From vetali100 at gmail.com  Sun May  1 00:07:39 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Sat, 30 Apr 2011 23:07:39 +0300
Subject: [Freeswitch-users] Trunking between Lync and FreeSWITCH
In-Reply-To: <761372A6-B060-42ED-8553-9EB3FFF13F7B@ipeva.fr>
References: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>
	<4DBBE861.9050906@xpirio.com>
	<761372A6-B060-42ED-8553-9EB3FFF13F7B@ipeva.fr>
Message-ID: <BANLkTinqTbEGQLqativwZhZi9te6h+ompQ@mail.gmail.com>

I saw the cases when call is disconnected after 32 seconds when one of the
parties does not receive the ACK message after sending the OK message when
call is answered.
It was not with Lync, but with some crappy sip client.
Maybe you can investigate in this direction.

Vitalie
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From ovvenkatesan at gmail.com  Sun May  1 11:08:41 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Sun, 1 May 2011 12:38:41 +0530
Subject: [Freeswitch-users] freeswitch installation error
Message-ID: <BANLkTikEYZvKAZfP9oMPFnX2Vug=6qOhuA@mail.gmail.com>

Hi to all,

Today, I am upgrading my platform and I used the following commands to
upgrade the platform

Goto the freeswitch source directory

1.$ git stash
2.$ git pull
3.$ git stash apply
4.$./bootstrap.sh
5.$./configure
6.$make current

Afte "make current" command executed, I am getting following errors..


.......
.......
.......
grep: /home/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la: No such file
or directory
/bin/sed: can't read /home/freeswitch/libs/apr-util/xml/expat/lib/
libexpat.la: No such file or directory
quiet_libtool: link: `/home/freeswitch/libs/apr-util/xml/expat/lib/
libexpat.la' is not a valid libtool archive
make[2]: *** [libfreeswitch.la] Error 1
make[2]: Leaving directory `/usr/src/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [current] Error 2

Since, its production server , please anyone help to get resolve this issue.




-- 


Regards
Venkatesan OV.
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From anton.vazir at gmail.com  Sun May  1 12:43:26 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 1 May 2011 13:43:26 +0500
Subject: [Freeswitch-users] freeswitch installation error
In-Reply-To: <BANLkTikEYZvKAZfP9oMPFnX2Vug=6qOhuA@mail.gmail.com>
References: <BANLkTikEYZvKAZfP9oMPFnX2Vug=6qOhuA@mail.gmail.com>
Message-ID: <BANLkTi=Xop=jkh_WWLTLz4KqWTZ3hxvtfQ@mail.gmail.com>

apt-get install libexpat1-dev

2011/5/1 ovvenkat <ovvenkatesan at gmail.com>:
> Hi to all,
>
> Today, I am upgrading my platform and I used the following commands to
> upgrade the platform
>
> Goto the freeswitch source directory
>
> 1.$ git stash
> 2.$ git pull
> 3.$ git stash apply
> 4.$./bootstrap.sh
> 5.$./configure
> 6.$make current
>
> Afte "make current" command executed, I am getting following errors..
>
>
> .......
> .......
> .......
> grep: /home/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la: No such file
> or directory
> /bin/sed: can't read
> /home/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la: No such file or
> directory
> quiet_libtool: link:
> `/home/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la' is not a valid
> libtool archive
> make[2]: *** [libfreeswitch.la] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [current] Error 2
>
> Since, its production server , please anyone help to get resolve this issue.
>
>
>
> --
>
>
> Regards
> Venkatesan OV.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From fieldpeak at gmail.com  Sun May  1 14:44:37 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 1 May 2011 18:44:37 +0800
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <4dbc5332.c9860e0a.668b.ffffa003@mx.google.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
	<4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
	<BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>
	<BANLkTikXtcGECS_tZPMcS+4Kq_mvASgnXA@mail.gmail.com>
	<BANLkTinhWi3Q7x4B3hM60w+1F96htTVpvA@mail.gmail.com>
	<BANLkTim1Bp=t-XMQJmi9ABrvfY=fr+CnGg@mail.gmail.com>
	<BANLkTikfsdWxJR5NnzqxFj6JfajcDxRF-A@mail.gmail.com>
	<4dbc5332.c9860e0a.668b.ffffa003@mx.google.com>
Message-ID: <BANLkTimpqcFg68xbBOUbeRAxA7qs8R9XnQ@mail.gmail.com>

hi garmt,
agree with you, in a production system...
thanks.

Regards,
Charles

? 2011-5-1 ??2:23?"Grmt" <garmt.noname at gmail.com>???
> I assume you mean GIT april 4th, 2011 ? I'm almost certain that I also
> experienced this problem after apr 21st, 2010.
>
> And indeed you have to restart/scan sofia sip profiles.
>
>
>
> I suspect it is related to
>
> - Having a multihomed system (multiple network cards, i.e. wifi +
> fixed Ethernet + virtual adaptors)
>
> o Possibly related to running virtual machines (with virtual adaptors) on
> the same host (e.g. any vmware installation)
>
> - Using ipv4 and ipv6 simultaneously
>
>
>
> On the other hand, I have not seen this problem recently. I changed my
> vmware player network card installation and also disabled ipv6 and updated
> the firmware of my router, while still using DHCP (and NAT). Changes to
the
> FS source code in the mean time may also have fixed the issue. As this
> problem occurred on my dev system, I did not bother going to the bottom of
> it . In a production system I would recommend using a fixed IP address and
> the settings as noted in the wiki.
>
>
>
>
>
> Garmt
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> fieldpeak
> Sent: Saturday, 30 April, 2011 07:39
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS - SIP profiles crashed
>
>
>
> sorry, my misunderstanding, it looks this issue fixed by FS-933 on
> 21/Apr/10, however, my GIT is April 4, 2001, it still exist... strange...
>
>
> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>
> it looks the same issue as
>
http://jira.freeswitch.org/browse/FS-933?page=com.atlassian.jira.plugin.syst
> em.issuetabpanels%3Aall-tabpanel
>
> however, where to add the new parameter: PFLAG_SKIP_RESTART
>
>
> ?
>
> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>
> oh, looks i found the root cause,
>
> after i unplug the network cable and wait around 20 seconds, FS show ' IP
> address changed to '0.0.0.0', then try to load profiles, however, due to
> wrong IP address, load failure.
> and then i re-plugin the cable, FS detected the IP change back, and
> reloadxml, however, did not restart the profiles, so caused profiles
> crashed... attached is the log for throughout the procedure...
>
> the resolution is set
>
> in the sofia.conf.xml file:
>
> <param name="auto-restart" value="false"/>
>
> welcome to any comment,
>
> thanks.
>
> Regards,
> Charles
>
>
>
> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>
> really want to know what caused that the IP address changed to '0.0.0.0',
> very odd...
> i tried even unplug and plugin the network cable, it will not change the
> IP...let alone 0.0.0.0...
>
>
>
> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>
> Thanks all for information, it definitely help a lot. i will try...
>
> meanwhile, i found this link
>
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003086.html
>
> it looks not so special resolve this issue...
>
> i would like to know what is use for "bind_server_ip=auto" and where and
> when FS get the value of $${local_ip_v4}? and if it will changed when
> physical IP changed?
>
> Thanks!
>
> Regards,
> Charles
>
> 2011/4/30 Grmt <garmt.noname at gmail.com>
>
>
http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP
> _address
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael
> Collins
> Sent: Friday, 29 April, 2011 19:35
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS - SIP profiles crashed
>
>
>
>
>
> On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> If your box is configured with a static IP do this.
>
> conf/autoload_configs/sofia.conf.xml --
>
>
>
> Jeff, was there more to this message?
>
> -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 Nabble at slickdeals.endjunk.com  Sun May  1 17:31:08 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 1 May 2011 06:31:08 -0700 (PDT)
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTinKmrDQGmXMyoK5MVn8Ttq14zJPww@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<1304179126964-6319822.post@n2.nabble.com>
	<BANLkTinYd64kyirHuOKV64Hc_SVsmO+pzA@mail.gmail.com>
	<1304183588768-6319970.post@n2.nabble.com>
	<BANLkTinbWcMX2jP5TdL0FOp_khCrNJpwcA@mail.gmail.com>
	<BANLkTinKmrDQGmXMyoK5MVn8Ttq14zJPww@mail.gmail.com>
Message-ID: <1304256668602-6321342.post@n2.nabble.com>


Giovanni Maruzzelli wrote:
> 
> On Sat, Apr 30, 2011 at 7:45 PM, jesse &lt;chat2jesse at gmail.com&gt; wrote:
>> You don't like debian on dockstar?
> 
> mazilo was disilluded because for mod_skypopen you still need a Skype
> client, and Skype client does not exists for Debian on Dockstar. So,
> he cannot use mod_skypopen.
Unless there is a Skype client version for an ARM platform mobile/smart
phone running on a Linux OS that I can  download and test on my Seagate
DockStar, I just moved on. 
-----FreeSWITCH hosted on a Seagate DockStar with OpenWRT.--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NEW-skypopen-installer-and-easy-Skype-client-download-Skype-calls-on-FreeSWITCH-in-one-minute-tp6319154p6321342.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Sun May  1 17:34:17 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 1 May 2011 06:34:17 -0700 (PDT)
Subject: [Freeswitch-users] Reviewers needed: FreeSWITCH Cookbook
In-Reply-To: <BANLkTikL+6i6ovRX1EsfzMoWzDiw3TvbcQ@mail.gmail.com>
References: <BANLkTikL+6i6ovRX1EsfzMoWzDiw3TvbcQ@mail.gmail.com>
Message-ID: <1304256857833-6321346.post@n2.nabble.com>


mercutioviz wrote:
> If you meet the above qualifications and would like to help out then
> please
> contact me off list for more information.
Well, I wished I could help, but I failed one of the qualifications. 
-----FreeSWITCH hosted on a Seagate DockStar with OpenWRT.--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Reviewers-needed-FreeSWITCH-Cookbook-tp6317742p6321346.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From engineerzuhairraza at gmail.com  Sun May  1 23:12:08 2011
From: engineerzuhairraza at gmail.com (zohair raza)
Date: Mon, 2 May 2011 00:12:08 +0500
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
Message-ID: <BANLkTimubrY_jTRptCbL0oQC3pJkvLpT9g@mail.gmail.com>

Wow That is great Sir,

i would like to add how i am passing caller id

i am using xml curl and in my php file i added this :

<?php

  $calld  =  $_REQUEST['Caller-Destination-Number'];
  $cally =str_replace("skype3439","",$calld) ; ( I am sending skype id like
'skype3439remoteskypeid'  so here i am removing skype3439)
  $callsk   = $_REQUEST['Caller-Caller-ID-Number'];
  $zr= ' api_result=${skypopen(interface1 MESSAGE '.$cally.' (mp) Call from
'.$callsk.')}';

header("Content-type: text/xml");
$response = "
<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
<document type=\"freeswitch/xml\">
  <section name=\"dialplan\" description=\"example_curl_dialplan\">
    <context name=\"public\">

         <extension name=\"skype\">
         <condition field=\"destination_number\"
expression=\"^skype3439(.*)$\">
         <action application=\"set\" data=\"skype_get_inband_dtmf=false\"/>
         <action application=\"set\" data=\"$zr\"/>
         <action application=\"set\" data=\"instant_ringback=true\"/>
         <action application=\"bridge\" data=\"skypopen/RR/$1\"/>
         </condition>
         </extension>

  </context>
  </section>
</document>
";
echo $response;
exit;
?>


This way caller id will be sent before the call as a skype chat message
which will only work if remoteskypeid has added the skype id which you are
using, otherwise it won't work.


On Sat, Apr 30, 2011 at 1:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Dear FreeSWITCHers,
>
> after a fair amount of effort, I ended up with a new way to install
> and use mod_skypopen on Linux.
>
> No more looking around the internet for the lost 2.0.0.72 Skype client for
> ALSA.
>
> First, we can use the readily available Skype client for OSS.
>
> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
> driver, that's very easy to compile and install, and do not need to
> mess with the operating system installation.)
>
> Second, I wrote an installer that automatically do all the tedious
> work for you: download and install the skype client, create the config
> directory for Skype clients, create the config file for mod_skypopen,
> create the script that launches the Skype clients.
>
> I hope those improvements will lower the barriers for Skype calls on
> FreeSWITCH.
>
> Actually is ludicrously simple now, and after you compile FreeSWITCH,
> mod_skypopen and the skypopen.ko OSS driver it will take like less
> than one minute to have a complete installation of mod_skypopen ready
> to make and receive calls.
>
> All automatic, no more need to fiddle around with the Skype client
> download, configurations, authorization, etc.
>
> Is all well tested, but maybe there are still some bugs, and maybe the
> docs are not clear/easy enough.
>
> Please have a look at the new and improved wiki page and let me know
> what do you think about (and maybe test the procedures).
>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>
> You must update to the latest git to have all the goodies.
>
> Thank you all for your support,
>
> -giovanni
>
> --
> 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
>



-- 
Regards,
Zohair Raza
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From david.ponzone at ipeva.fr  Sun May  1 23:24:07 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 1 May 2011 21:24:07 +0200
Subject: [Freeswitch-users] Trunking between Lync and FreeSWITCH
In-Reply-To: <BANLkTinqTbEGQLqativwZhZi9te6h+ompQ@mail.gmail.com>
References: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>
	<4DBBE861.9050906@xpirio.com>
	<761372A6-B060-42ED-8553-9EB3FFF13F7B@ipeva.fr>
	<BANLkTinqTbEGQLqativwZhZi9te6h+ompQ@mail.gmail.com>
Message-ID: <8B8AB46F-535E-4213-9BB5-82F957C7F96F@ipeva.fr>

Vitalie,

that was actually a good idea, and I just checked that on my trace.
Too bad, the ACK from FreeSWITCH is correctly received by Lync.

On my side, I think the next step will be to do a test without NAT, just to be on the safe side.

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

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

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




Le 30/04/2011 ? 22:07, Vitalie Colosov a ?crit :

> I saw the cases when call is disconnected after 32 seconds when one of the parties does not receive the ACK message after sending the OK message when call is answered.
> It was not with Lync, but with some crappy sip client.
> Maybe you can investigate in this direction.
> 
> Vitalie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gmaruzz at celliax.org  Sun May  1 23:32:41 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 1 May 2011 21:32:41 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTimubrY_jTRptCbL0oQC3pJkvLpT9g@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimubrY_jTRptCbL0oQC3pJkvLpT9g@mail.gmail.com>
Message-ID: <BANLkTikx7-vxWBU4bc0LFmfcpAZZR77iJA@mail.gmail.com>

Zohair,
please ***never*** answer to a mailing list  post  asking something
not immediately related to the original topic of tje original post.
Please make a new topic, write a new mail to the mailing list with a
"Subject" (the title of the mail) describing your question.
Thanks in advance,
-giovanni

On 5/1/11, zohair raza <engineerzuhairraza at gmail.com> wrote:
> Wow That is great Sir,
>
> i would like to add how i am passing caller id
>
> i am using xml curl and in my php file i added this :
>
> <?php
>
>   $calld  =  $_REQUEST['Caller-Destination-Number'];
>   $cally =str_replace("skype3439","",$calld) ; ( I am sending skype id like
> 'skype3439remoteskypeid'  so here i am removing skype3439)
>   $callsk   = $_REQUEST['Caller-Caller-ID-Number'];
>   $zr= ' api_result=${skypopen(interface1 MESSAGE '.$cally.' (mp) Call from
> '.$callsk.')}';
>
> header("Content-type: text/xml");
> $response = "
> <?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
> <document type=\"freeswitch/xml\">
>   <section name=\"dialplan\" description=\"example_curl_dialplan\">
>     <context name=\"public\">
>
>          <extension name=\"skype\">
>          <condition field=\"destination_number\"
> expression=\"^skype3439(.*)$\">
>          <action application=\"set\" data=\"skype_get_inband_dtmf=false\"/>
>          <action application=\"set\" data=\"$zr\"/>
>          <action application=\"set\" data=\"instant_ringback=true\"/>
>          <action application=\"bridge\" data=\"skypopen/RR/$1\"/>
>          </condition>
>          </extension>
>
>   </context>
>   </section>
> </document>
> ";
> echo $response;
> exit;
> ?>
>
>
> This way caller id will be sent before the call as a skype chat message
> which will only work if remoteskypeid has added the skype id which you are
> using, otherwise it won't work.
>
>
> On Sat, Apr 30, 2011 at 1:31 PM, Giovanni Maruzzelli
> <gmaruzz at gmail.com>wrote:
>
>> Dear FreeSWITCHers,
>>
>> after a fair amount of effort, I ended up with a new way to install
>> and use mod_skypopen on Linux.
>>
>> No more looking around the internet for the lost 2.0.0.72 Skype client for
>> ALSA.
>>
>> First, we can use the readily available Skype client for OSS.
>>
>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
>> driver, that's very easy to compile and install, and do not need to
>> mess with the operating system installation.)
>>
>> Second, I wrote an installer that automatically do all the tedious
>> work for you: download and install the skype client, create the config
>> directory for Skype clients, create the config file for mod_skypopen,
>> create the script that launches the Skype clients.
>>
>> I hope those improvements will lower the barriers for Skype calls on
>> FreeSWITCH.
>>
>> Actually is ludicrously simple now, and after you compile FreeSWITCH,
>> mod_skypopen and the skypopen.ko OSS driver it will take like less
>> than one minute to have a complete installation of mod_skypopen ready
>> to make and receive calls.
>>
>> All automatic, no more need to fiddle around with the Skype client
>> download, configurations, authorization, etc.
>>
>> Is all well tested, but maybe there are still some bugs, and maybe the
>> docs are not clear/easy enough.
>>
>> Please have a look at the new and improved wiki page and let me know
>> what do you think about (and maybe test the procedures).
>>
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>>
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>>
>> You must update to the latest git to have all the goodies.
>>
>> Thank you all for your support,
>>
>> -giovanni
>>
>> --
>> 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
>>
>
>
>
> --
> Regards,
> Zohair Raza
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Sun May  1 23:35:49 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 1 May 2011 21:35:49 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTikx7-vxWBU4bc0LFmfcpAZZR77iJA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimubrY_jTRptCbL0oQC3pJkvLpT9g@mail.gmail.com>
	<BANLkTikx7-vxWBU4bc0LFmfcpAZZR77iJA@mail.gmail.com>
Message-ID: <BANLkTi=xuE13V-mgGNPDXo0ixZsAXP7j+Q@mail.gmail.com>

Zohair,
anyway, many thanks for sharing this useful technique.
I will add it to the wiki page asap (or you can do it, if you like).
Thanks again
-giovanni

On 5/1/11, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Zohair,
> please ***never*** answer to a mailing list  post  asking something
> not immediately related to the original topic of tje original post.
> Please make a new topic, write a new mail to the mailing list with a
> "Subject" (the title of the mail) describing your question.
> Thanks in advance,
> -giovanni
>
> On 5/1/11, zohair raza <engineerzuhairraza at gmail.com> wrote:
>> Wow That is great Sir,
>>
>> i would like to add how i am passing caller id
>>
>> i am using xml curl and in my php file i added this :
>>
>> <?php
>>
>>   $calld  =  $_REQUEST['Caller-Destination-Number'];
>>   $cally =str_replace("skype3439","",$calld) ; ( I am sending skype id
>> like
>> 'skype3439remoteskypeid'  so here i am removing skype3439)
>>   $callsk   = $_REQUEST['Caller-Caller-ID-Number'];
>>   $zr= ' api_result=${skypopen(interface1 MESSAGE '.$cally.' (mp) Call
>> from
>> '.$callsk.')}';
>>
>> header("Content-type: text/xml");
>> $response = "
>> <?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
>> <document type=\"freeswitch/xml\">
>>   <section name=\"dialplan\" description=\"example_curl_dialplan\">
>>     <context name=\"public\">
>>
>>          <extension name=\"skype\">
>>          <condition field=\"destination_number\"
>> expression=\"^skype3439(.*)$\">
>>          <action application=\"set\"
>> data=\"skype_get_inband_dtmf=false\"/>
>>          <action application=\"set\" data=\"$zr\"/>
>>          <action application=\"set\" data=\"instant_ringback=true\"/>
>>          <action application=\"bridge\" data=\"skypopen/RR/$1\"/>
>>          </condition>
>>          </extension>
>>
>>   </context>
>>   </section>
>> </document>
>> ";
>> echo $response;
>> exit;
>> ?>
>>
>>
>> This way caller id will be sent before the call as a skype chat message
>> which will only work if remoteskypeid has added the skype id which you
>> are
>> using, otherwise it won't work.
>>
>>
>> On Sat, Apr 30, 2011 at 1:31 PM, Giovanni Maruzzelli
>> <gmaruzz at gmail.com>wrote:
>>
>>> Dear FreeSWITCHers,
>>>
>>> after a fair amount of effort, I ended up with a new way to install
>>> and use mod_skypopen on Linux.
>>>
>>> No more looking around the internet for the lost 2.0.0.72 Skype client
>>> for
>>> ALSA.
>>>
>>> First, we can use the readily available Skype client for OSS.
>>>
>>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
>>> driver, that's very easy to compile and install, and do not need to
>>> mess with the operating system installation.)
>>>
>>> Second, I wrote an installer that automatically do all the tedious
>>> work for you: download and install the skype client, create the config
>>> directory for Skype clients, create the config file for mod_skypopen,
>>> create the script that launches the Skype clients.
>>>
>>> I hope those improvements will lower the barriers for Skype calls on
>>> FreeSWITCH.
>>>
>>> Actually is ludicrously simple now, and after you compile FreeSWITCH,
>>> mod_skypopen and the skypopen.ko OSS driver it will take like less
>>> than one minute to have a complete installation of mod_skypopen ready
>>> to make and receive calls.
>>>
>>> All automatic, no more need to fiddle around with the Skype client
>>> download, configurations, authorization, etc.
>>>
>>> Is all well tested, but maybe there are still some bugs, and maybe the
>>> docs are not clear/easy enough.
>>>
>>> Please have a look at the new and improved wiki page and let me know
>>> what do you think about (and maybe test the procedures).
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>>>
>>> You must update to the latest git to have all the goodies.
>>>
>>> Thank you all for your support,
>>>
>>> -giovanni
>>>
>>> --
>>> 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
>>>
>>
>>
>>
>> --
>> Regards,
>> Zohair Raza
>>
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From engineerzuhairraza at gmail.com  Sun May  1 23:57:11 2011
From: engineerzuhairraza at gmail.com (zohair raza)
Date: Mon, 2 May 2011 00:57:11 +0500
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTikx7-vxWBU4bc0LFmfcpAZZR77iJA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimubrY_jTRptCbL0oQC3pJkvLpT9g@mail.gmail.com>
	<BANLkTikx7-vxWBU4bc0LFmfcpAZZR77iJA@mail.gmail.com>
Message-ID: <BANLkTim2zWP0miBjKvQx3m+tpMbwn+1o1g@mail.gmail.com>

sorry big boss.

noted !

Thanks

On Mon, May 2, 2011 at 12:32 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Zohair,
> please ***never*** answer to a mailing list  post  asking something
> not immediately related to the original topic of tje original post.
> Please make a new topic, write a new mail to the mailing list with a
> "Subject" (the title of the mail) describing your question.
> Thanks in advance,
> -giovanni
>
> On 5/1/11, zohair raza <engineerzuhairraza at gmail.com> wrote:
> > Wow That is great Sir,
> >
> > i would like to add how i am passing caller id
> >
> > i am using xml curl and in my php file i added this :
> >
> > <?php
> >
> >   $calld  =  $_REQUEST['Caller-Destination-Number'];
> >   $cally =str_replace("skype3439","",$calld) ; ( I am sending skype id
> like
> > 'skype3439remoteskypeid'  so here i am removing skype3439)
> >   $callsk   = $_REQUEST['Caller-Caller-ID-Number'];
> >   $zr= ' api_result=${skypopen(interface1 MESSAGE '.$cally.' (mp) Call
> from
> > '.$callsk.')}';
> >
> > header("Content-type: text/xml");
> > $response = "
> > <?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
> > <document type=\"freeswitch/xml\">
> >   <section name=\"dialplan\" description=\"example_curl_dialplan\">
> >     <context name=\"public\">
> >
> >          <extension name=\"skype\">
> >          <condition field=\"destination_number\"
> > expression=\"^skype3439(.*)$\">
> >          <action application=\"set\"
> data=\"skype_get_inband_dtmf=false\"/>
> >          <action application=\"set\" data=\"$zr\"/>
> >          <action application=\"set\" data=\"instant_ringback=true\"/>
> >          <action application=\"bridge\" data=\"skypopen/RR/$1\"/>
> >          </condition>
> >          </extension>
> >
> >   </context>
> >   </section>
> > </document>
> > ";
> > echo $response;
> > exit;
> > ?>
> >
> >
> > This way caller id will be sent before the call as a skype chat message
> > which will only work if remoteskypeid has added the skype id which you
> are
> > using, otherwise it won't work.
> >
> >
> > On Sat, Apr 30, 2011 at 1:31 PM, Giovanni Maruzzelli
> > <gmaruzz at gmail.com>wrote:
> >
> >> Dear FreeSWITCHers,
> >>
> >> after a fair amount of effort, I ended up with a new way to install
> >> and use mod_skypopen on Linux.
> >>
> >> No more looking around the internet for the lost 2.0.0.72 Skype client
> for
> >> ALSA.
> >>
> >> First, we can use the readily available Skype client for OSS.
> >>
> >> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
> >> driver, that's very easy to compile and install, and do not need to
> >> mess with the operating system installation.)
> >>
> >> Second, I wrote an installer that automatically do all the tedious
> >> work for you: download and install the skype client, create the config
> >> directory for Skype clients, create the config file for mod_skypopen,
> >> create the script that launches the Skype clients.
> >>
> >> I hope those improvements will lower the barriers for Skype calls on
> >> FreeSWITCH.
> >>
> >> Actually is ludicrously simple now, and after you compile FreeSWITCH,
> >> mod_skypopen and the skypopen.ko OSS driver it will take like less
> >> than one minute to have a complete installation of mod_skypopen ready
> >> to make and receive calls.
> >>
> >> All automatic, no more need to fiddle around with the Skype client
> >> download, configurations, authorization, etc.
> >>
> >> Is all well tested, but maybe there are still some bugs, and maybe the
> >> docs are not clear/easy enough.
> >>
> >> Please have a look at the new and improved wiki page and let me know
> >> what do you think about (and maybe test the procedures).
> >>
> >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
> >>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
> >>
> >>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
> >>
> >> You must update to the latest git to have all the goodies.
> >>
> >> Thank you all for your support,
> >>
> >> -giovanni
> >>
> >> --
> >> 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
> >>
> >
> >
> >
> > --
> > Regards,
> > Zohair Raza
> >
>
> --
> 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
>



-- 
Regards,
Zohair Raza
SuperTec Inc.
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From engineerzuhairraza at gmail.com  Mon May  2 00:35:27 2011
From: engineerzuhairraza at gmail.com (zohair raza)
Date: Mon, 2 May 2011 01:35:27 +0500
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTi=xuE13V-mgGNPDXo0ixZsAXP7j+Q@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimubrY_jTRptCbL0oQC3pJkvLpT9g@mail.gmail.com>
	<BANLkTikx7-vxWBU4bc0LFmfcpAZZR77iJA@mail.gmail.com>
	<BANLkTi=xuE13V-mgGNPDXo0ixZsAXP7j+Q@mail.gmail.com>
Message-ID: <BANLkTimo-h60MiD-33DDk4Kis230G-kbcQ@mail.gmail.com>

Thank you sir,

I have added it on wiki


On Mon, May 2, 2011 at 12:35 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Zohair,
> anyway, many thanks for sharing this useful technique.
> I will add it to the wiki page asap (or you can do it, if you like).
> Thanks again
> -giovanni
>
> On 5/1/11, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> > Zohair,
> > please ***never*** answer to a mailing list  post  asking something
> > not immediately related to the original topic of tje original post.
> > Please make a new topic, write a new mail to the mailing list with a
> > "Subject" (the title of the mail) describing your question.
> > Thanks in advance,
> > -giovanni
> >
> > On 5/1/11, zohair raza <engineerzuhairraza at gmail.com> wrote:
> >> Wow That is great Sir,
> >>
> >> i would like to add how i am passing caller id
> >>
> >> i am using xml curl and in my php file i added this :
> >>
> >> <?php
> >>
> >>   $calld  =  $_REQUEST['Caller-Destination-Number'];
> >>   $cally =str_replace("skype3439","",$calld) ; ( I am sending skype id
> >> like
> >> 'skype3439remoteskypeid'  so here i am removing skype3439)
> >>   $callsk   = $_REQUEST['Caller-Caller-ID-Number'];
> >>   $zr= ' api_result=${skypopen(interface1 MESSAGE '.$cally.' (mp) Call
> >> from
> >> '.$callsk.')}';
> >>
> >> header("Content-type: text/xml");
> >> $response = "
> >> <?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
> >> <document type=\"freeswitch/xml\">
> >>   <section name=\"dialplan\" description=\"example_curl_dialplan\">
> >>     <context name=\"public\">
> >>
> >>          <extension name=\"skype\">
> >>          <condition field=\"destination_number\"
> >> expression=\"^skype3439(.*)$\">
> >>          <action application=\"set\"
> >> data=\"skype_get_inband_dtmf=false\"/>
> >>          <action application=\"set\" data=\"$zr\"/>
> >>          <action application=\"set\" data=\"instant_ringback=true\"/>
> >>          <action application=\"bridge\" data=\"skypopen/RR/$1\"/>
> >>          </condition>
> >>          </extension>
> >>
> >>   </context>
> >>   </section>
> >> </document>
> >> ";
> >> echo $response;
> >> exit;
> >> ?>
> >>
> >>
> >> This way caller id will be sent before the call as a skype chat message
> >> which will only work if remoteskypeid has added the skype id which you
> >> are
> >> using, otherwise it won't work.
> >>
> >>
> >> On Sat, Apr 30, 2011 at 1:31 PM, Giovanni Maruzzelli
> >> <gmaruzz at gmail.com>wrote:
> >>
> >>> Dear FreeSWITCHers,
> >>>
> >>> after a fair amount of effort, I ended up with a new way to install
> >>> and use mod_skypopen on Linux.
> >>>
> >>> No more looking around the internet for the lost 2.0.0.72 Skype client
> >>> for
> >>> ALSA.
> >>>
> >>> First, we can use the readily available Skype client for OSS.
> >>>
> >>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
> >>> driver, that's very easy to compile and install, and do not need to
> >>> mess with the operating system installation.)
> >>>
> >>> Second, I wrote an installer that automatically do all the tedious
> >>> work for you: download and install the skype client, create the config
> >>> directory for Skype clients, create the config file for mod_skypopen,
> >>> create the script that launches the Skype clients.
> >>>
> >>> I hope those improvements will lower the barriers for Skype calls on
> >>> FreeSWITCH.
> >>>
> >>> Actually is ludicrously simple now, and after you compile FreeSWITCH,
> >>> mod_skypopen and the skypopen.ko OSS driver it will take like less
> >>> than one minute to have a complete installation of mod_skypopen ready
> >>> to make and receive calls.
> >>>
> >>> All automatic, no more need to fiddle around with the Skype client
> >>> download, configurations, authorization, etc.
> >>>
> >>> Is all well tested, but maybe there are still some bugs, and maybe the
> >>> docs are not clear/easy enough.
> >>>
> >>> Please have a look at the new and improved wiki page and let me know
> >>> what do you think about (and maybe test the procedures).
> >>>
> >>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
> >>>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
> >>>
> >>>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
> >>>
> >>> You must update to the latest git to have all the goodies.
> >>>
> >>> Thank you all for your support,
> >>>
> >>> -giovanni
> >>>
> >>> --
> >>> 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
> >>>
> >>
> >>
> >>
> >> --
> >> Regards,
> >> Zohair Raza
> >>
> >
> > --
> > Sent from my mobile device
> >
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
>
> --
> 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
>



-- 
Regards,
Zohair Raza
SuperTec Inc.
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From devel at omninet.eu  Mon May  2 11:05:01 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Mon, 2 May 2011 10:05:01 +0300
Subject: [Freeswitch-users] wrong end_stamp in CDRs in latest git
Message-ID: <A94CD7CAF7564127A95991C4694065B5@omni1.local>

Hello,

 

After updating to latest git I get in the CDRs as end_stamp (checked xml_cdr
and csv) instead of the real date and time: 1970-01-01 00:00:00

 

Has anybody else noticed this?

 

Thank you

Anestis

 

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From brad at tritelcomm.com  Mon May  2 11:11:54 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Mon, 2 May 2011 00:11:54 -0700
Subject: [Freeswitch-users] Reviewers needed: FreeSWITCH Cookbook
In-Reply-To: <1304256857833-6321346.post@n2.nabble.com>
References: <BANLkTikL+6i6ovRX1EsfzMoWzDiw3TvbcQ@mail.gmail.com>
	<1304256857833-6321346.post@n2.nabble.com>
Message-ID: <BANLkTinhb8VqELuJeJLeyQM5phBQd74QLQ@mail.gmail.com>

I meet the requirements and would love to test these things.

On Sun, May 1, 2011 at 6:34 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> mercutioviz wrote:
> > If you meet the above qualifications and would like to help out then
> > please
> > contact me off list for more information.
> Well, I wished I could help, but I failed one of the qualifications.
> -----FreeSWITCH hosted on a Seagate DockStar with OpenWRT.--
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Reviewers-needed-FreeSWITCH-Cookbook-tp6317742p6321346.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Mon May  2 11:23:26 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 2 May 2011 09:23:26 +0200
Subject: [Freeswitch-users] wrong end_stamp in CDRs in latest git
In-Reply-To: <A94CD7CAF7564127A95991C4694065B5@omni1.local>
References: <A94CD7CAF7564127A95991C4694065B5@omni1.local>
Message-ID: <64DFFD41-14D3-4642-8CC1-E7D7770AC25D@ipeva.fr>

Yes, I got the exact same thing since last friday.
Call duration is also wrong.

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

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

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




Le 02/05/2011 ? 09:05, Anestis Mavro a ?crit :

> Hello,
>  
> After updating to latest git I get in the CDRs as end_stamp (checked xml_cdr and csv) instead of the real date and time: 1970-01-01 00:00:00
>  
> Has anybody else noticed this?
>  
> Thank you
> 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 benkokakao at gmail.com  Mon May  2 12:24:59 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Mon, 2 May 2011 10:24:59 +0200
Subject: [Freeswitch-users] Looking for transcription of english sound-files
Message-ID: <BANLkTinGW_hip46XNqOd8YGrtL-qMhfCXg@mail.gmail.com>

Hi!

I'm looking for a transcript of the english sound-files. I've seen
some references scattered across the wiki(And references to missing
files) but have not yet found a complete transcript of the sound-files
in freeswitch-sounds-en-us-callie-8000-1.0.15.tar.gz.
Is there a transcript somewhere?

Best regards
Christian


From michal.bielicki at seventhsignal.de  Mon May  2 15:58:40 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Mon, 2 May 2011 13:58:40 +0200
Subject: [Freeswitch-users] Looking for transcription of english
	sound-files
In-Reply-To: <BANLkTinGW_hip46XNqOd8YGrtL-qMhfCXg@mail.gmail.com>
References: <BANLkTinGW_hip46XNqOd8YGrtL-qMhfCXg@mail.gmail.com>
Message-ID: <B7913E70-92B2-469D-BBB2-A275D484F4B8@seventhsignal.de>

Ich hab zwar kein Transscript aber das Budget fehlt gerade mal ;) W?rde es sinn machen ne sammlung zu machen, und diie Bestellung die wir bei GM Voices machen wollten mit einigen anderen gemeinsam zu machen ?

Gru? aus Berlin
cypromis

Am 02.05.2011 um 10:24 schrieb Christian Benke:

> Hi!
> 
> I'm looking for a transcript of the english sound-files. I've seen
> some references scattered across the wiki(And references to missing
> files) but have not yet found a complete transcript of the sound-files
> in freeswitch-sounds-en-us-callie-8000-1.0.15.tar.gz.
> Is there a transcript somewhere?
> 
> Best regards
> Christian
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



From benkokakao at gmail.com  Mon May  2 17:22:30 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Mon, 2 May 2011 15:22:30 +0200
Subject: [Freeswitch-users] Looking for transcription of english
	sound-files
In-Reply-To: <B7913E70-92B2-469D-BBB2-A275D484F4B8@seventhsignal.de>
References: <BANLkTinGW_hip46XNqOd8YGrtL-qMhfCXg@mail.gmail.com>
	<B7913E70-92B2-469D-BBB2-A275D484F4B8@seventhsignal.de>
Message-ID: <BANLkTim3G8ACrBckiwyO8qUQCFFOmfY9JA@mail.gmail.com>

Hu? Did i miss a discussion about doing prompts at GM Voices?! You are
talking about doing german prompts with them, right? Christian
Loeschenkohl showed similar interest a few weeks
ago(http://freeswitch-users.2379917.n2.nabble.com/German-voice-prompts-missing-exceptions-in-mod-say-de-c-td6174515.html).
We might be interested, it depends on the voice and the accent though...

Still, a transcript of the original english prompts would be great to
make sure we have a complete list before we go to a studio.

