-431e-154b-bed1-bea1ea8240a0,1001,fring,919902730542,sofia/external/919902730542,45a49ed1-b426-5f4d-8da6-5d7ae5934dc9
> ****
>
> 2011-10-14
> 13:52:34,1318593154,switch_ivr_signal_bridge,1001,1001,r008613986690504,sofia/external/
> 1001 at 172.16.100.155
> ,39dd44b8-1854-be40-8cd4-81c34ebbcc11,1001,fring,8613986690504,sofia/external/8613986690504,7d641511-0cd1-c54c-8058-568f7f0f67f0
> ****
>
> 2011-10-14
> 14:44:34,1318596274,switch_ivr_signal_bridge,1001,1001,r0020101,sofia/external/
> 1001 at 172.16.100.155
> ,08408567-1407-ea4a-be73-ce0219738962,1001,fring,20101,sofia/external/20101,aa1cbd89-f2f1-114a-a0a4-2d18fb647a1d
> ****
>
> 2011-10-14
> 19:30:55,1318613455,switch_ivr_signal_bridge,1001,1001,r00919953239015,sofia/external/
> 1001 at 172.16.100.155
> ,931bd3ec-a4a9-1b47-b6a3-7226cf83dc3b,1001,fring,919953239015,sofia/external/919953239015,37b98044-fbe2-e940-a368-fe9556fb810f
> ****
>
> 2011-10-15
> 01:42:58,1318635778,switch_ivr_signal_bridge,1001,1001,r00919032587213,sofia/external/
> 1001 at 172.16.100.155
> ,11537992-801d-eb42-ba4e-763fbff0f381,1001,fring,919032587213,sofia/external/919032587213,4d2e98a0-d4b4-d046-aa6c-11c7d7137c63
> ****
>
> 2011-10-15
> 05:32:32,1318649552,switch_ivr_signal_bridge,1001,1001,r00919885688639,sofia/external/
> 1001 at 172.16.100.155
> ,b2b70ad1-0754-8c49-82eb-c0ae5d051618,1001,fring,919885688639,sofia/external/919885688639,a35b33e7-08c0-f34f-b664-2c1b68be3e95
> ****
>
> 2011-10-15
> 06:40:56,1318653656,switch_ivr_signal_bridge,1001,1001,r00919886690120,sofia/external/
> 1001 at 172.16.100.155
> ,13a08888-c378-d949-8625-95e2239bac45,1001,fring,919886690120,sofia/external/919886690120,ff62df61-8354-0641-a1ce-cba74fe53448
> ****
>
> 2011-10-15
> 09:51:20,1318665080,switch_ivr_signal_bridge,1001,1001,r00265888891340,sofia/external/
> 1001 at 172.16.100.155
> ,41c4263a-18f2-d147-8b7a-3a14739eeb4f,1001,fring,265888891340,sofia/external/265888891340,1c19dbaa-d35c-3e4d-a913-355d2f9b6c0d
> ****
>
> 2011-10-15
> 10:06:41,1318666001,switch_ivr_signal_bridge,1001,1001,r00919820158704,sofia/external/
> 1001 at 172.16.100.155
> ,c1363ab8-2905-2549-876c-dd3cbc05563f,1001,fring,919820158704,sofia/external/919820158704,660b2675-974e-a249-8a24-349020a7449a
> ****
>
> 2011-10-15
> 11:19:21,1318670361,switch_ivr_signal_bridge,1001,1001,r0020197105643,sofia/external/
> 1001 at 172.16.100.155
> ,cc2f1ede-7d0c-8845-b455-4eb97ed2b12f,1001,fring,20197105643,sofia/external/20197105643,81cb54ff-b63d-6c4e-afa9-bfea68a553a9
> ****
>
> 2011-10-15
> 12:14:13,1318673653,switch_ivr_signal_bridge,1001,1001,r00918870183328,sofia/external/
> 1001 at 172.16.100.155
> ,81ba52e8-d157-4441-bfe1-c314e4f1b4c7,1001,fring,918870183328,sofia/external/918870183328,72306599-6017-c54a-a291-9b3eb1650d83
> ****
>
> 2011-10-15
> 14:15:42,1318680942,switch_ivr_signal_bridge,1001,1001,r00919539499709,sofia/external/
> 1001 at 172.16.100.155
> ,b24a486b-b798-c048-b0c5-9c624b0ebade,1001,fring,919539499709,sofia/external/919539499709,456681fa-c2bf-004f-875a-4d4b18ea1e99
> ****
>
> 2011-10-15
> 23:23:12,1318713792,switch_ivr_signal_bridge,1001,1001,r0020140088523,sofia/external/
> 1001 at 172.16.100.155
> ,c2f98b57-0c7b-1b41-a77c-05a34ab99b79,1001,fring,20140088523,sofia/external/20140088523,f681a50f-d263-a942-a48f-ab7eb2adff38
> ****
>
> 2011-10-16
> 02:16:44,1318724204,switch_ivr_signal_bridge,1001,1001,r00919886859264,sofia/external/
> 1001 at 172.16.100.155
> ,764923b2-29b4-fd4d-9de8-cd89f2654a9f,1001,fring,919886859264,sofia/external/919886859264,ecbe5234-bb44-1a40-a93c-4fd36307f3e9
> ****
>
> 2011-10-16
> 06:50:47,1318740647,switch_ivr_signal_bridge,1001,1001,r00919915919879,sofia/external/
> 1001 at 172.16.100.155
> ,27d55f39-86e3-f146-ad9b-2f77938ad868,1001,fring,919915919879,sofia/external/919915919879,2df4fa27-2896-6f4d-81ce-97532b2dee0c
> ****
>
> 2011-10-16
> 10:02:03,1318752123,switch_ivr_signal_bridge,1001,1001,r00965555751124,sofia/external/
> 1001 at 172.16.100.155
> ,7d69745c-89e1-ad49-8f57-ca146ecc769a,1001,fring,965555751124,sofia/external/965555751124,973c1c3b-0f50-684a-873b-b61bdd21a245
> ****
>
> 2011-10-16
> 11:37:30,1318757850,switch_ivr_signal_bridge,1001,1001,r00201001301440,sofia/external/
> 1001 at 172.16.100.155
> ,22fb4d57-3207-4d4c-8be1-145655dad6a5,1001,fring,201001301440,sofia/external/201001301440,397ee409-504c-244c-b11b-b693070ff7dd
> ****
>
> 2011-10-16