Best regards
Christian

2011/5/2 Michal Bielicki <michal.bielicki at seventhsignal.de>:
> Ich hab zwar kein Transscript aber das Budget fehlt gerade mal ;) W?rde es sinn machen ne sammlung zu machen, und diie Bestellung die wir bei GM Voices machen wollten mit einigen anderen gemeinsam zu machen ?
>
> Gru? aus Berlin
> cypromis
>
> Am 02.05.2011 um 10:24 schrieb Christian Benke:
>
>> Hi!
>>
>> I'm looking for a transcript of the english sound-files. I've seen
>> some references scattered across the wiki(And references to missing
>> files) but have not yet found a complete transcript of the sound-files
>> in freeswitch-sounds-en-us-callie-8000-1.0.15.tar.gz.
>> Is there a transcript somewhere?
>>
>> Best regards
>> Christian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
>
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Gesch?ftsf?hrer: Michal Bielicki
> Pers?nlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.: http://www.seventhsignal.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Travelling across the globe by bike - http://poab.org


From gourav at rentec.com  Mon May  2 17:33:52 2011
From: gourav at rentec.com (gourav at rentec.com)
Date: Mon, 02 May 2011 06:33:52 -0700
Subject: [Freeswitch-users] Reviewers needed: FreeSWITCH Cookbook
Message-ID: <11053664.13232.1304343232883.JavaMail.nabble@jim.nabble.com>

Would love to help you with this.  

We have multiple test installations of freeware version of freeswitch and one Cutadel test appliance.  


regards,
Gourav.-


From dave at dchorton.com  Mon May  2 17:57:57 2011
From: dave at dchorton.com (Dave Horton)
Date: Mon, 2 May 2011 09:57:57 -0400
Subject: [Freeswitch-users] help on creating an rpm
Message-ID: <B2FF20A3-3A39-4565-A287-01E719A65005@dchorton.com>

Can anyone provide some help on how I can create my own rpm of freeswitch?  What I want to do is package up freeswitch in an rpm, but with edited conf files that are suited to my customers.  I've seen that there is an rpm spec file (freeswitch.spec) in the base directory, so I took a shot at running that but didn't get very far, and I didn't see any relevant articles on the wiki.

What I've done so far:

- created the standard rpm build directory tree (BUILD  RPMS  SOURCES  SPECS  SRPMS)
- copied freeswitch.spec into SPECs
- ran rpmbuild

I get this:

[build at centos64-1 freeswitch-stock]$ rpmbuild -bb -v SPECS/freeswitch.spec 
error: File /usr/src/redhat/SOURCES/freeswitch-1.0.7.tar.bz2: No such file or directory

I was surprised at that error, since the Source directives in the spec file indicate that rpmbuild should grab the source via http urls.  (Eventually, I'm going to need to change this, since I'll it to use a source tarball that I create, with my edited conf files).

Given that someone did a great job of putting a spec file together, are there any docs on how to use it?

Dave

From mel0torme at gmail.com  Mon May  2 18:09:58 2011
From: mel0torme at gmail.com (Tom C)
Date: Mon, 2 May 2011 07:09:58 -0700
Subject: [Freeswitch-users] Call particular function in Lua script from
	console?
Message-ID: <BANLkTikH8htShcc9L5rkmrSmHkP4y9mjUA@mail.gmail.com>

Hi!  Is there any way to pass parameters to a Lua script, or to call a
particular function within a lua script, when calling it with the "lua"
command from the console?  (I can accomplish what I need using
global_setvar, but that's an extra step for the user.)

I'm looking for something more like this:

    freeswitch at internal> lua MyFunction("myContents") in MyLuaScriptFile.lua

Alternatively, is there a way to get input from the console, or otherwise
interact with the user?  Such as:
  stream:write("Enter user name:")
  cUserName = stream:readln()     --  :-)


For anyone who is curious, the global_setvar method of passing info to a
script can be done like this...

In the console, set the global variable(s), then call the script:

   freeswitch at internal> global_setvar myparameter=My Contents
   freeswitch at internal> lua MyLuaScriptFile.lua

(Note that there were no quotes around the string.)  And in the script:

   fsapi = freeswitch.API();
   myParameter = fsapi:execute("global_getvar", "myparameter")
   if string.len(myParameter) == 0 then
       -- Global variable does not exist or is empty!
       stream:write('ERROR: Please use "global_setvar myparameter=My
Contents'"to specify user.\n')
       return false
   else
       -- We retrieved contents of global.
       -- Might want to clean out the global to avoid later confusion.
       fsapi.:execute("global_setvar", "myparameter=")
   end
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From jcasale at activenetwerx.com  Mon May  2 18:16:31 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 2 May 2011 14:16:31 +0000
Subject: [Freeswitch-users] help on creating an rpm
In-Reply-To: <B2FF20A3-3A39-4565-A287-01E719A65005@dchorton.com>
References: <B2FF20A3-3A39-4565-A287-01E719A65005@dchorton.com>
Message-ID: <assp.01031a9203.92097A6A775D5147B1078E3F15430B92482D22@prato.activenetwerx.local>

> [build at centos64-1 freeswitch-stock]$ rpmbuild -bb -v SPECS/freeswitch.spec 
>error: File /usr/src/redhat/SOURCES/freeswitch-1.0.7.tar.bz2: No such file or directory

You never made an ~/.rpmmacros file so that rpmbuild knows about the new build root.


From michal.bielicki at seventhsignal.de  Mon May  2 18:28:21 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Mon, 2 May 2011 16:28:21 +0200
Subject: [Freeswitch-users] help on creating an rpm
In-Reply-To: <B2FF20A3-3A39-4565-A287-01E719A65005@dchorton.com>
References: <B2FF20A3-3A39-4565-A287-01E719A65005@dchorton.com>
Message-ID: <D043D32D-1EC6-4191-ABCE-7E0EAE6C9A1F@seventhsignal.de>

You seem to not be very experienced with RPM. You need to change the stuff in your sourcetree and than create the file freeswitch-1.0.7.tar.bz2 in SOURCES, containing all the source with all your changes.
rpm does not automagicall download anything, the source files have to be there.

Am 02.05.2011 um 15:57 schrieb Dave Horton:

> Can anyone provide some help on how I can create my own rpm of freeswitch?  What I want to do is package up freeswitch in an rpm, but with edited conf files that are suited to my customers.  I've seen that there is an rpm spec file (freeswitch.spec) in the base directory, so I took a shot at running that but didn't get very far, and I didn't see any relevant articles on the wiki.
> 
> What I've done so far:
> 
> - created the standard rpm build directory tree (BUILD  RPMS  SOURCES  SPECS  SRPMS)
> - copied freeswitch.spec into SPECs
> - ran rpmbuild
> 
> I get this:
> 
> [build at centos64-1 freeswitch-stock]$ rpmbuild -bb -v SPECS/freeswitch.spec 
> error: File /usr/src/redhat/SOURCES/freeswitch-1.0.7.tar.bz2: No such file or directory
> 
> I was surprised at that error, since the Source directives in the spec file indicate that rpmbuild should grab the source via http urls.  (Eventually, I'm going to need to change this, since I'll it to use a source tarball that I create, with my edited conf files).
> 
> Given that someone did a great job of putting a spec file together, are there any docs on how to use it?
> 
> Dave
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



From frank at rosengart.de  Mon May  2 18:47:30 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Mon, 02 May 2011 16:47:30 +0200
Subject: [Freeswitch-users] Play recording sound into conference
Message-ID: <4DBEC402.9090009@rosengart.de>

Hi,

I wonder if anyone has a recipe for the typical "please state your name
after the tone" ... "(play $recording) has joined the conference"
feature of conference systems.

The recording part is obvious. But how do I play audio into a conference
from the dialplan?

This is how I think it should work.
--snip--
<action application="playback" data="voicemail/vm-record_name1.wav"/>
<action application="record" data="/tmp/name-${uuid}.wav 5 200"/>
<action application="conference"
  data="$conference play /tmp/name-${uuid}.wav"/>
<action application="conference"
  data="$conference play conference/conf-has_joined.wav"/>
<action application="conference" data="$conference"/>

But the first 'conference play ..'  action already puts the call into
the conference.

Any idea?

Thanks!

Frank


From garmt.noname at gmail.com  Mon May  2 19:03:17 2011
From: garmt.noname at gmail.com (Grmt)
Date: Mon, 2 May 2011 17:03:17 +0200
Subject: [Freeswitch-users] Call particular function in Lua script from
	console?
In-Reply-To: <BANLkTikH8htShcc9L5rkmrSmHkP4y9mjUA@mail.gmail.com>
References: <BANLkTikH8htShcc9L5rkmrSmHkP4y9mjUA@mail.gmail.com>
Message-ID: <4dbec7b8.c9860e0a.7dbd.ffffdeb2@mx.google.com>

Why not use command line arguments and parse them in your lua script with
argv[x] (where x is 0,1,2 .)?

e.g.

lua call_report.lua yesterday

 

-- call_report.lua

print("enter a number")

a = io.read("*number")

 

if argv[1] == "yesterday" then

                print("Sorry we were closed, it was Sunday\n")

else

                print("calls today: \n" .. a * 12345)  -- we like to show
our boss that is was really busy

end

 

I suggest you start reading here:

http://www.lua.org/pil/1.html (lua getting started ...)

 

Note: If you want users to interact with freeswitch I recommend doing that
from outside fs and connect to fs with esl (and you could still use lua for
that).

 

Grmt

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tom C
Sent: Monday, 02 May, 2011 16:10
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Call particular function in Lua script from
console?

 

Hi!  Is there any way to pass parameters to a Lua script, or to call a
particular function within a lua script, when calling it with the "lua"
command from the console?  (I can accomplish what I need using
global_setvar, but that's an extra step for the user.)

 

I'm looking for something more like this:

 

    freeswitch at internal> lua MyFunction("myContents") in MyLuaScriptFile.lua

 

Alternatively, is there a way to get input from the console, or otherwise
interact with the user?  Such as:

  stream:write("Enter user name:")

  cUserName = stream:readln()     --  :-)

 

 

For anyone who is curious, the global_setvar method of passing info to a
script can be done like this...

 

In the console, set the global variable(s), then call the script:

 

   freeswitch at internal> global_setvar myparameter=My Contents

   freeswitch at internal> lua MyLuaScriptFile.lua

 

(Note that there were no quotes around the string.)  And in the script:

 

   fsapi = freeswitch.API();

   myParameter = fsapi:execute("global_getvar", "myparameter")

   if string.len(myParameter) == 0 then

       -- Global variable does not exist or is empty!

       stream:write('ERROR: Please use "global_setvar myparameter=My
Contents'"to specify user.\n')

       return false

   else

       -- We retrieved contents of global.  

       -- Might want to clean out the global to avoid later confusion.

       fsapi.:execute("global_setvar", "myparameter=")

   end

 

 

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From mel0torme at gmail.com  Mon May  2 19:20:54 2011
From: mel0torme at gmail.com (Tom C)
Date: Mon, 2 May 2011 08:20:54 -0700
Subject: [Freeswitch-users] Call particular function in Lua script from
	console?
In-Reply-To: <4dbec7b8.c9860e0a.7dbd.ffffdeb2@mx.google.com>
References: <BANLkTikH8htShcc9L5rkmrSmHkP4y9mjUA@mail.gmail.com>
	<4dbec7b8.c9860e0a.7dbd.ffffdeb2@mx.google.com>
Message-ID: <BANLkTi=xvS3cj9D=ARNBrtFrForY+7QsEA@mail.gmail.com>

That's exactly what I need.  Thank you!!!!



On Mon, May 2, 2011 at 8:03 AM, Grmt <garmt.noname at gmail.com> wrote:

>  Why not use command line arguments and parse them in your lua script with
> argv[x] (where x is 0,1,2 ?)?
>
> e.g.
>
> lua call_report.lua yesterday
>
>
>
> -- call_report.lua
>
> print(?enter a number?)
>
> a = io.read(?*number?)
>
>
>
> if argv[1] == ?yesterday? then
>
>                 print(?Sorry we were closed, it was Sunday\n?)
>
> else
>
>                 print(?calls today: \n? .. a * 12345)  -- we like to show
> our boss that is was really busy
>
> end
>
>
>
> I suggest you start reading here:
>
> http://www.lua.org/pil/1.html (lua getting started ...)
>
>
>
> Note: If you want users to interact with freeswitch I recommend doing that
> from outside fs and connect to fs with esl (and you could still use lua for
> that).
>
>
>
> Grmt
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tom C
> *Sent:* Monday, 02 May, 2011 16:10
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Call particular function in Lua script from
> console?
>
>
>
> Hi!  Is there any way to pass parameters to a Lua script, or to call a
> particular function within a lua script, when calling it with the "lua"
> command from the console?  (I can accomplish what I need using
> global_setvar, but that's an extra step for the user.)
>
>
>
> I'm looking for something more like this:
>
>
>
>     freeswitch at internal> lua MyFunction("myContents") in
> MyLuaScriptFile.lua
>
>
>
> Alternatively, is there a way to get input from the console, or otherwise
> interact with the user?  Such as:
>
>   stream:write("Enter user name:")
>
>   cUserName = stream:readln()     --  :-)
>
>
>
>
>
> For anyone who is curious, the global_setvar method of passing info to a
> script can be done like this...
>
>
>
> In the console, set the global variable(s), then call the script:
>
>
>
>    freeswitch at internal> global_setvar myparameter=My Contents
>
>    freeswitch at internal> lua MyLuaScriptFile.lua
>
>
>
> (Note that there were no quotes around the string.)  And in the script:
>
>
>
>    fsapi = freeswitch.API();
>
>    myParameter = fsapi:execute("global_getvar", "myparameter")
>
>    if string.len(myParameter) == 0 then
>
>        -- Global variable does not exist or is empty!
>
>        stream:write('ERROR: Please use "global_setvar myparameter=My
> Contents'"to specify user.\n')
>
>        return false
>
>    else
>
>        -- We retrieved contents of global.
>
>        -- Might want to clean out the global to avoid later confusion.
>
>        fsapi.:execute("global_setvar", "myparameter=")
>
>    end
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May  2 23:04:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 May 2011 12:04:11 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook: resources
Message-ID: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>

Hello all!

The response to my request for reviewers is overwhelming - thank you so
much! We have more than enough reviewers. However, any of you who wish to
read the draft and offer feedback are still welcome to do so. You just won't
be "official" Packt reviewers.

I'd like to focus attention on testing resources. For those of you who have
data center resources and such I would like to ask for your assistance. It
would be helpful to have a spare server (low power == okay, cuz it's only
for a few simultaneous calls max) with a DID & public IP address. This will
help us test various scenarios and make sure that we can do box2box and NAT
scenarios.

Please contact me off list if you have any servers, DIDs, etc. that could be
used for this purpose.

Thanks again to everyone who has volunteered to help! I would much rather
have to sift through dozens of emails from people who want to help than have
to keep begging and have no one step up. ;)  You guys are awesome - keep up
the good work.

-MC
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From eric at loopfx.com  Tue May  3 00:14:42 2011
From: eric at loopfx.com (Eric Beard)
Date: Mon, 2 May 2011 16:14:42 -0400
Subject: [Freeswitch-users] Call quality issues
In-Reply-To: <BANLkTi=sGLYWpnV-rmn78MngdsKq0x6hQg@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
	<BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A11A8@doctorbob.loop.local>
	<BANLkTind4gbC5p1RM1tk_Te=E-HngaMLJA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A123A@doctorbob.loop.local>
	<BANLkTi=sGLYWpnV-rmn78MngdsKq0x6hQg@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DF6A19C5@doctorbob.loop.local>

So, we finally figured this out.  The NIC in that machine is faulty.  We switched over from the add-in Intel NIC (which runs flawlessly for us in dozens of other machines) to the cheap onboard NIC and everything runs great.  I hate it when things are just sort of broken, and not completely broken - makes it hard to see where the problem really is.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, April 29, 2011 2:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call quality issues

Build skew is a possibility. We always recommend a make current for updating, just in case.
-MC
On Fri, Apr 29, 2011 at 11:03 AM, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
On the bad machine, I just did a make clean, re-ran bootstrap.sh, configure, make, then make install, and now it works fine.  I'm mystified.  I really wish I knew what the problem was so I can avoid it in the future.

The only possible cause I can think of is that at some point I needed to rebuild to use an extra module (like curl), and I did a make, make install, without doing a make clean, and somehow that was related?  Sounds iffy.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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: Friday, April 29, 2011 1:33 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call quality issues

What about the operating system? Same rev, same kernel, same timing?
-MC
On Fri, Apr 29, 2011 at 10:10 AM, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
It seems to be a CPU issue.

I started from scratch today with a new machine, same exact hardware, and I am running 20 calls at less than 1% CPU, with perfect audio quality (same as the bad machine before it went bad).

On the bad machine, 20 calls causes the CPU to be at 20%, and quality is shaky.  30 calls and it's terrible.  Only the freeswitch process is eating CPU.

I have diffed the conf folders and everything is the same.  This is driving me nuts.  I have no idea  what I did to that machine to ruin the performance.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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: Friday, April 29, 2011 12:57 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call quality issues


On Thu, Apr 28, 2011 at 2:52 PM, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
Hello,

I ran into some major call quality issues this week, and I'm trying to figure out how to troubleshoot things.  I've been running FreeSwitch for a few weeks, and suddenly a few days ago my call quality dropped drastically.  I had been running more than 100 concurrent calls, with the CPU at less than 20%, but now at 20 concurrent calls, the CPU is still at a little less than 20%, and the call quality is bad - any higher and calls go almost completely silent.  There is a direct correlation between the number of simultaneous calls and call quality.

I have tested against multiple gateways, same results against each, so it's not an issue with the gateway.

I have captured packets on the machine and analyzed them with Wireshark.  It seems like the inbound packets are all fine, no jitter or loss.  But the packets being sent by FreeSwitch are degraded.

One sample call showed:

Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%)

What is the topology of the network path for the above call? Also, where on the LAN/WAN did you capture these packets? Wrong timestamps and 14% dropped packets suggests that something on the network is interfering with the delivery of these packets in a timely manner.
-MC


_______________________________________________
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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From msc at freeswitch.org  Tue May  3 00:22:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 May 2011 13:22:08 -0700
Subject: [Freeswitch-users] Call quality issues
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DF6A19C5@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
	<BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A11A8@doctorbob.loop.local>
	<BANLkTind4gbC5p1RM1tk_Te=E-HngaMLJA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A123A@doctorbob.loop.local>
	<BANLkTi=sGLYWpnV-rmn78MngdsKq0x6hQg@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A19C5@doctorbob.loop.local>
Message-ID: <BANLkTinuhHxhE56BGbFVkPCzLbOFnCu8vw@mail.gmail.com>

On Mon, May 2, 2011 at 1:14 PM, Eric Beard <eric at loopfx.com> wrote:

> So, we finally figured this out.  The NIC in that machine is faulty.  We
> switched over from the add-in Intel NIC (which runs flawlessly for us in
> dozens of other machines) to the cheap onboard NIC and everything runs
> great.  I hate it when things are just sort of broken, and not completely
> broken ? makes it hard to see where the problem really is.
>

Thanks for the followup. This is a great example of a "problem with
FreeSWITCH" that really isn't a problem with FreeSWITCH. :)  I think I will
bookmark this thread for future reference for when someone says, "I'd like
to report a bug in FreeSWITCH..." This is also validation of the rule of
thumb we have for people who have FS in production: HAVE A SPARE MACHINE FOR
TESTING!!!

Nice work, Eric. Keep on FreeSWITCHin'!
-MC
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From msc at freeswitch.org  Tue May  3 01:11:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 May 2011 14:11:00 -0700
Subject: [Freeswitch-users] Play recording sound into conference
In-Reply-To: <4DBEC402.9090009@rosengart.de>
References: <4DBEC402.9090009@rosengart.de>
Message-ID: <BANLkTim3b7qdbweH3meXHruhnwEpuqCwbg@mail.gmail.com>

If you know the conference name then you can use the API-from-dialplan
syntax:
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan

something like:
<action application="set" data="tmp=${conference $conf_name play
${recording} async}"/>

Give it a try and let us know what happens.
-MC

On Mon, May 2, 2011 at 7:47 AM, Frank Rosengart <frank at rosengart.de> wrote:

> Hi,
>
> I wonder if anyone has a recipe for the typical "please state your name
> after the tone" ... "(play $recording) has joined the conference"
> feature of conference systems.
>
> The recording part is obvious. But how do I play audio into a conference
> from the dialplan?
>
> This is how I think it should work.
> --snip--
> <action application="playback" data="voicemail/vm-record_name1.wav"/>
> <action application="record" data="/tmp/name-${uuid}.wav 5 200"/>
> <action application="conference"
>  data="$conference play /tmp/name-${uuid}.wav"/>
> <action application="conference"
>  data="$conference play conference/conf-has_joined.wav"/>
> <action application="conference" data="$conference"/>
>
> But the first 'conference play ..'  action already puts the call into
> the conference.
>
> Any idea?
>
> Thanks!
>
> Frank
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue May  3 01:11:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 May 2011 14:11:49 -0700
Subject: [Freeswitch-users] Looking for transcription of english
	sound-files
In-Reply-To: <BANLkTim3G8ACrBckiwyO8qUQCFFOmfY9JA@mail.gmail.com>
References: <BANLkTinGW_hip46XNqOd8YGrtL-qMhfCXg@mail.gmail.com>
	<B7913E70-92B2-469D-BBB2-A275D484F4B8@seventhsignal.de>
	<BANLkTim3G8ACrBckiwyO8qUQCFFOmfY9JA@mail.gmail.com>
Message-ID: <BANLkTimG_PFQL5=eODK_gFakHdPUGF8eMw@mail.gmail.com>

Under FreeSWITCH source directory: docs/phrase/phrase_en.xml

-MC

On Mon, May 2, 2011 at 6:22 AM, Christian Benke <benkokakao at gmail.com>wrote:

> Hu? Did i miss a discussion about doing prompts at GM Voices?! You are
> talking about doing german prompts with them, right? Christian
> Loeschenkohl showed similar interest a few weeks
> ago(
> http://freeswitch-users.2379917.n2.nabble.com/German-voice-prompts-missing-exceptions-in-mod-say-de-c-td6174515.html
> ).
> We might be interested, it depends on the voice and the accent though...
>
> Still, a transcript of the original english prompts would be great to
> make sure we have a complete list before we go to a studio.
>
> Best regards
> Christian
>
> 2011/5/2 Michal Bielicki <michal.bielicki at seventhsignal.de>:
> > Ich hab zwar kein Transscript aber das Budget fehlt gerade mal ;) W?rde
> es sinn machen ne sammlung zu machen, und diie Bestellung die wir bei GM
> Voices machen wollten mit einigen anderen gemeinsam zu machen ?
> >
> > Gru? aus Berlin
> > cypromis
> >
> > Am 02.05.2011 um 10:24 schrieb Christian Benke:
> >
> >> Hi!
> >>
> >> I'm looking for a transcript of the english sound-files. I've seen
> >> some references scattered across the wiki(And references to missing
> >> files) but have not yet found a complete transcript of the sound-files
> >> in freeswitch-sounds-en-us-callie-8000-1.0.15.tar.gz.
> >> Is there a transcript somewhere?
> >>
> >> Best regards
> >> Christian
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > Michal Bielicki
> > Gesch?ftsf?hrer / CEO
> >
> > Seventh Signal Ltd. & Co. KG
> > Weigandufer 45, B?ro 115, D-12059 Berlin
> > Voice: +49 30 60988730
> >
> > Amtsgericht Charlottenburg HRA 44413 B
> > Ust.-ID: DE266981999
> > Gesch?ftsf?hrer: Michal Bielicki
> > Pers?nlich Haftende Gesellschafterin:
> > Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> > B18 6EW, GB, Company Nr.: 06889439
> > WWW.: http://www.seventhsignal.de
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Travelling across the globe by bike - http://poab.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue May  3 01:12:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 May 2011 14:12:41 -0700
Subject: [Freeswitch-users] wrong end_stamp in CDRs in latest git
In-Reply-To: <64DFFD41-14D3-4642-8CC1-E7D7770AC25D@ipeva.fr>
References: <A94CD7CAF7564127A95991C4694065B5@omni1.local>
	<64DFFD41-14D3-4642-8CC1-E7D7770AC25D@ipeva.fr>
Message-ID: <BANLkTimVh_UkjP=DOv30y4SzahHaOYJOLg@mail.gmail.com>

Update to latest and try again. Tony added a patch late last week.
-MC

2011/5/2 David Ponzone <david.ponzone at ipeva.fr>

> Yes, I got the exact same thing since last friday.
> Call duration is also wrong.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 02/05/2011 ? 09:05, Anestis Mavro a ?crit :
>
> Hello,
>
> After updating to latest git I get in the CDRs as end_stamp (checked
> xml_cdr and csv) instead of the real date and time: 1970-01-01 00:00:00
>
> Has anybody else noticed this?
>
> Thank you
> 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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From garmt.noname at gmail.com  Tue May  3 01:26:48 2011
From: garmt.noname at gmail.com (Grmt)
Date: Mon, 2 May 2011 23:26:48 +0200
Subject: [Freeswitch-users] wrong end_stamp in CDRs in latest git
In-Reply-To: <BANLkTimVh_UkjP=DOv30y4SzahHaOYJOLg@mail.gmail.com>
References: <A94CD7CAF7564127A95991C4694065B5@omni1.local>
	<64DFFD41-14D3-4642-8CC1-E7D7770AC25D@ipeva.fr>
	<BANLkTimVh_UkjP=DOv30y4SzahHaOYJOLg@mail.gmail.com>
Message-ID: <4dbf219a.90870e0a.258b.fffff10a@mx.google.com>

And Jeff  ?gitted? another patch today


 

http://jira.freeswitch.org/browse/FS-3280

 

another example of how quickly and adequately the community fixes issues
together, good stuff!

 

Grmt

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, 02 May, 2011 23:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] wrong end_stamp in CDRs in latest git

 

Update to latest and try again. Tony added a patch late last week.

-MC

2011/5/2 David Ponzone <david.ponzone at ipeva.fr>

Yes, I got the exact same thing since last friday.

Call duration is also wrong.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

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

 

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

 

 

 

Le 02/05/2011 ? 09:05, Anestis Mavro a ?crit :





Hello,

 

After updating to latest git I get in the CDRs as end_stamp (checked xml_cdr
and csv) instead of the real date and time: 1970-01-01 00:00:00

 

Has anybody else noticed this?

 

Thank you

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

 


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

 

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From Info at KennedySoftware.ie  Tue May  3 02:25:30 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Mon, 02 May 2011 23:25:30 +0100
Subject: [Freeswitch-users] FreeSWITCH Cookbook: resources
In-Reply-To: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
References: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
Message-ID: <4DBF2F5A.30908@KennedySoftware.ie>

Michael,

I probably cannot help much with your current "proofing" needs, but, I'm 
offering!!

   - I've never installed FS anywhere - but I will be doing so in the 
next few days (under Ubuntu server, 8.04/10.04).

   - I am a "proofer" (with the "R"!!) for a few linux pubs, the best 
known being "FullCircleMagazine.org".

   - I'm a bit of a PITA regarding syntax, spelling, punctuation, etc.

   - I got the FS book a few months ago - very good publication, though 
I would have liked a few chapters on legacy telephony technology, 
products & services, and converting to FS, etc...

   - Formally, I'm an electronics engineer (digital), but I've been 
full-time programming since 1968 - started in IBM then, on various 
Assemblers.

So... if I can help.... I'll be very happy to oblige.

I'm in Dublin, Ireland, so my other query would be where that Collins 
surname came from, but that's for another thread ;-)

Very best regards, and very many thanks for the huge efforts - by all of 
you - into the FS project...

   - Mike Kennedy

On 02/05/2011 20:04, Michael Collins wrote:
> Hello all!
>
> The response to my request for reviewers is overwhelming - thank you so
> much! We have more than enough reviewers. However, any of you who wish
> to read the draft and offer feedback are still welcome to do so. You
> just won't be "official" Packt reviewers.
>
> I'd like to focus attention on testing resources. For those of you who
> have data center resources and such I would like to ask for your
> assistance. It would be helpful to have a spare server (low power ==
> okay, cuz it's only for a few simultaneous calls max) with a DID &
> public IP address. This will help us test various scenarios and make
> sure that we can do box2box and NAT scenarios.
>
> Please contact me off list if you have any servers, DIDs, etc. that
> could be used for this purpose.
>
> Thanks again to everyone who has volunteered to help! I would much
> rather have to sift through dozens of emails from people who want to
> help than have to keep begging and have no one step up. ;)  You guys are
> awesome - keep up the good work.
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From avi at avimarcus.net  Tue May  3 02:27:09 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 3 May 2011 01:27:09 +0300
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
Message-ID: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>

I've tried this cool formula for streaming TTS via google:

http://wiki.freeswitch.org/wiki/TTS

and while the link produces a pretty darn nice sounding MP3, I get:
2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at
/usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123: Invalid
mpg123 handle. (code 10)

Is anyone else using this reliably?

Thanks,
Avi
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From msc at freeswitch.org  Tue May  3 02:42:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 May 2011 15:42:39 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook: resources
In-Reply-To: <4DBF2F5A.30908@KennedySoftware.ie>
References: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
	<4DBF2F5A.30908@KennedySoftware.ie>
Message-ID: <BANLkTikBHQoUY73xTM1mVh9aQ0MRXWnoXQ@mail.gmail.com>

Mike,

Thanks for checking in! I'll add you to the list of people who receive the
draft. I have a nice list. :)

As for my name, I definitely am of Irish decent. My father's paternal
grandfather emigrated from Ireland in early 20th century. I suppose with a
name like "Michael Sean Collins" I'd better have at least a little Irish
blood. :)

Thanks,
MC

On Mon, May 2, 2011 at 3:25 PM, Michael Kennedy <Info at kennedysoftware.ie>wrote:

> Michael,
>
> I probably cannot help much with your current "proofing" needs, but, I'm
> offering!!
>
>   - I've never installed FS anywhere - but I will be doing so in the
> next few days (under Ubuntu server, 8.04/10.04).
>
>   - I am a "proofer" (with the "R"!!) for a few linux pubs, the best
> known being "FullCircleMagazine.org".
>
>   - I'm a bit of a PITA regarding syntax, spelling, punctuation, etc.
>
>   - I got the FS book a few months ago - very good publication, though
> I would have liked a few chapters on legacy telephony technology,
> products & services, and converting to FS, etc...
>
>   - Formally, I'm an electronics engineer (digital), but I've been
> full-time programming since 1968 - started in IBM then, on various
> Assemblers.
>
> So... if I can help.... I'll be very happy to oblige.
>
> I'm in Dublin, Ireland, so my other query would be where that Collins
> surname came from, but that's for another thread ;-)
>
> Very best regards, and very many thanks for the huge efforts - by all of
> you - into the FS project...
>
>   - Mike Kennedy
>
> On 02/05/2011 20:04, Michael Collins wrote:
> > Hello all!
> >
> > The response to my request for reviewers is overwhelming - thank you so
> > much! We have more than enough reviewers. However, any of you who wish
> > to read the draft and offer feedback are still welcome to do so. You
> > just won't be "official" Packt reviewers.
> >
> > I'd like to focus attention on testing resources. For those of you who
> > have data center resources and such I would like to ask for your
> > assistance. It would be helpful to have a spare server (low power ==
> > okay, cuz it's only for a few simultaneous calls max) with a DID &
> > public IP address. This will help us test various scenarios and make
> > sure that we can do box2box and NAT scenarios.
> >
> > Please contact me off list if you have any servers, DIDs, etc. that
> > could be used for this purpose.
> >
> > Thanks again to everyone who has volunteered to help! I would much
> > rather have to sift through dozens of emails from people who want to
> > help than have to keep begging and have no one step up. ;)  You guys are
> > awesome - keep up the good work.
> >
> > -MC
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Tue May  3 04:17:56 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 3 May 2011 08:17:56 +0800
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
Message-ID: <C99843A16043492594E175C6A597474F@gmail.com>

I tested a few days ago on a Mac, and it was fine.

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
On Tuesday, May 3, 2011 at 6:27 AM, Avi Marcus wrote: 
> I've tried this cool formula for streaming TTS via google:
> 
> http://wiki.freeswitch.org/wiki/TTS 
> 
> and while the link produces a pretty darn nice sounding MP3, I get:
> 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123: Invalid mpg123 handle. (code 10)
> 
> 
> Is anyone else using this reliably?
> 
> Thanks,
> Avi
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
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From anthony.minessale at gmail.com  Tue May  3 04:21:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 May 2011 19:21:59 -0500
Subject: [Freeswitch-users] Play recording sound into conference
In-Reply-To: <BANLkTim3b7qdbweH3meXHruhnwEpuqCwbg@mail.gmail.com>
References: <4DBEC402.9090009@rosengart.de>
	<BANLkTim3b7qdbweH3meXHruhnwEpuqCwbg@mail.gmail.com>
Message-ID: <BANLkTikY_axius8iTCx9R6ZKJXPEgnskhQ@mail.gmail.com>

or use the read application from the dp to record the name then set
conference_enter_sound variable to the path to the file
and use api_hangup_hook to delete it


On Mon, May 2, 2011 at 4:11 PM, Michael Collins <msc at freeswitch.org> wrote:
> If you know the conference name then you can use the API-from-dialplan
> syntax:
> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
> something like:
> <action application="set" data="tmp=${conference $conf_name play
> ${recording} async}"/>
> Give it a try and let us know what happens.
> -MC
>
> On Mon, May 2, 2011 at 7:47 AM, Frank Rosengart <frank at rosengart.de> wrote:
>>
>> Hi,
>>
>> I wonder if anyone has a recipe for the typical "please state your name
>> after the tone" ... "(play $recording) has joined the conference"
>> feature of conference systems.
>>
>> The recording part is obvious. But how do I play audio into a conference
>> from the dialplan?
>>
>> This is how I think it should work.
>> --snip--
>> <action application="playback" data="voicemail/vm-record_name1.wav"/>
>> <action application="record" data="/tmp/name-${uuid}.wav 5 200"/>
>> <action application="conference"
>> ?data="$conference play /tmp/name-${uuid}.wav"/>
>> <action application="conference"
>> ?data="$conference play conference/conf-has_joined.wav"/>
>> <action application="conference" data="$conference"/>
>>
>> But the first 'conference play ..' ?action already puts the call into
>> the conference.
>>
>> Any idea?
>>
>> Thanks!
>>
>> Frank
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 dujinfang at gmail.com  Tue May  3 04:50:50 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 3 May 2011 08:50:50 +0800
Subject: [Freeswitch-users] skypopen scalability tests
In-Reply-To: <BANLkTimnJ1wRTmVRJ8sw1ed57HdKUZxDZA@mail.gmail.com>
References: <BANLkTinMytjFaMc4zA0pfmgV6dMqEHu7WQ@mail.gmail.com>
	<BANLkTimnJ1wRTmVRJ8sw1ed57HdKUZxDZA@mail.gmail.com>
Message-ID: <A3C80138B34C4F2897F8374A374B611A@gmail.com>

Perhaps there are two ways

1) each one interested in this list can config 20 different users on their server, and then you can call 200 if we get 10 people. The other party just need to create a dialplan and plan MOH.

2) 200 different clients can call your server at a certain time, with same username, say Bob, is that reliable?

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
On Sunday, May 1, 2011 at 1:09 AM, Giovanni Maruzzelli wrote: 
> Ooops, I was wrong in my previous mail (my memory no more what was
> used to be :) ).
> 
> You cannot have simultaneus calls from many instances of Bob to many
> instances of Alice. It just does not works reliably.
> 
> I was testing simultaneus calls from one client skypeusername (Bob) to
> a server with many skypeusernames (Alice, Adam, etc).
> 
> So, we're back in the situation I described (need to register many
> different skypeusernames) with the added drawback that simultaneus
> calls from one Skype client does not work well.
> 
> Any other ideas?
> -giovanni
> 
> On Sat, Apr 30, 2011 at 7:01 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> > On Sat, Apr 30, 2011 at 6:52 PM, Anton VG <anton.vazir at gmail.com> wrote:
> > > > Any other ideas on how to proceed for scalability tests?
> > > 
> > > Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to
> > > register lots of usernames, just 2 of them for 2 PC's
> > > than make 100 calls from 1 machine to another
> > 
> > I was going that way, couple years ago, but the Skype client was
> > scaling kind of badly for me. No more than a bunch of simultaneus
> > calls.
> > But at that time I was testing using a standard ALSA sound driver, and
> > Skype client for ALSA.
> > Maybe using Skype for OSS and skypopen.ko it will scale better.
> > 
> > Good hint, at least I will check into it in the future.
> > 
> > Other ideas?
> > 
> > -giovanni
> > 
> > --
> > Sincerely,
> > 
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> 
> 
> 
> -- 
> 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 marcdecorny at gmail.com  Tue May  3 04:57:52 2011
From: marcdecorny at gmail.com (Marc De Corny)
Date: Tue, 3 May 2011 01:57:52 +0100
Subject: [Freeswitch-users] FreeSWITCH Cookbook: resources
In-Reply-To: <BANLkTikBHQoUY73xTM1mVh9aQ0MRXWnoXQ@mail.gmail.com>
References: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
	<4DBF2F5A.30908@KennedySoftware.ie>
	<BANLkTikBHQoUY73xTM1mVh9aQ0MRXWnoXQ@mail.gmail.com>
Message-ID: <A2473FE2-75CC-47BC-944E-D87952886CF9@gmail.com>

Hi Michael

I have a small production instance and two test setups. I bought the first book and was very impressed by the detail though could have done with a bit more about fifo etc but maybe it was thick enough :)

I m from a nortel/genband carrier voip / SBC background looking at integrating more advanced features into the network and i have so far never experienced so few interworking issues.

I heard today that one of our customers is trying to hook one up to the network in order to carry out proper testing as PBX

So
1. I would be pleased to proof read and test what I can.

2. Do you have an interop doc that i could complete if you don't already have it as proof that it was tested with a nortel/genband cs2000?

Have a good evening

Thanks
Marc



On 2 May 2011, at 23:42, Michael Collins <msc at freeswitch.org> wrote:

> Mike,
> 
> Thanks for checking in! I'll add you to the list of people who receive the draft. I have a nice list. :)
> 
> As for my name, I definitely am of Irish decent. My father's paternal grandfather emigrated from Ireland in early 20th century. I suppose with a name like "Michael Sean Collins" I'd better have at least a little Irish blood. :)
> 
> Thanks,
> MC
> 
> On Mon, May 2, 2011 at 3:25 PM, Michael Kennedy <Info at kennedysoftware.ie> wrote:
> Michael,
> 
> I probably cannot help much with your current "proofing" needs, but, I'm
> offering!!
> 
>   - I've never installed FS anywhere - but I will be doing so in the
> next few days (under Ubuntu server, 8.04/10.04).
> 
>   - I am a "proofer" (with the "R"!!) for a few linux pubs, the best
> known being "FullCircleMagazine.org".
> 
>   - I'm a bit of a PITA regarding syntax, spelling, punctuation, etc.
> 
>   - I got the FS book a few months ago - very good publication, though
> I would have liked a few chapters on legacy telephony technology,
> products & services, and converting to FS, etc...
> 
>   - Formally, I'm an electronics engineer (digital), but I've been
> full-time programming since 1968 - started in IBM then, on various
> Assemblers.
> 
> So... if I can help.... I'll be very happy to oblige.
> 
> I'm in Dublin, Ireland, so my other query would be where that Collins
> surname came from, but that's for another thread ;-)
> 
> Very best regards, and very many thanks for the huge efforts - by all of
> you - into the FS project...
> 
>   - Mike Kennedy
> 
> On 02/05/2011 20:04, Michael Collins wrote:
> > Hello all!
> >
> > The response to my request for reviewers is overwhelming - thank you so
> > much! We have more than enough reviewers. However, any of you who wish
> > to read the draft and offer feedback are still welcome to do so. You
> > just won't be "official" Packt reviewers.
> >
> > I'd like to focus attention on testing resources. For those of you who
> > have data center resources and such I would like to ask for your
> > assistance. It would be helpful to have a spare server (low power ==
> > okay, cuz it's only for a few simultaneous calls max) with a DID &
> > public IP address. This will help us test various scenarios and make
> > sure that we can do box2box and NAT scenarios.
> >
> > Please contact me off list if you have any servers, DIDs, etc. that
> > could be used for this purpose.
> >
> > Thanks again to everyone who has volunteered to help! I would much
> > rather have to sift through dozens of emails from people who want to
> > help than have to keep begging and have no one step up. ;)  You guys are
> > awesome - keep up the good work.
> >
> > -MC
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From OSchenk at wnr.com.au  Tue May  3 08:02:52 2011
From: OSchenk at wnr.com.au (Schenk, Oliver)
Date: Tue, 3 May 2011 12:02:52 +0800
Subject: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin
Message-ID: <A4544360CBA2AC4298EE6AA1350E53AEC8A5B5@WNRCOLEX01.wnr.internal>

Hi All,

 

I'm completely new to FreeSWITCH, but in a week I managed to achieve
quite a bit. It was quite a learning curve to say the least. This is my
current setup:

 

-          FreeSWITCH 1.0.7 built from tarball.

-          Developing on Windows XP with VS2008 in C#.

-          Configured extension 1024 to connect to a managed dll file,
which implements IAppPlugin.

-          Using X-Lite softphone I can connect to FreeSWITCH as user
1001 and dial 1024 and start going through a menu that I built in C#. It
successfully retrieves records from an SQL database and so forth. No
problems there.

 

My first step was to create a module that gets called whenever a user
dials IN bound. They will hear the menu just described.

 

 

My struggle now is relating to OUT bound calls. What I want is a module
that is started as soon as FreeSWITCH is started and begins executing on
an endless processing loop in the background. This will continuously
monitor a database and if certain conditions occur an outbound call
should be queued and then made. If multiple calls need to be made I
guess they will be queued and processed one by one. I can handle the
queuing part.

 

At this stage I will be testing using extension 1001 as the receiver of
the call using my softphone.

 

 

Question 1:

 

I've been trying to use a class that implements IApiPlugin, but how do I
get it to start when FreeSWITCH starts? As I said it should simply be a
never ending thread as long as FreeSWITCH is running. I can't find any
information regarding how to "Execute" a managed module immediately when
FreeSWITCH has started.

 

 

Question 2:

 

If IApiPlugin can do this, how do I get the session object? Like this?

ManagedSession session = new ManagedSession();

session.Originate(???);

 

I can't find any help at all on this.

 

 

Thanks very much!

 

 

 

Oliver Schenk

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.
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From adminjew at gmail.com  Tue May  3 09:56:22 2011
From: adminjew at gmail.com (Yitzchok)
Date: Tue, 3 May 2011 01:56:22 -0400
Subject: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin
In-Reply-To: <A4544360CBA2AC4298EE6AA1350E53AEC8A5B5@WNRCOLEX01.wnr.internal>
References: <A4544360CBA2AC4298EE6AA1350E53AEC8A5B5@WNRCOLEX01.wnr.internal>
Message-ID: <BANLkTi==m6_WYtNE5bXHVywBWSgb3mzubg@mail.gmail.com>

The correct way to do this is to connect to FreeSWITCH using
ESL<http://wiki.freeswitch.org/wiki/ESL> from
your application.


Have your application that polls your database run outside of freeswitch and
when you need to create a new call connect to FS using ESL (there is a .NET
lib in \libs\esl\managed\) and create a new call with the Originate command
that bridges to an extension or application something like this "originate
sofia/gateway/name/8885551234 *1024*" or "originate
sofia/gateway/name/8885551234 *&managed(ClassNameToIApiPlugin)*".


Yitzchok


On Tue, May 3, 2011 at 12:02 AM, Schenk, Oliver <OSchenk at wnr.com.au> wrote:

>  Hi All,
>
>
>
> I?m completely new to FreeSWITCH, but in a week I managed to achieve quite
> a bit. It was quite a learning curve to say the least. This is my current
> setup:
>
>
>
> -          FreeSWITCH 1.0.7 built from tarball.
>
> -          Developing on Windows XP with VS2008 in C#.
>
> -          Configured extension 1024 to connect to a managed dll file,
> which implements IAppPlugin.
>
> -          Using X-Lite softphone I can connect to FreeSWITCH as user 1001
> and dial 1024 and start going through a menu that I built in C#. It
> successfully retrieves records from an SQL database and so forth. No
> problems there.
>
>
>
> My first step was to create a module that gets called whenever a user dials
> IN bound. They will hear the menu just described.
>
>
>
>
>
> My struggle now is relating to OUT bound calls. What I want is a module
> that is started as soon as FreeSWITCH is started and begins executing on an
> endless processing loop in the background. This will continuously monitor a
> database and if certain conditions occur an outbound call should be queued
> and then made. If multiple calls need to be made I guess they will be queued
> and processed one by one. I can handle the queuing part.
>
>
>
> At this stage I will be testing using extension 1001 as the receiver of the
> call using my softphone.
>
>
>
>
>
> Question 1:
>
>
>
> I?ve been trying to use a class that implements IApiPlugin, but how do I
> get it to start when FreeSWITCH starts? As I said it should simply be a
> never ending thread as long as FreeSWITCH is running. I can?t find any
> information regarding how to ?Execute? a managed module immediately when
> FreeSWITCH has started.
>
>
>
>
>
> Question 2:
>
>
>
> If IApiPlugin can do this, how do I get the session object? Like this?
>
> ManagedSession session = new ManagedSession();
>
> session.Originate(???);
>
>
>
> I can?t find any help at all on this.
>
>
>
>
>
> Thanks very much!
>
>
>
>
>
>
>
> *Oliver Schenk*
>
>
>
> NOTICE - This e-mail and any files transmitted with it are confidential and
> are only for the use of the person to whom they are addressed.
> If you are not the intended recipient then you have received this e-mail in
> error; please advise us immediately if this is the case.
> Any views expressed in this message are those of the individual sender,
> except where the sender specifically states them to be the views of WestNet
> Rail.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From simon0922 at gmail.com  Tue May  3 10:01:51 2011
From: simon0922 at gmail.com (Simon Leck)
Date: Tue, 3 May 2011 14:01:51 +0800
Subject: [Freeswitch-users] FreeSWITCH  calls unable to be released issue
Message-ID: <01ca01cc0957$9ab1f3a0$d015dae0$@gmail.com>

Dear All,

 

Re: FreeSWITCH  calls unable to be released issue:

 

At this moment we are facing some major issue in Freeswitch hope somebody
out there can give us some kind advice on how to rectify the issue mentioned
below.

 

The major problem we are facing at this stage is "Channels or Calls unable
to be released", and we are sure the response of command we used"show calls"
and "show channels" are correct and reliable.

 

We found this situation, after we pass through massive traffic to
FreeSWITCH, and then, get into the fs_cli, and exec "status", "show
channels" and "show calls", we will have different numbers for each
response, even when we turn off the source traffic, the "status" response 0
sessions, and type "show calls" we will still see 5, 10 or more calls on it,
and when we type "show channels" it would show 6,15 or more channels; we did
try to modify mod_command to retrieve the status from core.db, and the
results are still the same, the source data from core.db do not matched,
some calls / channels still could not be released.

 

We had studied the wiki to find solutions, put core.db to tmpfs, setting
higher memory. etc, but all still could not worked, till now we are still
plague by these problems, We then more a step further by trying to figure
out how to put core.db into mysql or some database via doing research on the
internet, but we have no luck as there are no further information about how
to put core.db into mysql. Please somebody out there if you know how to do
this kindly give us a guidance or point us to somebody who could advise us
on how to rectify this issue. 

 

Yet another issue we have on hand is Freeswitch could not stop fluently when
there are calls has not been fully released and this cause our status
Monitoring system to fail and could not restart Freeswitch. 

 

Looking forward to getting some prompt reply soonest. 

 

Thanks again Everybody.

Simon Leck

IM: simonleck at worldteltech.com
Cell: +886918981838

 

 

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From peter.olsson at visionutveckling.se  Tue May  3 10:09:38 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 3 May 2011 08:09:38 +0200
Subject: [Freeswitch-users] FreeSWITCH calls unable to be released issue
In-Reply-To: <01ca01cc0957$9ab1f3a0$d015dae0$@gmail.com>
References: <01ca01cc0957$9ab1f3a0$d015dae0$@gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F12761A4@cooper>

Hi,

First of all - are you running latest GIT HEAD? If you're not, please upgrade first. If you are, please file a Jira with debug information attached to it, so we can find the cause for this.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Simon Leck [simon0922 at gmail.com]
Skickat: den 3 maj 2011 08:01
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] FreeSWITCH  calls unable to be released issue

Dear All,

Re: FreeSWITCH  calls unable to be released issue:

At this moment we are facing some major issue in Freeswitch hope somebody out there can give us some kind advice on how to rectify the issue mentioned below.

The major problem we are facing at this stage is ?Channels or Calls unable to be released?, and we are sure the response of command we used?show calls? and ?show channels? are correct and reliable.

We found this situation, after we pass through massive traffic to FreeSWITCH, and then, get into the fs_cli, and exec ?status?, ?show channels? and ?show calls?, we will have different numbers for each response, even when we turn off the source traffic, the ?status? response 0 sessions, and type ?show calls? we will still see 5, 10 or more calls on it, and when we type ?show channels? it would show 6,15 or more channels; we did try to modify mod_command to retrieve the status from core.db, and the results are still the same, the source data from core.db do not matched, some calls / channels still could not be released.

We had studied the wiki to find solutions, put core.db to tmpfs, setting higher memory? etc, but all still could not worked, till now we are still plague by these problems, We then more a step further by trying to figure out how to put core.db into mysql or some database via doing research on the internet, but we have no luck as there are no further information about how to put core.db into mysql. Please somebody out there if you know how to do this kindly give us a guidance or point us to somebody who could advise us on how to rectify this issue.

Yet another issue we have on hand is Freeswitch could not stop fluently when there are calls has not been fully released and this cause our status Monitoring system to fail and could not restart Freeswitch.

Looking forward to getting some prompt reply soonest.

Thanks again Everybody.
Simon Leck
IM: simonleck at worldteltech.com<mailto:simonleck at worldteltech.com>
Cell: +886918981838


!DSPAM:4dbf9acc32761034989110!


From zetruger at gmail.com  Tue May  3 10:36:48 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Tue, 3 May 2011 10:36:48 +0400
Subject: [Freeswitch-users] mod_radius_cdr and Acct-Delay-Time
In-Reply-To: <b3b623bc7e6d035914ce1a15716da299@mail.gmail.com>
References: <9b86b6d280317812f3b7de59136aa457@mail.gmail.com>
	<BANLkTi=Dxk4pdG7nMDHugEjoOhpDsEqVYQ@mail.gmail.com>
	<b3b623bc7e6d035914ce1a15716da299@mail.gmail.com>
Message-ID: <BANLkTinvaJ_HiMWXUqOrEfPo3o4p5+Nmog@mail.gmail.com>

/src/mod/event_handlers/mod_radius_cdr/mod_radius_cdr.c
Your changes you must write self.

2011/4/29 Camila Troncoso <ctroncoso at redvoiss.net>:
> Thanks,
>
> But where do I find these changes, are they in the git source?
>
> Regardas,
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ????
> ????????
> Sent: viernes, 29 de abril de 2011 8:28
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] mod_radius_cdr and Acct-Delay-Time
>
> Just fix mod_radius_cdr )) change format of Freeswitch-Callstartdat.
> Use PW_ACCT_DELAY_TIME for adding Acct-Delay-Time field.
>
> 2011/4/27 Camila Troncoso <ctroncoso at redvoiss.net>:
>>
>>
>> Hi,
>>
>>
>>
>> I?m working with radius accounting using the mod_radius_cdr module. I?m
>> having trouble with the date format that Radius server receive. ?An
> example
>> of what the radius server receive is:
>>
>>
>>
>> ?Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"?
>>
>>
>>
>> This date format is very difficult to read and I want to change it? to
> make
>> the accounting easier. I search all around for some param or
> configuration
>> that allows me to do so, but I only find that this date format is define
> in
>> mod_radius_cdr.c module.
>>
>>
>>
>> I?m also having problems with the parameter Acct-Delay-Time, it is not
>> increasing when the client resend the radius packet. I read the
> buildreq.c
>> code but didn?t find the problem.
>>
>>
>>
>> Please some help with this matter.
>>
>>
>>
>> Regards,
>>
>>
>>
>> Camila Troncoso |?Ingeniero de Desarrollo
>>
>> RedVoiss |ctroncoso at redvoiss.net
>>
>> Santiago - Chile | +56 2 2408535
>>
>> www.redvoiss.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
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From OSchenk at wnr.com.au  Tue May  3 10:52:18 2011
From: OSchenk at wnr.com.au (Schenk, Oliver)
Date: Tue, 3 May 2011 14:52:18 +0800
Subject: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin
In-Reply-To: <BANLkTi==m6_WYtNE5bXHVywBWSgb3mzubg@mail.gmail.com>
References: <A4544360CBA2AC4298EE6AA1350E53AEC8A5B5@WNRCOLEX01.wnr.internal>
	<BANLkTi==m6_WYtNE5bXHVywBWSgb3mzubg@mail.gmail.com>
Message-ID: <A4544360CBA2AC4298EE6AA1350E53AEC8A65A@WNRCOLEX01.wnr.internal>

 Thank you!

 

It worked ? at least in the basic sense. I just created a very quick example as follows:

 

 

 

ESLconnection eslConnection = new ESLConnection(?127.0.0.1?, ?8021?, ?ClueCon?);

 

if (eslConnection.Connected() != ESL_SUCCESS)

{

    Console.WriteLine(?Error connecting to FreeSwitch?);

    return;
}

 

eslConnection.SendRecv(?api originate sofia/xxx.xxx.xxx.xxx/1001 &managed(myAppClassName)?);

 

 

 

 

xxx.xxx.xxx.xxx ? is my server IP address

1001 ? is one of the default sample extensions

 

 

 

Thanks!

 

 

 

From: Yitzchok [mailto:adminjew at gmail.com] 
Sent: 03/05/11 13:56
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin

 

The correct way to do this is to connect to FreeSWITCH using ESL <http://wiki.freeswitch.org/wiki/ESL>  from your application.

 

 

Have your application that polls your database run outside of freeswitch and when you need to create a new call connect to FS using ESL (there is a .NET lib in \libs\esl\managed\) and create a new call with the Originate command that bridges to an extension or application something like this "originate sofia/gateway/name/8885551234 1024" or "originate sofia/gateway/name/8885551234 &managed(ClassNameToIApiPlugin)".

 

 

Yitzchok



On Tue, May 3, 2011 at 12:02 AM, Schenk, Oliver <OSchenk at wnr.com.au> wrote:

Hi All,

 

I?m completely new to FreeSWITCH, but in a week I managed to achieve quite a bit. It was quite a learning curve to say the least. This is my current setup:

 

-          FreeSWITCH 1.0.7 built from tarball.

-          Developing on Windows XP with VS2008 in C#.

-          Configured extension 1024 to connect to a managed dll file, which implements IAppPlugin.

-          Using X-Lite softphone I can connect to FreeSWITCH as user 1001 and dial 1024 and start going through a menu that I built in C#. It successfully retrieves records from an SQL database and so forth. No problems there.

 

My first step was to create a module that gets called whenever a user dials IN bound. They will hear the menu just described.

 

 

My struggle now is relating to OUT bound calls. What I want is a module that is started as soon as FreeSWITCH is started and begins executing on an endless processing loop in the background. This will continuously monitor a database and if certain conditions occur an outbound call should be queued and then made. If multiple calls need to be made I guess they will be queued and processed one by one. I can handle the queuing part.

 

At this stage I will be testing using extension 1001 as the receiver of the call using my softphone.

 

 

Question 1:

 

I?ve been trying to use a class that implements IApiPlugin, but how do I get it to start when FreeSWITCH starts? As I said it should simply be a never ending thread as long as FreeSWITCH is running. I can?t find any information regarding how to ?Execute? a managed module immediately when FreeSWITCH has started.

 

 

Question 2:

 

If IApiPlugin can do this, how do I get the session object? Like this?

ManagedSession session = new ManagedSession();

session.Originate(???);

 

I can?t find any help at all on this.

 

 

Thanks very much!

 

 

 

Oliver Schenk

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.


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

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.
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From OSchenk at wnr.com.au  Tue May  3 11:08:06 2011
From: OSchenk at wnr.com.au (Schenk, Oliver)
Date: Tue, 3 May 2011 15:08:06 +0800
Subject: [Freeswitch-users] Freeswitch Windows CLI exits but process is
	still running
Message-ID: <A4544360CBA2AC4298EE6AA1350E53AEC8A667@WNRCOLEX01.wnr.internal>

Hey All.

 

 

I usually start FreeSWITCH like this while testing:

 

> FreeSWITCH.exe -nonat

 

While freeswitch is running and I am testing things a lot of notices and
things will start being printed.

 

 

However, often the command line returns back to cmd.exe. I can see in
the task manager that FreeSWITCH.exe is still running. When I try to run
freeswitch again I get an error that the process pid is already locked.

 

Does anyone know how to connect back to the freeswitch command line? At
the moment I have to Cntr-Alt-Delete and terminate the process in order
to restart it.

 

 

Thanks,

 

 

Oliver Schenk

 

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.
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From avi at avimarcus.net  Tue May  3 11:31:52 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 3 May 2011 10:31:52 +0300
Subject: [Freeswitch-users] FreeSWITCH Cookbook: resources
In-Reply-To: <A2473FE2-75CC-47BC-944E-D87952886CF9@gmail.com>
References: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
	<4DBF2F5A.30908@KennedySoftware.ie>
	<BANLkTikBHQoUY73xTM1mVh9aQ0MRXWnoXQ@mail.gmail.com>
	<A2473FE2-75CC-47BC-944E-D87952886CF9@gmail.com>
Message-ID: <BANLkTim1xHHZ8i0EE9KRSq=Ni5zLXwEXAw@mail.gmail.com>

On Tue, May 3, 2011 at 3:57 AM, Marc De Corny <marcdecorny at gmail.com> wrote:

> Hi Michael
>
> 2. Do you have an interop doc that i could complete if you don't already
> have it as proof that it was tested with a nortel/genband cs2000?
>
> Have a good evening
>
> Thanks
> Marc
>

There's no standard interop doc afaik, but you can try the basic few listed
on http://wiki.freeswitch.org/wiki/Interop_List and update the wiki.

-Avi
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From benkokakao at gmail.com  Tue May  3 11:43:43 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 3 May 2011 09:43:43 +0200
Subject: [Freeswitch-users] Looking for transcription of english
	sound-files
In-Reply-To: <BANLkTimG_PFQL5=eODK_gFakHdPUGF8eMw@mail.gmail.com>
References: <BANLkTinGW_hip46XNqOd8YGrtL-qMhfCXg@mail.gmail.com>
	<B7913E70-92B2-469D-BBB2-A275D484F4B8@seventhsignal.de>
	<BANLkTim3G8ACrBckiwyO8qUQCFFOmfY9JA@mail.gmail.com>
	<BANLkTimG_PFQL5=eODK_gFakHdPUGF8eMw@mail.gmail.com>
Message-ID: <BANLkTikvcuGmeGraHnWxSyb7snbnRsfKxA@mail.gmail.com>

n 2 May 2011 23:11, Michael Collins <msc at freeswitch.org> wrote:
> Under FreeSWITCH source directory: docs/phrase/phrase_en.xml
> -MC

Perfect, thanks!


From steveayre at gmail.com  Tue May  3 11:54:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 3 May 2011 08:54:00 +0100
Subject: [Freeswitch-users] Freeswitch Windows CLI exits but process is
 still running
In-Reply-To: <A4544360CBA2AC4298EE6AA1350E53AEC8A667@WNRCOLEX01.wnr.internal>
References: <A4544360CBA2AC4298EE6AA1350E53AEC8A667@WNRCOLEX01.wnr.internal>
Message-ID: <BANLkTi=uO+_qQnuXjFCU2i-CcX9VLN-P8g@mail.gmail.com>

Look at loading mod_event_socket and connecting using fs_cli, which provide
you a remote CLI.

This is the normal way to connect to FS when it's running in the background
as a service.

-Steve


On 3 May 2011 08:08, Schenk, Oliver <OSchenk at wnr.com.au> wrote:

>  Hey All.
>
>
>
>
>
> I usually start FreeSWITCH like this while testing:
>
>
>
> > FreeSWITCH.exe ?nonat
>
>
>
> While freeswitch is running and I am testing things a lot of notices and
> things will start being printed.
>
>
>
>
>
> However, often the command line returns back to cmd.exe. I can see in the
> task manager that FreeSWITCH.exe is still running. When I try to run
> freeswitch again I get an error that the process pid is already locked.
>
>
>
> Does anyone know how to connect back to the freeswitch command line? At the
> moment I have to Cntr-Alt-Delete and terminate the process in order to
> restart it.
>
>
>
>
>
> Thanks,
>
>
>
>
>
> *Oliver Schenk*
>
>
>
>
>
> NOTICE - This e-mail and any files transmitted with it are confidential and
> are only for the use of the person to whom they are addressed.
> If you are not the intended recipient then you have received this e-mail in
> error; please advise us immediately if this is the case.
> Any views expressed in this message are those of the individual sender,
> except where the sender specifically states them to be the views of WestNet
> Rail.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Tue May  3 15:44:38 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 03 May 2011 13:44:38 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Cookbook:
	resources
In-Reply-To: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
References: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
Message-ID: <4DBFEAA6.8010300@gmail.com>

ready to ofer you a quad core Del Server with 4GB of ram and a did of 
your choice
please contact me this evening.
anything else needed?
thank you.
On 02/05/2011 21:04, Michael Collins wrote:
> Hello all!
>
> The response to my request for reviewers is overwhelming - thank you 
> so much! We have more than enough reviewers. However, any of you who 
> wish to read the draft and offer feedback are still welcome to do so. 
> You just won't be "official" Packt reviewers.
>
> I'd like to focus attention on testing resources. For those of you who 
> have data center resources and such I would like to ask for your 
> assistance. It would be helpful to have a spare server (low power == 
> okay, cuz it's only for a few simultaneous calls max) with a DID & 
> public IP address. This will help us test various scenarios and make 
> sure that we can do box2box and NAT scenarios.
>
> Please contact me off list if you have any servers, DIDs, etc. that 
> could be used for this purpose.
>
> Thanks again to everyone who has volunteered to help! I would much 
> rather have to sift through dozens of emails from people who want to 
> help than have to keep begging and have no one step up. ;)  You guys 
> are awesome - keep up the good work.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883

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From dave at dchorton.com  Tue May  3 16:09:29 2011
From: dave at dchorton.com (Dave Horton)
Date: Tue, 3 May 2011 08:09:29 -0400
Subject: [Freeswitch-users] help on creating an rpm
References: <B2FF20A3-3A39-4565-A287-01E719A65005@dchorton.com>
Message-ID: <6DDD2468-B228-4E86-9A32-7018697DD8D0@dchorton.com>


Thanks for some of the feedback, I have gotten further.  (As noted, I am not all that experienced with rpms, but have read up on various articles).  Now, I have created my .rpmmacros to set the top level directory, and gotten my source tree in the correct place to be located, and when I run rpmbuild against the spec file it runs for quite a while but eventually fails with this error below.  

 gcc -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python/mod_python.c -o mod_python.o >/dev/null 2>&1
Creating mod_python.so...
/usr/bin/ld: /usr/local/lib/libpython2.7.a(abstract.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
/usr/local/lib/libpython2.7.a: could not read symbols: Bad value
collect2: ld returned 1 exit status
g++ -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -D_GNU_SOURCE -shared -o .libs/mod_python.so -shared -Wl,-x .libs/mod_python.o freeswitch_python.o mod_python_wrap.o  /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/.libs/libfreeswitch.so -lz -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib -lpq /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib/.libs/libexpat.a /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/srtp -lssl -lcrypto -lncurses -L/usr/local/lib -lpthread -ldl -lutil -lm -lpython2.7   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath -Wl,/opt/freeswitch/mod
make[5]: *** [mod_python.so] Error 1
make[5]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
make[4]: *** [all] Error 1
make[4]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
make[3]: *** [languages/mod_python-all] Error 1
make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod'
make[3]: Entering directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
make[3]: Nothing to be done for `all-am'.
make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7'
make: *** [all] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.67849 (%build)

Any thoughts about what might be wrong?  

From sid.kshatriya at gmail.com  Tue May  3 17:00:00 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Tue, 3 May 2011 18:30:00 +0530
Subject: [Freeswitch-users] Is it a good idea to deploy Freeswitch on Xen
	Instances/Virtual Machines?
Message-ID: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>

Dear All,

I'm interested in installing Freeswitch on Xen servers (e.g. linode.com or
slicehost or maybe even Amazon EC2). I'll be purchasing a few SIP trunks to
my freeswitch installation and running an an IVR system there.

I'd like to know whether this is a recommended or desirable thing to do.
I've heard that the latency on Xen can be bad for "real time" applications
like IVRs. Don't want to install freeswitch on my server and later hear
choppy voice on the IVR!! :-)

Thanks,

Sidharth


-- 
Sidharth Kshatriya
www.sidk.info
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From randy.andrade at gmail.com  Tue May  3 17:22:03 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Tue, 3 May 2011 09:22:03 -0400
Subject: [Freeswitch-users] Is it a good idea to deploy Freeswitch on
 Xen Instances/Virtual Machines?
In-Reply-To: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>
References: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>
Message-ID: <BANLkTi=uYpBin6GsAERcv2vnE72zaC50Zw@mail.gmail.com>

While I can't really speak to a hosted Xen setup, I ran freeswitch on a
premise-based ESXi system rather successfully for a period of time. I did
end up moving it to a dedicated hardware box, however, due to the issues
with (slightly) choppy audio, and my biggest issue, the sender --> receiver
latency whenever someone would speak. The latter might not be an issue, if
you're just using it for an IVR, however what I did find to be wonderful was
how easy the migration was. I literally just installed the same OS on my
hardware system, built & compiled FS (with the same options / modules), then
copied the entire /conf directory from my VM to the hardware system and
restarted FS there. I should also note that all other variables (hostname,
IP, etc) remained the same between the VM and the hardware box.

Randy

On Tue, May 3, 2011 at 9:00 AM, Sidharth Kshatriya
<sid.kshatriya at gmail.com>wrote:

> Dear All,
>
> I'm interested in installing Freeswitch on Xen servers (e.g. linode.com or
> slicehost or maybe even Amazon EC2). I'll be purchasing a few SIP trunks to
> my freeswitch installation and running an an IVR system there.
>
> I'd like to know whether this is a recommended or desirable thing to do.
> I've heard that the latency on Xen can be bad for "real time" applications
> like IVRs. Don't want to install freeswitch on my server and later hear
> choppy voice on the IVR!! :-)
>
> Thanks,
>
> Sidharth
>
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gchen00 at insightbb.com  Tue May  3 17:23:07 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Tue, 03 May 2011 09:23:07 -0400
Subject: [Freeswitch-users] How to use fs_path?
Message-ID: <fc76c1d1132f6.4dbfc97b@insightbb.com>

FreeSWITCH?Version 1.0.head (git-f17e962 2011-04-19 03-05-32 -0400)

I am tring?to send call through a sip proxy like this:
<action application="bridge" data="sofia/internal/5038443019 at XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/>

It does not work. Whenever I make call, it immediately send to voicemail.

What is the correct way to use fs_path?

Gary

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From sid.kshatriya at gmail.com  Tue May  3 17:48:32 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Tue, 3 May 2011 19:18:32 +0530
Subject: [Freeswitch-users] Is it a good idea to deploy Freeswitch on
 Xen Instances/Virtual Machines?
In-Reply-To: <BANLkTi=uYpBin6GsAERcv2vnE72zaC50Zw@mail.gmail.com>
References: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>
	<BANLkTi=uYpBin6GsAERcv2vnE72zaC50Zw@mail.gmail.com>
Message-ID: <BANLkTim+k0g9eJ6vutxvfdUi=H7986fM-Q@mail.gmail.com>

Thanks for taking the time -- you seem to confirm my fears about choppy
audio due to high latency on virtual machines... :-)

My site is going to be streaming a lot of audio via red5 servers (internet)
and running an IVR using freeswitch

Thanks,

Sidharth

On Tue, May 3, 2011 at 6:52 PM, Randy Andrade <randy.andrade at gmail.com>wrote:

> While I can't really speak to a hosted Xen setup, I ran freeswitch on a
> premise-based ESXi system rather successfully for a period of time. I did
> end up moving it to a dedicated hardware box, however, due to the issues
> with (slightly) choppy audio, and my biggest issue, the sender --> receiver
> latency whenever someone would speak. The latter might not be an issue, if
> you're just using it for an IVR, however what I did find to be wonderful was
> how easy the migration was. I literally just installed the same OS on my
> hardware system, built & compiled FS (with the same options / modules), then
> copied the entire /conf directory from my VM to the hardware system and
> restarted FS there. I should also note that all other variables (hostname,
> IP, etc) remained the same between the VM and the hardware box.
>
> Randy
>
> On Tue, May 3, 2011 at 9:00 AM, Sidharth Kshatriya <
> sid.kshatriya at gmail.com> wrote:
>
>> Dear All,
>>
>> I'm interested in installing Freeswitch on Xen servers (e.g. linode.comor slicehost or maybe even Amazon EC2). I'll be purchasing a few SIP trunks
>> to my freeswitch installation and running an an IVR system there.
>>
>> I'd like to know whether this is a recommended or desirable thing to do.
>> I've heard that the latency on Xen can be bad for "real time" applications
>> like IVRs. Don't want to install freeswitch on my server and later hear
>> choppy voice on the IVR!! :-)
>>
>> Thanks,
>>
>> Sidharth
>>
>>
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From Prometheus001 at gmx.net  Tue May  3 17:49:36 2011
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 03 May 2011 15:49:36 +0200
Subject: [Freeswitch-users] How to use fs_path?
In-Reply-To: <fc76c1d1132f6.4dbfc97b@insightbb.com>
References: <fc76c1d1132f6.4dbfc97b@insightbb.com>
Message-ID: <4DC007F0.7070401@gmx.net>

Hello Gary,

you are sure about sofia/internal/?
Don't you want to use sofia/external for external routing?


Best regards
Peter

Gary Chen schrieb:
> FreeSWITCH Version 1.0.head (git-f17e962 2011-04-19 03-05-32 -0400)
>
> I am tring to send call through a sip proxy like this:
> <action application="bridge"
> data="sofia/internal/5038443019 at XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/>
>
> It does not work. Whenever I make call, it immediately send to voicemail.
>
> What is the correct way to use fs_path?
>
> Gary
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From jeff at jefflenk.com  Tue May  3 17:56:49 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 3 May 2011 06:56:49 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch Windows CLI exits but process is
	still running
In-Reply-To: <A4544360CBA2AC4298EE6AA1350E53AEC8A667@WNRCOLEX01.wnr.internal>
References: <A4544360CBA2AC4298EE6AA1350E53AEC8A667@WNRCOLEX01.wnr.internal>
Message-ID: <1304431009358-6327364.post@n2.nabble.com>

FreeSWITCH.exe ?nonat should not exit the console by itself. There must be a
critical error or exception occurring that is causing that.

If you can get a stack trace with a debug log please post that to
http://jira.freeswitch.org

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Windows-CLI-exits-but-process-is-still-running-tp6326379p6327364.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From frank at rosengart.de  Tue May  3 17:59:50 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Tue, 03 May 2011 15:59:50 +0200
Subject: [Freeswitch-users] Play recording sound into conference
In-Reply-To: <BANLkTim3b7qdbweH3meXHruhnwEpuqCwbg@mail.gmail.com>
References: <4DBEC402.9090009@rosengart.de>
	<BANLkTim3b7qdbweH3meXHruhnwEpuqCwbg@mail.gmail.com>
Message-ID: <4DC00A56.1000901@rosengart.de>

On 05/02/2011 11:11 PM, Michael Collins wrote:

> something like:
> <action application="set" data="tmp=${conference $conf_name play
> ${recording} async}"/>

This works great without the 'async'. I was not aware of the power of
the API-from-dialplan feature.

Maybe this could be included in the demo files?


Thanks a lot!
Frank



From gmaruzz at gmail.com  Tue May  3 19:49:17 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 3 May 2011 17:49:17 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTimo-h60MiD-33DDk4Kis230G-kbcQ@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimubrY_jTRptCbL0oQC3pJkvLpT9g@mail.gmail.com>
	<BANLkTikx7-vxWBU4bc0LFmfcpAZZR77iJA@mail.gmail.com>
	<BANLkTi=xuE13V-mgGNPDXo0ixZsAXP7j+Q@mail.gmail.com>
	<BANLkTimo-h60MiD-33DDk4Kis230G-kbcQ@mail.gmail.com>
Message-ID: <BANLkTinJ5ykgTWNGGF3A62bceTYKOwpeog@mail.gmail.com>

On Sun, May 1, 2011 at 10:35 PM, zohair raza
<engineerzuhairraza at gmail.com> wrote:
>
> Thank you sir,
>
> I have added it on wiki

Thanks Zohair,

I've put it into a new section, "Use cases, integration, special
implementations", so now it has better visibility.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gchen00 at insightbb.com  Tue May  3 20:07:58 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Tue, 03 May 2011 12:07:58 -0400
Subject: [Freeswitch-users] How to use fs_path?
Message-ID: <fc6db3bb17ddc.4dbff01e@insightbb.com>

I also tried sofia/external and got the same problem.

Gary? 




From: 
freeswitch-users-bounces at lists.freeswitch.org 
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter 
P GMX
Sent: Tuesday, May 03, 2011 9:50 AM
To: FreeSWITCH 
Users Help
Subject: Re: [Freeswitch-users] How to use 
fs_path?


Hello Gary,

you are sure about sofia/internal/? 
Don't you 
want to use sofia/external for external routing?


Best 
regards
Peter

Gary Chen schrieb: 
FreeSWITCH?Version 1.0.head (git-f17e962 2011-04-19 03-05-32 
-0400)

I am tring?to send call through a sip proxy like 
this:
<action application="bridge" data="sofia/internal/5038443019 at XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/>

It 
does not work. Whenever I make call, it immediately send to 
voicemail.

What is the correct way to use fs_path?