> 14:55:05,1318769705,switch_ivr_signal_bridge,1001,1001,r00919742055333,sofia/external/
> 1001 at 172.16.100.155
> ,1817b98a-e036-1746-bd7d-437152ed6b0a,1001,fring,919742055333,sofia/external/919742055333,ab2b04ce-a454-a041-a40e-7ee5ddbb29a4
> ****
>
> 2011-10-16
> 16:20:52,1318774852,switch_ivr_signal_bridge,1001,1001,r00998662245737,sofia/external/
> 1001 at 172.16.100.155
> ,986c8dc4-831b-0041-9b3c-1bee42818ab5,1001,fring,998662245737,sofia/external/998662245737,a1eaeb3c-6b1a-ea4c-a3b1-f3144572bf72
> ****
>
> 2011-10-16
> 16:51:24,1318776684,switch_ivr_signal_bridge,1001,1001,r008801712054706,sofia/external/
> 1001 at 172.16.100.155
> ,84a34d92-8be5-1644-a9cd-487023eee234,1001,fring,8801712054706,sofia/external/8801712054706,1930550c-d3e3-f645-bda9-e4a0d39a6397
> ****
>
> 2011-10-16
> 18:14:22,1318781662,switch_ivr_signal_bridge,1001,1001,r00919390177288,sofia/external/
> 1001 at 172.16.100.155
> ,2878943a-56ee-be47-a5fc-632a92c3734a,1001,fring,919390177288,sofia/external/919390177288,4a8de922-86ab-ba41-8e9c-05a79087a79a
> ****
>
> 2011-10-16
> 20:06:31,1318788391,switch_ivr_signal_bridge,1001,1001,r00201224955898,sofia/external/
> 1001 at 172.16.100.155
> ,fc68f803-781d-c347-a6dc-f50a8651f61f,1001,fring,201224955898,sofia/external/201224955898,1e873400-c1ca-da42-be32-837931d2c845
> ****
>
> 2011-10-16
> 22:52:28,1318798348,switch_ivr_signal_bridge,1001,1001,r00914567242736,sofia/external/
> 1001 at 172.16.100.155
> ,d8b4385a-bcf9-ad4a-ae00-0539fe7c56f8,1001,fring,914567242736,sofia/external/914567242736,5eb0ca19-155a-3047-a480-8f951f26993f
> ****
>
> 2011-10-16
> 23:06:09,1318799169,switch_ivr_signal_bridge,1001,1001,r00966552321650,sofia/external/
> 1001 at 172.16.100.155
> ,f9998988-f7f1-9543-afd5-1cd76e035594,1001,fring,966552321650,sofia/external/966552321650,c01c89f6-212b-5a48-90ce-c79587ecdbac
> ****
>
> 2011-10-17
> 00:38:08,1318804688,switch_ivr_signal_bridge,1001,1001,r00911821277451,sofia/external/
> 1001 at 172.16.100.155
> ,54365caa-8f25-8545-9010-db155e4bdde8,1001,fring,911821277451,sofia/external/911821277451,8df3edfb-3b35-a047-8c8d-51d57eeb56b3
> ****
>
> 2011-10-17
> 06:02:00,1318824120,switch_ivr_signal_bridge,1001,1001,r00919746008642,sofia/external/
> 1001 at 172.16.100.155
> ,ba807588-4536-d94f-81a7-e1d0e8434e05,1001,fring,919746008642,sofia/external/919746008642,6252a3b1-a0a6-ff42-8188-69e771094104
> ****
>
> 2011-10-17
> 06:30:17,1318825817,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/
> 1001 at 172.16.100.155
> ,2e099d56-5e90-2f48-816f-d66e410fc8b9,1001,fring,919824645363,sofia/external/919824645363,7d18d6e9-dfb9-c949-b00b-5a9e62ecc21c
> ****
>
> 2011-10-17
> 07:42:55,1318830175,switch_ivr_signal_bridge,1001,1001,r00919824645363,sofia/external/
> 1001 at 172.16.100.155
> ,54960c11-5352-e34b-ae35-672923413dae,1001,fring,919824645363,sofia/external/919824645363,bbc5cdc4-7a02-2945-b685-28baae5fe280
> ****
>
> 2011-10-17
> 10:11:26,1318839086,switch_ivr_signal_bridge,1001,1001,r00919711534925,sofia/external/
> 1001 at 172.16.100.155
> ,583ffded-6229-6345-bd46-4ef09652a8aa,1001,fring,919711534925,sofia/external/919711534925,9d101a1a-0e4e-7144-bff1-a132378a3d9f
> ****
>
> 2011-10-17
> 17:50:55,1318866655,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/
> 1001 at 172.16.100.155
> ,b6cd78b7-7da4-b44e-a4f4-60e404fafca3,1001,fring,919988175489,sofia/external/919988175489,602bd41d-3f60-d144-b2bb-36d4da232ddc
> ****
>
> 2011-10-17
> 22:52:13,1318884733,switch_ivr_signal_bridge,1001,1001,r00919999233886,sofia/external/
> 1001 at 172.16.100.155
> ,eefd784e-9c04-084f-888b-b0a033a5ee5b,1001,fring,919999233886,sofia/external/919999233886,389a824e-e4c5-b74e-ac8a-9315d9821117
> ****
>
> 2011-10-17
> 23:24:02,1318886642,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/
> 1001 at 172.16.100.155
> ,fcf32cd6-9148-cc40-bc90-534145199ba9,1001,fring,919313467728,sofia/external/919313467728,7d13a986-8454-be4d-a056-c47276ba9f5b
> ****
>
> 2011-10-18
> 00:41:19,1318891279,switch_ivr_signal_bridge,1001,1001,r00919747072485,sofia/external/
> 1001 at 172.16.100.155
> ,25f36536-4606-1942-825f-5bfaf28a66ca,1001,fring,919747072485,sofia/external/919747072485,ee6298c7-a6d4-3547-9578-7c95f96ede0f
> ****
>
> 2011-10-18
> 01:05:08,1318892708,switch_ivr_signal_bridge,1001,1001,r00919313467728,sofia/external/
> 1001 at 172.16.100.155