Gary

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

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From anthony.minessale at gmail.com  Tue May  3 20:53:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 3 May 2011 11:53:33 -0500
Subject: [Freeswitch-users] How to use fs_path?
In-Reply-To: <fc6db3bb17ddc.4dbff01e@insightbb.com>
References: <fc6db3bb17ddc.4dbff01e@insightbb.com>
Message-ID: <BANLkTim0WbVRrpqU9P5XmapUdbw9PAU9eA@mail.gmail.com>

start the dial string with sip: to tell mod_sofia not to parse any of the params

sofia/internal/sip:foo at bar.com;param=val


On Tue, May 3, 2011 at 11:07 AM, Gary Chen <gchen00 at insightbb.com> wrote:
> I also tried sofia/external and got the same problem.
>
> Gary
>
> ________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P
> GMX
> Sent: Tuesday, May 03, 2011 9:50 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] How to use fs_path?
>
> Hello Gary,
>
> you are sure about sofia/internal/?
> Don't you want to use sofia/external for external routing?
>
>
> Best regards
> Peter
>
> Gary Chen schrieb:
>
> FreeSWITCH?Version 1.0.head (git-f17e962 2011-04-19 03-05-32 -0400)
>
> I am tring?to send call through a sip proxy like this:
> <action application="bridge"
> data="sofia/internal/5038443019 at XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/>
>
> It does not work. Whenever I make call, it immediately send to voicemail.
>
> What is the correct way to use fs_path?
>
> Gary
>
> ________________________________
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 gchen00 at insightbb.com  Tue May  3 21:17:09 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Tue, 03 May 2011 13:17:09 -0400
Subject: [Freeswitch-users] How to use fs_path?
Message-ID: <fc83a48d138cc.4dc00055@insightbb.com>

So what is the correct way to do this?


I also tried this without 'sip:' in the dialstr like following and it also failed:


<action application="bridge" data="sofia/internal/5038443019 at XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/>





-----Original Message-----


From: freeswitch-users-bounces at lists.freeswitch.org 
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
Anthony Minessale


Sent: Tuesday, May 03, 2011 12:54 PM


To: FreeSWITCH Users Help


Subject: Re: [Freeswitch-users] How to use fs_path?





start the dial string with sip: to tell mod_sofia not to parse any of the params





sofia/internal/sip:foo at bar.com;param=val








On Tue, May 3, 2011 at 11:07 AM, Gary Chen <gchen00 at insightbb.com> wrote:


> I also tried sofia/external and got the same problem.


>


> Gary


>


> ________________________________


> From: freeswitch-users-bounces at lists.freeswitch.org


> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 


> Peter P GMX


> Sent: Tuesday, May 03, 2011 9:50 AM


> To: FreeSWITCH Users Help


> Subject: Re: [Freeswitch-users] How to use fs_path?


>


> Hello Gary,


>


> you are sure about sofia/internal/?


> Don't you want to use sofia/external for external routing?


>


>


> Best regards


> Peter


>


> Gary Chen schrieb:


>


> FreeSWITCH?Version 1.0.head (git-f17e962 2011-04-19 03-05-32 -0400)


>


> I am tring?to send call through a sip proxy like this:


> <action application="bridge"


> data="sofia/internal/5038443019 at XXX.XXX.XXX.225:5060;fs_path=sip:XXX.X


> XX.XXX.230:5060"/>


>


> It does not work. Whenever I make call, it immediately send to voicemail.


>


> What is the correct way to use fs_path?


>


> Gary


>


> ________________________________


> _______________________________________________


> FreeSWITCH-users mailing list


> FreeSWITCH-users at lists.freeswitch.org


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


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


>


>











--


Anthony Minessale II





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 gchen00 at insightbb.com  Tue May  3 21:27:35 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Tue, 03 May 2011 13:27:35 -0400
Subject: [Freeswitch-users] How to use fs_path?
Message-ID: <fc7fe88611802.4dc002c7@insightbb.com>

Never mind. I got it.

Thanks.

Gary

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From michal.bielicki at seventhsignal.de  Tue May  3 22:25:40 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 3 May 2011 20:25:40 +0200
Subject: [Freeswitch-users] help on creating an rpm
In-Reply-To: <6DDD2468-B228-4E86-9A32-7018697DD8D0@dchorton.com>
References: <B2FF20A3-3A39-4565-A287-01E719A65005@dchorton.com>
	<6DDD2468-B228-4E86-9A32-7018697DD8D0@dchorton.com>
Message-ID: <8B767EE1-0CB1-41B5-99F3-F543EC547D1B@seventhsignal.de>

What platoform is this ? The spec is made for centos 5.X. there is no python 2.7 on centos so I assume you are trying fedora ?

Am 03.05.2011 um 14:09 schrieb Dave Horton:

> 
> Thanks for some of the feedback, I have gotten further.  (As noted, I am not all that experienced with rpms, but have read up on various articles).  Now, I have created my .rpmmacros to set the top level directory, and gotten my source tree in the correct place to be located, and when I run rpmbuild against the spec file it runs for quite a while but eventually fails with this error below.  
> 
> gcc -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python/mod_python.c -o mod_python.o >/dev/null 2>&1
> Creating mod_python.so...
> /usr/bin/ld: /usr/local/lib/libpython2.7.a(abstract.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
> /usr/local/lib/libpython2.7.a: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> g++ -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -D_GNU_SOURCE -shared -o .libs/mod_python.so -shared -Wl,-x .libs/mod_python.o freeswitch_python.o mod_python_wrap.o  /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/.libs/libfreeswitch.so -lz -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib -lpq /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib/.libs/libexpat.a /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr/.libs/libapr-1.a -luuid -lrt -lcr
> ypt -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/srtp -lssl -lcrypto -lncurses -L/usr/local/lib -lpthread -ldl -lutil -lm -lpython2.7   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath -Wl,/opt/freeswitch/mod
> make[5]: *** [mod_python.so] Error 1
> make[5]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[4]: *** [all] Error 1
> make[4]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[3]: *** [languages/mod_python-all] Error 1
> make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod'
> make[3]: Entering directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[3]: Nothing to be done for `all-am'.
> make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7'
> make: *** [all] Error 2
> error: Bad exit status from /var/tmp/rpm-tmp.67849 (%build)
> 
> Any thoughts about what might be wrong?  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From brad at tritelcomm.com  Tue May  3 22:39:24 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Tue, 3 May 2011 11:39:24 -0700
Subject: [Freeswitch-users] Is it a good idea to deploy Freeswitch on
 Xen Instances/Virtual Machines?
In-Reply-To: <BANLkTim+k0g9eJ6vutxvfdUi=H7986fM-Q@mail.gmail.com>
References: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>
	<BANLkTi=uYpBin6GsAERcv2vnE72zaC50Zw@mail.gmail.com>
	<BANLkTim+k0g9eJ6vutxvfdUi=H7986fM-Q@mail.gmail.com>
Message-ID: <BANLkTinaBXqvE7bX3TLSKd=MQgG8eMmDUw@mail.gmail.com>

The testing I've done with linode specifically have been pretty good. I've
not seen any major problems with audio quality. However, my tests were not
intensive and I did not do any call-volume testing. For $20, I'd recommend
trying it out for a month. Linode support is pretty good as well, their IRC
channel is fairly active.

On Tue, May 3, 2011 at 6:48 AM, Sidharth Kshatriya
<sid.kshatriya at gmail.com>wrote:

> Thanks for taking the time -- you seem to confirm my fears about choppy
> audio due to high latency on virtual machines... :-)
>
> My site is going to be streaming a lot of audio via red5 servers (internet)
> and running an IVR using freeswitch
>
> Thanks,
>
> Sidharth
>
>
> On Tue, May 3, 2011 at 6:52 PM, Randy Andrade <randy.andrade at gmail.com>wrote:
>
>> While I can't really speak to a hosted Xen setup, I ran freeswitch on a
>> premise-based ESXi system rather successfully for a period of time. I did
>> end up moving it to a dedicated hardware box, however, due to the issues
>> with (slightly) choppy audio, and my biggest issue, the sender --> receiver
>> latency whenever someone would speak. The latter might not be an issue, if
>> you're just using it for an IVR, however what I did find to be wonderful was
>> how easy the migration was. I literally just installed the same OS on my
>> hardware system, built & compiled FS (with the same options / modules), then
>> copied the entire /conf directory from my VM to the hardware system and
>> restarted FS there. I should also note that all other variables (hostname,
>> IP, etc) remained the same between the VM and the hardware box.
>>
>> Randy
>>
>> On Tue, May 3, 2011 at 9:00 AM, Sidharth Kshatriya <
>> sid.kshatriya at gmail.com> wrote:
>>
>>> Dear All,
>>>
>>> I'm interested in installing Freeswitch on Xen servers (e.g. linode.comor slicehost or maybe even Amazon EC2). I'll be purchasing a few SIP trunks
>>> to my freeswitch installation and running an an IVR system there.
>>>
>>> I'd like to know whether this is a recommended or desirable thing to do.
>>> I've heard that the latency on Xen can be bad for "real time" applications
>>> like IVRs. Don't want to install freeswitch on my server and later hear
>>> choppy voice on the IVR!! :-)
>>>
>>> Thanks,
>>>
>>> Sidharth
>>>
>>>
>>> --
>>> Sidharth Kshatriya
>>> www.sidk.info
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mkopacki at gmail.com  Wed May  4 00:31:08 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Tue, 3 May 2011 22:31:08 +0200
Subject: [Freeswitch-users] few questions about multi-tenant configuration
Message-ID: <20110503223108.29d1dc94@ankh.reapnet.com>

    Hi,

I'm working on scenario, which include resellers. Below
short description:

There is one instance of freeswitch and some logical layer for
resellers. I would like to be able to handle multiple domains on one FS
and additionally few domains per reseller. I've already read multitenant
description on wiki but it barely scratch the surface so i have few
questions and I'll appreciate any thoughts:

I would like to separate domains in different contexts:

domain A have context domain-a and users in it
domain B have context domain-b and users in it

For internal calls (within domain) it's simple, similar for outbound
calls (gateways defined in directory). But how to handle calls
between extensions ?

Is there a way to add context name to bridge instruction ?

And what about inbound calls ?

As far I understand all inbound calls goes through sip_profile and in
definition of profile there is param like context.

Is it possible to have this param dynamic (based on incoming call for
example) or it's better to have separate profile per domain ?

-- 
Regards,
Michal


From dave at dchorton.com  Wed May  4 00:33:18 2011
From: dave at dchorton.com (Dave Horton)
Date: Tue, 3 May 2011 16:33:18 -0400
Subject: [Freeswitch-users] help on creating an rpm
References: <6DDD2468-B228-4E86-9A32-7018697DD8D0@dchorton.com>
Message-ID: <B0D2ACF7-8C5D-498E-A2D5-1B7DA62F9B4D@dchorton.com>

Hmm..actually this is centos:

[root at centos64-1 ~]# uname -a
Linux centos64-1 2.6.18-194.26.1.el5 #1 SMP Tue Nov 9 12:54:20 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
[root at centos64-1 ~]# cat /etc/redhat-release 
CentOS release 5.5 (Final)




Begin forwarded message:

From: Dave Horton <dave at dchorton.com>
Date: May 3, 2011 8:09:29 AM EDT
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: help on creating an rpm


Thanks for some of the feedback, I have gotten further.  (As noted, I am not all that experienced with rpms, but have read up on various articles).  Now, I have created my .rpmmacros to set the top level directory, and gotten my source tree in the correct place to be located, and when I run rpmbuild against the spec file it runs for quite a while but eventually fails with this error below.  

gcc -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python/mod_python.c -o mod_python.o >/dev/null 2>&1
Creating mod_python.so...
/usr/bin/ld: /usr/local/lib/libpython2.7.a(abstract.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
/usr/local/lib/libpython2.7.a: could not read symbols: Bad value
collect2: ld returned 1 exit status
g++ -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -D_GNU_SOURCE -shared -o .libs/mod_python.so -shared -Wl,-x .libs/mod_python.o freeswitch_python.o mod_python_wrap.o  /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/.libs/libfreeswitch.so -lz -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib -lpq /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib/.libs/libexpat.a /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/srtp -lssl -lcrypto -lncurses -L/usr/local/lib -lpthread -ldl -lutil -lm -lpython2.7   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath -Wl,/opt/freeswitch/mod
make[5]: *** [mod_python.so] Error 1
make[5]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
make[4]: *** [all] Error 1
make[4]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
make[3]: *** [languages/mod_python-all] Error 1
make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod'
make[3]: Entering directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
make[3]: Nothing to be done for `all-am'.
make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7'
make: *** [all] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.67849 (%build)

Any thoughts about what might be wrong?  



From eagle.antonio at gmail.com  Wed May  4 00:34:55 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 3 May 2011 21:34:55 +0100
Subject: [Freeswitch-users] few questions about multi-tenant
	configuration
In-Reply-To: <20110503223108.29d1dc94@ankh.reapnet.com>
References: <20110503223108.29d1dc94@ankh.reapnet.com>
Message-ID: <BANLkTi=mDHJUWAtOkARkweHuYRAKJygbng@mail.gmail.com>

2011/5/3 Michal Kopacki <mkopacki at gmail.com>

>    Hi,
>
> I'm working on scenario, which include resellers. Below
> short description:
>
> There is one instance of freeswitch and some logical layer for
> resellers. I would like to be able to handle multiple domains on one FS
> and additionally few domains per reseller. I've already read multitenant
> description on wiki but it barely scratch the surface so i have few
> questions and I'll appreciate any thoughts:
>
> I would like to separate domains in different contexts:
>
> domain A have context domain-a and users in it
> domain B have context domain-b and users in it
>
> For internal calls (within domain) it's simple, similar for outbound
> calls (gateways defined in directory). But how to handle calls
> between extensions ?
>
> Is there a way to add context name to bridge instruction ?
>
> And what about inbound calls ?
>
> As far I understand all inbound calls goes through sip_profile and in
> definition of profile there is param like context.
>
> Is it possible to have this param dynamic (based on incoming call for
> example) or it's better to have separate profile per domain ?
>
> --
> Regards,
> Michal
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From michal.bielicki at seventhsignal.de  Wed May  4 00:46:35 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 3 May 2011 22:46:35 +0200
Subject: [Freeswitch-users] help on creating an rpm
In-Reply-To: <B0D2ACF7-8C5D-498E-A2D5-1B7DA62F9B4D@dchorton.com>
References: <6DDD2468-B228-4E86-9A32-7018697DD8D0@dchorton.com>
	<B0D2ACF7-8C5D-498E-A2D5-1B7DA62F9B4D@dchorton.com>
Message-ID: <3FFE5FF0-4F80-4042-88B9-54BCBD394294@seventhsignal.de>

It seems to try to link a 32bit version of libpython to mod_pthon instead of the 64bit version. In dtail it seems to be linking in a version you installed by hand in/usr/local and that seems to be a 32bit version and that is why it fails.

Am 03.05.2011 um 22:33 schrieb Dave Horton:

> Hmm..actually this is centos:
> 
> [root at centos64-1 ~]# uname -a
> Linux centos64-1 2.6.18-194.26.1.el5 #1 SMP Tue Nov 9 12:54:20 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
> [root at centos64-1 ~]# cat /etc/redhat-release 
> CentOS release 5.5 (Final)
> 
> 
> 
> 
> Begin forwarded message:
> 
> From: Dave Horton <dave at dchorton.com>
> Date: May 3, 2011 8:09:29 AM EDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: help on creating an rpm
> 
> 
> Thanks for some of the feedback, I have gotten further.  (As noted, I am not all that experienced with rpms, but have read up on various articles).  Now, I have created my .rpmmacros to set the top level directory, and gotten my source tree in the correct place to be located, and when I run rpmbuild against the spec file it runs for quite a while but eventually fails with this error below.  
> 
> gcc -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python/mod_python.c -o mod_python.o >/dev/null 2>&1
> Creating mod_python.so...
> /usr/bin/ld: /usr/local/lib/libpython2.7.a(abstract.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
> /usr/local/lib/libpython2.7.a: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> g++ -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -D_GNU_SOURCE -shared -o .libs/mod_python.so -shared -Wl,-x .libs/mod_python.o freeswitch_python.o mod_python_wrap.o  /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/.libs/libfreeswitch.so -lz -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib -lpq /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib/.libs/libexpat.a /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr/.libs/libapr-1.a -luuid -lrt -lcr
> ypt -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/srtp -lssl -lcrypto -lncurses -L/usr/local/lib -lpthread -ldl -lutil -lm -lpython2.7   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath -Wl,/opt/freeswitch/mod
> make[5]: *** [mod_python.so] Error 1
> make[5]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[4]: *** [all] Error 1
> make[4]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[3]: *** [languages/mod_python-all] Error 1
> make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod'
> make[3]: Entering directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[3]: Nothing to be done for `all-am'.
> make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7'
> make: *** [all] Error 2
> error: Bad exit status from /var/tmp/rpm-tmp.67849 (%build)
> 
> Any thoughts about what might be wrong?  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From krice at freeswitch.org  Wed May  4 00:49:52 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 03 May 2011 15:49:52 -0500
Subject: [Freeswitch-users] few questions about multi-tenant
	configuration
In-Reply-To: <20110503223108.29d1dc94@ankh.reapnet.com>
Message-ID: <C9E5D4A0.14BD3%krice@freeswitch.org>

While the multi-tenant stuff is open-source, please see
http://www.fusionpbx.com/support.php for support.


On 5/3/11 3:31 PM, "Michal Kopacki" <mkopacki at gmail.com> wrote:

>     Hi,
> 
> I'm working on scenario, which include resellers. Below
> short description:
> 
> There is one instance of freeswitch and some logical layer for
> resellers. I would like to be able to handle multiple domains on one FS
> and additionally few domains per reseller. I've already read multitenant
> description on wiki but it barely scratch the surface so i have few
> questions and I'll appreciate any thoughts:
> 
> I would like to separate domains in different contexts:
> 
> domain A have context domain-a and users in it
> domain B have context domain-b and users in it
> 
> For internal calls (within domain) it's simple, similar for outbound
> calls (gateways defined in directory). But how to handle calls
> between extensions ?
> 
> Is there a way to add context name to bridge instruction ?
> 
> And what about inbound calls ?
> 
> As far I understand all inbound calls goes through sip_profile and in
> definition of profile there is param like context.
> 
> Is it possible to have this param dynamic (based on incoming call for
> example) or it's better to have separate profile per domain ?




From krice at freeswitch.org  Wed May  4 00:50:24 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 03 May 2011 15:50:24 -0500
Subject: [Freeswitch-users] few questions about multi-tenant
	configuration
In-Reply-To: <20110503223108.29d1dc94@ankh.reapnet.com>
Message-ID: <C9E5D4C0.14BD4%krice@freeswitch.org>

Crap IGNORE THAT... READING THE WRONG MAILING LIST


On 5/3/11 3:31 PM, "Michal Kopacki" <mkopacki at gmail.com> wrote:

>     Hi,
> 
> I'm working on scenario, which include resellers. Below
> short description:
> 
> There is one instance of freeswitch and some logical layer for
> resellers. I would like to be able to handle multiple domains on one FS
> and additionally few domains per reseller. I've already read multitenant
> description on wiki but it barely scratch the surface so i have few
> questions and I'll appreciate any thoughts:
> 
> I would like to separate domains in different contexts:
> 
> domain A have context domain-a and users in it
> domain B have context domain-b and users in it
> 
> For internal calls (within domain) it's simple, similar for outbound
> calls (gateways defined in directory). But how to handle calls
> between extensions ?
> 
> Is there a way to add context name to bridge instruction ?
> 
> And what about inbound calls ?
> 
> As far I understand all inbound calls goes through sip_profile and in
> definition of profile there is param like context.
> 
> Is it possible to have this param dynamic (based on incoming call for
> example) or it's better to have separate profil



From jaybinks at gmail.com  Wed May  4 01:59:02 2011
From: jaybinks at gmail.com (Jay Binks)
Date: Wed, 4 May 2011 07:59:02 +1000
Subject: [Freeswitch-users] How to use fs_path?
In-Reply-To: <fc7fe88611802.4dc002c7@insightbb.com>
References: <fc7fe88611802.4dc002c7@insightbb.com>
Message-ID: <6A7C8F16-0C97-4A70-9751-F9F59C78AE0F@gmail.com>

Post your working result please



On 04/05/2011, at 3:27 AM, Gary Chen <gchen00 at insightbb.com> wrote:

> Never mind. I got it.
> 
> Thanks.
> 
> Gary
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From chad at apartmentlines.com  Wed May  4 02:31:56 2011
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Tue, 3 May 2011 15:31:56 -0700
Subject: [Freeswitch-users] Is it a good idea to deploy Freeswitch on
	Xen Instances/Virtual Machines?
In-Reply-To: <BANLkTinaBXqvE7bX3TLSKd=MQgG8eMmDUw@mail.gmail.com>
References: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>
	<BANLkTi=uYpBin6GsAERcv2vnE72zaC50Zw@mail.gmail.com>
	<BANLkTim+k0g9eJ6vutxvfdUi=H7986fM-Q@mail.gmail.com>
	<BANLkTinaBXqvE7bX3TLSKd=MQgG8eMmDUw@mail.gmail.com>
Message-ID: <9A2340F2-B677-4DB4-BF33-46683146A551@apartmentlines.com>


On May 3, 2011, at 11:39 AM, Brad Mina wrote:

> The testing I've done with linode specifically have been pretty good. I've not seen any major problems with audio quality. However, my tests were not intensive and I did not do any call-volume testing.

i use it Xen for testing setups, but would definitely not recommend it for any production environment that involves a lot of calls.  i used a Xen instance for a get-out-the-vote campaign last november, and had repeated issues with choppy audio, even with almost all of the hardware resources assigned to that single Xen instance.

my understanding is that OpenVZ has been used successfully in these circumstances, so you might want to check that out if you're serious about pushing a lot of calls.

chad



From adam.kelloway at newpace.ca  Tue May  3 22:15:05 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Tue, 03 May 2011 15:15:05 -0300
Subject: [Freeswitch-users] org.freeswitch.esl.client warns of resource
	misuse
Message-ID: <4DC04629.7000200@newpace.ca>

Hi there,

This is directed at those involved in the development of the 
org.freeswitch.esl.client library. Apologies if this has already been 
mentioned, I couldn't find any references to it.

In evaluating this library, I noticed that 
AbstractOutboundPipelineFactory implementation of 
ChannelPipelineFactory.getPipeline() creates a new ExecutionHandler each 
time it is called, which results in a increasing number of thread pools 
being used that eventually triggers the following warning:

org.jboss.netty.util.internal.SharedResourceMisuseDetector
WARNING: You are creating too many MemoryAwareThreadPoolExecutor 
instances.  MemoryAwareThreadPoolExecutor is a shared resource that must 
be reused across the application, so that only a few instances are created.

Looking at the documentation for ExecutionHandler, I noticed that the 
example implementation provides a dedicated ExecutionHandler instance, 
which is passed to all pipelines returned in getPipeline(). Note the 
comment in the example that states it "Must be shared".

>From 
http://docs.jboss.org/netty/3.2/api/org/jboss/netty/handler/execution/ExecutionHandler.html:
--------------
public class DatabaseGatewayPipelineFactory implements 
|ChannelPipelineFactory| 
<http://docs.jboss.org/netty/3.2/api/org/jboss/netty/channel/ChannelPipelineFactory.html> 
{

*private final |ExecutionHandler| 
<http://docs.jboss.org/netty/3.2/api/org/jboss/netty/handler/execution/ExecutionHandler.html> 
executionHandler;*

public DatabaseGatewayPipelineFactory(|ExecutionHandler| 
<http://docs.jboss.org/netty/3.2/api/org/jboss/netty/handler/execution/ExecutionHandler.html> 
executionHandler) {
   this.executionHandler = executionHandler;
}

public |ChannelPipeline| 
<http://docs.jboss.org/netty/3.2/api/org/jboss/netty/channel/ChannelPipeline.html> 
getPipeline() {
   return |Channels| 
<http://docs.jboss.org/netty/3.2/api/org/jboss/netty/channel/Channels.html>.pipeline(
     new DatabaseGatewayProtocolEncoder(),
     new DatabaseGatewayProtocolDecoder(),
*    executionHandler, // Must be shared*
     new DatabaseQueryingHandler());
   }
}
-------------

Is there any particular reason why AbstractOutboundPipelineFactory does 
not provide a dedicated ExecutionHandler that may be shared by all 
pipelines, or is this simply an oversight?

Thanks and keep up the good work,

Adam Kelloway
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From msc at freeswitch.org  Wed May  4 04:21:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 3 May 2011 17:21:48 -0700
Subject: [Freeswitch-users] few questions about multi-tenant
	configuration
In-Reply-To: <C9E5D4C0.14BD4%krice@freeswitch.org>
References: <20110503223108.29d1dc94@ankh.reapnet.com>
	<C9E5D4C0.14BD4%krice@freeswitch.org>
Message-ID: <BANLkTimFERTUQX45Lpz6fx705XSvypRCAQ@mail.gmail.com>

On Tue, May 3, 2011 at 1:50 PM, Ken Rice <krice at freeswitch.org> wrote:

> Crap IGNORE THAT... READING THE WRONG MAILING LIST
>
Yeah, this isn't 4chan you FREAK! :P
-MC
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From brad at tritelcomm.com  Wed May  4 06:04:03 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Tue, 3 May 2011 19:04:03 -0700
Subject: [Freeswitch-users] How to use fs_path?
In-Reply-To: <6A7C8F16-0C97-4A70-9751-F9F59C78AE0F@gmail.com>
References: <fc7fe88611802.4dc002c7@insightbb.com>
	<6A7C8F16-0C97-4A70-9751-F9F59C78AE0F@gmail.com>
Message-ID: <BANLkTikg9SJvogn9VVRyym57OjSJ35hH=w@mail.gmail.com>

Yes! Wiki update please? :P

On Tue, May 3, 2011 at 2:59 PM, Jay Binks <jaybinks at gmail.com> wrote:

> Post your working result please
>
>
>
> On 04/05/2011, at 3:27 AM, Gary Chen <gchen00 at insightbb.com> wrote:
>
> > Never mind. I got it.
> >
> > Thanks.
> >
> > Gary
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From frankie.k.yiu at gmail.com  Wed May  4 07:48:22 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 3 May 2011 20:48:22 -0700
Subject: [Freeswitch-users] Build error related to "libsofia_sip_ua_static"
Message-ID: <BANLkTimvO6dv_07uuO6KQ3AeG3DkZwPNFA@mail.gmail.com>

Hi there,

I just download the latest freeSwitch code through TortoiseGit x64 on
Windows.  When I downloaded it I had the option "AutoCrlf" unchecked.  I am
building it on Windows 7 x64 with Visual Studio 2008 SP1.  Could someone
please let me know why I am getting these errors (not having the
libsofia_sip_ua_static.lib generated) ?  And when I build it on the other
machine (Windows Server x64, Windows Server Win32) I don't have the same
error.  All these machines are using Visual Studio 2008 SP1.


I got the following errors:

Error 1 error C2220: warning treated as error - no 'object' file
generated c:\BA\freeswitch\libs\sofia-sip\libsofia-sip-ua\su\su_string.c
1 libsofia_sip_ua_static
Error 4 fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib'
mod_unimrcp mod_unimrcp
Error 21 fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib'
mod_sofia mod_sofia

This is the build log:

1>------ Build started: Project: Download OPENSSL, Configuration: Debug
Win32 ------
1>Downloading OPENSSL.
1>Build log was saved at "
file://c:\Brightarrow\BASIPServices\freeswitch\libs\win32\Debug\BuildLogDownload
OPENSSL.htm"
1>Download OPENSSL - 0 error(s), 0 warning(s)
2>------ Build started: Project: Download CELT, Configuration: Debug Win32
------
2>Downloading CELT.
3>------ Build started: Project: Download JSON, Configuration: Debug Win32
------
3>Downloading JSON.
2>0 File(s) copied
3>Build log was saved at "
file://c:\Brightarrow\BASIPServices\freeswitch\libs\win32\Debug\BuildLogDownload
JSON.htm"
3>Download JSON - 0 error(s), 0 warning(s)
2>0 File(s) copied
2>Build log was saved at "
file://c:\Brightarrow\BASIPServices\freeswitch\libs\win32\Debug\BuildLogDownload
CELT.htm"
2>Download CELT - 0 error(s), 0 warning(s)
4>------ Build started: Project: libsofia_sip_ua_static, Configuration:
Debug Win32 ------
4>Performing Pre-Build Event...
5>------ Skipped Build: Project: Download 16khz music, Configuration: Debug
Win32 ------
5>Project not selected to build for this solution configuration
6>------ Skipped Build: Project: Download 32khz music, Configuration: Debug
Win32 ------
6>Project not selected to build for this solution configuration
7>------ Skipped Build: Project: Download 16khzsound, Configuration: Debug
Win32 ------
7>Project not selected to build for this solution configuration
8>------ Skipped Build: Project: Download 32khzsound, Configuration: Debug
Win32 ------
8>Project not selected to build for this solution configuration
9>------ Build started: Project: FreeSWITCH.Managed, Configuration: Debug
Any CPU ------
9>FreeSWITCH.Managed ->
C:\Brightarrow\BASIPServices\freeswitch\Debug\mod\FreeSWITCH.Managed.dll
10>------ Skipped Build: Project: mod_flite, Configuration: Debug Win32
------
10>Project not selected to build for this solution configuration
11>------ Skipped Build: Project: 32khz, Configuration: Debug Win32 ------
11>Project not selected to build for this solution configuration
12>------ Skipped Build: Project: 16khz, Configuration: Debug Win32 ------
12>Project not selected to build for this solution configuration
13>------ Build started: Project: 8khz, Configuration: Debug Win32 ------
13>Performing Post-Build Event...
13>0 File(s) copied
13>0 File(s) copied
4>multipart mismatch with Recursive multipart ()
13>0 File(s) copied
13>0 File(s) copied
13>0 File(s) copied
13>0 File(s) copied
4>Suppress-Body-If-Match header is experimental
4>Suppress-Notify-If-Match header is experimental
13>0 File(s) copied
13>0 File(s) copied
13>0 File(s) copied
13>0 File(s) copied
13>Build log was saved at "
file://c:\Brightarrow\BASIPServices\freeswitch\libs\win32\Sound_Files\Debug\BuildLog.htm
"
13>8khz - 0 error(s), 0 warning(s)
14>------ Skipped Build: Project: mod_opal, Configuration: Debug Win32
------
14>Project not selected to build for this solution configuration
15>------ Skipped Build: Project: mod_h323, Configuration: Debug Win32
------
15>Project not selected to build for this solution configuration
16>------ Skipped Build: Project: mod_skinny, Configuration: Debug Win32
------
16>Project not selected to build for this solution configuration
17>------ Skipped Build: Project: mod_skel, Configuration: Debug Win32
------
17>Project not selected to build for this solution configuration
18>------ Skipped Build: Project: mod_skypopen, Configuration: Debug Win32
------
18>Project not selected to build for this solution configuration
19>------ Build started: Project: 8khz music, Configuration: Debug Win32
------
19>Performing Post-Build Event...
4>Suppress-Body-If-Match header is experimental
4>Suppress-Notify-If-Match header is experimental
4>Suppress-Body-If-Match header is experimental
4>Suppress-Notify-If-Match header is experimental
19>0 File(s) copied
19>Build log was saved at "
file://c:\Brightarrow\BASIPServices\freeswitch\libs\win32\Sound_Files\Debug\BuildLog.htm
"
19>8khz music - 0 error(s), 0 warning(s)
20>------ Skipped Build: Project: 16khz music, Configuration: Debug Win32
------
20>Project not selected to build for this solution configuration
21>------ Skipped Build: Project: 32khz music, Configuration: Debug Win32
------
21>Project not selected to build for this solution configuration
22>------ Skipped Build: Project: FSComm, Configuration: Debug Win32 ------
22>Project not selected to build for this solution configuration
23>------ Skipped Build: Project: mod_cepstral, Configuration: Debug Win32
------
23>Project not selected to build for this solution configuration
24>------ Skipped Build: Project: mod_directory, Configuration: Debug Win32
------
24>Project not selected to build for this solution configuration
25>------ Skipped Build: Project: docs (Docs\docs), Configuration: Debug
Win32 ------
25>Project not selected to build for this solution configuration
4>NOTE:
4>NOTE: Remember to install pthreadVC2.dll to your path, too!
4>NOTE:
4>Compiling...
4>su_string.c
4>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : error C2220: warning
treated as error - no 'object' file generated
4>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : warning C4819: The file
contains a character that cannot be represented in the current code page
(950). Save the file in Unicode format to prevent data loss
4>Creating browse information file...
4>Microsoft Browse Information Maintenance Utility Version 9.00.21022
4>Copyright (C) Microsoft Corporation. All rights reserved.
4>Build log was saved at "
file://c:\Brightarrow\BASIPServices\freeswitch\libs\win32\sofia\Debug\BuildLog.htm
"
4>libsofia_sip_ua_static - 1 error(s), 1 warning(s)
26>------ Build started: Project: mrcpsofiasip, Configuration: Debug Win32
------
27>------ Build started: Project: mod_sofia, Configuration: Debug Win32
------
26>Compiling...
27>Compiling...
26>mrcp_sofiasip_client_agent.c
27>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler
27>sofia_sla.c
26>mrcp_sofiasip_server_agent.c
26>Generating Code...
26>Creating library...
26>Build log was saved at "
file://c:\BA\freeswitch\libs\unimrcp\modules\mrcp-sofiasip\Debug\BuildLog.htm
"
26>mrcpsofiasip - 0 error(s), 0 warning(s)
28>------ Build started: Project: mod_unimrcp, Configuration: Debug Win32
------
28>Linking...
27>sofia_reg.c
28>LINK : fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib'
28>Build log was saved at "
file://c:\BA\freeswitch\src\mod\asr_tts\mod_unimrcp\Debug\BuildLog.htm"
28>mod_unimrcp - 1 error(s), 0 warning(s)
27>sofia_presence.c
27>sofia_glue.c
27>.\sofia_glue.c(559) : warning C4244: 'function' : conversion from
'uint32_t' to 'switch_port_t', possible loss of data
27>.\sofia_glue.c(563) : warning C4244: 'function' : conversion from
'uint32_t' to 'switch_port_t', possible loss of data
27>.\sofia_glue.c(581) : warning C4244: 'function' : conversion from
'uint32_t' to 'switch_port_t', possible loss of data
27>.\sofia_glue.c(585) : warning C4244: 'function' : conversion from
'uint32_t' to 'switch_port_t', possible loss of data
27>.\sofia_glue.c(3173) : warning C4133: 'function' : incompatible types -
from 'switch_rtp_bug_flag_t *' to 'uint32_t *'
27>.\sofia_glue.c(3557) : warning C4244: 'function' : conversion from
'unsigned int' to 'switch_payload_t', possible loss of data
27>.\sofia_glue.c(4627) : warning C4244: 'function' : conversion from
'unsigned int' to 'switch_payload_t', possible loss of data
27>.\sofia_glue.c(4743) : warning C4244: '=' : conversion from 'unsigned
int' to 'switch_payload_t', possible loss of data
27>.\sofia_glue.c(4867) : warning C4244: '=' : conversion from 'unsigned
int' to 'switch_payload_t', possible loss of data
27>.\sofia_glue.c(5436) : warning C4244: 'function' : conversion from 'int'
to 'switch_payload_t', possible loss of data
27>.\sofia_glue.c(5442) : warning C4244: 'function' : conversion from 'int'
to 'switch_payload_t', possible loss of data
27>sofia.c
27>.\sofia.c(2436) : warning C4133: 'function' : incompatible types - from
'switch_rtp_bug_flag_t *' to 'uint32_t *'
27>.\sofia.c(2438) : warning C4133: 'function' : incompatible types - from
'switch_rtp_bug_flag_t *' to 'uint32_t *'
27>.\sofia.c(3233) : warning C4133: 'function' : incompatible types - from
'switch_rtp_bug_flag_t *' to 'uint32_t *'
27>.\sofia.c(3235) : warning C4133: 'function' : incompatible types - from
'switch_rtp_bug_flag_t *' to 'uint32_t *'
27>mod_sofia.c
27>Generating Code...
27>c:\BA\freeswitch\src\mod\endpoints\mod_sofia\sofia_glue.c(4129) : warning
C4702: unreachable code
27>Linking...
27>LINK : fatal error LNK1181: cannot open input file
'..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib'
27>Build log was saved at "
file://c:\BA\freeswitch\src\mod\endpoints\mod_sofia\Win32\Debug\BuildLog.htm
"
27>mod_sofia - 1 error(s), 17 warning(s)
========== Build: 7 succeeded, 3 failed, 134 up-to-date, 18 skipped
==========


Thanks,

Frankie
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From mkopacki at gmail.com  Wed May  4 10:50:21 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Wed, 04 May 2011 08:50:21 +0200
Subject: [Freeswitch-users] few questions about
	multi-tenant	configuration
In-Reply-To: <C9E5D4C0.14BD4%krice@freeswitch.org>
References: <C9E5D4C0.14BD4%krice@freeswitch.org>
Message-ID: <4DC0F72D.1060806@gmail.com>

    sure, it happens :)

-- 
Michal

On 2011-05-03 22:50, Ken Rice wrote:
> Crap IGNORE THAT... READING THE WRONG MAILING LIST
>
>
> On 5/3/11 3:31 PM, "Michal Kopacki"<mkopacki at gmail.com>  wrote:
>
>>      Hi,
>>
>> I'm working on scenario, which include resellers. Below
>> short description:
>>
>> There is one instance of freeswitch and some logical layer for
>> resellers. I would like to be able to handle multiple domains on one FS
>> and additionally few domains per reseller. I've already read multitenant
>> description on wiki but it barely scratch the surface so i have few
>> questions and I'll appreciate any thoughts:
>>
>> I would like to separate domains in different contexts:
>>
>> domain A have context domain-a and users in it
>> domain B have context domain-b and users in it
>>
>> For internal calls (within domain) it's simple, similar for outbound
>> calls (gateways defined in directory). But how to handle calls
>> between extensions ?
>>
>> Is there a way to add context name to bridge instruction ?
>>
>> And what about inbound calls ?
>>
>> As far I understand all inbound calls goes through sip_profile and in
>> definition of profile there is param like context.
>>
>> Is it possible to have this param dynamic (based on incoming call for
>> example) or it's better to have separate profil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From zetruger at gmail.com  Wed May  4 12:49:09 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Wed, 4 May 2011 12:49:09 +0400
Subject: [Freeswitch-users] how to implement execute_on_hold/unhold ?
Message-ID: <BANLkTi=k98NdfukR=urcjeA9-m5cW78AVw@mail.gmail.com>

how to implement execute_on_hold/unhold ?


From sid.kshatriya at gmail.com  Wed May  4 14:02:38 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 4 May 2011 15:32:38 +0530
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
Message-ID: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>

Dear All,

I'm thinking of buying a sip trunk from https://store.freepbx.com/ . A
single trunk is $24.99 a month.

I'd like to be able to support 2-3 simultaneous calls to my IVR on a single
US DID number. So that is ~$75 a month. Any cheaper options?  Other
providers ?

Thanks,

Sidharth

-- 
Sidharth Kshatriya
www.sidk.info
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From hesser4900 at gmail.com  Wed May  4 14:41:23 2011
From: hesser4900 at gmail.com (hesser4900 at gmail.com)
Date: Wed, 04 May 2011 10:41:23 +0000
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
Message-ID: <000e0cd4c6d0e6bd6104a270e53b@google.com>

Try Teliax.com. They give you a pay as you go model. Or Vitelity.com. same  
model

Thx
Holger

On May 4, 2011 5:02am, Sidharth Kshatriya <sid.kshatriya at gmail.com> wrote:
> Dear All,

> I'm thinking of buying a sip trunk from https://store.freepbx.com/ . A  
> single trunk is $24.99 a month.


> I'd like to be able to support 2-3 simultaneous calls to my IVR on a  
> single US DID number. So that is ~$75 a month. Any cheaper options? Other  
> providers ?




> Thanks,


> Sidharth

> --
> Sidharth Kshatriya
> www.sidk.info





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From Nabble at slickdeals.endjunk.com  Wed May  4 14:46:35 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 4 May 2011 03:46:35 -0700 (PDT)
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
Message-ID: <1304505994974-6330365.post@n2.nabble.com>

Have a look  http://www.dslreports.com/gbu here .

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-anybody-recommend-some-good-sip-trunk-providers-tp6330281p6330365.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Wed May  4 15:03:38 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 4 May 2011 14:03:38 +0300
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <1304505994974-6330365.post@n2.nabble.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
Message-ID: <BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>

Is this just for inbound? The benefit of the freepbx trunks is that it
includes outgoing for that price.. and therefore costs a lot more.

1) Have a look at one of FS's sponsors, VoiceNetwork.ca's pricing:
http://www.voicenetwork.ca/voip.php?page=4
 <http://www.voicenetwork.ca/voip.php?page=4>
USA DID Origination
Unlimited Incoming USA DID's for $3.95 MRC (2 Channels)
Per-Minute Incoming USA DID's for $0.99 MRC @ $0.011 per minute (10
Channels)

2) Or try http://www.didforsale.com for multi-channel incoming.

-Avi



On Wed, May 4, 2011 at 1:46 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

> Have a look  http://www.dslreports.com/gbu here .
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Can-anybody-recommend-some-good-sip-trunk-providers-tp6330281p6330365.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sid.kshatriya at gmail.com  Wed May  4 15:06:51 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 4 May 2011 16:36:51 +0530
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
Message-ID: <BANLkTinLFs78LQ2sU2szkOwMWeJVeZBnEw@mail.gmail.com>

Yes, just for inbound.

On Wed, May 4, 2011 at 4:33 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Is this just for inbound? The benefit of the freepbx trunks is that it
> includes outgoing for that price.. and therefore costs a lot more.
>
> 1) Have a look at one of FS's sponsors, VoiceNetwork.ca's pricing:
> http://www.voicenetwork.ca/voip.php?page=4
>  <http://www.voicenetwork.ca/voip.php?page=4>
> USA DID Origination
> Unlimited Incoming USA DID's for $3.95 MRC (2 Channels)
> Per-Minute Incoming USA DID's for $0.99 MRC @ $0.011 per minute (10
> Channels)
>
> 2) Or try http://www.didforsale.com for multi-channel incoming.
>
> -Avi
>
>
>
> On Wed, May 4, 2011 at 1:46 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:
>
>> Have a look  http://www.dslreports.com/gbu here .
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Can-anybody-recommend-some-good-sip-trunk-providers-tp6330281p6330365.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
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From Nabble at slickdeals.endjunk.com  Wed May  4 16:04:36 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 4 May 2011 05:04:36 -0700 (PDT)
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTinLFs78LQ2sU2szkOwMWeJVeZBnEw@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
	<BANLkTinLFs78LQ2sU2szkOwMWeJVeZBnEw@mail.gmail.com>
Message-ID: <1304510676628-6330552.post@n2.nabble.com>


Sidharth Kshatriya wrote:
> Yes, just for inbound.
Just for inbound, you can use Google Voice,  http://www.sipgate.com SIPGate
US ,  http://www.voxox.com VoXoX ,  http://whistlephone.com WhistlePhone ,
etc., and they are FREE.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-anybody-recommend-some-good-sip-trunk-providers-tp6330281p6330552.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tayeb.meftah at gmail.com  Wed May  4 16:24:50 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 04 May 2011 14:24:50 +0200
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
 providers?
In-Reply-To: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
Message-ID: <4DC14592.90500@gmail.com>

http://www.voicenetwork.ca/
what do you think?
On 04/05/2011 12:02, Sidharth Kshatriya wrote:
> Dear All,
>
> I'm thinking of buying a sip trunk from https://store.freepbx.com/ . A 
> single trunk is $24.99 a month.
>
> I'd like to be able to support 2-3 simultaneous calls to my IVR on a 
> single US DID number. So that is ~$75 a month. Any cheaper options? 
>  Other providers ?
>
> Thanks,
>
> Sidharth
>
> -- 
> Sidharth Kshatriya
> www.sidk.info <http://www.sidk.info>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883

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From avi at avimarcus.net  Wed May  4 16:32:33 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 4 May 2011 15:32:33 +0300
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <1304510676628-6330552.post@n2.nabble.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
	<BANLkTinLFs78LQ2sU2szkOwMWeJVeZBnEw@mail.gmail.com>
	<1304510676628-6330552.post@n2.nabble.com>
Message-ID: <BANLkTi=+JWPxWS-s8t4tjq=JaXsnoiD5CQ@mail.gmail.com>

They are nice.. for personal stuff. But if you want several channels.. or if
it's for business usage - it may be against the TOS and more importantly,
since you're not paying it's *out of your control*.
e.g. whistle started charging for "long" calls as per
http://whistlephone.com/naywa/

 <http://whistlephone.com/naywa/>Often, full price is the cheapest prices...

-Avi

On Wed, May 4, 2011 at 3:04 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> Sidharth Kshatriya wrote:
> > Yes, just for inbound.
> Just for inbound, you can use Google Voice,  http://www.sipgate.comSIPGate
> US ,  http://www.voxox.com VoXoX ,  http://whistlephone.com WhistlePhone ,
> etc., and they are FREE.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Can-anybody-recommend-some-good-sip-trunk-providers-tp6330281p6330552.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 Nabble at slickdeals.endjunk.com  Wed May  4 16:59:59 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 4 May 2011 05:59:59 -0700 (PDT)
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTi=+JWPxWS-s8t4tjq=JaXsnoiD5CQ@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
	<BANLkTinLFs78LQ2sU2szkOwMWeJVeZBnEw@mail.gmail.com>
	<1304510676628-6330552.post@n2.nabble.com>
	<BANLkTi=+JWPxWS-s8t4tjq=JaXsnoiD5CQ@mail.gmail.com>
Message-ID: <1304513999838-6330705.post@n2.nabble.com>


Avi Marcus-2 wrote:
> They are nice.. for personal stuff. But if you want several channels.. or
> if
> it's for business usage - it may be against the TOS and more importantly,
> since you're not paying it's *out of your control*.
> e.g. whistle started charging for "long" calls as per
> http://whistlephone.com/naywa/
> 
>  &lt;http://whistlephone.com/naywa/&gt;Often, full price is the cheapest
> prices...
> 
> -Avi
I agree with you. Since OP is looking for some cheaper inbound ONLY options,
I thought to throw in some freebies that cost nothing for inbound calls with
no guarantee in quality. We know that cheap equates with lemon in a lesser
quality.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-anybody-recommend-some-good-sip-trunk-providers-tp6330281p6330705.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Wed May  4 17:48:13 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 4 May 2011 06:48:13 -0700 (PDT)
Subject: [Freeswitch-users] Build error related to
	"libsofia_sip_ua_static"
In-Reply-To: <BANLkTimvO6dv_07uuO6KQ3AeG3DkZwPNFA@mail.gmail.com>
References: <BANLkTimvO6dv_07uuO6KQ3AeG3DkZwPNFA@mail.gmail.com>
Message-ID: <1304516893189-6330890.post@n2.nabble.com>

If you can help me diagnose what is wrong with the code in your locale maybe
we can fix this.

4>su_string.c
4>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : error C2220: warning
treated as error - no 'object' file generated
 4>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : warning C4819: The file
contains a character that cannot be represented in the current code page
(950). Save the file in Unicode format to prevent data loss


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-error-related-to-libsofia-sip-ua-static-tp6329599p6330890.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From pkelly at gmail.com  Wed May  4 18:05:12 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 4 May 2011 15:05:12 +0100
Subject: [Freeswitch-users] Continue streaming RTP after BYE
Message-ID: <BANLkTimbsYExeK3e7tSWrjtsdUuds9btmQ@mail.gmail.com>

Hi, I don't have high hopes that this is possible, however I'll give it a
shot here :)

I have an IVR application which plays a prompt and then hangs up with a
BYE.

When the BYE is sent, it's intercepted by an opensips b2bua and a reINVITE
is issued upstream, followed by a downstream invite (possibly to another IVR
or PSTN).

The problem I have is that between the BYE from the IVR and the placing of
the next call there is often silence, the user will only hear ringing if the
downstream INVITE responds by sending RTP to the media IP and port of the
caller.

My question is, is it possible to have freeswitch/another program somehow
continue to send media to the origination IP/port for a few seconds after
the BYE has been issued? If it's not possible with freeswitch does anyone
know of a third party app which could be invoked to send RTP media to a
known IP and port?

I know it's a bizarre request but if anyone could point me in a direction I
would be grateful.

Thanks

Pete
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From nico at clickfono.com  Wed May  4 19:26:40 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Wed, 4 May 2011 11:26:40 -0400
Subject: [Freeswitch-users] Is it a good idea to deploy Freeswitch on
 Xen Instances/Virtual Machines?
In-Reply-To: <9A2340F2-B677-4DB4-BF33-46683146A551@apartmentlines.com>
References: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>
	<BANLkTi=uYpBin6GsAERcv2vnE72zaC50Zw@mail.gmail.com>
	<BANLkTim+k0g9eJ6vutxvfdUi=H7986fM-Q@mail.gmail.com>
	<BANLkTinaBXqvE7bX3TLSKd=MQgG8eMmDUw@mail.gmail.com>
	<9A2340F2-B677-4DB4-BF33-46683146A551@apartmentlines.com>
Message-ID: <BANLkTikdgKM-AiNpdWdQYtyupcOJouf36w@mail.gmail.com>

We have a production server on EC2 and it works great, we haven't had any
issues with call quality at all. But I guess it all depends on the call
volume you get and what other processes are running on the same machine.

Probably for what you are going to do though, your biggest issue will be the
connection quality (latency) between the flash app and the red5 server.


On Tue, May 3, 2011 at 6:31 PM, Chad Phillips -- Apartment Lines <
chad at apartmentlines.com> wrote:

>
> On May 3, 2011, at 11:39 AM, Brad Mina wrote:
>
> > The testing I've done with linode specifically have been pretty good.
> I've not seen any major problems with audio quality. However, my tests were
> not intensive and I did not do any call-volume testing.
>
> i use it Xen for testing setups, but would definitely not recommend it for
> any production environment that involves a lot of calls.  i used a Xen
> instance for a get-out-the-vote campaign last november, and had repeated
> issues with choppy audio, even with almost all of the hardware resources
> assigned to that single Xen instance.
>
> my understanding is that OpenVZ has been used successfully in these
> circumstances, so you might want to check that out if you're serious about
> pushing a lot of calls.
>
> 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 msc at freeswitch.org  Wed May  4 19:27:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 May 2011 08:27:31 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTi=H5hdhzP8nh8JgnX3trEjBjsUtvQ@mail.gmail.com>

Hey folks,

We're having an easy conference call today. The agenda is light:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_04

Feel free to join and hang out with us!

-MC
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From steveayre at gmail.com  Wed May  4 19:55:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 4 May 2011 16:55:21 +0100
Subject: [Freeswitch-users] Is it a good idea to deploy Freeswitch on
 Xen Instances/Virtual Machines?
In-Reply-To: <9A2340F2-B677-4DB4-BF33-46683146A551@apartmentlines.com>
References: <BANLkTiknAj+M-2HqiBSxjJmZoJCEZERi_w@mail.gmail.com>
	<BANLkTi=uYpBin6GsAERcv2vnE72zaC50Zw@mail.gmail.com>
	<BANLkTim+k0g9eJ6vutxvfdUi=H7986fM-Q@mail.gmail.com>
	<BANLkTinaBXqvE7bX3TLSKd=MQgG8eMmDUw@mail.gmail.com>
	<9A2340F2-B677-4DB4-BF33-46683146A551@apartmentlines.com>
Message-ID: <BANLkTin=G41V3-fR+OXXnWrZEHUx+qCu_w@mail.gmail.com>

The issue with running within a virtual machine is that your guest OS has no
control over timing. If another guest or the host does a lot of CPU or I/O
that can mean your guest can get blocked for an unpredictable amount of time
making maintaining a steady timer difficult. FS needs a reliable timer for
its RTP media to work reliably.

However having said that, if the server (both host and guests) isn't highly
loaded you probably won't have any major issues - timing will probably be
'reliable enough' and your jitter buffers will hide some of the jitter
you'll get so that you don't notice some of the more minor timing glitches.

I have run FS within virtual machines (VirtualBox) for testing purposes
without problems, and with problems. I don't run FS in a VM on any of my
production servers and wouldn't - but I do know others do.

OpenVZ takes a different approach to virtual machines - rather than
emulating an entire virtual server it partitions off your host system.
Everything is still running on the same system with the kernel knowing
exactly what's going on so timing is far more reliable. OpenVZ is the way
I've seen the FS developers recommend you run it if you must run it within a
VM, but that doesn't mean it won't work under Xen/EC2/etc.

-Steve




On 3 May 2011 23:31, Chad Phillips -- Apartment Lines <
chad at apartmentlines.com> wrote:

>
> On May 3, 2011, at 11:39 AM, Brad Mina wrote:
>
> > The testing I've done with linode specifically have been pretty good.
> I've not seen any major problems with audio quality. However, my tests were
> not intensive and I did not do any call-volume testing.
>
> i use it Xen for testing setups, but would definitely not recommend it for
> any production environment that involves a lot of calls.  i used a Xen
> instance for a get-out-the-vote campaign last november, and had repeated
> issues with choppy audio, even with almost all of the hardware resources
> assigned to that single Xen instance.
>
> my understanding is that OpenVZ has been used successfully in these
> circumstances, so you might want to check that out if you're serious about
> pushing a lot of calls.
>
> 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 dujinfang at gmail.com  Wed May  4 21:05:37 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 5 May 2011 01:05:37 +0800
Subject: [Freeswitch-users] FS segfault when fs_cli -u xxxx
Message-ID: <3931B7235A3040AA80A84D31E12E0413@gmail.com>

I know the right syntax is fs_cli -u user at domain, but it really kills FS when do fs_cli -u xxxx without @

Since I'm not running the latest git so not sure should I report a bug.

I'm on a Mac 10.6.7 with git-3d73e23 2011-04-20 16-26-47 -0500

Any one can repeat this?

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
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From steveayre at gmail.com  Wed May  4 21:52:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 4 May 2011 18:52:32 +0100
Subject: [Freeswitch-users] FS segfault when fs_cli -u xxxx
In-Reply-To: <3931B7235A3040AA80A84D31E12E0413@gmail.com>
References: <3931B7235A3040AA80A84D31E12E0413@gmail.com>
Message-ID: <BANLkTik3LE=_mxicNHvhYCuqiBBuR-470A@mail.gmail.com>

You can report it as a bug if you can reproduce it on the latest git.

-Steve


On 4 May 2011 18:05, Seven Du <dujinfang at gmail.com> wrote:

>  I know the right syntax is fs_cli -u user at domain, but it really kills FS
> when do fs_cli -u xxxx without @
>
> Since I'm not running the latest git so not sure should I report a bug.
>
> I'm on a Mac 10.6.7 with git-3d73e23 2011-04-20 16-26-47 -0500
>
> Any one can repeat this?
>
> --
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> Sent with Sparrow <http://www.sparrowmailapp.com>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed May  4 21:55:48 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 4 May 2011 12:55:48 -0500
Subject: [Freeswitch-users] FS segfault when fs_cli -u xxxx
In-Reply-To: <BANLkTik3LE=_mxicNHvhYCuqiBBuR-470A@mail.gmail.com>
References: <3931B7235A3040AA80A84D31E12E0413@gmail.com>
	<BANLkTik3LE=_mxicNHvhYCuqiBBuR-470A@mail.gmail.com>
Message-ID: <AEC55FB3-8F7E-42CC-9CC1-C640AA981B13@freeswitch.org>

Sounds like a bug in user lookup in FreeSWITCH please give the backtrace and info on a Jira once you confirm it still happens on the latest git rev.. 


On May 4, 2011, at 12:52 PM, Steven Ayre wrote:

> You can report it as a bug if you can reproduce it on the latest git.
> 
> -Steve
> 
> 
> On 4 May 2011 18:05, Seven Du <dujinfang at gmail.com> wrote:
> 
>> I know the right syntax is fs_cli -u user at domain, but it really kills FS
>> when do fs_cli -u xxxx without @
>> 
>> Since I'm not running the latest git so not sure should I report a bug.
>> 
>> I'm on a Mac 10.6.7 with git-3d73e23 2011-04-20 16-26-47 -0500
>> 
>> Any one can repeat this?

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From frankie.k.yiu at gmail.com  Wed May  4 22:00:52 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 4 May 2011 11:00:52 -0700
Subject: [Freeswitch-users] Build error related to
	"libsofia_sip_ua_static"
Message-ID: <BANLkTimHGBsGZc2qmVKPsPUZKv7PxfLx=A@mail.gmail.com>

My problem is solved.

Under Region and Language, I have the "Language for non-Unicode programs"
set to Chinese when I could not compile successfully.  Now I set it to
English and it works fine.

So I guess I can not set it to other language while working on freeSWITCH?

Thanks,
Frankie




> ---------- Forwarded message ----------
> From: Jeff Lenk <jeff at jefflenk.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 4 May 2011 06:48:13 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Build error related to
> "libsofia_sip_ua_static"
> If you can help me diagnose what is wrong with the code in your locale
> maybe
> we can fix this.
>
> 4>su_string.c
> 4>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : error C2220: warning
> treated as error - no 'object' file generated
>  4>..\..\sofia-sip\libsofia-sip-ua\su\su_string.c : warning C4819: The file
> contains a character that cannot be represented in the current code page
> (950). Save the file in Unicode format to prevent data loss
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Build-error-related-to-libsofia-sip-ua-static-tp6329599p6330890.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
>
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From msc at freeswitch.org  Wed May  4 23:42:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 May 2011 12:42:18 -0700
Subject: [Freeswitch-users] how to implement execute_on_hold/unhold ?
In-Reply-To: <BANLkTi=k98NdfukR=urcjeA9-m5cW78AVw@mail.gmail.com>
References: <BANLkTi=k98NdfukR=urcjeA9-m5cW78AVw@mail.gmail.com>
Message-ID: <BANLkTinJ-8Ae1CP-qX_O+a-RwKj64huNcg@mail.gmail.com>

Could you supply more details on what you are trying to accomplish?
-MC

On Wed, May 4, 2011 at 1:49 AM, ???? ???????? <zetruger at gmail.com> wrote:

> how to implement execute_on_hold/unhold ?
>
>
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From jeff at jefflenk.com  Thu May  5 01:59:03 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 4 May 2011 14:59:03 -0700 (PDT)
Subject: [Freeswitch-users] Build error related to
	"libsofia_sip_ua_static"
In-Reply-To: <BANLkTimHGBsGZc2qmVKPsPUZKv7PxfLx=A@mail.gmail.com>
References: <BANLkTimvO6dv_07uuO6KQ3AeG3DkZwPNFA@mail.gmail.com>
	<BANLkTimHGBsGZc2qmVKPsPUZKv7PxfLx=A@mail.gmail.com>
Message-ID: <1304546343916-6332522.post@n2.nabble.com>

Please git pull and try this again I submitted a fix. Please let me know if
it works now or if there are more problems.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Build-error-related-to-libsofia-sip-ua-static-tp6329599p6332522.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.varnes at gmail.com  Thu May  5 03:37:53 2011
From: david.varnes at gmail.com (david varnes)
Date: Thu, 5 May 2011 09:37:53 +1000
Subject: [Freeswitch-users] org.freeswitch.esl.client warns of resource
	misuse
In-Reply-To: <4DC04629.7000200@newpace.ca>
References: <4DC04629.7000200@newpace.ca>
Message-ID: <BANLkTin=6FrG8_G0ozvaC+XJ5EyzTxZ2gA@mail.gmail.com>

Hi Adam,

On 4 May 2011 04:15, Adam Kelloway <adam.kelloway at newpace.ca> wrote:
> Hi there,
>
> This is directed at those involved in the development of the
> org.freeswitch.esl.client library. Apologies if this has already been
> mentioned, I couldn't find any references to it.
>
> In evaluating this library, I noticed that AbstractOutboundPipelineFactory
> implementation of ChannelPipelineFactory.getPipeline() creates a new
> ExecutionHandler each time it is called, which results in a increasing
> number of thread pools being used that eventually triggers the following
> warning:
>
> org.jboss.netty.util.internal.SharedResourceMisuseDetector
> WARNING: You are creating too many MemoryAwareThreadPoolExecutor instances.
> MemoryAwareThreadPoolExecutor is a shared resource that must be reused
> across the application, so that only a few instances are created.
>
> Looking at the documentation for ExecutionHandler, I noticed that the
> example implementation provides a dedicated ExecutionHandler instance, which
> is passed to all pipelines returned in getPipeline(). Note the comment in
> the example that states it "Must be shared".
>
> From
> http://docs.jboss.org/netty/3.2/api/org/jboss/netty/handler/execution/ExecutionHandler.html:
> --------------
> public class DatabaseGatewayPipelineFactory implements
> ChannelPipelineFactory {
>
> private final ExecutionHandler executionHandler;
>
> public DatabaseGatewayPipelineFactory(ExecutionHandler executionHandler) {
> ? this.executionHandler = executionHandler;
> }
>
> public ChannelPipeline getPipeline() {
> ? return Channels.pipeline(
> ??? new DatabaseGatewayProtocolEncoder(),
> ??? new DatabaseGatewayProtocolDecoder(),
> ??? executionHandler, // Must be shared
> ??? new DatabaseQueryingHandler());
> ? }
> }
> -------------
>
> Is there any particular reason why AbstractOutboundPipelineFactory does not
> provide a dedicated ExecutionHandler that may be shared by all pipelines, or
> is this simply an oversight?

Very good question.  Off the top of my head I would say this is an oversight.

Which version of esl-client and netty are you seeing this with ?

Should be straight forward to sort this out.
I have a few pending changes to commit into the esl-client, and will
try to get to it asap.

Thanks for the post.

Also, I should find somewhere to log bugs (and patches :-), not sure
if the official JIRA
system is available.

regards
davidv


> Thanks and keep up the good work,
>
> Adam Kelloway
>



-- 
david varnes

e: david.varnes at gmail.com


From dave at dchorton.com  Thu May  5 06:28:14 2011
From: dave at dchorton.com (Dave Horton)
Date: Wed, 4 May 2011 22:28:14 -0400
Subject: [Freeswitch-users] help on creating an rpm
In-Reply-To: <3FFE5FF0-4F80-4042-88B9-54BCBD394294@seventhsignal.de>
References: <6DDD2468-B228-4E86-9A32-7018697DD8D0@dchorton.com>
	<B0D2ACF7-8C5D-498E-A2D5-1B7DA62F9B4D@dchorton.com>
	<3FFE5FF0-4F80-4042-88B9-54BCBD394294@seventhsignal.de>
Message-ID: <BC50171E-8E38-48E4-B9B0-46C960947014@dchorton.com>

Thanks, that was the problem.  I had installed python 2.7 because some other software needed it; not sure exactly how it was causing problems but I removed it so that configure found the older python install (2.4) and the rpm built fine.  Many thanks.  
On May 3, 2011, at 4:46 PM, Michal Bielicki wrote:

It seems to try to link a 32bit version of libpython to mod_pthon instead of the 64bit version. In dtail it seems to be linking in a version you installed by hand in/usr/local and that seems to be a 32bit version and that is why it fails.

Am 03.05.2011 um 22:33 schrieb Dave Horton:

> Hmm..actually this is centos:
> 
> [root at centos64-1 ~]# uname -a
> Linux centos64-1 2.6.18-194.26.1.el5 #1 SMP Tue Nov 9 12:54:20 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
> [root at centos64-1 ~]# cat /etc/redhat-release 
> CentOS release 5.5 (Final)
> 
> 
> 
> 
> Begin forwarded message:
> 
> From: Dave Horton <dave at dchorton.com>
> Date: May 3, 2011 8:09:29 AM EDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: help on creating an rpm
> 
> 
> Thanks for some of the feedback, I have gotten further.  (As noted, I am not all that experienced with rpms, but have read up on various articles).  Now, I have created my .rpmmacros to set the top level directory, and gotten my source tree in the correct place to be located, and when I run rpmbuild against the spec file it runs for quite a while but eventually fails with this error below.  
> 
> gcc -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python/mod_python.c -o mod_python.o >/dev/null 2>&1
> Creating mod_python.so...
> /usr/bin/ld: /usr/local/lib/libpython2.7.a(abstract.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
> /usr/local/lib/libpython2.7.a: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> g++ -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -D_GNU_SOURCE -shared -o .libs/mod_python.so -shared -Wl,-x .libs/mod_python.o freeswitch_python.o mod_python_wrap.o  /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/.libs/libfreeswitch.so -lz -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib -lpq /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib/.libs/libexpat.a /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr/.libs/libapr-1.a -luuid -lrt -lcr
> ypt -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/srtp -lssl -lcrypto -lncurses -L/usr/local/lib -lpthread -ldl -lutil -lm -lpython2.7   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath -Wl,/opt/freeswitch/mod
> make[5]: *** [mod_python.so] Error 1
> make[5]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[4]: *** [all] Error 1
> make[4]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[3]: *** [languages/mod_python-all] Error 1
> make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod'
> make[3]: Entering directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[3]: Nothing to be done for `all-am'.
> make[3]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7'
> make: *** [all] Error 2
> error: Bad exit status from /var/tmp/rpm-tmp.67849 (%build)
> 
> Any thoughts about what might be wrong?  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de


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From fieldpeak at gmail.com  Thu May  5 08:24:38 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 5 May 2011 12:24:38 +0800
Subject: [Freeswitch-users] FS as outbound conference server
Message-ID: <BANLkTi=8zUG2Y=GD8UkfVupVyO__ngN=Dg@mail.gmail.com>

Hi Friends,

i need configure FS as  outbound conference server, and control FS to
realize it by php webpages and DTMF.
if there any open source php webpages will be much better, thanks for any
hints.

Regards,
Charles
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From frankie.k.yiu at gmail.com  Thu May  5 11:01:05 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Thu, 5 May 2011 00:01:05 -0700
Subject: [Freeswitch-users] Build error related to
	"libsofia_sip_ua_static"
Message-ID: <BANLkTin0EaC7sVgAY_bosx1+icAj+1aiCg@mail.gmail.com>

It works now.  Thanks Jeff.



> ---------- Forwarded message ----------
> From: Jeff Lenk <jeff at jefflenk.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 4 May 2011 14:59:03 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Build error related to
> "libsofia_sip_ua_static"
> Please git pull and try this again I submitted a fix. Please let me know if
> it works now or if there are more problems.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Build-error-related-to-libsofia-sip-ua-static-tp6329599p6332522.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
>
> ---------- Forwarded message ----------
> From: david varnes <david.varnes at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 5 May 2011 09:37:53 +1000
> Subject: Re: [Freeswitch-users] org.freeswitch.esl.client warns of resource
> misuse
> Hi Adam,
>
> On 4 May 2011 04:15, Adam Kelloway <adam.kelloway at newpace.ca> wrote:
> > Hi there,
> >
> > This is directed at those involved in the development of the
> > org.freeswitch.esl.client library. Apologies if this has already been
> > mentioned, I couldn't find any references to it.
> >
> > In evaluating this library, I noticed that
> AbstractOutboundPipelineFactory
> > implementation of ChannelPipelineFactory.getPipeline() creates a new
> > ExecutionHandler each time it is called, which results in a increasing
> > number of thread pools being used that eventually triggers the following
> > warning:
> >
> > org.jboss.netty.util.internal.SharedResourceMisuseDetector
> > WARNING: You are creating too many MemoryAwareThreadPoolExecutor
> instances.
> > MemoryAwareThreadPoolExecutor is a shared resource that must be reused
> > across the application, so that only a few instances are created.
> >
> > Looking at the documentation for ExecutionHandler, I noticed that the
> > example implementation provides a dedicated ExecutionHandler instance,
> which
> > is passed to all pipelines returned in getPipeline(). Note the comment in
> > the example that states it "Must be shared".
> >
> > From
> >
> http://docs.jboss.org/netty/3.2/api/org/jboss/netty/handler/execution/ExecutionHandler.html
> :
> > --------------
> > public class DatabaseGatewayPipelineFactory implements
> > ChannelPipelineFactory {
> >
> > private final ExecutionHandler executionHandler;
> >
> > public DatabaseGatewayPipelineFactory(ExecutionHandler executionHandler)
> {
> >   this.executionHandler = executionHandler;
> > }
> >
> > public ChannelPipeline getPipeline() {
> >   return Channels.pipeline(
> >     new DatabaseGatewayProtocolEncoder(),
> >     new DatabaseGatewayProtocolDecoder(),
> >     executionHandler, // Must be shared
> >     new DatabaseQueryingHandler());
> >   }
> > }
> > -------------
> >
> > Is there any particular reason why AbstractOutboundPipelineFactory does
> not
> > provide a dedicated ExecutionHandler that may be shared by all pipelines,
> or
> > is this simply an oversight?
>
> Very good question.  Off the top of my head I would say this is an
> oversight.
>
> Which version of esl-client and netty are you seeing this with ?
>
> Should be straight forward to sort this out.
> I have a few pending changes to commit into the esl-client, and will
> try to get to it asap.
>
> Thanks for the post.
>
> Also, I should find somewhere to log bugs (and patches :-), not sure
> if the official JIRA
> system is available.
>
> regards
> davidv
>
>
> > Thanks and keep up the good work,
> >
> > Adam Kelloway
> >
>
>
>
> --
> david varnes
>
> e: david.varnes at gmail.com
>
>
>
>
> ---------- Forwarded message ----------
> From: Dave Horton <dave at dchorton.com>
> To: Michal Bielicki <michal.bielicki at seventhsignal.de>
> Date: Wed, 4 May 2011 22:28:14 -0400
> Subject: Re: [Freeswitch-users] help on creating an rpm
> Thanks, that was the problem.  I had installed python 2.7 because some
> other software needed it; not sure exactly how it was causing problems but I
> removed it so that configure found the older python install (2.4) and the
> rpm built fine.  Many thanks.
> On May 3, 2011, at 4:46 PM, Michal Bielicki wrote:
>
> It seems to try to link a 32bit version of libpython to mod_pthon instead
> of the 64bit version. In dtail it seems to be linking in a version you
> installed by hand in/usr/local and that seems to be a 32bit version and that
> is why it fails.
>
> Am 03.05.2011 um 22:33 schrieb Dave Horton:
>
> Hmm..actually this is centos:
>
> [root at centos64-1 ~]# uname -a
> Linux centos64-1 2.6.18-194.26.1.el5 #1 SMP Tue Nov 9 12:54:20 EST 2010
> x86_64 x86_64 x86_64 GNU/Linux
> [root at centos64-1 ~]# cat /etc/redhat-release
> CentOS release 5.5 (Final)
>
>
>
>
> Begin forwarded message:
>
> From: Dave Horton <dave at dchorton.com>
> Date: May 3, 2011 8:09:29 AM EDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: help on creating an rpm
>
>
> Thanks for some of the feedback, I have gotten further.  (As noted, I am
> not all that experienced with rpms, but have read up on various articles).
>  Now, I have created my .rpmmacros to set the top level directory, and
> gotten my source tree in the correct place to be located, and when I run
> rpmbuild against the spec file it runs for quite a while but eventually
> fails with this error below.
>
> gcc -I/usr/local/include/python2.7 -I/usr/local/include/python2.7
> -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall
> -Wstrict-prototypes
> -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include
> -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include
> -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC
> -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
> -ggdb -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
> -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wall
> -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE
> -DHAVE_CONFIG_H -c
> /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python/mod_python.c
> -o mod_python.o >/dev/null 2>&1
> Creating mod_python.so...
> /usr/bin/ld: /usr/local/lib/libpython2.7.a(abstract.o): relocation
> R_X86_64_32 against `a local symbol' can not be used when making a shared
> object; recompile with -fPIC
> /usr/local/lib/libpython2.7.a: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> g++ -I/usr/local/include/python2.7 -I/usr/local/include/python2.7
> -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv -O3 -Wall
> -Wstrict-prototypes
> -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include
> -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/include
> -I/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/libteletone/src -fPIC
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -O2 -g
> -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector
> --param=ssp-buffer-size=4 -m64 -mtune=generic -D_GNU_SOURCE -shared -o
> .libs/mod_python.so -shared -Wl,-x .libs/mod_python.o freeswitch_python.o
> mod_python_wrap.o
>  /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/.libs/libfreeswitch.so -lz
> -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib
> -lpq
> /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/apr/.libs/libapr-1.a
> -luuid -lrt -lcr
> ypt -L/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/libs/srtp -lssl
> -lcrypto -lncurses -L/usr/local/lib -lpthread -ldl -lutil -lm -lpython2.7
>   -Wl,--rpath -Wl,/opt/freeswitch/lib -Wl,--rpath -Wl,/opt/freeswitch/mod
> make[5]: *** [mod_python.so] Error 1
> make[5]: Leaving directory
> `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[4]: *** [all] Error 1
> make[4]: Leaving directory
> `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod/languages/mod_python'
> make[3]: *** [languages/mod_python-all] Error 1
> make[3]: Leaving directory
> `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src/mod'
> make[3]: Entering directory
> `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[3]: Nothing to be done for `all-am'.
> make[3]: Leaving directory
> `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory
> `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory
> `/data/home/build/tmp/rpm/BUILD/freeswitch-1.0.7'
> make: *** [all] Error 2
> error: Bad exit status from /var/tmp/rpm-tmp.67849 (%build)
>
> Any thoughts about what might be wrong?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> *Michal Bielicki*
> Gesch?ftsf?hrer / CEO
>
> *Seventh Signal Ltd. & Co. KG*
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Gesch?ftsf?hrer: Michal Bielicki
> Pers?nlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.: http://www.seventhsignal.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Thu May  5 13:41:07 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 5 May 2011 11:41:07 +0200
Subject: [Freeswitch-users] FS as outbound conference server
In-Reply-To: <BANLkTi=8zUG2Y=GD8UkfVupVyO__ngN=Dg@mail.gmail.com>
References: <BANLkTi=8zUG2Y=GD8UkfVupVyO__ngN=Dg@mail.gmail.com>
Message-ID: <DC5E5E9C-FEE7-4A55-940C-6AEE0FF2E9F7@ipeva.fr>

Charles,

check the wiki for info on PHP ESL.

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

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

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




Le 05/05/2011 ? 06:24, fieldpeak a ?crit :

> Hi Friends,
> 
> i need configure FS as  outbound conference server, and control FS to realize it by php webpages and DTMF.
> if there any open source php webpages will be much better, thanks for any hints.
> 
> 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 adam.kelloway at newpace.ca  Thu May  5 16:11:11 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Thu, 05 May 2011 09:11:11 -0300
Subject: [Freeswitch-users] org.freeswitch.esl.client warns of resource
 misuse
In-Reply-To: <BANLkTin=6FrG8_G0ozvaC+XJ5EyzTxZ2gA@mail.gmail.com>
References: <4DC04629.7000200@newpace.ca>
	<BANLkTin=6FrG8_G0ozvaC+XJ5EyzTxZ2gA@mail.gmail.com>
Message-ID: <4DC293DF.5060103@newpace.ca>

Hi David,

This was with version 0.9.2 and netty-3.2.1.Final.jar.