> ,8c89657f-9ce7-404a-9d5c-a3559f66b141,1001,fring,919313467728,sofia/external/919313467728,b6a3eb92-6419-4643-930f-e7bebeabb33a
> ****
>
> 2011-10-18
> 03:25:04,1318901104,switch_ivr_signal_bridge,1001,1001,r006596112485,sofia/external/
> 1001 at 172.16.100.155
> ,b5189c01-0df2-4e47-af11-8a3e96e247a8,1001,fring,6596112485,sofia/external/6596112485,a834eb93-3744-be49-8165-3b39966c9502
> ****
>
> 2011-10-18
> 06:25:34,1318911934,switch_ivr_signal_bridge,1001,1001,r00919876855863,sofia/external/
> 1001 at 172.16.100.155
> ,d6589f30-ba06-5c45-9d20-1e99664a10d8,1001,fring,919876855863,sofia/external/919876855863,4999eac4-e560-fe40-b795-fc5d5cf8ac44
> ****
>
> 2011-10-18
> 09:41:35,1318923695,switch_ivr_signal_bridge,1001,1001,r00919922965741,sofia/external/
> 1001 at 172.16.100.155
> ,57335341-bda9-3943-8f0d-59c7a7903c15,1001,fring,919922965741,sofia/external/919922965741,d201c220-2c1f-bb4e-b725-bada9895035b
> ****
>
> 2011-10-18
> 12:52:37,1318935157,switch_ivr_signal_bridge,1001,1001,r00919967359493,sofia/external/
> 1001 at 172.16.100.155
> ,66fadc05-a756-a848-bb84-5d39c9998fc5,1001,fring,919967359493,sofia/external/919967359493,71bcd983-d4a4-de4c-b615-f0dbf1095739
> ****
>
> 2011-10-18
> 16:41:02,1318948862,switch_ivr_signal_bridge,1001,1001,r00919831094384,sofia/external/
> 1001 at 172.16.100.155
> ,2a96ebc1-a6be-8e49-b508-8b3993b62f9a,1001,fring,919831094384,sofia/external/919831094384,86e74690-fca2-6247-9d6a-c7732445a1be
> ****
>
> 2011-10-18
> 21:01:18,1318964478,switch_ivr_signal_bridge,1001,1001,r00919729947506,sofia/external/
> 1001 at 172.16.100.155
> ,96995af7-e82c-3e40-addb-1b62c1f05747,1001,fring,919729947506,sofia/external/919729947506,6a82a0ec-e4a2-be4f-85ee-87b34b4ea3e2
> ****
>
> 2011-10-19
> 00:20:24,1318976424,switch_ivr_signal_bridge,1001,1001,r00919605053363,sofia/external/
> 1001 at 172.16.100.155
> ,8f9d6670-2aff-c242-b01e-3d9f0e493d12,1001,fring,919605053363,sofia/external/919605053363,7fd8cee5-cd6b-7b44-848f-8d756f26fd4a
> ****
>
> 2011-10-19
> 04:30:30,1318991430,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/
> 1001 at 172.16.100.155
> ,1a6895e1-57a5-564c-bf08-5780df1d79c7,1001,fring,919814950511,sofia/external/919814950511,595fe521-1673-ea44-8bc3-c9d69c2f2de6
> ****
>
> 2011-10-19
> 04:40:27,1318992027,switch_ivr_signal_bridge,1001,1001,r00919814950511,sofia/external/
> 1001 at 172.16.100.155
> ,905be5af-844a-d44f-bbfd-a160ff1f1759,1001,fring,919814950511,sofia/external/919814950511,8a41744e-b19e-954a-a68f-7d16e29976e5
> ****
>
> 2011-10-19
> 06:23:43,1318998223,switch_ivr_signal_bridge,1001,1001,r00919990249401,sofia/external/
> 1001 at 172.16.100.155
> ,d009a643-b1b3-3f46-b194-c971e2a17b64,1001,fring,919990249401,sofia/external/919990249401,f26ae13f-de35-2f40-92cb-232449391e3e
> ****
>
> 2011-10-19
> 09:38:43,1319009923,switch_ivr_signal_bridge,1001,1001,r00919761333537,sofia/external/
> 1001 at 172.16.100.155
> ,353ede21-8d68-b74e-8711-0ddce2a82928,1001,fring,919761333537,sofia/external/919761333537,d0567650-c2f4-9542-9a20-d8bc1718cc80
> ****
>
> 2011-10-19
> 10:59:56,1319014796,switch_ivr_signal_bridge,1001,1001,r00918144890121,sofia/external/
> 1001 at 172.16.100.155
> ,8593dc8b-8d4f-bc40-8e3a-daf85d7c7702,1001,fring,918144890121,sofia/external/918144890121,ec3fda7c-cc8e-e041-b4a0-171928ade343
> ****
>
> 2011-10-19
> 13:00:41,1319022041,switch_ivr_signal_bridge,1001,1001,r00201060198003,sofia/external/
> 1001 at 172.16.100.155
> ,bba15b1f-785a-674f-a023-bb8f64f57c53,1001,fring,201060198003,sofia/external/201060198003,2c90f16e-a4ec-6d42-9bd1-4f06bf7c2da9
> ****
>
> 2011-10-19
> 13:45:11,1319024711,switch_ivr_signal_bridge,1001,1001,r00914312341722,sofia/external/
> 1001 at 172.16.100.155
> ,845903b7-2770-6847-a116-3d0a8cb7b096,1001,fring,914312341722,sofia/external/914312341722,aa20b33f-eb4d-714a-b236-59afbfe158e7
> ****
>
> 2011-10-19
> 19:11:01,1319044261,switch_ivr_signal_bridge,1001,1001,r009845967072,sofia/external/
> 1001 at 172.16.100.155
> ,349ef121-64b9-dc44-a22d-3d12acbb6ece,1001,fring,9845967072,sofia/external/9845967072,d89904a7-daf7-ea45-bae3-57c914e88b48
> ****
>
> 2011-10-20
> 02:36:13,1319070973,switch_ivr_signal_bridge,1001,1001,r0060166298037,sofia/external/
> 1001 at 172.16.100.155
> ,f6ff096b-612e-0544-a733-9a566b08fc6d,1001,fring,60166298037,sofia/external/60166298037,1c35f10e-3695-2845-9f4c-dd5a517dd6f7
> ****
>
> 2011-10-20