Thanks very much,

Adam

On 3:59 PM, david varnes wrote:
> Hi Adam,
>
> On 4 May 2011 04:15, Adam Kelloway<adam.kelloway at newpace.ca>  wrote:
>> Hi there,
>>
>> This is directed at those involved in the development of the
>> org.freeswitch.esl.client library. Apologies if this has already been
>> mentioned, I couldn't find any references to it.
>>
>> In evaluating this library, I noticed that AbstractOutboundPipelineFactory
>> implementation of ChannelPipelineFactory.getPipeline() creates a new
>> ExecutionHandler each time it is called, which results in a increasing
>> number of thread pools being used that eventually triggers the following
>> warning:
>>
>> org.jboss.netty.util.internal.SharedResourceMisuseDetector
>> WARNING: You are creating too many MemoryAwareThreadPoolExecutor instances.
>> MemoryAwareThreadPoolExecutor is a shared resource that must be reused
>> across the application, so that only a few instances are created.
>>
>> Looking at the documentation for ExecutionHandler, I noticed that the
>> example implementation provides a dedicated ExecutionHandler instance, which
>> is passed to all pipelines returned in getPipeline(). Note the comment in
>> the example that states it "Must be shared".
>>
>> From
>> http://docs.jboss.org/netty/3.2/api/org/jboss/netty/handler/execution/ExecutionHandler.html:
>> --------------
>> public class DatabaseGatewayPipelineFactory implements
>> ChannelPipelineFactory {
>>
>> private final ExecutionHandler executionHandler;
>>
>> public DatabaseGatewayPipelineFactory(ExecutionHandler executionHandler) {
>>    this.executionHandler =xecutionHandler;
>> }
>>
>> public ChannelPipeline getPipeline() {
>>    return Channels.pipeline(
>>      new DatabaseGatewayProtocolEncoder(),
>>      new DatabaseGatewayProtocolDecoder(),
>>      executionHandler, // Must be shared
>>      new DatabaseQueryingHandler());
>>    }
>> }
>> -------------
>>
>> Is there any particular reason why AbstractOutboundPipelineFactory does not
>> provide a dedicated ExecutionHandler that may be shared by all pipelines, or
>> is this simply an oversight?
> Very good question.  Off the top of my head I would say this is an oversight.
>
> Which version of esl-client and netty are you seeing this with ?
>
> Should be straight forward to sort this out.
> I have a few pending changes to commit into the esl-client, and will
> try to get to it asap.
>
> Thanks for the post.
>
> Also, I should find somewhere to log bugs (and patches :-), not sure
> if the official JIRA
> system is available.
>
> regards
> davidv
>
>
>> Thanks and keep up the good work,
>>
>> Adam Kelloway
>>
>
>


-- 
Adam Kelloway
Software Engineer
NewPace Technology Development Inc.
adam.kelloway at newpace.ca
+1 902-406-8375 x1031
www.newpace.ca



From jcasale at activenetwerx.com  Thu May  5 23:57:46 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 5 May 2011 19:57:46 +0000
Subject: [Freeswitch-users] Multi tenant sofia profile issue
Message-ID: <assp.0106754fd1.92097A6A775D5147B1078E3F15430B92492363@prato.activenetwerx.local>

I am trying to sort out my sofia profile setup for what will be a multihomed fs box
with all the different internal domains each in its own subnet serviced by a unique
interface.

There will not be any external clients connecting, just one DID for each domain so
the external profile loads all DIDs and each of those routes into the appropriate context.

The internal profile setup has me a bit puzzled, originally the Name/Type made sense
where I had Foo/profile and Bar/profile each on its set interface ip and port. Now
I have one showing up as an alias for the other? Logfile says profile is started and
it creates an alias for it, yet "sofia status" shows only an alias for Bar to Foo, and the
only internal profile is Foo.

Any idea what I may have overlooked?

Thanks!
jlc


From jcasale at activenetwerx.com  Fri May  6 03:15:55 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 5 May 2011 23:15:55 +0000
Subject: [Freeswitch-users] Multi tenant sofia profile issue
In-Reply-To: <92097A6A775D5147B1078E3F15430B92492363@prato.activenetwerx.local>
References: <92097A6A775D5147B1078E3F15430B92492363@prato.activenetwerx.local>
Message-ID: <assp.01067d0750.92097A6A775D5147B1078E3F15430B92492D67@prato.activenetwerx.local>

>Any idea what I may have overlooked?

Sorry guys, error in my directory conf...


From scvikic at dotix.hr  Fri May  6 03:41:58 2011
From: scvikic at dotix.hr (Srdjan Cvikic)
Date: Fri, 6 May 2011 01:41:58 +0200 (CEST)
Subject: [Freeswitch-users] How to add INVITE arguments?
Message-ID: <005f01cc0b7e$09f97530$1dec5f90$@hr>

Hi all,

How can I add INVITE line or header arguments as mentioned in RFC4715?

INVITE tel:+17005554141;isub=12345;isub-encoding=nsap-ia5 SIP/2.0
To: <tel:+17005554141;isub=12345;isub-encoding=nsap-ia5>
From: "Bob"<sip:bob at biloxi.example.com>;tag=1928301774

TIA!


From admin at blindi.net  Fri May  6 05:47:17 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 6 May 2011 03:47:17 +0200 (CEST)
Subject: [Freeswitch-users] Problem originate conference hangup
In-Reply-To: <E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
Message-ID: <Pine.LNX.4.64.1105060307050.9252@debian>

Hi all, i have setup a admingroup in a conference.
My section in conference.conf.xml is:
<group name="admin">
  <control action="execute_application" digits="2"
  data="bridge loopback/427/default"/>

my 427:
<extension name="DISA for conference dialout">
<condition field="destination_number" expression="^(427)$">
  <action application="answer"/>
  <action application="sleep" data="100"/>
<action application="read"
data="2 30 'tone_stream://%(10000,0,3 50,440)' digits 30000 #"/>
<action application="set"
data="konfnum=${hash(select/confout/thomas)}"/>
<action application="export"
data="konfnum1=${hash(select/confout/thomas)}"/>

<action application="system"
data="$${base_dir}/scripts/callme.sh ${konfnum}  ${digits}" />

My callme.sh:
#!/bin/sh
##callout for a conference
/usr/local/freeswitch/bin/fs_cli -x
  "bgapi originate  {ignore_early_media=true,originate_retries=100,
  origination_caller_id_name=Callback,originate_retry_sleep_ms=10000,
  originate_timeout=900}loopback/$2/disa &conference($1 at admin)"

My Adminextension:
     <extension name="local admin  konferenz">
       <condition field="destination_number" expression="^(7676)$">
         <action application="answer"/>
      <action application="sleep" data="1000"/>
<action application="hash"
data="insert/confout/thomas/$1-${domain_name}"/>
<action application="conference" data="$1-${domain_name}@admin"/>
  </condition>
  </extension>

These works fine from my sipadapters.
I  enter the adminextension, press 2 heare the dialtone, enter the outside 
numer, press # and go back to the conference. This is ok.

The problem: i use the originate command to call me self.
I enter the admin extension. Then i press 2 and hear the dialtone, i enter 
the phonenumber and press #. Freeswitch hangup my callbackline.
I like to go back to my conference.
the same from my sip-adapters.
Can you help me please?
I neet a parameter on the originate command or so?

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 andrew.keil at askinteractive.net  Fri May  6 09:14:11 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Fri, 6 May 2011 15:14:11 +1000
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>

To Freeswitch developers,

RE: MOD_SPANDSP

I have a client that wishes to send 20,000 to 30,000 outbound faxes daily and has requested me to quote for a Freeswitch setup that will support this.

I notice the comment on the mod spandsp wiki page: "We are finishing mod_spandsp. It requires full field testing now."  Well this could be a perfect opportunity :)

I need to understand the following:


1.       Does the mod spandsp support the TDM cards (Sangoma) to send faxes out over E1 (Euro ISDN)?

2.       How many concurrent faxes out have been tested using mod_spandsp ?  Was this over SIP or over ISDN?

3.       What is the preferred Operating System to use (since basically I would prefer to use the same one as the main developer of mod_spandsp)?

4.       Where do I source the latest email2pdf to test that out?

That should be enough to kick this question off.

Kind Regards,

Andrew Keil

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From markmorreny at gmail.com  Fri May  6 11:50:42 2011
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 6 May 2011 15:50:42 +0800
Subject: [Freeswitch-users] freeswitch interop issue
Message-ID: <BANLkTin0KGmp25O2Bpzkp_OLJ-7pqCEhYQ@mail.gmail.com>

Hi,

I am having trouble receiving calls from a sippy voip server.
Freeswitch returns 400 for a normal INVITE I received from my vendor.
Does anyone know what is wrong?  Is there anyway to configure
freeswitch to get this interoperability issue resolved?



U x.x.x.x:5061 -> :5060
INVITE sip:13453334444 at y.y.y.y:5060 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.x:5061;branch=z9hG4bK-cjuna7hd7oy4rvgl;rport.
Max-Forwards: 70.
From: ;tag=uyfnqeaupzqsi3az.o.
To: .
Call-ID: YTRhZTRjMzAyZTg0ZjgxNTg4MmI5ZjRjYjljNjY4YTY.~1o.
CSeq: 621 INVITE.
Contact: Anonymous .
Expires: 300.
User-Agent: Sippy.
cisco-GUID: 1714481388-1998655968-2851059884-1870679645.
h323-conf-id: 1714481388-1998655968-2851059884-1870679645.
Content-disposition: session.
Content-Length: 403.
Content-Type: application/sdp.
.
v=0.
o=Sippy 476627024 0 IN IP4 x.x.x.x.
s=CounterPath X-Lite 4.0.
t=0 0.
a=ice-ufrag:e60b22.
a=ice-pwd:3dc27abbae9db2484de85b0a2ac295fe.
m=audio 34354 RTP/AVP 0 8 101.
c=IN IP4 70.52.58.139.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=candidate:1 1 UDP 659136 192.168.1.104 56716 typ host.
a=candidate:1 2 UDP 659134 192.168.1.104 56717 typ host.
a=oldmediaip:192.168.1.105.


U y.y.y.y:5060 -> x.x.x.x:5061
SIP/2.0 400 Bad Session Description.
Via: SIP/2.0/UDP x.x.x.x:5061;branch=z9hG4bK-cjuna7hd7oy4rvgl;rport=5061.
From: ;tag=uyfnqeaupzqsi3az.o.
To: ;tag=QggHUaHjv0SSr.
Call-ID: YTRhZTRjMzAyZTg0ZjgxNTg4MmI5ZjRjYjljNjY4YTY.~1o.
CSeq: 621 INVITE.
User-Agent: freeswitch.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Length: 0.
.

Thanks,
Mark


From david.ponzone at ipeva.fr  Fri May  6 12:40:57 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 6 May 2011 10:40:57 +0200
Subject: [Freeswitch-users] freeswitch interop issue
In-Reply-To: <BANLkTin0KGmp25O2Bpzkp_OLJ-7pqCEhYQ@mail.gmail.com>
References: <BANLkTin0KGmp25O2Bpzkp_OLJ-7pqCEhYQ@mail.gmail.com>
Message-ID: <B56F1A65-790B-40E3-94DD-59F5B6DA8279@ipeva.fr>

Mark,

it seems strange to me that the SIP From and To fields are empty.

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

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

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




Le 06/05/2011 ? 09:50, mark morreny a ?crit :

> Hi,
> 
> I am having trouble receiving calls from a sippy voip server.
> Freeswitch returns 400 for a normal INVITE I received from my vendor.
> Does anyone know what is wrong?  Is there anyway to configure
> freeswitch to get this interoperability issue resolved?
> 
> 
> 
> U x.x.x.x:5061 -> :5060
> INVITE sip:13453334444 at y.y.y.y:5060 SIP/2.0.
> Via: SIP/2.0/UDP x.x.x.x:5061;branch=z9hG4bK-cjuna7hd7oy4rvgl;rport.
> Max-Forwards: 70.
> From: ;tag=uyfnqeaupzqsi3az.o.
> To: .
> Call-ID: YTRhZTRjMzAyZTg0ZjgxNTg4MmI5ZjRjYjljNjY4YTY.~1o.
> CSeq: 621 INVITE.
> Contact: Anonymous .
> Expires: 300.
> User-Agent: Sippy.
> cisco-GUID: 1714481388-1998655968-2851059884-1870679645.
> h323-conf-id: 1714481388-1998655968-2851059884-1870679645.
> Content-disposition: session.
> Content-Length: 403.
> Content-Type: application/sdp.
> .
> v=0.
> o=Sippy 476627024 0 IN IP4 x.x.x.x.
> s=CounterPath X-Lite 4.0.
> t=0 0.
> a=ice-ufrag:e60b22.
> a=ice-pwd:3dc27abbae9db2484de85b0a2ac295fe.
> m=audio 34354 RTP/AVP 0 8 101.
> c=IN IP4 70.52.58.139.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
> a=candidate:1 1 UDP 659136 192.168.1.104 56716 typ host.
> a=candidate:1 2 UDP 659134 192.168.1.104 56717 typ host.
> a=oldmediaip:192.168.1.105.
> 
> 
> U y.y.y.y:5060 -> x.x.x.x:5061
> SIP/2.0 400 Bad Session Description.
> Via: SIP/2.0/UDP x.x.x.x:5061;branch=z9hG4bK-cjuna7hd7oy4rvgl;rport=5061.
> From: ;tag=uyfnqeaupzqsi3az.o.
> To: ;tag=QggHUaHjv0SSr.
> Call-ID: YTRhZTRjMzAyZTg0ZjgxNTg4MmI5ZjRjYjljNjY4YTY.~1o.
> CSeq: 621 INVITE.
> User-Agent: freeswitch.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, refer.
> Content-Length: 0.
> .
> 
> Thanks,
> Mark
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tayeb.meftah at gmail.com  Fri May  6 14:35:08 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 06 May 2011 12:35:08 +0200
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
Message-ID: <4DC3CEDC.6030105@gmail.com>

Os: CentOs (Recomanded)
Sangoma: yes, supported over any protocol. fax is endpoint indepande, 
could use sangoma card, tdm (FXO/FXS).
thank you
On 06/05/2011 07:14, Andrew Keil wrote:
>
> To Freeswitch developers,
>
> * *
>
> *RE: MOD_SPANDSP*
>
> I have a client that wishes to send 20,000 to 30,000 outbound faxes 
> daily and has requested me to quote for a Freeswitch setup that will 
> support this.
>
> I notice the comment on the mod spandsp wiki page: "We are finishing 
> mod_spandsp. It requires full field testing now."  Well this could be 
> a perfect opportunity J
>
> _I need to understand the following:_
>
> 1. Does the mod spandsp support the TDM cards (Sangoma) to send faxes 
> out over E1 (Euro ISDN)?
>
> 2. How many concurrent faxes out have been tested using mod_spandsp ?  
> Was this over SIP or over ISDN?
>
> 3. What is the preferred Operating System to use (since basically I 
> would prefer to use the same one as the main developer of mod_spandsp)?
>
> 4. Where do I source the latest *email2pdf* to test that out?
>
> That should be enough to kick this question off.
>
> Kind Regards,
>
> Andrew Keil
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883

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From steve at justfone.com  Fri May  6 15:45:04 2011
From: steve at justfone.com (Steven Brown)
Date: Fri, 6 May 2011 12:45:04 +0100
Subject: [Freeswitch-users] Dingaling on Ubuntu 10.04
Message-ID: <BANLkTimN7v+ANCfmepuS9_YQph1_CNqpGQ@mail.gmail.com>

Hi

I'm having problems getting dingaling running on Ubuntu 10.04 LTS to
talk to googletalk, a quick look at the logs revealed that the issue
seems to be " TLS NOT SUPPORTED IN THIS BUILD! " .

I can see this exact problem has been covered before both in this list
and the wiki so apologies for raising it again but I'm stummped,  I
think I have covered everything off but despite this and resorting to
several fresh builds from scratch the problem persists.

The fresh builds followed the process :

installing libgnutls26 and libgnutls-dev
download fs from git
./bootstrap.sh
./configure
enable dingaling in modules.conf
make
make install
enable dingaling in modules.conf.xml and set up a client profile
run fs

also tried manually editing the -lgnutls flag in mod_dingaling.la as
per the wiki,  followed by a make mod_dingaling install but still the
same problem.

Any suggestions much appreciated.

Thanks

Steve


From tayeb.meftah at gmail.com  Fri May  6 17:29:32 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 06 May 2011 15:29:32 +0200
Subject: [Freeswitch-users] Dingaling on Ubuntu 10.04
In-Reply-To: <BANLkTimN7v+ANCfmepuS9_YQph1_CNqpGQ@mail.gmail.com>
References: <BANLkTimN7v+ANCfmepuS9_YQph1_CNqpGQ@mail.gmail.com>
Message-ID: <4DC3F7BC.2000401@gmail.com>

1. install gnutls and gnutls devel i don't know the package name in ubuntu
2. reconfigure your git head using ./configure
3. make && make install and retry
thank you
On 06/05/2011 13:45, Steven Brown wrote:
> Hi
>
> I'm having problems getting dingaling running on Ubuntu 10.04 LTS to
> talk to googletalk, a quick look at the logs revealed that the issue
> seems to be " TLS NOT SUPPORTED IN THIS BUILD! " .
>
> I can see this exact problem has been covered before both in this list
> and the wiki so apologies for raising it again but I'm stummped,  I
> think I have covered everything off but despite this and resorting to
> several fresh builds from scratch the problem persists.
>
> The fresh builds followed the process :
>
> installing libgnutls26 and libgnutls-dev
> download fs from git
> ./bootstrap.sh
> ./configure
> enable dingaling in modules.conf
> make
> make install
> enable dingaling in modules.conf.xml and set up a client profile
> run fs
>
> also tried manually editing the -lgnutls flag in mod_dingaling.la as
> per the wiki,  followed by a make mod_dingaling install but still the
> same problem.
>
> Any suggestions much appreciated.
>
> Thanks
>
> 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
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883



From brett.maxfield+freeswitch at gmail.com  Fri May  6 17:30:40 2011
From: brett.maxfield+freeswitch at gmail.com (Brett Maxfield)
Date: Fri, 6 May 2011 23:30:40 +1000
Subject: [Freeswitch-users] FreeSWITCH and SOHO systems / HPET timers
In-Reply-To: <BANLkTinAbN-3bCNodmO4_01-h1mg9MA20w@mail.gmail.com>
References: <BANLkTinAbN-3bCNodmO4_01-h1mg9MA20w@mail.gmail.com>
Message-ID: <BANLkTikdHVCTGVM7XDG-K3Z_YNN-02vo4w@mail.gmail.com>

Hi List,

I have been looking for a small SOHO software switch to manage one or two
phone extensions to initially one upstream SIP provider.

I have installed the demo/example system on a few pieces of hardware which
while are not high end, nevertheless are OK considering the very small
capacity required for a SOHO system, and me wanting to reserve my better
hardware for more high end loads. The embedded fanless system i have runs
ok, but the voice for the test IVR is very choppy, and the same choppy
result on a atom dual core net-top (both centos 5.6). I understand that
tuning is a complex issue on a non-RT system such as Linux, and so
FreeSWITCH is probably only tuned for high end systems. This choppiness,
specifically on low end systems, is "mentioned" in previous list discussions
as being a timer granularity issue with the kernel.

I've had some googling around and it seems that the newest centos version
supported by FreeSWITCH is CentOS 5.3, perhaps due to issues with timer
resolution in newer kernels. The stock kernel for 5.4 and 5.5 does not
appear to have a 1000Hz kernel, from what i can see, and xen stock kernels
that i have seen have 250Hz, which is different again, and would also be a
problem for FreeSWITCH running in a xen guest.

So my question is, given the contentious issue this seems to have been in
the past, should i even bother trying/using FreeSWITCH for these sorts of
SOHO / low end systems ?

Is there any plan to make FreeSWITCH tunable (in a non-default way) for low
end SOHO systems, which might have a typical 3-4 (say 10 at max)
connections, rather than being tuned for 100's or even 1000's of connections
by default.

>From what i have seen i quite like FreeSWITCH, and would like to run it for
both SOHO and learning purposes.

I note that centos 5.6 does not explicitly mention CONFIG_HZ_* anymore in
the kernel config (/boot/config-2.6.18-238.9.1.el5.centos.plus), but it does
appear to have support for HPET :

CONFIG_HPET_TIMER=y
CONFIG_HPET_EMULATE_RTC=y
CONFIG_TICK_DIVIDER=y

There was talk of high resolution timer patches for kernel 2.6.17, and hpet
timers seem to be present in the 2.6.18 kernel, at least in centos 5.6 above
(i have not checked previous versions).

Given the increasing availability of HPET timers in newer kernels, could
features such as nanosleep() and HPET timers be used to avoid the problems
with the kernel's timer granularity / HZ issues ?

Cheers
Brett
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From michele.garribba at gmail.com  Fri May  6 10:29:55 2011
From: michele.garribba at gmail.com (Michele Garribba)
Date: Fri, 6 May 2011 08:29:55 +0200
Subject: [Freeswitch-users] freeswitch configure file empty
Message-ID: <BA2E8AE6-F9E1-4CDA-99B1-6967B371F4D8@gmail.com>

Hi,

I'm tryin  to setup a system with freeswitch and zrtp.
Using ubuntu server 10.10

Everything works during the install process following the wiki
but when i got to the ./configure --enable-zrtp command nothing
happens. Looking the configure file, it's empty!

I downloaded freeswitch with git, as the guide told.

Any suggetion?
My i try with wget and pickup the tar? In the zrtp wiki says to
use only git to do this.

Thanks
smaikol


From max.clark at gmail.com  Fri May  6 18:25:41 2011
From: max.clark at gmail.com (Max Clark)
Date: Fri, 6 May 2011 07:25:41 -0700
Subject: [Freeswitch-users] Moving on from CentOS
Message-ID: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>

Hello all,

Recent developments (or absolute lack of) within the CentOS project
and its perceived long term viability has forced an internal
discussion to select a successor distribution. The most likely
candidates at this point are Ubuntu and its LTS releases for servers,
and Scientific Linux with the new 6.x releases. The pro/con lists for
each are growing and the issue is complicated.

With CentOS 5.x being the reference distro for FreeSWITCH development
I'm curious if this conversation has started among the FreeSWITCH
developers, and if it has, what is the project leaning to for a
successor distribution?

Thanks,
Max


From gmaruzz at gmail.com  Fri May  6 18:26:32 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 6 May 2011 16:26:32 +0200
Subject: [Freeswitch-users] freeswitch configure file empty
In-Reply-To: <BA2E8AE6-F9E1-4CDA-99B1-6967B371F4D8@gmail.com>
References: <BA2E8AE6-F9E1-4CDA-99B1-6967B371F4D8@gmail.com>
Message-ID: <BANLkTikPeCeu2zocHOc6RaqVZGx8Y9n30A@mail.gmail.com>

You must first do:
./bootsrap.sh
then configure

-giovanni

On 5/6/11, Michele Garribba <michele.garribba at gmail.com> wrote:
> Hi,
>
> I'm tryin  to setup a system with freeswitch and zrtp.
> Using ubuntu server 10.10
>
> Everything works during the install process following the wiki
> but when i got to the ./configure --enable-zrtp command nothing
> happens. Looking the configure file, it's empty!
>
> I downloaded freeswitch with git, as the guide told.
>
> Any suggetion?
> My i try with wget and pickup the tar? In the zrtp wiki says to
> use only git to do this.
>
> Thanks
> smaikol
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveu at coppice.org  Fri May  6 18:58:14 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 06 May 2011 22:58:14 +0800
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
Message-ID: <4DC40C86.9000503@coppice.org>

On 05/06/2011 01:14 PM, Andrew Keil wrote:
>
> To Freeswitch developers,
>
> **
>
> *RE: MOD_SPANDSP*
>
> I have a client that wishes to send 20,000 to 30,000 outbound faxes 
> daily and has requested me to quote for a Freeswitch setup that will 
> support this.
>
> I notice the comment on the mod spandsp wiki page: ?We are finishing 
> mod_spandsp. It requires full field testing now.? Well this could be a 
> perfect opportunity J
>
> _I need to understand the following:_
>
> 1.Does the mod spandsp support the TDM cards (Sangoma) to send faxes 
> out over E1 (Euro ISDN)?
>
> 2.How many concurrent faxes out have been tested using mod_spandsp ? 
> Was this over SIP or over ISDN?
>
> 3.What is the preferred Operating System to use (since basically I 
> would prefer to use the same one as the main developer of mod_spandsp)?
>
> 4.Where do I source the latest *email2pdf* to test that out?
>
> That should be enough to kick this question off.
>
> Kind Regards,
>
> Andrew Keil
>
>
What is the average number of pages per FAX that you expect to send? 
Unless its quite big, this will be a fairly small scale system, so you 
won't need any fancy hardware.

You can send the FAXes over ISDN or SIP. The Sangoma cards will do a 
good job if you want to use ISDN E1s. If the average FAX is <=4 pages, 
you'll probably only need a single E1, and you won't need anything very 
exotic for the server, unless you are doing quite a lot of works besides 
sending the FAXes. If this is a FAX only system you can use an E1 card 
that does not have an echo canceller (EC) module. That saves money. If 
you will have voice traffic as well, you had better consider buying an 
EC equipped card.

Centos 5.x is the OS usually recommended for FreeSwitch, as it is what 
most of the core developers test on.

I'm not clear why you are looking for email2pdf.

Is this going to be some kind of small scale e-mail to FAX gateway? FAX 
to e-mail is a more common setup, but quite a few people run e-mail to 
FAX gateways, too.

Steve



From steve at justfone.com  Fri May  6 19:04:59 2011
From: steve at justfone.com (Steven Brown)
Date: Fri, 6 May 2011 16:04:59 +0100
Subject: [Freeswitch-users] Dingaling on Ubuntu 10.04
In-Reply-To: <BANLkTimqjrn2688F6V_7M-RVRtbb=yfasw@mail.gmail.com>
References: <BANLkTimN7v+ANCfmepuS9_YQph1_CNqpGQ@mail.gmail.com>
	<4DC3F7BC.2000401@gmail.com>
	<BANLkTimqjrn2688F6V_7M-RVRtbb=yfasw@mail.gmail.com>
Message-ID: <BANLkTi=2WbCfkH_XTKB9XaYVVAf_P3=m3w@mail.gmail.com>

Thanks, I suspected as you do that I didn't have gnutls properly
installed, however from what I can gather the correct packages for
ubuntu are libgnutls26 and libgnutls-dev and I installed them before
running ./configure

Perhaps I have the wrong ubuntu packages but these are the ones
advised in the wiki,

I wonder if there is there anyone that confirm they have dingaling
running ok on Ubuntu 10.04 ?

Steve



>
> On 6 May 2011 14:29, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>> 1. install gnutls and gnutls devel i don't know the package name in ubuntu
>> 2. reconfigure your git head using ./configure
>> 3. make && make install and retry
>> thank you
>> On 06/05/2011 13:45, Steven Brown wrote:
>>>
>>> Hi
>>>
>>> I'm having problems getting dingaling running on Ubuntu 10.04 LTS to
>>> talk to googletalk, a quick look at the logs revealed that the issue
>>> seems to be " TLS NOT SUPPORTED IN THIS BUILD! " .
>>>
>>> I can see this exact problem has been covered before both in this list
>>> and the wiki so apologies for raising it again but I'm stummped, ?I
>>> think I have covered everything off but despite this and resorting to
>>> several fresh builds from scratch the problem persists.
>>>
>>> The fresh builds followed the process :
>>>
>>> installing libgnutls26 and libgnutls-dev
>>> download fs from git
>>> ./bootstrap.sh
>>> ./configure
>>> enable dingaling in modules.conf
>>> make
>>> make install
>>> enable dingaling in modules.conf.xml and set up a client profile
>>> run fs
>>>
>>> also tried manually editing the -lgnutls flag in mod_dingaling.la as
>>> per the wiki, ?followed by a make mod_dingaling install but still the
>>> same problem.
>>>
>>> Any suggestions much appreciated.
>>>
>>> Thanks
>>>
>>> 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
>>>
>>
>>
>> --
>> Meftah Tayeb
>> inum: +883510001288000
>> phone: +13477595883
>>
>>
>
>
>
> --
> Steven Brown
>
> email?? steve at justfone.com
> office?? 08707706968
> mobile 07768755409
> fax? ?01896242000
>
> Justfone - Company Reg. No.? : 3926817
>
> Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW
>
> The contents of this e-mail may be privileged and are confidential. It
> may not be disclosed to or used by anyone other than the addressee(s),
> nor copied in any way. If received in error, please advise sender,
> then delete it from your system. Internet email communications are not
> secure and therefore Justfone do not accept legal responsibility for
> the contents of this message. Any views or opinions presented are
> solely those of the author and do not necessarily represent those of
> Justfone unless otherwise specifically stated.
>



-- 
Steven Brown

email?? steve at justfone.com
office?? 08707706968
mobile 07768755409
fax? ?01896242000

Justfone - Company Reg. No.? : 3926817

Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW

The contents of this e-mail may be privileged and are confidential. It
may not be disclosed to or used by anyone other than the addressee(s),
nor copied in any way. If received in error, please advise sender,
then delete it from your system. Internet email communications are not
secure and therefore Justfone do not accept legal responsibility for
the contents of this message. Any views or opinions presented are
solely those of the author and do not necessarily represent those of
Justfone unless otherwise specifically stated.


From Nabble at slickdeals.endjunk.com  Fri May  6 19:34:38 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 6 May 2011 08:34:38 -0700 (PDT)
Subject: [Freeswitch-users] Dingaling on Ubuntu 10.04
In-Reply-To: <BANLkTi=2WbCfkH_XTKB9XaYVVAf_P3=m3w@mail.gmail.com>
References: <BANLkTimN7v+ANCfmepuS9_YQph1_CNqpGQ@mail.gmail.com>
	<4DC3F7BC.2000401@gmail.com>
	<BANLkTi=2WbCfkH_XTKB9XaYVVAf_P3=m3w@mail.gmail.com>
Message-ID: <1304696078558-6337932.post@n2.nabble.com>

Perhaps, do a make update-clean first then make to see if that will resolve
the issue.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dingaling-on-Ubuntu-10-04-tp6337317p6337932.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Dennis.Young at supportkids.com  Fri May  6 19:46:42 2011
From: Dennis.Young at supportkids.com (Dennis Young)
Date: Fri, 6 May 2011 10:46:42 -0500
Subject: [Freeswitch-users] Trillium Error
Message-ID: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49C@jehuty.supportkids.com>

All,

I'm seeing this error message in the command prompt window that is running Freeswitch but it's not showing up in the freeswitch log file or remote console.  I running GIT HEAD 5/1/11 on WIN32 system.

UNTSS: sw error:  ent: 010  inst: 000  proc id: 000
                file: ..\..\trillium\in\in_bdy4.c line: 483 errcode: 14536 errcls: ERRCLS_DEBUG
                errval: 00018  errdesc: inUsrT302S25() failed, timer not defined for switch.

It seem to happen right before in inbound freetdm call rolls to voicemail.  No calls in the switch are failing to my knowledge.

Any ideas?

...dly

Dennis Young |CIO |Phone 512.437.3901 | Fax 512.437.7202
Supportkids Services, Inc. | P.O. Box 18988 | Austin | TX 78760 | Phone 512.437.6000 | Fax 512.437.6030

[cid:image001.gif at 01CC0BD9.49D052E0]


Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law.  Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited.  If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.
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From brad at tritelcomm.com  Fri May  6 20:35:01 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Fri, 6 May 2011 09:35:01 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
Message-ID: <BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>

I'd personally like to know some opinions on distros, being that I've been
wanting to move FROM Ubuntu and TO CentOS for stability reasons. Ubuntu's
packages update so frequently I'm afraid of newer libraries and such
breaking FreeSwitch and/or the way things operate.

On Fri, May 6, 2011 at 7:25 AM, Max Clark <max.clark at gmail.com> wrote:

> Hello all,
>
> Recent developments (or absolute lack of) within the CentOS project
> and its perceived long term viability has forced an internal
> discussion to select a successor distribution. The most likely
> candidates at this point are Ubuntu and its LTS releases for servers,
> and Scientific Linux with the new 6.x releases. The pro/con lists for
> each are growing and the issue is complicated.
>
> With CentOS 5.x being the reference distro for FreeSWITCH development
> I'm curious if this conversation has started among the FreeSWITCH
> developers, and if it has, what is the project leaning to for a
> successor distribution?
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mitch.capper at gmail.com  Fri May  6 20:56:40 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 6 May 2011 09:56:40 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
Message-ID: <BANLkTinNYCnNQH4F7e_XjZd87wvzCcVMLQ@mail.gmail.com>

For what its worth freeswitch is not dependent on too many things.
Freeswitch (despite some peoples hatred of it) fetches the known good
versions of libraries that it uses.   This helps ensure stability,
everyone on an equal platform, and in this case means that if a system
library is upgrade it would not affect freeswitch.   With that said I
personally use Gentoo and have had great results with freeswitch on
it.

~Mitch

On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:
> I'd personally like to know some opinions on distros, being that I've been
> wanting to move FROM Ubuntu and TO CentOS for stability reasons. Ubuntu's
> packages update so frequently I'm afraid of newer libraries and such
> breaking FreeSwitch and/or the way things operate.
>
> On Fri, May 6, 2011 at 7:25 AM, Max Clark <max.clark at gmail.com> wrote:
>>
>> Hello all,
>>
>> Recent developments (or absolute lack of) within the CentOS project
>> and its perceived long term viability has forced an internal
>> discussion to select a successor distribution. The most likely
>> candidates at this point are Ubuntu and its LTS releases for servers,
>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>> each are growing and the issue is complicated.
>>
>> With CentOS 5.x being the reference distro for FreeSWITCH development
>> I'm curious if this conversation has started among the FreeSWITCH
>> developers, and if it has, what is the project leaning to for a
>> successor distribution?
>>
>> Thanks,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri May  6 21:13:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 6 May 2011 10:13:44 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
Message-ID: <BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>

On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:

> I'd personally like to know some opinions on distros, being that I've been
> wanting to move FROM Ubuntu and TO CentOS for stability reasons. Ubuntu's
> packages update so frequently I'm afraid of newer libraries and such
> breaking FreeSwitch and/or the way things operate.


I don't know what's up with CentOS either, but I'm in no hurry to run away
from 5.6 right now. That being said, we've been having reports of success
with various distros. TJ (IRC: bougyman) has been having what he calls
"brilliant success" with Arch Linux, even when running AMD hardware. ;)
 I've personally used Debian Lenny for a while and I find it to be stable
and predictable. Moc has been experimenting with Scientific Linux and he has
also reported good success.

As it stands now, CentOS is still the "ol' reliable" of the distros and
still continues to be boring and predictable - just what we like. :)  I'm
about to migrate one of my test boxes to CentOS 5.6 and do some testing.
I'll report back any issues.

-MC
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From krice at freeswitch.org  Fri May  6 21:35:22 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 06 May 2011 12:35:22 -0500
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
Message-ID: <C9E99B8A.16DCC%krice@freeswitch.org>

I?ve been running Centos5.6 for a bit now with not real issues to report
with FS... 


On 5/6/11 12:13 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> As it stands now, CentOS is still the "ol' reliable" of the distros and still
> continues to be boring and predictable - just what we like. :) ?I'm about to
> migrate one of my test boxes to CentOS 5.6 and do some testing. I'll report
> back any issues.?

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From oseslija at gmail.com  Fri May  6 21:40:59 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Fri, 6 May 2011 19:40:59 +0200
Subject: [Freeswitch-users] FreeSWITCH and SOHO systems / HPET timers
In-Reply-To: <BANLkTikdHVCTGVM7XDG-K3Z_YNN-02vo4w@mail.gmail.com>
References: <BANLkTinAbN-3bCNodmO4_01-h1mg9MA20w@mail.gmail.com>
	<BANLkTikdHVCTGVM7XDG-K3Z_YNN-02vo4w@mail.gmail.com>
Message-ID: <BANLkTinxoqxOHZRZ8XZyeAoyb1Kb0-7oXA@mail.gmail.com>

5.6 has 1000HZ set and so do 5.4 and 5.5 on my machines.

On Fri, May 6, 2011 at 3:30 PM, Brett Maxfield <
brett.maxfield+freeswitch at gmail.com> wrote:

> Hi List,
>
> I have been looking for a small SOHO software switch to manage one or two
> phone extensions to initially one upstream SIP provider.
>
> I have installed the demo/example system on a few pieces of hardware which
> while are not high end, nevertheless are OK considering the very small
> capacity required for a SOHO system, and me wanting to reserve my better
> hardware for more high end loads. The embedded fanless system i have runs
> ok, but the voice for the test IVR is very choppy, and the same choppy
> result on a atom dual core net-top (both centos 5.6). I understand that
> tuning is a complex issue on a non-RT system such as Linux, and so
> FreeSWITCH is probably only tuned for high end systems. This choppiness,
> specifically on low end systems, is "mentioned" in previous list discussions
> as being a timer granularity issue with the kernel.
>
> I've had some googling around and it seems that the newest centos version
> supported by FreeSWITCH is CentOS 5.3, perhaps due to issues with timer
> resolution in newer kernels. The stock kernel for 5.4 and 5.5 does not
> appear to have a 1000Hz kernel, from what i can see, and xen stock kernels
> that i have seen have 250Hz, which is different again, and would also be a
> problem for FreeSWITCH running in a xen guest.
>
> So my question is, given the contentious issue this seems to have been in
> the past, should i even bother trying/using FreeSWITCH for these sorts of
> SOHO / low end systems ?
>
> Is there any plan to make FreeSWITCH tunable (in a non-default way) for low
> end SOHO systems, which might have a typical 3-4 (say 10 at max)
> connections, rather than being tuned for 100's or even 1000's of connections
> by default.
>
> From what i have seen i quite like FreeSWITCH, and would like to run it for
> both SOHO and learning purposes.
>
> I note that centos 5.6 does not explicitly mention CONFIG_HZ_* anymore in
> the kernel config (/boot/config-2.6.18-238.9.1.el5.centos.plus), but it does
> appear to have support for HPET :
>
> CONFIG_HPET_TIMER=y
> CONFIG_HPET_EMULATE_RTC=y
> CONFIG_TICK_DIVIDER=y
>
> There was talk of high resolution timer patches for kernel 2.6.17, and hpet
> timers seem to be present in the 2.6.18 kernel, at least in centos 5.6 above
> (i have not checked previous versions).
>
> Given the increasing availability of HPET timers in newer kernels, could
> features such as nanosleep() and HPET timers be used to avoid the problems
> with the kernel's timer granularity / HZ issues ?
>
> Cheers
> Brett
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri May  6 22:16:02 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 6 May 2011 20:16:02 +0200
Subject: [Freeswitch-users] FreeSWITCH and SOHO systems / HPET timers
In-Reply-To: <BANLkTinxoqxOHZRZ8XZyeAoyb1Kb0-7oXA@mail.gmail.com>
References: <BANLkTinAbN-3bCNodmO4_01-h1mg9MA20w@mail.gmail.com>
	<BANLkTikdHVCTGVM7XDG-K3Z_YNN-02vo4w@mail.gmail.com>
	<BANLkTinxoqxOHZRZ8XZyeAoyb1Kb0-7oXA@mail.gmail.com>
Message-ID: <BANLkTikwi4ya9=xFQOXW1cjJspkYHDMRzA@mail.gmail.com>

all centos 5 kernels are 1000hz with tick, only their xen version are
250hz with tick. If you look around you find a 1000hz kernel for
centos xen too

On Fri, May 6, 2011 at 7:40 PM, Ognjen Seslija <oseslija at gmail.com> wrote:
> 5.6 has 1000HZ set and so do 5.4 and 5.5 on my machines.
>
> On Fri, May 6, 2011 at 3:30 PM, Brett Maxfield
> <brett.maxfield+freeswitch at gmail.com> wrote:
>>
>> Hi List,
>>
>> I have been looking for a small SOHO software switch to manage one or two
>> phone extensions to initially one upstream SIP provider.
>>
>> I have installed the demo/example system on a few pieces of hardware which
>> while are not high end, nevertheless are OK considering the very small
>> capacity required for a SOHO system, and me wanting to reserve my better
>> hardware for more high end loads. The embedded fanless system i have runs
>> ok, but the voice for the test IVR is very choppy, and the same choppy
>> result on a atom dual core net-top (both centos 5.6). I understand that
>> tuning is a complex issue on a non-RT system such as Linux, and so
>> FreeSWITCH is probably only tuned for high end systems. This choppiness,
>> specifically on low end systems, is "mentioned" in previous list discussions
>> as being a timer granularity issue with the kernel.
>>
>> I've had some googling around and it seems that the newest centos version
>> supported by FreeSWITCH is CentOS 5.3, perhaps due to issues with timer
>> resolution in newer kernels. The stock kernel for 5.4 and 5.5 does not
>> appear to have a 1000Hz kernel, from what i can see, and xen stock kernels
>> that i have seen have 250Hz, which is different again, and would also be a
>> problem for FreeSWITCH running in a xen guest.
>>
>> So my question is, given the contentious issue this seems to have been in
>> the past, should i even bother trying/using FreeSWITCH for these sorts of
>> SOHO / low end systems ?
>>
>> Is there any plan to make FreeSWITCH tunable (in a non-default way) for
>> low end SOHO systems, which might have a typical 3-4 (say 10 at max)
>> connections, rather than being tuned for 100's or even 1000's of connections
>> by default.
>>
>> From what i have seen i quite like FreeSWITCH, and would like to run it
>> for both SOHO and learning purposes.
>>
>> I note that centos 5.6 does not explicitly mention CONFIG_HZ_* anymore in
>> the kernel config (/boot/config-2.6.18-238.9.1.el5.centos.plus), but it does
>> appear to have support for HPET :
>>
>> CONFIG_HPET_TIMER=y
>> CONFIG_HPET_EMULATE_RTC=y
>> CONFIG_TICK_DIVIDER=y
>>
>> There was talk of high resolution timer patches for kernel 2.6.17, and
>> hpet timers seem to be present in the 2.6.18 kernel, at least in centos 5.6
>> above (i have not checked previous versions).
>>
>> Given the increasing availability of HPET timers in newer kernels, could
>> features such as nanosleep() and HPET timers be used to avoid the problems
>> with the kernel's timer granularity / HZ issues ?
>>
>> Cheers
>> Brett
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 roger.castaldo at gmail.com  Fri May  6 22:19:26 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Fri, 6 May 2011 14:19:26 -0400
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <C9E99B8A.16DCC%krice@freeswitch.org>
References: <BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
	<C9E99B8A.16DCC%krice@freeswitch.org>
Message-ID: <BANLkTikNyhr=eBEJ+9CoKRziuLz9Gd3OSg@mail.gmail.com>

I have been running freeswitch on a test server (old p4 laptop) using a
custom made Slitaz live cd with no apparent issues.

On Fri, May 6, 2011 at 1:35 PM, Ken Rice <krice at freeswitch.org> wrote:

>  I?ve been running Centos5.6 for a bit now with not real issues to report
> with FS...
>
>
> On 5/6/11 12:13 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
> As it stands now, CentOS is still the "ol' reliable" of the distros and
> still continues to be boring and predictable - just what we like. :)  I'm
> about to migrate one of my test boxes to CentOS 5.6 and do some testing.
> I'll report back any issues.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From andrew.keil at askinteractive.net  Sat May  7 00:02:00 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Sat, 7 May 2011 06:02:00 +1000
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <4DC40C86.9000503@coppice.org>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>

Steve,

Thanks for your response.

Further clarification on my part:

1) 20,000 to 30,000 pages per day to be sent out.
2) It will be an e-mail to fax style gateway (not fax to e-mail since that would involve inbound faxes)
3) The reason I asked about e-mail to PDF is the initial comments from my client requested the ability to send PDFs and WORD documents (I guess from attachments to the original e-mail), I understand the format that gets faxed should be TIFF so I saw on the freeswitch wiki email2pdf mentioned.  Then ImageMagick can help get a PDF to TIFF.

Can I ask some more questions:

Q1) Based on your experience what would be the average time (in seconds) to send a single fax page (TIFF file) via Freeswitch & Sangoma TDM?  You can quote a TIFF file size to make it more accurate.  From there I should be able to do the math to calculate my Client's requirements better.
Q2) Running on CentOS and using mod_spandsp/Freeswitch & Sangoma TDM what percentage CPU usage would I expect to see if 30 concurrent faxes are being sent at the same time (ie. All channels of my E1 are faxing)?  (The hardware would be a new 1U rack server from a major hardware vendor)
Q3) What version of CentOS 5.x would you recommend?  Would the latest version 5.6 be fine?
Q4) From memory there used to be different fax quality modes on fax machines (STANDARD, FINE & SUPER FINE or something like that).  Is it possible to set the fax send quality from mod_spandsp (also can you provide an example)?  If this is the case could you also answer question (Q1) based on the different fax send quality modes.
Q5) From exisiting deployments of Freeswitch using mod_spandsp (& Sagoma TDM cards (although this is not critical)) what is the largest number of concurrent outbound faxes done on a single box that you know of?

I appreciate your feedback and experience.  It sounds like this will work fine with the mod_spandsp/Freeswitch & Sangoma TDM combination on CentOS.  

I will most likely go for two servers with at least 2 x E1s in each, that way I future proof it a little and add redundancy.  Plus my Client can start using Freeswitch for Voice related services also.  On my side I can also test faxes going out on the first server and via a cross-over cable I can terminate them on the second server (for testing) - much nicer.

Thanks again,

Andrew



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood
Sent: Saturday, 7 May 2011 12:58 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Major deployment of outbound FAX on latest version of Freeswitch question

On 05/06/2011 01:14 PM, Andrew Keil wrote:
>
> To Freeswitch developers,
>
> **
>
> *RE: MOD_SPANDSP*
>
> I have a client that wishes to send 20,000 to 30,000 outbound faxes 
> daily and has requested me to quote for a Freeswitch setup that will 
> support this.
>
> I notice the comment on the mod spandsp wiki page: "We are finishing 
> mod_spandsp. It requires full field testing now." Well this could be a 
> perfect opportunity J
>
> _I need to understand the following:_
>
> 1.Does the mod spandsp support the TDM cards (Sangoma) to send faxes 
> out over E1 (Euro ISDN)?
>
> 2.How many concurrent faxes out have been tested using mod_spandsp ? 
> Was this over SIP or over ISDN?
>
> 3.What is the preferred Operating System to use (since basically I 
> would prefer to use the same one as the main developer of mod_spandsp)?
>
> 4.Where do I source the latest *email2pdf* to test that out?
>
> That should be enough to kick this question off.
>
> Kind Regards,
>
> Andrew Keil
>
>
What is the average number of pages per FAX that you expect to send? 
Unless its quite big, this will be a fairly small scale system, so you won't need any fancy hardware.

You can send the FAXes over ISDN or SIP. The Sangoma cards will do a good job if you want to use ISDN E1s. If the average FAX is <=4 pages, you'll probably only need a single E1, and you won't need anything very exotic for the server, unless you are doing quite a lot of works besides sending the FAXes. If this is a FAX only system you can use an E1 card that does not have an echo canceller (EC) module. That saves money. If you will have voice traffic as well, you had better consider buying an EC equipped card.

Centos 5.x is the OS usually recommended for FreeSwitch, as it is what most of the core developers test on.

I'm not clear why you are looking for email2pdf.

Is this going to be some kind of small scale e-mail to FAX gateway? FAX to e-mail is a more common setup, but quite a few people run e-mail to FAX gateways, too.

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
 

__________ Information from ESET NOD32 Antivirus, version of virus signature database 6101 (20110506) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 


From raul at etellicom.com  Sat May  7 00:23:31 2011
From: raul at etellicom.com (Raul Fragoso)
Date: Fri, 06 May 2011 17:23:31 -0300
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
	<BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
Message-ID: <1304713411.4953.21.camel@raul-laptop>

FYI: I've been running FS with latest wanpipe drivers on CentOS 5.6
(x86_64) for about 2 weeks now in 3 servers with no issues whatsoever. I
can confirm that the FS RPM's are built successfully with that
combination as well.

Regards,

-- 
---------------------
Raul Fragoso
Software Engineer
eTellicom Pty Ltd
---------------------


On Fri, 2011-05-06 at 10:13 -0700, Michael Collins wrote:
> 
> 
> On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:
>         I'd personally like to know some opinions on distros, being
>         that I've been wanting to move FROM Ubuntu and TO CentOS for
>         stability reasons. Ubuntu's packages update so frequently I'm
>         afraid of newer libraries and such breaking FreeSwitch and/or
>         the way things operate.
> 
> 
> I don't know what's up with CentOS either, but I'm in no hurry to run
> away from 5.6 right now. That being said, we've been having reports of
> success with various distros. TJ (IRC: bougyman) has been having what
> he calls "brilliant success" with Arch Linux, even when running AMD
> hardware. ;)  I've personally used Debian Lenny for a while and I find
> it to be stable and predictable. Moc has been experimenting with
> Scientific Linux and he has also reported good success. 
> 
> 
> As it stands now, CentOS is still the "ol' reliable" of the distros
> and still continues to be boring and predictable - just what we
> like. :)  I'm about to migrate one of my test boxes to CentOS 5.6 and
> do some testing. I'll report back any issues. 
> 
> 
> -MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From norm at voicenetwork.ca  Sat May  7 01:14:05 2011
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Fri, 6 May 2011 17:14:05 -0400
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <1304713411.4953.21.camel@raul-laptop>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
	<BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
	<1304713411.4953.21.camel@raul-laptop>
Message-ID: <BANLkTi=F6faH5HYkeRwLi2Eg5vCfazFaUg@mail.gmail.com>

We have been running CentOS for about 3 years now without any issues.
 CentOS 5.6 has been
working great as well.

I am not planning on switching any time soon.

Norman Tomlins
Voice Network Inc.
http://www.VoiceNetwork.ca


From ckmonkey158 at yahoo.com  Sat May  7 03:07:19 2011
From: ckmonkey158 at yahoo.com (Chris Monkey)
Date: Fri, 6 May 2011 16:07:19 -0700 (PDT)
Subject: [Freeswitch-users] Skypopen: start_skype_clients script permissions
Message-ID: <254712.55188.qm@web59401.mail.ac4.yahoo.com>

First, Giovanni, Thank You for the new skypopen installer, it really makes things easy... and stable.

I am curious though, why must skype run as root? I'd prefer that it didn't, but I can't quite get it working when run as a normal user. I've given all rw permissions to /dev/dsp and the skype-clients-configuration-dir, which allows me to load mod_skypeopen and a skype call to ring in, but as soon as I answer, skypopen reports a hangup with normal clearing.

Am I missing a permission somewhere else, or is there a fundamental reason why skype must be root for this to work?

Thanks,

Chris
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From max.clark at gmail.com  Sat May  7 04:43:38 2011
From: max.clark at gmail.com (Max Clark)
Date: Fri, 6 May 2011 17:43:38 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
Message-ID: <BANLkTinNi78ekMenx4wr+R=O5FtVR3+zBA@mail.gmail.com>

Brad,

Out of curiosity is this on LTS?

-Max

On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:
> I'd personally like to know some opinions on distros, being that I've been
> wanting to move FROM Ubuntu and TO CentOS for stability reasons. Ubuntu's
> packages update so frequently I'm afraid of newer libraries and such
> breaking FreeSwitch and/or the way things operate.
>
> On Fri, May 6, 2011 at 7:25 AM, Max Clark <max.clark at gmail.com> wrote:
>>
>> Hello all,
>>
>> Recent developments (or absolute lack of) within the CentOS project
>> and its perceived long term viability has forced an internal
>> discussion to select a successor distribution. The most likely
>> candidates at this point are Ubuntu and its LTS releases for servers,
>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>> each are growing and the issue is complicated.
>>
>> With CentOS 5.x being the reference distro for FreeSWITCH development
>> I'm curious if this conversation has started among the FreeSWITCH
>> developers, and if it has, what is the project leaning to for a
>> successor distribution?
>>
>> Thanks,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From max.clark at gmail.com  Sat May  7 04:45:48 2011
From: max.clark at gmail.com (Max Clark)
Date: Fri, 6 May 2011 17:45:48 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
	<BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
Message-ID: <BANLkTimJLoB-za6efWxygJNk7q5jOpBM0w@mail.gmail.com>

Michael,

Any experience with RHEL 6.x builds yet?

-Max

On Fri, May 6, 2011 at 10:13 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:
>>
>> I'd personally like to know some opinions on distros, being that I've been
>> wanting to move FROM Ubuntu and TO CentOS for stability reasons. Ubuntu's
>> packages update so frequently I'm afraid of newer libraries and such
>> breaking FreeSwitch and/or the way things operate.
>
> I don't know what's up with CentOS either, but I'm in no hurry to run away
> from 5.6 right now. That being said, we've been having reports of success
> with various distros. TJ (IRC: bougyman) has been having what he calls
> "brilliant success" with Arch Linux, even when running AMD hardware. ;)
> ?I've personally used Debian Lenny for a while and I find it to be stable
> and predictable. Moc has been experimenting with Scientific Linux and he has
> also reported good success.
> As it stands now, CentOS is still the "ol' reliable" of the distros and
> still continues to be boring and predictable - just what we like. :) ?I'm
> about to migrate one of my test boxes to CentOS 5.6 and do some testing.
> I'll report back any issues.
> -MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brad at tritelcomm.com  Sat May  7 05:07:30 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Fri, 6 May 2011 18:07:30 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTinNi78ekMenx4wr+R=O5FtVR3+zBA@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
	<BANLkTinNi78ekMenx4wr+R=O5FtVR3+zBA@mail.gmail.com>
Message-ID: <BANLkTimy3ERJbf=1yMxXduUOa2dEkrE9yw@mail.gmail.com>

Yes, Ubuntu 10.04LTS

On Fri, May 6, 2011 at 5:43 PM, Max Clark <max.clark at gmail.com> wrote:

> Brad,
>
> Out of curiosity is this on LTS?
>
> -Max
>
> On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:
> > I'd personally like to know some opinions on distros, being that I've
> been
> > wanting to move FROM Ubuntu and TO CentOS for stability reasons. Ubuntu's
> > packages update so frequently I'm afraid of newer libraries and such
> > breaking FreeSwitch and/or the way things operate.
> >
> > On Fri, May 6, 2011 at 7:25 AM, Max Clark <max.clark at gmail.com> wrote:
> >>
> >> Hello all,
> >>
> >> Recent developments (or absolute lack of) within the CentOS project
> >> and its perceived long term viability has forced an internal
> >> discussion to select a successor distribution. The most likely
> >> candidates at this point are Ubuntu and its LTS releases for servers,
> >> and Scientific Linux with the new 6.x releases. The pro/con lists for
> >> each are growing and the issue is complicated.
> >>
> >> With CentOS 5.x being the reference distro for FreeSWITCH development
> >> I'm curious if this conversation has started among the FreeSWITCH
> >> developers, and if it has, what is the project leaning to for a
> >> successor distribution?
> >>
> >> Thanks,
> >> Max
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fvillarroel at yahoo.com  Sat May  7 06:00:28 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 6 May 2011 19:00:28 -0700 (PDT)
Subject: [Freeswitch-users] Gateways ACL Context
Message-ID: <310161.61604.qm@web34307.mail.mud.yahoo.com>

Dear All.

How i can change to a private context like "my_context" for inbound traffic from gateways authorized on ACL and how i can assign different private context for each gateway authorized in ACL, like:

 <list name="domains" default="deny">
      <node type="allow" cidr="190.XXX.XX.XXX/32"/> context foo
      <node type="allow" cidr="66.X.XXX.XX/32"/>    context foo1


 </list>

Regards.



From jcasale at activenetwerx.com  Sat May  7 07:07:01 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 7 May 2011 03:07:01 +0000
Subject: [Freeswitch-users] Routing based on alias
Message-ID: <assp.0108dc260c.92097A6A775D5147B1078E3F15430B92492D6E@prato.activenetwerx.local>

I assumed one could route a call to a ua via its alias in addition to
its actual domain, but that is not working for me. I have a sofia profile
setup as an ip in the subnet it services and its aliased to a more
descriptive name, a dialplan tries to bridge a call to 1000@<ipaddr>
which works as expected, but when it attempts to bridge to the
alias, 1000 at foo, it fails?

Thanks,
jlc


From fvillarroel at yahoo.com  Sat May  7 07:23:34 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 6 May 2011 20:23:34 -0700 (PDT)
Subject: [Freeswitch-users] mod_limit Error
Message-ID: <636535.38388.qm@web34305.mail.mud.yahoo.com>

Hi All.

I rebuild my FS box and now i am getting the following error when i load the module limit:

 2011-05-06 23:56:11.229308 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_limit.so
**/usr/local/freeswitch/mod/mod_limit.so: undefined symbol: switch_channel_get_variable**

How i can solve?

Fernando.


From brad at tritelcomm.com  Sat May  7 09:36:10 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Fri, 6 May 2011 22:36:10 -0700
Subject: [Freeswitch-users] Routing based on alias
In-Reply-To: <assp.0108dc260c.92097A6A775D5147B1078E3F15430B92492D6E@prato.activenetwerx.local>
References: <assp.0108dc260c.92097A6A775D5147B1078E3F15430B92492D6E@prato.activenetwerx.local>
Message-ID: <BANLkTinqd1UFskjS+ocKvozPU5g2xosouQ@mail.gmail.com>

I'm not exactly sure what you're trying to accomplish, but it rings bells to
me with something along the lines of a multi-tenant setup? Either way, have
you tried creating a context for which to assign to the extension and route
the inbound call through?

On Fri, May 6, 2011 at 8:07 PM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> I assumed one could route a call to a ua via its alias in addition to
> its actual domain, but that is not working for me. I have a sofia profile
> setup as an ip in the subnet it services and its aliased to a more
> descriptive name, a dialplan tries to bridge a call to 1000@<ipaddr>
> which works as expected, but when it attempts to bridge to the
> alias, 1000 at foo, it fails?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brad at tritelcomm.com  Sat May  7 10:04:16 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Fri, 6 May 2011 23:04:16 -0700
Subject: [Freeswitch-users] Gateways ACL Context
In-Reply-To: <310161.61604.qm@web34307.mail.mud.yahoo.com>
References: <310161.61604.qm@web34307.mail.mud.yahoo.com>
Message-ID: <BANLkTin+SUgCuWJOLwrBrH9kUQ8tpZ6weQ@mail.gmail.com>

Your ACLs just allow and disallow connections most commonly for
non-registering endpoints that Freeswitch communicates with.
In your external sip profile you can assign a context with which traffic can
be routed with using your inbound dialplan. You should define your context
within your dialplan profile. With this, you can either write your entire
dialplan or use includes to create a subdirectory that holds all the
specific settings instead of a more "global" context definition.

<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer>

On Fri, May 6, 2011 at 7:00 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear All.
>
> How i can change to a private context like "my_context" for inbound traffic
> from gateways authorized on ACL and how i can assign different private
> context for each gateway authorized in ACL, like:
>
>  <list name="domains" default="deny">
>      <node type="allow" cidr="190.XXX.XX.XXX/32"/> context foo
>      <node type="allow" cidr="66.X.XXX.XX/32"/>    context foo1
>
>
>  </list>
>
> 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 jpatten at co.brazos.tx.us  Sat May  7 10:18:26 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Sat, 7 May 2011 06:18:26 +0000
Subject: [Freeswitch-users] Add Call-Info header
Message-ID: <8C8A3D4965236A42BDFF1758727F049A339E0C@ITEX1.bc.local>

Hi there

I'm trying to add add a Call-Info parameter to perform an auto-answer for phones that are not directly registered to FreeSWITCH (they are registered on a sipXecs server). None of the normal methods of auto-answer seem to work so I am attempting to copy the method that sipXecs uses to force phones to auto-answer. Upon analysis of how sipXecs performs an auto-answer, I find the following information in the SIP headers:

Call-Info: <sip:7012 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>;answer-after=0

The key here is the answer-after=0 part. What I've done in my dialplan is to add the following (In this example 7003 is initiating a call to a conference bridge which is then outcalling 7001):

<action application="set" data="sip_h_Call-Info=answer-after=0"/>
<action application="conference_set_auto_outcall" data="sofia/custom_dialplan/7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/>

However in analysis of the SIP messages I never see any Call-Info header being set, though FreeSWITCH processes the entry:
EXECUTE sofia/custom_dialplan/7003 at sipxpbx.bc.local set(sip_h_Call-Info=answer-after=0)
2011-05-07 00:56:46.563506 [DEBUG] mod_dptools.c:1059 sofia/custom_dialplan/7003 at sipxpbx.bc.local SET [sip_h_Call-Info]=[answer-after=0]

However as you can see in the following SIP message, the header just isn't there:

   ------------------------------------------------------------------------
   INVITE sip:7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0
   Via: SIP/2.0/UDP 10.200.129.102;rport;branch=z9hG4bK7aF5c90Dmecce
   Max-Forwards: 70
   From: "FreeSWITCH" <sip:0000000000 at 10.200.129.102>;tag=D3Q2t45cZvcar
   To: <sip:7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>
   Call-ID: 308e7013-f311-122e-448b-463bb5d10d94
   CSeq: 12037743 INVITE
   Contact: <sip:mod_sofia at 10.200.129.102:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-bfd0ba9 2011-03-07 13-02-41 -0600
   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: 239
   X-FS-Support: update_display
   Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 10.200.129.102>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1304728354 1304728355 IN IP4 10.200.129.102
   s=FreeSWITCH
   c=IN IP4 10.200.129.102
   t=0 0
   m=audio 19452 RTP/AVP 9 0 8 98 10 101 13
   a=rtpmap:98 SPEEX/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------

Thanks for your help :-)

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From jpatten at co.brazos.tx.us  Sat May  7 10:58:24 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Sat, 7 May 2011 06:58:24 +0000
Subject: [Freeswitch-users] Add Call-Info header
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A339E0C@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A339E0C@ITEX1.bc.local>
Message-ID: <8C8A3D4965236A42BDFF1758727F049A339E80@ITEX1.bc.local>

Nevermind, this fixed it:

<action application="conference_set_auto_outcall" data="{alert_info=sipXpage}sofia/custom_dialplan/7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/>

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten [jpatten at co.brazos.tx.us]
Sent: Saturday, May 07, 2011 1:18 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Add Call-Info header

Hi there

I'm trying to add add a Call-Info parameter to perform an auto-answer for phones that are not directly registered to FreeSWITCH (they are registered on a sipXecs server). None of the normal methods of auto-answer seem to work so I am attempting to copy the method that sipXecs uses to force phones to auto-answer. Upon analysis of how sipXecs performs an auto-answer, I find the following information in the SIP headers:

Call-Info: <sip:7012 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>;answer-after=0

The key here is the answer-after=0 part. What I've done in my dialplan is to add the following (In this example 7003 is initiating a call to a conference bridge which is then outcalling 7001):

<action application="set" data="sip_h_Call-Info=answer-after=0"/>
<action application="conference_set_auto_outcall" data="sofia/custom_dialplan/7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/>

However in analysis of the SIP messages I never see any Call-Info header being set, though FreeSWITCH processes the entry:
EXECUTE sofia/custom_dialplan/7003 at sipxpbx.bc.local set(sip_h_Call-Info=answer-after=0)
2011-05-07 00:56:46.563506 [DEBUG] mod_dptools.c:1059 sofia/custom_dialplan/7003 at sipxpbx.bc.local SET [sip_h_Call-Info]=[answer-after=0]

However as you can see in the following SIP message, the header just isn't there:

   ------------------------------------------------------------------------
   INVITE sip:7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0
   Via: SIP/2.0/UDP 10.200.129.102;rport;branch=z9hG4bK7aF5c90Dmecce
   Max-Forwards: 70
   From: "FreeSWITCH" <sip:0000000000 at 10.200.129.102>;tag=D3Q2t45cZvcar
   To: <sip:7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>
   Call-ID: 308e7013-f311-122e-448b-463bb5d10d94
   CSeq: 12037743 INVITE
   Contact: <sip:mod_sofia at 10.200.129.102:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-bfd0ba9 2011-03-07 13-02-41 -0600
   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: 239
   X-FS-Support: update_display
   Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 10.200.129.102>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1304728354 1304728355 IN IP4 10.200.129.102
   s=FreeSWITCH
   c=IN IP4 10.200.129.102
   t=0 0
   m=audio 19452 RTP/AVP 9 0 8 98 10 101 13
   a=rtpmap:98 SPEEX/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------

Thanks for your help :-)

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From brad at tritelcomm.com  Sat May  7 11:10:12 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Sat, 7 May 2011 00:10:12 -0700
Subject: [Freeswitch-users] Add Call-Info header
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A339E80@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A339E0C@ITEX1.bc.local>
	<8C8A3D4965236A42BDFF1758727F049A339E80@ITEX1.bc.local>
Message-ID: <BANLkTi=VB03w9-4X2=AkhwTkOxgz-VMJUg@mail.gmail.com>

Sounds like a wiki edit is in order!

On Fri, May 6, 2011 at 11:58 PM, Josh M. Patten <jpatten at co.brazos.tx.us>wrote:

>  Nevermind, this fixed it:
>
> <action application="conference_set_auto_outcall"
> data="{alert_info=sipXpage}sofia/custom_dialplan/7001 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false"/>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten
> [jpatten at co.brazos.tx.us]
> *Sent:* Saturday, May 07, 2011 1:18 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Add Call-Info header
>
>   Hi there
>
> I'm trying to add add a Call-Info parameter to perform an auto-answer for
> phones that are not directly registered to FreeSWITCH (they are registered
> on a sipXecs server). None of the normal methods of auto-answer seem to work
> so I am attempting to copy the method that sipXecs uses to force phones to
> auto-answer. Upon analysis of how sipXecs performs an auto-answer, I find
> the following information in the SIP headers:
>
> Call-Info: <sip:7012 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false>;answer-after=0
>
> The key here is the answer-after=0 part. What I've done in my dialplan is
> to add the following (In this example 7003 is initiating a call to a
> conference bridge which is then outcalling 7001):
>
> <action application="set" data="sip_h_Call-Info=answer-after=0"/>
> <action application="conference_set_auto_outcall"
> data="sofia/custom_dialplan/7001 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false"/>
>
> However in analysis of the SIP messages I never see any Call-Info header
> being set, though FreeSWITCH processes the entry:
> EXECUTE sofia/custom_dialplan/7003 at sipxpbx.bc.localset(sip_h_Call-Info=answer-after=0)
> 2011-05-07 00:56:46.563506 [DEBUG] mod_dptools.c:1059
> sofia/custom_dialplan/7003 at sipxpbx.bc.local SET
> [sip_h_Call-Info]=[answer-after=0]
>
> However as you can see in the following SIP message, the header just isn't
> there:
>
>    ------------------------------------------------------------------------
>    INVITE sip:7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false
> SIP/2.0
>    Via: SIP/2.0/UDP 10.200.129.102;rport;branch=z9hG4bK7aF5c90Dmecce
>    Max-Forwards: 70
>    From: "FreeSWITCH" <sip:0000000000 at 10.200.129.102>;tag=D3Q2t45cZvcar
>    To: <sip:7001 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false>
>    Call-ID: 308e7013-f311-122e-448b-463bb5d10d94
>    CSeq: 12037743 INVITE
>    Contact: <sip:mod_sofia at 10.200.129.102:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-bfd0ba9 2011-03-07
> 13-02-41 -0600
>    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: 239
>    X-FS-Support: update_display
>    Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 10.200.129.102
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1304728354 1304728355 IN IP4 10.200.129.102
>    s=FreeSWITCH
>    c=IN IP4 10.200.129.102
>    t=0 0
>    m=audio 19452 RTP/AVP 9 0 8 98 10 101 13
>    a=rtpmap:98 SPEEX/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
>
> Thanks for your help :-)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jpatten at co.brazos.tx.us  Sat May  7 11:22:39 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Sat, 7 May 2011 07:22:39 +0000
Subject: [Freeswitch-users] Slow dialplan execution
Message-ID: <8C8A3D4965236A42BDFF1758727F049A339E98@ITEX1.bc.local>

I seem to be having a big problem when using the "Mad-Boss" Paging setup described in: http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom

To connect 5 phones to the conference bridge takes anywhere from 3 - 4 seconds which is entirely too long to be useful. Here is the dialplan entry:

<extension name="Test Paging Group">
      <condition field="destination_number" expression="^(3400)$" break="never">
        <action application="set" data="api_hangup_hook=conference 3400 kick all"/>
        <action application="answer"/>
        <action application="export" data="sip_invite_params=intercom=true"/>
        <action application="export" data="sip_auto_answer=true"/>
        <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
        <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
        <action application="set" data="conference_auto_outcall_timeout=60"/>
        <action application="set" data="conference_auto_outcall_flags=none"/>
        <action application="conference_set_auto_outcall" data="{alert_info=sipXpage}sofia/custom_dialplan/7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7002 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7004 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7005 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7006 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/>
      </condition>
      <condition field="destination_number" expression="^(3400)$">
        <action application="conference" data="3400 at default"/>
        <action application="conference" data="3400 kick all"/>
      </condition>
    </extension>

Here are my questions:

Is it possible to speed up XML dialplan execution to overcome this issue? I promise I have enough CPU, RAM, and bandwidth to handle the flurry of SIP messages.

If it is not possible to speed up the XML dialplan execution is there a way to "thread" the dialplan execution so that all of these conference_set_auto_outcall applications can be run simultaneously?

If not, would attempting to use a database based dialplan be worth my time?

Because of the complexity of the networks I'm dealing with I'd rather not try to fight with multicast.

Thanks!!
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From fieldpeak at gmail.com  Sat May  7 12:23:42 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 7 May 2011 16:23:42 +0800
Subject: [Freeswitch-users] FS as outbound conference server
In-Reply-To: <DC5E5E9C-FEE7-4A55-940C-6AEE0FF2E9F7@ipeva.fr>
References: <BANLkTi=8zUG2Y=GD8UkfVupVyO__ngN=Dg@mail.gmail.com>
	<DC5E5E9C-FEE7-4A55-940C-6AEE0FF2E9F7@ipeva.fr>
Message-ID: <BANLkTimwaWy=vcoQ-VDgizX0KngrQMxW2A@mail.gmail.com>

Hi David,

Thanks for your valuable info, it help much!

Regards,
Charles

2011/5/5 David Ponzone <david.ponzone at ipeva.fr>

> Charles,
>
> check the wiki for info on PHP ESL.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 05/05/2011 ? 06:24, fieldpeak a ?crit :
>
> Hi Friends,
>
> i need configure FS as  outbound conference server, and control FS to
> realize it by php webpages and DTMF.
> if there any open source php webpages will be much better, thanks for any
> hints.
>
> 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
>
>
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From jaybinks at gmail.com  Sat May  7 12:26:02 2011
From: jaybinks at gmail.com (jay binks)
Date: Sat, 7 May 2011 18:26:02 +1000
Subject: [Freeswitch-users] can I overwrite channel var with response
	headers ?
Message-ID: <BANLkTi=+duWX7PMsgs0jxLd3_A4mfp-7zQ@mail.gmail.com>

I have 2 FS boxes ... BoxA & BoxB
BoxA invites to BoxB and I want BoxB to respond with some information to
place in the channel vars on BoxA ( to store in the CDR there )

ive found I can use sip_ph_X-blah & sip_rh_X-blah to pass the data back ( in
different messages obviously )
however neither of these overwrite the "blah" channel variable..

is there a way to tell FS to overwrite channel var "blah" with these sip
responses ?

-- 
Sincerely

Jay
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From devel at omninet.eu  Sat May  7 13:52:55 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Sat, 7 May 2011 12:52:55 +0300
Subject: [Freeswitch-users] many outgoing calls with mod_lcr at the same
	time possible?
Message-ID: <E6513A570B0E4F47A248C4E0735EAD99@omni1.local>

Hello Freeswitchers,

 

Is it possible to make many outgoing calls with mod_lcr at the same time? An
example would be to redirect an incoming call to two mobile phones and ring
them simultaneously.

With a "normal" bridge we can achieve this, but what about having also LCR
(billing and limit included) involved?

 

Regards

Anestis

 

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From jcasale at activenetwerx.com  Sat May  7 16:36:06 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 7 May 2011 12:36:06 +0000
Subject: [Freeswitch-users] Routing based on alias
In-Reply-To: <BANLkTinqd1UFskjS+ocKvozPU5g2xosouQ@mail.gmail.com>
References: <92097A6A775D5147B1078E3F15430B92492D6E@prato.activenetwerx.local>
	<BANLkTinqd1UFskjS+ocKvozPU5g2xosouQ@mail.gmail.com>
Message-ID: <assp.0108a00ca0.92097A6A775D5147B1078E3F15430B924B1764@prato.activenetwerx.local>

>I'm not exactly sure what you're trying to accomplish, but it rings bells to me with something
>along the lines of a multi-tenant setup? Either way, have you tried creating a context for
>which to assign to the extension and route the inbound call through?

Hi,
Yes it is multi-tenant, and there is specific contexts for each profile. Each profile has its own
interface, and the phones register to the ip address of that interface, probably I have done this
wrong.

The sip profiles look like:

<profile name="foo.local">
  <aliases>
    <alias name="foo.local"/>
  </aliases>
  <domains>
    <domain name="10.1.4.1" alias="true" parse="false"/> 
  </domains>

The directory is set to the ip.

So when a dialplan attempts 1000 at 10.1.4.1 it routes correct, but if a dialplan routes to
1000 at foo.local it fails? A 'sofia status' at the cli shows:

freeswitch at internal> sofia status
                     Name          Type                                       Data      State
=================================================================================================
                 external       profile               sip:mod_sofia at 10.0.0.13:5080      RUNNING (0)
external::foo.local-outbound     gateway          sip:xxx5551212 at sip.provider.com      REGED
      bar.local       profile                sip:mod_sofia at 10.1.3.1:5060      RUNNING (0)
         foo.local       profile                sip:mod_sofia at 10.1.4.1:5060      RUNNING (0)
                 10.1.4.1         alias                           foo.local      ALIASED
                 10.1.3.1         alias                        bar.local      ALIASED
=================================================================================================
3 profiles 2 aliases

Thanks!
jlc


From Nabble at slickdeals.endjunk.com  Sat May  7 18:12:51 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 7 May 2011 07:12:51 -0700 (PDT)
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
Message-ID: <1304777571644-6340156.post@n2.nabble.com>

I wondered if this  http://developer.skype.com/public/skypekit SkypeKit  will
help making mod_skypopen more useable. Has any FS developers here gotten a
chance to take a look at  http://developer.skype.com/public/skypekit
SkypeKit  and tries to incorporate into FS for mod_skypopen to replace the
need for a Skype client?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SkypeKit-for-mo-skyopen-tp6340156p6340156.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Sat May  7 19:02:57 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 7 May 2011 08:02:57 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTimJLoB-za6efWxygJNk7q5jOpBM0w@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
	<BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
	<BANLkTimJLoB-za6efWxygJNk7q5jOpBM0w@mail.gmail.com>
Message-ID: <BANLkTinjUtGQ=aLaq5Uq1SX9e8P1s+sPrA@mail.gmail.com>

Yes, I am running Scientific Linux 6 here. No noticeable difference in
terms with running it on a Fedora 12/13 box. Runs pretty rock solid,
really.