> 03:15:43,1319073343,switch_ivr_signal_bridge,1001,1001,r00919988498802,sofia/external/
> 1001 at 172.16.100.155
> ,1a9da86a-bc16-3e49-9b08-94facc657843,1001,fring,919988498802,sofia/external/919988498802,9430553a-ee00-ed4f-b109-41eabdd4dc7e
> ****
>
> 2011-10-20
> 04:35:24,1319078124,switch_ivr_signal_bridge,1001,1001,r00919976131008,sofia/external/
> 1001 at 172.16.100.155
> ,12d0429e-a539-6944-831e-a862183d905b,1001,fring,919976131008,sofia/external/919976131008,f672283c-a37e-e947-b45a-b7e66ef7881a
> ****
>
> 2011-10-20
> 05:22:23,1319080943,switch_ivr_signal_bridge,1001,1001,r00919973840848,sofia/external/
> 1001 at 172.16.100.155
> ,42e29646-c0c4-d64a-9cf5-7c37aeabc1a6,1001,fring,919973840848,sofia/external/919973840848,1d2ef437-17aa-db4b-86d9-6d3d5310e280
> ****
>
> 2011-10-20
> 05:37:34,1319081854,switch_ivr_signal_bridge,1001,1001,r00919840288799,sofia/external/
> 1001 at 172.16.100.155
> ,f48fed29-8041-3545-bffc-5050bc19cbc6,1001,fring,919840288799,sofia/external/919840288799,df29a0df-3ef8-4b43-807e-73e0c0e25f3b
> ****
>
> 2011-10-20
> 10:00:26,1319097626,switch_ivr_signal_bridge,1001,1001,r0091955445666,sofia/external/
> 1001 at 172.16.100.155
> ,931102bc-ac35-3140-acab-988224d502b0,1001,fring,91955445666,sofia/external/91955445666,9283eb63-bd0a-6d43-9d8a-7985ec48e23c
> ****
>
> 2011-10-20
> 12:16:27,1319105787,switch_ivr_signal_bridge,1001,1001,r00919855120931,sofia/external/
> 1001 at 172.16.100.155
> ,db0f66aa-69f1-fe47-85af-de610a77d517,1001,fring,919855120931,sofia/external/919855120931,ceab5ae5-9b8f-3d4c-b149-c248567ab5d4
> ****
>
> 2011-10-20
> 12:42:02,1319107322,switch_ivr_signal_bridge,1001,1001,r00918872855601,sofia/external/
> 1001 at 172.16.100.155
> ,95249509-e793-0d42-9948-5ef53898b942,1001,fring,918872855601,sofia/external/918872855601,8c9fc203-8d54-aa4f-983f-889fbf0ca883
> ****
>
> 2011-10-20
> 13:45:10,1319111110,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/
> 1001 at 172.16.100.155
> ,7b645b14-5f89-a64c-b059-2574b2e4e69d,1001,fring,919988175489,sofia/external/919988175489,55f50da5-5679-fe4f-9a7f-b96111f3cd67
> ****
>
> 2011-10-20
> 14:49:50,1319114990,switch_ivr_signal_bridge,1001,1001,r00919663195202,sofia/external/
> 1001 at 172.16.100.155
> ,3601f4b8-c0ff-1a4c-9f76-47e4fa95b549,1001,fring,919663195202,sofia/external/919663195202,a22e2d11-0463-c845-a953-a12b49cfe863
> ****
>
> 2011-10-20
> 15:10:47,1319116247,switch_ivr_signal_bridge,1001,1001,r00919988175489,sofia/external/
> 1001 at 172.16.100.155
> ,e09d3de0-c759-fc4b-96f1-f01353a4297d,1001,fring,919988175489,sofia/external/919988175489,23d4b490-cbb7-7a45-86af-705c45d07fc7
> ****
>
> 2011-10-20
> 20:52:17,1319136737,switch_ivr_signal_bridge,1001,1001,r009888560750,sofia/external/
> 1001 at 172.16.100.155
> ,ff8ab1f6-f1c6-224f-928b-1c042b0ed908,1001,fring,9888560750,sofia/external/9888560750,c37b8ebe-2264-934f-a187-6b14c7dcb308
> ****
>
> 2011-10-20
> 22:32:38,1319142758,switch_ivr_signal_bridge,1001,1001,r008801816630096,sofia/external/
> 1001 at 172.16.100.155
> ,a38182cb-b661-854d-b243-a830ab0bf992,1001,fring,8801816630096,sofia/external/8801816630096,f88dc530-303e-d041-83cb-eca118780ca2
> ****
>
> 2011-10-20
> 22:34:32,1319142872,switch_ivr_signal_bridge,1001,1001,r00919161209997,sofia/external/
> 1001 at 172.16.100.155
> ,0add3c1c-7fb5-2447-a9d6-d0622886fcf9,1001,fring,919161209997,sofia/external/919161209997,2b91f5de-4668-0244-ac76-d031fe827a25
> ****
>
> 2011-10-20
> 23:56:16,1319147776,switch_ivr_signal_bridge,1001,1001,r00985468962,sofia/external/
> 1001 at 172.16.100.155
> ,f7137d11-0a87-6a4b-b5b3-b43eeeb2b462,1001,fring,985468962,sofia/external/985468962,025f197b-c5aa-5b4b-8606-20fb6f17e888
> ****
>
> 2011-10-21
> 04:36:00,1319164560,switch_ivr_signal_bridge,1001,1001,r00917597234252,sofia/external/
> 1001 at 172.16.100.155
> ,edd2f16b-9a6a-f749-bd62-4d303bc09442,1001,fring,917597234252,sofia/external/917597234252,1c55b21b-9c27-b647-8ba5-1ffce398c747
> ****
>
> 2011-10-21
> 04:47:03,1319165223,switch_ivr_signal_bridge,1001,1001,r00919541622040,sofia/external/
> 1001 at 172.16.100.155
> ,336c146b-c96d-494e-b378-c024bcfc0c94,1001,fring,919541622040,sofia/external/919541622040,746877e1-c509-8a45-934e-f3f4b7f82db2
> ****
>
> 2011-10-21
> 07:59:54,1319176794,switch_ivr_signal_bridge,1001,1001,r009613235498,sofia/external/
> 1001 at 172.16.100.155
> ,16575979-c92a-5842-879f-e9f6448c7d54,1001,fring,9613235498,sofia/external/9613235498,bb05fd5a-4a6f-3248-81a9-a81074a9eacd
> ****
>
> 2011-10-21
> 08:02:05,1319176925,switch_ivr_signal_bridge,1001,1001,r00912026336917,sofia/external/
> 1001 at 172.16.100.155
> ,ebb234aa-8edf-2e4b-b641-602478b25782,1001,fring,912026336917,sofia/external/912026336917,f6a0e4b1-69f7-dc49-a5bf-41e6d8c7c96d
> ****
>
> 2011-10-21
> 08:51:57,1319179917,switch_ivr_signal_bridge,1001,1001,r00919372875359,sofia/external/
> 1001 at 172.16.100.155
> ,b090dc91-932e-1246-abb8-8fc949131ae6,1001,fring,919372875359,sofia/external/919372875359,3a4702a6-e7cc-cb41-ad99-defa80b14680
> ****
>
> 2011-10-21
> 09:15:47,1319181347,switch_ivr_signal_bridge,1001,1001,r00919915709069,sofia/external/
> 1001 at 172.16.100.155
> ,ae308194-ce89-4945-b8de-6923ce6d0dc0,1001,fring,919915709069,sofia/external/919915709069,502ffe82-0fb9-b14a-9421-4acd25a0b4df
> ****
>
> 2011-10-21
> 11:35:16,1319189716,switch_ivr_signal_bridge,1001,1001,r00919847023725,sofia/external/
> 1001 at 172.16.100.155
> ,c7cb7640-7313-d742-b756-a7713273ccea,1001,fring,919847023725,sofia/external/919847023725,4ab50403-04ac-bc40-a759-40f2a62f1699
> ****
>
> 2011-10-21
> 14:48:46,1319201326,switch_ivr_signal_bridge,1001,1001,r00919878738103,sofia/external/
> 1001 at 172.16.100.155
> ,91626ac0-9e14-a34d-94f5-b1fbfe7e7629,1001,fring,919878738103,sofia/external/919878738103,39c58424-b087-6749-9ba9-c9a821f46895
> ****
>
> 2011-10-21
> 17:41:28,1319211688,switch_ivr_signal_bridge,1001,1001,r009843648082,sofia/external/
> 1001 at 172.16.100.155
> ,3929dc6b-b7de-c648-8eae-3d749f0ac0fa,1001,fring,9843648082,sofia/external/9843648082,6653a9ed-d23b-564e-98e5-9a56a40e499c
> ****
>
> 2011-10-22
> 05:07:21,1319252841,switch_ivr_signal_bridge,1001,1001,r00919036392951,sofia/external/
> 1001 at 172.16.100.155
> ,7cd56510-2983-554a-9fa2-a27c3fb1f9e6,1001,fring,919036392951,sofia/external/919036392951,e63642ec-d43a-c243-9291-0ea6725d04bb
> ****
>
> 2011-10-22
> 06:42:56,1319258576,switch_ivr_signal_bridge,1001,1001,r00918870521722,sofia/external/
> 1001 at 172.16.100.155
> ,7dd85bc6-e00e-c44e-bd42-89c6d438485e,1001,fring,918870521722,sofia/external/918870521722,8d7e1c17-2ad9-2447-b91b-d2f2b2e27e4c
> ****
>
> 2011-10-22
> 08:36:26,1319265386,switch_ivr_signal_bridge,1001,1001,r00911874289543,sofia/external/
> 1001 at 172.16.100.155
> ,1214a817-f235-1a4e-a21c-d2363e9e5760,1001,fring,911874289543,sofia/external/911874289543,13f85a5d-eb37-0c4a-a15e-2c86dbc34817
> ****
>
> 2011-10-22
> 12:14:32,1319278472,switch_ivr_signal_bridge,1001,1001,r00918988262666,sofia/external/
> 1001 at 172.16.100.155
> ,1b5cb871-e043-f244-9942-251a7a492136,1001,fring,918988262666,sofia/external/918988262666,21f328fb-0dfe-2643-957b-302f6a80f17f
> ****
>
> 2011-10-22
> 16:28:35,1319293715,switch_ivr_signal_bridge,1001,1001,r00918722222400,sofia/external/
> 1001 at 172.16.100.155
> ,912e1b07-a99b-bf4d-9e2f-0b5990a68821,1001,fring,918722222400,sofia/external/918722222400,8b417629-3161-f141-93d7-bb0ddd7982d5
> ****
>
> 2011-10-22
> 17:32:10,1319297530,switch_ivr_signal_bridge,1001,1001,r00919839884297,sofia/external/
> 1001 at 172.16.100.155
> ,ec7decdd-c378-6c45-a888-9b962e4ecece,1001,fring,919839884297,sofia/external/919839884297,f12896b7-9605-9b40-bdc9-face2181146b
> ****
>
> 2011-10-23
> 06:26:42,1319344002,switch_ivr_signal_bridge,1001,1001,r00919567382898,sofia/external/
> 1001 at 172.16.100.155
> ,db2e199b-7b9f-fe4d-9742-7415dded69c7,1001,fring,919567382898,sofia/external/919567382898,dd8bb627-c804-2f49-8b5b-5acf5b3423d4
> ****
>
> 2011-10-23
> 07:21:29,1319347289,switch_ivr_signal_bridge,1001,1001,r00966537646047,sofia/external/
> 1001 at 172.16.100.155
> ,e97ba53e-1ef6-2944-b53b-b3abe5dfef36,1001,fring,966537646047,sofia/external/966537646047,9eea7d3f-5092-de4b-9291-e7f0a37db2a1
> ****
>
> 2011-10-23
> 08:12:33,1319350353,switch_ivr_signal_bridge,1001,1001,r00914942588715,sofia/external/
> 1001 at 172.16.100.155
> ,30ca9d2d-5cae-6645-a16c-6ae8735b6316,1001,fring,914942588715,sofia/external/914942588715,6774d23c-f131-7d41-854d-2fcff77c2cf6
> ****
>
> 2011-10-23
> 13:04:33,1319367873,switch_ivr_signal_bridge,1001,1001,r00919625833433,sofia/external/
> 1001 at 172.16.100.155
> ,284691f9-dd3b-cc4a-97c7-9465bd3a6b6c,1001,fring,919625833433,sofia/external/919625833433,4efb7446-11f2-c84c-88c4-10ab97eddedf
> ****
>
> 2011-10-23
> 14:35:20,1319373320,switch_ivr_signal_bridge,1001,1001,r00919888880401,sofia/external/
> 1001 at 172.16.100.155
> ,475abe11-2137-c94a-ada7-825ad8524bd8,1001,fring,919888880401,sofia/external/919888880401,5e06efb5-2c72-ba4f-9443-c15568ad2608
> ****
>
> 2011-10-23
> 15:59:20,1319378360,switch_ivr_signal_bridge,1001,1001,r009878539161,sofia/external/
> 1001 at 172.16.100.155
> ,aa5c4f8f-5a69-e947-a239-75f4a584b9f0,1001,fring,9878539161,sofia/external/9878539161,366b56d1-3ce4-fa43-9d3d-872c04f5229f
> ****
>
> 2011-10-23
> 20:52:23,1319395943,switch_ivr_signal_bridge,1001,1001,r0017787079272,sofia/external/
> 1001 at 172.16.100.155
> ,3f9c8b11-2e7a-e641-8cfc-3b7dc65e7118,1001,fring,17787079272,sofia/external/17787079272,d5b4a2e0-ee48-7048-b54c-f6ea968428af
> ****
>
> 2011-10-24
> 10:23:29,1319444609,switch_ivr_signal_bridge,1001,1001,r00919881375673,sofia/external/
> 1001 at 172.16.100.155
> ,527d1ca5-3f05-a54b-8b0c-192b7515eebb,1001,fring,919881375673,sofia/external/919881375673,4d1da2aa-87c1-7d41-b57d-05034e503ede
> ****
>
> 2011-10-24
> 10:29:44,1319444984,switch_ivr_signal_bridge,1001,1001,r0061430594924,sofia/external/
> 1001 at 172.16.100.155
> ,6ce52143-219d-ed40-a27f-fc519eef5092,1001,fring,61430594924,sofia/external/61430594924,11ef495a-d651-564e-9479-94fa579a5286
> ****
>
> 2011-10-24
> 10:56:32,1319446592,switch_ivr_signal_bridge,1001,1001,r00919003574979,sofia/external/
> 1001 at 172.16.100.155
> ,8d83282b-f96b-0a43-ac58-1f3d3e4e26f0,1001,fring,919003574979,sofia/external/919003574979,2bd6e807-c5ef-8c47-807f-1e5171bcbc3d
> ****
>
> 2011-10-24
> 13:58:59,1319457539,switch_ivr_signal_bridge,1001,1001,r00919876395511,sofia/external/
> 1001 at 172.16.100.155
> ,68fdbb19-f76f-0a40-b46b-e0296eed8690,1001,fring,919876395511,sofia/external/919876395511,6cf77cf5-8cd9-f340-98c0-bb896304dbf9
> ****
>
> 2011-10-24
> 14:16:45,1319458605,switch_ivr_signal_bridge,1001,1001,r00919988170711,sofia/external/
> 1001 at 172.16.100.155
> ,57e0187b-9d25-044c-bffb-9d11431c251a,1001,fring,919988170711,sofia/external/919988170711,b71d6822-289f-614b-8e5b-2bc20225d9ef
> ****
>
> 2011-10-25
> 00:16:09,1319494569,switch_ivr_signal_bridge,1001,1001,r005016630551,sofia/external/
> 1001 at 172.16.100.155
> ,f5e81b48-8f86-4644-b206-cb066e15418d,1001,fring,5016630551,sofia/external/5016630551,b1467ed2-56e3-bb42-ba6c-cdc272b2533b
> ****
>
> 2011-10-25
> 02:30:30,1319502630,switch_ivr_signal_bridge,1001,1001,r00919729325377,sofia/external/
> 1001 at 172.16.100.155
> ,f6eb7c48-cc1d-9b42-bbac-46ac759de0e0,1001,fring,919729325377,sofia/external/919729325377,d8eb68b9-ab39-814b-a70d-915347e2afe7
> ****
>
> 2011-10-25
> 09:35:05,1319528105,switch_ivr_signal_bridge,1001,1001,r00966554656169,sofia/external/
> 1001 at 172.16.100.155
> ,03c9a95e-4d36-8847-a2a2-3545c38e233e,1001,fring,966554656169,sofia/external/966554656169,1d2927f5-7202-914c-a463-6dcd00588cfc
> ****
>
> 2011-10-25
> 12:53:58,1319540038,switch_ivr_signal_bridge,1001,1001,r00201094811534,sofia/external/
> 1001 at 172.16.100.155
> ,3bfe5634-dcf9-c84d-ba72-26f8fee4a8eb,1001,fring,201094811534,sofia/external/201094811534,d0c58bb9-53a3-0549-9b78-4ca5f14fcb63
> ****
>
> 2011-10-26
> 04:22:12,1319595732,switch_ivr_signal_bridge,1001,1001,r00913871398117,sofia/external/
> 1001 at 172.16.100.155
> ,1479c10c-9922-854c-8030-6f2a6ce4f605,1001,fring,913871398117,sofia/external/913871398117,fbdc8cf2-e898-0340-bc48-b4ad7c54ac0e
> ****
>
> 2011-10-26
> 10:01:27,1319616087,switch_ivr_signal_bridge,1001,1001,r00201113961426,sofia/external/
> 1001 at 172.16.100.155
> ,db18777e-aefa-c248-84ea-06150fe6989f,1001,fring,201113961426,sofia/external/201113961426,0a5572c6-54b7-1d40-bf4e-918e6880b3ce
> ****
>
> ** **
>
> 147 total.****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Vellayappan
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From michaelt at callall.co.za  Mon Oct 31 23:00:52 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Mon, 31 Oct 2011 22:00:52 +0200
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <6D16158B-07AB-4FF6-8817-4166F855DC62@freeswitch.org>
References: <20111027072637.26979.qmail@community22.interfree.it>
	<CAFmdODgcPWv2T5LmaYtxHRDMeHsQzTdfCmNOW4exZQ1LMyaihw@mail.gmail.com>
	<CAFmdODhkFaCdk7JRpA1tm76_j8LoCp2FO_dgoaDZMFFT05qX8g@mail.gmail.com>
	<6D16158B-07AB-4FF6-8817-4166F855DC62@freeswitch.org>
Message-ID: <CAFmdODhJEvxS4Kg1WVZRB70kB4mMd8-MFKxDxT+DmLXq4CmuwQ@mail.gmail.com>

Hi Brian,

Surely FS should tear down the call if the media is not suitable?  I now
see that 'G729a' does not conform to the IANA standard and is wrong.

Let me get the m attribute fixed in the SDP & I will let you know.

Thanks for the input.

Regards,

Michael

On Mon, Oct 31, 2011 at 4:34 PM, Brian West <brian at freeswitch.org> wrote:

> Its not a bug.  zero is an indication that we rejected the media.
> /b
>
> On Oct 31, 2011, at 6:07 AM, Michael Toop wrote:
>
> Hi,
>
> Is there someone in the list interested in some consulting work to assist
> me to get to the bottom of this problem?
>
> Happy to pay for assistance here.
>
> Thanks,
>
> Michael
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From spencer at 5ninesolutions.com  Mon Oct 31 23:13:05 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 31 Oct 2011 13:13:05 -0700
Subject: [Freeswitch-users] Using Presence with Valet Parking
In-Reply-To: <CAKbxfG9sLMRea7ADb5HrxxgxndrFjkjUfTZjnSnXjZzZBh9xAA@mail.gmail.com>
References: <E572C020-3CF0-40A9-83A1-3F8EC4EEA639@5ninesolutions.com>
	<CAKbxfG9sLMRea7ADb5HrxxgxndrFjkjUfTZjnSnXjZzZBh9xAA@mail.gmail.com>
Message-ID: <74536C50-3AF4-4456-8F19-E8DB58663BE3@5ninesolutions.com>

Hi Anthony,
Thanks for your help!  This is working great very well now.

Spencer

On Oct 31, 2011, at 8:03 AM, Anthony Minessale wrote:

> make sure DOMAIN is the default domain set in vars.xml otherwise name the lot with @domain.com right in it.
> Also make sure its the most recent rev of the mod.
> 
> 
> On Sun, Oct 30, 2011 at 9:44 PM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello,
> I noticed there have been some recent improvements to mod_valet_parking regarding presence.  I was wondering if anyone had used this already and could shed some light on how to subscribe to the status of a particular slot.  I'm testing this with a Cisco SPA-509G and can't seem to get it working. My dialplan looks like this:
> 
>    <extension name="parking_slots">
>        <condition field="destination_number" expression="^(7[1-4])$" break="on-false">
>          <action application="answer"/>
>          <action application="valet_park" data="parking_slots $1"/>
>        </condition>
>    </extension>
> 
> The phone is sending a SUBSCRIBE to park+71@<DOMAIN> and gets a 202 Accepted response.
> 
> I do have presence configured and working correctly for user extensions.  When a call is transferred to 71, the parking works as expected but a NOTIFY is never sent to the phone.  What am I missing?
> 
> Thanks for any help!
> 
> Spencer
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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 msc at freeswitch.org  Mon Oct 31 23:17:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Oct 2011 13:17:13 -0700
Subject: [Freeswitch-users] Using Presence with Valet Parking
In-Reply-To: <74536C50-3AF4-4456-8F19-E8DB58663BE3@5ninesolutions.com>
References: <E572C020-3CF0-40A9-83A1-3F8EC4EEA639@5ninesolutions.com>
	<CAKbxfG9sLMRea7ADb5HrxxgxndrFjkjUfTZjnSnXjZzZBh9xAA@mail.gmail.com>
	<74536C50-3AF4-4456-8F19-E8DB58663BE3@5ninesolutions.com>
Message-ID: <CAKzWOxWfYy1N9_AYzZad78pT8Svwof_BnzEfHmJ=iRxdGwG5fQ@mail.gmail.com>

Would you mind paying the wiki tax and writing up your scenario and posting
a sample config? You can starting adding content on this page, since it
needs some love anyway:

http://wiki.freeswitch.org/wiki/Mod_valet_park

Thanks,
MC

On Mon, Oct 31, 2011 at 1:13 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hi Anthony,
> Thanks for your help!  This is working great very well now.
>
> Spencer
>
> On Oct 31, 2011, at 8:03 AM, Anthony Minessale wrote:
>
> make sure DOMAIN is the default domain set in vars.xml otherwise name the
> lot with @domain.com right in it.
> Also make sure its the most recent rev of the mod.
>
>
> On Sun, Oct 30, 2011 at 9:44 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> Hello,
>> I noticed there have been some recent improvements to mod_valet_parking
>> regarding presence.  I was wondering if anyone had used this already and
>> could shed some light on how to subscribe to the status of a particular
>> slot.  I'm testing this with a Cisco SPA-509G and can't seem to get it
>> working. My dialplan looks like this:
>>
>>    <extension name="parking_slots">
>>        <condition field="destination_number" expression="^(7[1-4])$"
>> break="on-false">
>>          <action application="answer"/>
>>          <action application="valet_park" data="parking_slots $1"/>
>>        </condition>
>>    </extension>
>>
>> The phone is sending a SUBSCRIBE to park+71@<DOMAIN> and gets a 202
>> Accepted response.
>>
>> I do have presence configured and working correctly for user extensions.
>>  When a call is transferred to 71, the parking works as expected but a
>> NOTIFY is never sent to the phone.  What am I missing?
>>
>> Thanks for any help!
>>
>> Spencer
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steve.kurzeja at gmail.com  Mon Oct 31 23:58:49 2011
From: steve.kurzeja at gmail.com (Steve Kurzeja)
Date: Tue, 1 Nov 2011 09:58:49 +1300
Subject: [Freeswitch-users] RTP Media Port = 0 Bug
In-Reply-To: <CAFmdODhJEvxS4Kg1WVZRB70kB4mMd8-MFKxDxT+DmLXq4CmuwQ@mail.gmail.com>
References: <20111027072637.26979.qmail@community22.interfree.it>
	<CAFmdODgcPWv2T5LmaYtxHRDMeHsQzTdfCmNOW4exZQ1LMyaihw@mail.gmail.com>
	<CAFmdODhkFaCdk7JRpA1tm76_j8LoCp2FO_dgoaDZMFFT05qX8g@mail.gmail.com>
	<6D16158B-07AB-4FF6-8817-4166F855DC62@freeswitch.org>
	<CAFmdODhJEvxS4Kg1WVZRB70kB4mMd8-MFKxDxT+DmLXq4CmuwQ@mail.gmail.com>
Message-ID: <CAENkwGu3hD-UmNrnxHv73pkrM8X+wjAuvecagtVRfQ=aKqYbAQ@mail.gmail.com>

On Tue, Nov 1, 2011 at 9:00 AM, Michael Toop <michaelt at callall.co.za> wrote:

> Hi Brian,
>
> Surely FS should tear down the call if the media is not suitable?  I now
> see that 'G729a' does not conform to the IANA standard and is wrong.
>
>
Yeh, I've actually seen this issue before in proxy media mode with bad iana
name flag enabled (which actually means fix specifically the string "g729a"
to "g729") where the 200 OK or 183 response with SDP gets broken on the way
back. By broken the SDP on the response to the B leg changes to dynamic
payload type 96 and rtp port 0 instead of payload type 18/non-zero rtp
port.

There's something I think in the sofia libs which is validating the SDP -
even though technically in proxy media mode the codec list should just get
passed through. I haven't yet found exactly where this is yet.



> Let me get the m attribute fixed in the SDP & I will let you know.
>
>

With the correct iana name in the rtpmap then I think your problem will go
away, assuming you control the endpoints in your environment and can change
them to originate the call with the correct iana name in the rtpmap then I
think your problem will go away.

Or you could turn on full media mode too which would solve it but you just
need to be careful with codec negotiation at that point.



> Thanks for the input.
>
> Regards,
>
> Michael
>
> On Mon, Oct 31, 2011 at 4:34 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Its not a bug.  zero is an indication that we rejected the media.
>> /b
>>
>> On Oct 31, 2011, at 6:07 AM, Michael Toop wrote:
>>
>> Hi,
>>
>> Is there someone in the list interested in some consulting work to assist
>> me to get to the bottom of this problem?
>>
>> Happy to pay for assistance here.
>>
>> Thanks,
>>
>> Michael
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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