On Fri, May 6, 2011 at 5:45 PM, Max Clark <max.clark at gmail.com> wrote:
> Michael,
>
> Any experience with RHEL 6.x builds yet?
>
> -Max
>
> On Fri, May 6, 2011 at 10:13 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>> On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:
>>>
>>> I'd personally like to know some opinions on distros, being that I've been
>>> wanting to move FROM Ubuntu and TO CentOS for stability reasons. Ubuntu's
>>> packages update so frequently I'm afraid of newer libraries and such
>>> breaking FreeSwitch and/or the way things operate.
>>
>> I don't know what's up with CentOS either, but I'm in no hurry to run away
>> from 5.6 right now. That being said, we've been having reports of success
>> with various distros. TJ (IRC: bougyman) has been having what he calls
>> "brilliant success" with Arch Linux, even when running AMD hardware. ;)
>> ?I've personally used Debian Lenny for a while and I find it to be stable
>> and predictable. Moc has been experimenting with Scientific Linux and he has
>> also reported good success.
>> As it stands now, CentOS is still the "ol' reliable" of the distros and
>> still continues to be boring and predictable - just what we like. :) ?I'm
>> about to migrate one of my test boxes to CentOS 5.6 and do some testing.
>> I'll report back any issues.
>> -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 fieldpeak at gmail.com  Sat May  7 19:04:55 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 7 May 2011 23:04:55 +0800
Subject: [Freeswitch-users] How to manipulate destination number before
	routing to PSTN GW
Message-ID: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>

Gurus,


i met an issue for dial plan, it sounds easy but puzzled me a few days not
fix it yet...i belive gurus here could help me...


 i want to remove the + of destination number before routing to PSTN GW,
e.g. when i dial +9123, i would like FS remove +, and then route to the PSTN
GW, below is the dial plan and log,

See the log, FS did remove '+' of '+9123', but then when it checks the rule
?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then failed.

Thanks.



*Dial plan:*

<extension name="remove_plus_of_dst_num" continue="true">

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

       <action application="set" data="effective_destination_number=$1"/>

   </condition>

</extension>

?...

<extension name="9_to_GW" >

   <condition field="destination_number" expression="^(9\d+)$">

       <action application="bridge" data="sofia/internal/$1 at 192.168.200.101
"/>

   </condition>

</extension>



*Log:*

freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
192.168.200.201

 Approved by acl "192.168.0.0/16[]". Access Granted.

2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
sofia/inter

nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]

2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
sofia/internal/+4001 at 192

.168.200.100 entering state [received][100]

2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:

v=0

o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201

s=Phone-Call

c=IN IP4 192.168.200.201

t=0 0

m=audio 6060 RTP/AVP 8 0 96

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

a=ptime:20



2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
(sofia/internal/+4001 at 192.168.20

0.100) State Change CS_NEW -> CS_INIT

2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send signal
sofia/

internal/+4001 at 192.168.200.100 [BREAK]

2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
(sofia/intern

al/+4001 at 192.168.200.100) Running State Change CS_INIT

2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
(sofia/intern

al/+4001 at 192.168.200.100) State INIT

2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
sofia/internal/+4001 at 192.168.2

00.100 SOFIA INIT

2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
(sofia/internal/+4001 at 192.168

.200.100) State Change CS_INIT -> CS_ROUTING

2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send signal
sofia/

internal/+4001 at 192.168.200.100 [BREAK]

2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
(sofia/intern

al/+4001 at 192.168.200.100) State INIT going to sleep

2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
(sofia/intern

al/+4001 at 192.168.200.100) Running State Change CS_ROUTING

2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
(sofia/internal/+4001 at 1

92.168.200.100) Callstate Change DOWN -> RINGING

2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
(sofia/intern

al/+4001 at 192.168.200.100) State ROUTING

2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
sofia/internal/+4001 at 192.168.

200.100 SOFIA ROUTING

2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
sofia/internal

/+4001 at 192.168.200.100 Standard ROUTING

2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing +4001
<+4001

>->+9123 in context default

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->unloop]
continu

e=false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
${unroll_lo

ops}(true) =~ /^true$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
${sip_loope

d_call}() =~ /^true$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
[default->remove_plus_of_

dst_num] continue=true

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
[remove_plus_of_dst_

num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
set(destination_number=912

3)

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
[default->remove_plus_of_

src_num] continue=true

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
[remove_plus_of_src_

num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
set(effective_caller_id_na

me=4001)

Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
set(effective_caller_id_nu

mber=4001)

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
[default->7_8_to_Lync] co

ntinue=false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
destin

ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->9_to_GW]
contin

ue=false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
destinatio

n_number(+9123) =~ /^(9\d+)$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->1_to_IPP]
conti

nue=false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
destinati

on_number(+9123) =~ /^(1\d{3})$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA]
continue=

false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
destination_n

umber(+9123) =~ /^\*(3472)$/ break=on-false

Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->Recordings]
con

tinue=false

Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [Recordings]
destina

tion_number(+9123) =~ /^\*(732673)$/ break=on-false

2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
(sofia/intern

al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE

2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send signal
sofia/

internal/+4001 at 192.168.200.100 [BREAK]

2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
(sofia/intern

al/+4001 at 192.168.200.100) State ROUTING going to sleep

2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
(sofia/intern

al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE

2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
(sofia/intern

al/+4001 at 192.168.200.100) State EXECUTE

2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
sofia/internal/+4001 at 192.168.

200.100 SOFIA EXECUTE

2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
sofia/interna

l/+4001 at 192.168.200.100 Standard EXECUTE

EXECUTE sofia/internal/+4001 at 192.168.200.100 set(destination_number=9123)

2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
sofia/internal/+4001 at 192.1

68.200.100 SET [destination_number]=[9123]

EXECUTE sofia/internal/+4001 at 192.168.200.100set(effective_caller_id_name=4001)

2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
sofia/internal/+4001 at 192.1

68.200.100 SET [effective_caller_id_name]=[4001]

EXECUTE sofia/internal/+4001 at 192.168.200.100set(effective_caller_id_number=4001

)

2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
sofia/internal/+4001 at 192.1

68.200.100 SET [effective_caller_id_number]=[4001]

2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
sofia/intern

al/+4001 at 192.168.200.100 has executed the last dialplan instruction, hanging
up.



2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
(sofia/internal/+4001 at 1

92.168.200.100) Callstate Change RINGING -> HANGUP

2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia

/internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]

2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
sofia/inter

nal/+4001 at 192.168.200.100 [KILL]

2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send signal
sofia/

internal/+4001 at 192.168.200.100 [BREAK]

2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
(sofia/intern

al/+4001 at 192.168.200.100) State EXECUTE going to sleep

2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
(sofia/intern

al/+4001 at 192.168.200.100) Running State Change CS_HANGUP

2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
(sofia/intern

al/+4001 at 192.168.200.100) State HANGUP

2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
sofia/internal/+4001@

192.168.200.100 hanging up, cause: NORMAL_CLEARING

2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE
with: 48

0

2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
sofia/internal

/+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING

2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
(sofia/intern

al/+4001 at 192.168.200.100) State HANGUP going to sleep

2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
(sofia/intern

al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING

2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send signal
sofia/

internal/+4001 at 192.168.200.100 [BREAK]

2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
(sofia/intern

al/+4001 at 192.168.200.100) Running State Change CS_REPORTING

2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
(sofia/intern

al/+4001 at 192.168.200.100) State REPORTING

2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
sofia/internal

/+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING

2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
(sofia/intern

al/+4001 at 192.168.200.100) State REPORTING going to sleep

2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
(sofia/intern

al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY

2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send signal
sofia/

internal/+4001 at 192.168.200.100 [BREAK]

2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2
(sofia/i

nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities

2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session 2
(sofia/

internal/+4001 at 192.168.200.100) Ended

2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close Channel
sof

ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]

2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
(sofia/intern

al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN

2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
(sofia/intern

al/+4001 at 192.168.200.100) Running State Change CS_DESTROY

2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
(sofia/intern

al/+4001 at 192.168.200.100) State DESTROY

2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
sofia/internal/+4001 at 192.168.

200.100 SOFIA DESTROY

2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
sofia/internal

/+4001 at 192.168.200.100 Standard DESTROY

2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
(sofia/intern

al/+4001 at 192.168.200.100) State DESTROY going to sleep
-------------- next part --------------
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From infos at madovsky.org  Sat May  7 19:15:58 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 7 May 2011 11:15:58 -0400
Subject: [Freeswitch-users] How to manipulate destination number
	beforerouting to PSTN GW
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
Message-ID: <097FACC852C04BEB8F518E754B64EB15@e1705>

use expression="(+)(\d)"
and $2 as result
  ----- Original Message ----- 
  From: fieldpeak 
  To: FreeSWITCH-users 
  Sent: Saturday, May 07, 2011 11:04 AM
  Subject: [Freeswitch-users] How to manipulate destination number beforerouting to PSTN GW


  Gurus,




  i met an issue for dial plan, it sounds easy but puzzled me a few days not fix it yet...i belive gurus here could help me...





  i want to remove the + of destination number before routing to PSTN GW, e.g. when i dial +9123, i would like FS remove +, and then route to the PSTN GW, below is the dial plan and log, 

  See the log, FS did remove '+' of '+9123', but then when it checks the rule ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then failed.

  Thanks.



  Dial plan:

  <extension name="remove_plus_of_dst_num" continue="true">

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

         <action application="set" data="effective_destination_number=$1"/>

     </condition>

  </extension>

  ?...

  <extension name="9_to_GW" >

     <condition field="destination_number" expression="^(9\d+)$">

         <action application="bridge" data="sofia/internal/$1 at 192.168.200.101"/>

     </condition>

  </extension>



  Log:

  freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP 192.168.200.201

   Approved by acl "192.168.0.0/16[]". Access Granted.

  2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel sofia/inter

  nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]

  2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel sofia/internal/+4001 at 192

  .168.200.100 entering state [received][100]

  2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:

  v=0

  o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201

  s=Phone-Call

  c=IN IP4 192.168.200.201

  t=0 0

  m=audio 6060 RTP/AVP 8 0 96

  a=rtpmap:8 PCMA/8000

  a=rtpmap:0 PCMU/8000

  a=rtpmap:96 telephone-event/8000

  a=fmtp:96 0-15

  a=ptime:20



  2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908 (sofia/internal/+4001 at 192.168.20

  0.100) State Change CS_NEW -> CS_INIT

  2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send signal sofia/

  internal/+4001 at 192.168.200.100 [BREAK]

  2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325 (sofia/intern

  al/+4001 at 192.168.200.100) Running State Change CS_INIT

  2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361 (sofia/intern

  al/+4001 at 192.168.200.100) State INIT

  2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84 sofia/internal/+4001 at 192.168.2

  00.100 SOFIA INIT

  2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124 (sofia/internal/+4001 at 192.168

  .200.100) State Change CS_INIT -> CS_ROUTING

  2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send signal sofia/

  internal/+4001 at 192.168.200.100 [BREAK]

  2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361 (sofia/intern

  al/+4001 at 192.168.200.100) State INIT going to sleep

  2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325 (sofia/intern

  al/+4001 at 192.168.200.100) Running State Change CS_ROUTING

  2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668 (sofia/internal/+4001 at 1

  92.168.200.100) Callstate Change DOWN -> RINGING

  2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364 (sofia/intern

  al/+4001 at 192.168.200.100) State ROUTING

  2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147 sofia/internal/+4001 at 192.168.

  200.100 SOFIA ROUTING

  2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77 sofia/internal

  /+4001 at 192.168.200.100 Standard ROUTING

  2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing +4001 <+4001

  >->+9123 in context default

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->unloop] continu

  e=false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop] ${unroll_lo

  ops}(true) =~ /^true$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop] ${sip_loope

  d_call}() =~ /^true$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->remove_plus_of_

  dst_num] continue=true

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [remove_plus_of_dst_

  num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Action set(destination_number=912

  3)

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->remove_plus_of_

  src_num] continue=true

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [remove_plus_of_src_

  num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Action set(effective_caller_id_na

  me=4001)

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Action set(effective_caller_id_nu

  mber=4001)

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->7_8_to_Lync] co

  ntinue=false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync] destin

  ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->9_to_GW] contin

  ue=false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW] destinatio

  n_number(+9123) =~ /^(9\d+)$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->1_to_IPP] conti

  nue=false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP] destinati

  on_number(+9123) =~ /^(1\d{3})$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA] continue=

  false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA] destination_n

  umber(+9123) =~ /^\*(3472)$/ break=on-false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->Recordings] con

  tinue=false

  Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [Recordings] destina

  tion_number(+9123) =~ /^\*(732673)$/ break=on-false

  2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119 (sofia/intern

  al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE

  2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send signal sofia/

  internal/+4001 at 192.168.200.100 [BREAK]

  2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364 (sofia/intern

  al/+4001 at 192.168.200.100) State ROUTING going to sleep

  2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325 (sofia/intern

  al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE

  2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371 (sofia/intern

  al/+4001 at 192.168.200.100) State EXECUTE

  2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240 sofia/internal/+4001 at 192.168.

  200.100 SOFIA EXECUTE

  2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157 sofia/interna

  l/+4001 at 192.168.200.100 Standard EXECUTE

  EXECUTE sofia/internal/+4001 at 192.168.200.100 set(destination_number=9123)

  2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060 sofia/internal/+4001 at 192.1

  68.200.100 SET [destination_number]=[9123]

  EXECUTE sofia/internal/+4001 at 192.168.200.100 set(effective_caller_id_name=4001)

  2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060 sofia/internal/+4001 at 192.1

  68.200.100 SET [effective_caller_id_name]=[4001]

  EXECUTE sofia/internal/+4001 at 192.168.200.100 set(effective_caller_id_number=4001

  )

  2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060 sofia/internal/+4001 at 192.1

  68.200.100 SET [effective_caller_id_number]=[4001]

  2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189 sofia/intern

  al/+4001 at 192.168.200.100 has executed the last dialplan instruction, hanging up.



  2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563 (sofia/internal/+4001 at 1

  92.168.200.100) Callstate Change RINGING -> HANGUP

  2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191 Hangup sofia

  /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]

  2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal sofia/inter

  nal/+4001 at 192.168.200.100 [KILL]

  2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send signal sofia/

  internal/+4001 at 192.168.200.100 [BREAK]

  2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371 (sofia/intern

  al/+4001 at 192.168.200.100) State EXECUTE going to sleep

  2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325 (sofia/intern

  al/+4001 at 192.168.200.100) Running State Change CS_HANGUP

  2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565 (sofia/intern

  al/+4001 at 192.168.200.100) State HANGUP

  2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel sofia/internal/+4001@

  192.168.200.100 hanging up, cause: NORMAL_CLEARING

  2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 48

  0

  2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46 sofia/internal

  /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING

  2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565 (sofia/intern

  al/+4001 at 192.168.200.100) State HANGUP going to sleep

  2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356 (sofia/intern

  al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING

  2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send signal sofia/

  internal/+4001 at 192.168.200.100 [BREAK]

  2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325 (sofia/intern

  al/+4001 at 192.168.200.100) Running State Change CS_REPORTING

  2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625 (sofia/intern

  al/+4001 at 192.168.200.100) State REPORTING

  2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53 sofia/internal

  /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING

  2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625 (sofia/intern

  al/+4001 at 192.168.200.100) State REPORTING going to sleep

  2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350 (sofia/intern

  al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY

  2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send signal sofia/

  internal/+4001 at 192.168.200.100 [BREAK]

  2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2 (sofia/i

  nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities

  2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session 2 (sofia/

  internal/+4001 at 192.168.200.100) Ended

  2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close Channel sof

  ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]

  2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454 (sofia/intern

  al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN

  2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457 (sofia/intern

  al/+4001 at 192.168.200.100) Running State Change CS_DESTROY

  2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467 (sofia/intern

  al/+4001 at 192.168.200.100) State DESTROY

  2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362 sofia/internal/+4001 at 192.168.

  200.100 SOFIA DESTROY

  2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60 sofia/internal

  /+4001 at 192.168.200.100 Standard DESTROY

  2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467 (sofia/intern

  al/+4001 at 192.168.200.100) State DESTROY going to sleep



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


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From infos at madovsky.org  Sat May  7 19:23:36 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 7 May 2011 11:23:36 -0400
Subject: [Freeswitch-users] Moving on from CentOS
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com><BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com><BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com><BANLkTimJLoB-za6efWxygJNk7q5jOpBM0w@mail.gmail.com>
	<BANLkTinjUtGQ=aLaq5Uq1SX9e8P1s+sPrA@mail.gmail.com>
Message-ID: <E11D14B90F4C4F8C9269544024CC434B@e1705>

same on Fedora10 64bits

----- Original Message ----- 
From: "curriegrad2004" <curriegrad2004 at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 07, 2011 11:02 AM
Subject: Re: [Freeswitch-users] Moving on from CentOS


> Yes, I am running Scientific Linux 6 here. No noticeable difference in
> terms with running it on a Fedora 12/13 box. Runs pretty rock solid,
> really.
>
> On Fri, May 6, 2011 at 5:45 PM, Max Clark <max.clark at gmail.com> wrote:
>> Michael,
>>
>> Any experience with RHEL 6.x builds yet?
>>
>> -Max
>>
>> On Fri, May 6, 2011 at 10:13 AM, Michael Collins <msc at freeswitch.org> 
>> wrote:
>>>
>>>
>>> On Fri, May 6, 2011 at 9:35 AM, Brad Mina <brad at tritelcomm.com> wrote:
>>>>
>>>> I'd personally like to know some opinions on distros, being that I've 
>>>> been
>>>> wanting to move FROM Ubuntu and TO CentOS for stability reasons. 
>>>> Ubuntu's
>>>> packages update so frequently I'm afraid of newer libraries and such
>>>> breaking FreeSwitch and/or the way things operate.
>>>
>>> I don't know what's up with CentOS either, but I'm in no hurry to run 
>>> away
>>> from 5.6 right now. That being said, we've been having reports of 
>>> success
>>> with various distros. TJ (IRC: bougyman) has been having what he calls
>>> "brilliant success" with Arch Linux, even when running AMD hardware. ;)
>>> I've personally used Debian Lenny for a while and I find it to be stable
>>> and predictable. Moc has been experimenting with Scientific Linux and he 
>>> has
>>> also reported good success.
>>> As it stands now, CentOS is still the "ol' reliable" of the distros and
>>> still continues to be boring and predictable - just what we like. :) I'm
>>> about to migrate one of my test boxes to CentOS 5.6 and do some testing.
>>> I'll report back any issues.
>>> -MC
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From krice at freeswitch.org  Sat May  7 19:23:42 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 07 May 2011 10:23:42 -0500
Subject: [Freeswitch-users] How to manipulate destination number
 beforerouting to PSTN GW
In-Reply-To: <097FACC852C04BEB8F518E754B64EB15@e1705>
Message-ID: <C9EACE2E.16E20%krice@freeswitch.org>

His originatl expression should work fine since the + is not in the () in
his regex, however set_effective_destination number does not over write the
destination_number field you need to transfer the call to get
destination_number reset
Or use the ${effective_destination_number} as your field in the 9 to gw
extension


On 5/7/11 10:15 AM, "Madovsky" <infos at madovsky.org> wrote:

> use expression="(+)(\d)"
> and $2 as result
>>  
>> ----- Original Message -----
>>  
>> From:  fieldpeak <mailto:fieldpeak at gmail.com>
>>  
>> To: FreeSWITCH-users <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>  
>> Sent: Saturday, May 07, 2011 11:04  AM
>>  
>> Subject: [Freeswitch-users] How to  manipulate destination number
>> beforerouting to PSTN GW
>>  
>> 
>>  
>> 
>> Gurus,
>>  
>> 
>>  
>> i met an issue for dial plan, it sounds easy but puzzled me  a few days not
>> fix it yet...i belive gurus here could help me...
>>  
>> 
>>  
>> i want to remove the + of destination number before routing  to PSTN GW, e.g.
>> when i dial +9123, i would like FS remove +, and then route  to the PSTN GW,
>> below is the dial plan and log,
>>  
>> See the log, FS did remove '+' of '+9123', but then when it  checks the rule
>> ?9_to_GW?, it still checks '+9123' (I expect it is '9123'),  and then failed.
>>  
>> Thanks.
>>  
>>  
>>  
>> Dial plan:
>>  
>> <extension name="remove_plus_of_dst_num"  continue="true">
>>  
>>     <condition field="destination_number" expression="^\+(\d+)$">
>>  
>>         <action application="set"  data="effective_destination_number=$1"/>
>>  
>>     </condition>
>>  
>> </extension>
>>  
>> ?...
>>  
>> <extension name="9_to_GW" >
>>  
>>     <condition field="destination_number" expression="^(9\d+)$">
>>  
>>         <action application="bridge"
>> data="sofia/internal/$1 at 192.168.200.101"/>
>>  
>>     </condition>
>>  
>> </extension>
>>  
>>  
>>  
>> Log:
>>  
>> freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG]  sofia.c:6488 IP
>> 192.168.200.201
>>  
>>  Approved by  acl "192.168.0.0/16[] <http://192.168.0.0/16[]> ". Access
>> Granted.
>>  
>> 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812  New Channel
>> sofia/inter
>>  
>> nal/+4001 at 192.168.200.100  [92568c57-ff7b-4dbc-b322-a9babf893e62]
>>  
>> 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
>> sofia/internal/+4001 at 192
>>  
>> .168.200.100 entering state [received][100]
>>  
>> 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote  SDP:
>>  
>> v=0
>>  
>> o=AudiocodesGW 1062502113 1062501981 IN IP4  192.168.200.201
>>  
>> s=Phone-Call
>>  
>> c=IN IP4 192.168.200.201
>>  
>> t=0 0
>>  
>> m=audio 6060 RTP/AVP 8 0 96
>>  
>> a=rtpmap:8 PCMA/8000
>>  
>> a=rtpmap:0 PCMU/8000
>>  
>> a=rtpmap:96 telephone-event/8000
>>  
>> a=fmtp:96 0-15
>>  
>> a=ptime:20
>>  
>>  
>>  
>> 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
>> (sofia/internal/+4001 at 192.168.20
>>  
>> 0.100) State Change CS_NEW -> CS_INIT
>>  
>> 2011-05-07 22:48:05.719537 [DEBUG]  switch_core_session.c:1116 Send signal
>> sofia/
>>  
>> internal/+4001 at 192.168.200.100 [BREAK]
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG]  switch_core_state_machine.c:325
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) Running State  Change CS_INIT
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG]  switch_core_state_machine.c:361
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State INIT
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
>> sofia/internal/+4001 at 192.168.2
>>  
>> 00.100 SOFIA INIT
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
>> (sofia/internal/+4001 at 192.168
>>  
>> .200.100) State Change CS_INIT -> CS_ROUTING
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG]  switch_core_session.c:1116 Send signal
>> sofia/
>>  
>> internal/+4001 at 192.168.200.100 [BREAK]
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG]  switch_core_state_machine.c:361
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State INIT going  to sleep
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG]  switch_core_state_machine.c:325
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) Running State  Change CS_ROUTING
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
>> (sofia/internal/+4001 at 1
>>  
>> 92.168.200.100) Callstate Change DOWN -> RINGING
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG]  switch_core_state_machine.c:364
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State ROUTING
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
>> sofia/internal/+4001 at 192.168.
>>  
>> 200.100 SOFIA ROUTING
>>  
>> 2011-05-07 22:48:05.722537 [DEBUG]  switch_core_state_machine.c:77
>> sofia/internal
>>  
>> /+4001 at 192.168.200.100 Standard  ROUTING
>>  
>> 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331  Processing +4001
>> <+4001
>>  
>>> >->+9123 in context default
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing  [default->unloop]
>> continu
>>  
>> e=false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)  [unloop]
>> ${unroll_lo
>>  
>> ops}(true) =~ /^true$/ break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)  [unloop]
>> ${sip_loope
>>  
>> d_call}() =~ /^true$/ break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> [default->remove_plus_of_
>>  
>> dst_num]  continue=true
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>> [remove_plus_of_dst_
>>  
>> num]  destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>> set(destination_number=912
>>  
>> 3)
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> [default->remove_plus_of_
>>  
>> src_num] continue=true
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>> [remove_plus_of_src_
>>  
>> num] caller_id_number(+4001) =~ /^\+(\d+)$/  break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>> set(effective_caller_id_na
>>  
>> me=4001)
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>> set(effective_caller_id_nu
>>  
>> mber=4001)
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> [default->7_8_to_Lync] co
>>  
>> ntinue=false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)  [7_8_to_Lync]
>> destin
>>  
>> ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing  [default->9_to_GW]
>> contin
>>  
>> ue=false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)  [9_to_GW]
>> destinatio
>>  
>> n_number(+9123) =~ /^(9\d+)$/  break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing  [default->1_to_IPP]
>> conti
>>  
>> nue=false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)  [1_to_IPP]
>> destinati
>>  
>> on_number(+9123) =~ /^(1\d{3})$/ break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing  [default->DISA]
>> continue=
>>  
>> false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)  [DISA]
>> destination_n
>>  
>> umber(+9123) =~ /^\*(3472)$/ break=on-false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing  [default->Recordings]
>> con
>>  
>> tinue=false
>>  
>> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)  [Recordings]
>> destina
>>  
>> tion_number(+9123) =~ /^\*(732673)$/ break=on-false
>>  
>> 2011-05-07 22:48:05.727537 [DEBUG]  switch_core_state_machine.c:119
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State Change  CS_ROUTING -> CS_EXECUTE
>>  
>> 2011-05-07 22:48:05.727537 [DEBUG]  switch_core_session.c:1116 Send signal
>> sofia/
>>  
>> internal/+4001 at 192.168.200.100 [BREAK]
>>  
>> 2011-05-07 22:48:05.727537 [DEBUG]  switch_core_state_machine.c:364
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State ROUTING  going to sleep
>>  
>> 2011-05-07 22:48:05.727537 [DEBUG]  switch_core_state_machine.c:325
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) Running State  Change CS_EXECUTE
>>  
>> 2011-05-07 22:48:05.727537 [DEBUG]  switch_core_state_machine.c:371
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State EXECUTE
>>  
>> 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
>> sofia/internal/+4001 at 192.168.
>>  
>> 200.100 SOFIA EXECUTE
>>  
>> 2011-05-07 22:48:05.727537 [DEBUG]  switch_core_state_machine.c:157
>> sofia/interna
>>  
>> l/+4001 at 192.168.200.100 Standard  EXECUTE
>>  
>> EXECUTE sofia/internal/+4001 at 192.168.200.100  set(destination_number=9123)
>>  
>> 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>> sofia/internal/+4001 at 192.1
>>  
>> 68.200.100 SET [destination_number]=[9123]
>>  
>> EXECUTE sofia/internal/+4001 at 192.168.200.100
>> set(effective_caller_id_name=4001)
>>  
>> 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>> sofia/internal/+4001 at 192.1
>>  
>> 68.200.100 SET [effective_caller_id_name]=[4001]
>>  
>> EXECUTE sofia/internal/+4001 at 192.168.200.100
>> set(effective_caller_id_number=4001
>>  
>> )
>>  
>> 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
>> sofia/internal/+4001 at 192.1
>>  
>> 68.200.100 SET [effective_caller_id_number]=[4001]
>>  
>> 2011-05-07 22:48:05.729537 [NOTICE]  switch_core_state_machine.c:189
>> sofia/intern
>>  
>> al/+4001 at 192.168.200.100 has executed the  last dialplan instruction, hanging
>> up.
>>  
>>  
>>  
>> 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
>> (sofia/internal/+4001 at 1
>>  
>> 92.168.200.100) Callstate Change RINGING -> HANGUP
>>  
>> 2011-05-07 22:48:05.729537 [NOTICE]  switch_core_state_machine.c:191 Hangup
>> sofia
>>  
>> /internal/+4001 at 192.168.200.100 [CS_EXECUTE]  [NORMAL_CLEARING]
>>  
>> 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579  Send signal
>> sofia/inter
>>  
>> nal/+4001 at 192.168.200.100 [KILL]
>>  
>> 2011-05-07 22:48:05.730538 [DEBUG]  switch_core_session.c:1116 Send signal
>> sofia/
>>  
>> internal/+4001 at 192.168.200.100 [BREAK]
>>  
>> 2011-05-07 22:48:05.730538 [DEBUG]  switch_core_state_machine.c:371
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State EXECUTE  going to sleep
>>  
>> 2011-05-07 22:48:05.730538 [DEBUG]  switch_core_state_machine.c:325
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) Running State  Change CS_HANGUP
>>  
>> 2011-05-07 22:48:05.731538 [DEBUG]  switch_core_state_machine.c:565
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State HANGUP
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
>> sofia/internal/+4001@
>>  
>> 192.168.200.100 hanging up, cause: NORMAL_CLEARING
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519  Responding to INVITE
>> with: 48
>>  
>> 0
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG]  switch_core_state_machine.c:46
>> sofia/internal
>>  
>> /+4001 at 192.168.200.100 Standard HANGUP,  cause: NORMAL_CLEARING
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG]  switch_core_state_machine.c:565
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State HANGUP  going to sleep
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG]  switch_core_state_machine.c:356
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State Change  CS_HANGUP -> CS_REPORTING
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG]  switch_core_session.c:1116 Send signal
>> sofia/
>>  
>> internal/+4001 at 192.168.200.100 [BREAK]
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG]  switch_core_state_machine.c:325
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) Running State  Change CS_REPORTING
>>  
>> 2011-05-07 22:48:05.732538 [DEBUG]  switch_core_state_machine.c:625
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State  REPORTING
>>  
>> 2011-05-07 22:48:06.017554 [DEBUG]  switch_core_state_machine.c:53
>> sofia/internal
>>  
>> /+4001 at 192.168.200.100 Standard  REPORTING, cause: NORMAL_CLEARING
>>  
>> 2011-05-07 22:48:06.017554 [DEBUG]  switch_core_state_machine.c:625
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State REPORTING  going to sleep
>>  
>> 2011-05-07 22:48:06.017554 [DEBUG]  switch_core_state_machine.c:350
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State Change  CS_REPORTING -> CS_DESTROY
>>  
>> 2011-05-07 22:48:06.017554 [DEBUG]  switch_core_session.c:1116 Send signal
>> sofia/
>>  
>> internal/+4001 at 192.168.200.100 [BREAK]
>>  
>> 2011-05-07 22:48:06.017554 [DEBUG]  switch_core_session.c:1288 Session 2
>> (sofia/i
>>  
>> nternal/+4001 at 192.168.200.100) Locked, Waiting  on external entities
>>  
>> 2011-05-07 22:48:06.017554 [NOTICE]  switch_core_session.c:1306 Session 2
>> (sofia/
>>  
>> internal/+4001 at 192.168.200.100) Ended
>>  
>> 2011-05-07 22:48:06.017554 [NOTICE]  switch_core_session.c:1308 Close Channel
>> sof
>>  
>> ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
>>  
>> 2011-05-07 22:48:06.018554 [DEBUG]  switch_core_state_machine.c:454
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) Callstate Change  HANGUP -> DOWN
>>  
>> 2011-05-07 22:48:06.018554 [DEBUG]  switch_core_state_machine.c:457
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) Running State  Change CS_DESTROY
>>  
>> 2011-05-07 22:48:06.018554 [DEBUG]  switch_core_state_machine.c:467
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State DESTROY
>>  
>> 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
>> sofia/internal/+4001 at 192.168.
>>  
>> 200.100 SOFIA DESTROY
>>  
>> 2011-05-07 22:48:06.018554 [DEBUG]  switch_core_state_machine.c:60
>> sofia/internal
>>  
>> /+4001 at 192.168.200.100 Standard  DESTROY
>>  
>> 2011-05-07 22:48:06.018554 [DEBUG]  switch_core_state_machine.c:467
>> (sofia/intern
>>  
>> al/+4001 at 192.168.200.100) State DESTROY  going to sleep
>>  
>>  
>> 
>>  
>> 
>> _______________________________________________
>> FreeSWITCH-users  mailing  list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kris at kriskinc.com  Sat May  7 20:13:55 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Sat, 7 May 2011 12:13:55 -0400
Subject: [Freeswitch-users] How to manipulate destination number before
 routing to PSTN GW
In-Reply-To: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
Message-ID: <BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>

There is certainly more than one way to skin this cat...

However, to keep with your current method you should read up on
dialplan hunting vs. execution and "inline" execution:

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

In short you need to add inline="true" to your first extension.

On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> Gurus,
>
> i met an issue for dial plan, it sounds easy but puzzled me a few days not
> fix it yet...i belive gurus here could help me...
>
> i want to remove the + of destination number before routing to PSTN GW, e.g.
> when i dial +9123, i would like FS remove +, and then route to the PSTN GW,
> below is the dial plan and log,
>
> See the log, FS did remove '+' of '+9123', but then when it checks the rule
> ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then failed.
>
> Thanks.
>
>
>
> Dial plan:
>
> <extension name="remove_plus_of_dst_num" continue="true">
>
> ?? <condition field="destination_number" expression="^\+(\d+)$">
>
> ?????? <action application="set" data="effective_destination_number=$1"/>
>
> ?? </condition>
>
> </extension>
>
> ?...
>
> <extension name="9_to_GW" >
>
> ?? <condition field="destination_number" expression="^(9\d+)$">
>
> ?????? <action application="bridge"
> data="sofia/internal/$1 at 192.168.200.101"/>
>
> ?? </condition>
>
> </extension>
>
>
>
> Log:
>
> freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
> 192.168.200.201
>
> ?Approved by acl "192.168.0.0/16[]". Access Granted.
>
> 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
> sofia/inter
>
> nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
>
> 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
> sofia/internal/+4001 at 192
>
> .168.200.100 entering state [received][100]
>
> 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
>
> v=0
>
> o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
>
> s=Phone-Call
>
> c=IN IP4 192.168.200.201
>
> t=0 0
>
> m=audio 6060 RTP/AVP 8 0 96
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:96 telephone-event/8000
>
> a=fmtp:96 0-15
>
> a=ptime:20
>
>
>
> 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
> (sofia/internal/+4001 at 192.168.20
>
> 0.100) State Change CS_NEW -> CS_INIT
>
> 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/+4001 at 192.168.200.100 [BREAK]
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/+4001 at 192.168.200.100) Running State Change CS_INIT
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State INIT
>
> 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
> sofia/internal/+4001 at 192.168.2
>
> 00.100 SOFIA INIT
>
> 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
> (sofia/internal/+4001 at 192.168
>
> .200.100) State Change CS_INIT -> CS_ROUTING
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/+4001 at 192.168.200.100 [BREAK]
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State INIT going to sleep
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
> (sofia/internal/+4001 at 1
>
> 92.168.200.100) Callstate Change DOWN -> RINGING
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State ROUTING
>
> 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
> sofia/internal/+4001 at 192.168.
>
> 200.100 SOFIA ROUTING
>
> 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
> sofia/internal
>
> /+4001 at 192.168.200.100 Standard ROUTING
>
> 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing +4001
> <+4001
>
>>->+9123 in context default
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->unloop]
> continu
>
> e=false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
> ${unroll_lo
>
> ops}(true) =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
> ${sip_loope
>
> d_call}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> [default->remove_plus_of_
>
> dst_num] continue=true
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
> [remove_plus_of_dst_
>
> num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> set(destination_number=912
>
> 3)
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> [default->remove_plus_of_
>
> src_num] continue=true
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
> [remove_plus_of_src_
>
> num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> set(effective_caller_id_na
>
> me=4001)
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> set(effective_caller_id_nu
>
> mber=4001)
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> [default->7_8_to_Lync] co
>
> ntinue=false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
> destin
>
> ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->9_to_GW]
> contin
>
> ue=false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
> destinatio
>
> n_number(+9123) =~ /^(9\d+)$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->1_to_IPP]
> conti
>
> nue=false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
> destinati
>
> on_number(+9123) =~ /^(1\d{3})$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA]
> continue=
>
> false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
> destination_n
>
> umber(+9123) =~ /^\*(3472)$/ break=on-false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->Recordings]
> con
>
> tinue=false
>
> Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [Recordings]
> destina
>
> tion_number(+9123) =~ /^\*(732673)$/ break=on-false
>
> 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>
> 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/+4001 at 192.168.200.100 [BREAK]
>
> 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State ROUTING going to sleep
>
> 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
>
> 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State EXECUTE
>
> 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
> sofia/internal/+4001 at 192.168.
>
> 200.100 SOFIA EXECUTE
>
> 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
> sofia/interna
>
> l/+4001 at 192.168.200.100 Standard EXECUTE
>
> EXECUTE sofia/internal/+4001 at 192.168.200.100 set(destination_number=9123)
>
> 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
> sofia/internal/+4001 at 192.1
>
> 68.200.100 SET [destination_number]=[9123]
>
> EXECUTE sofia/internal/+4001 at 192.168.200.100
> set(effective_caller_id_name=4001)
>
> 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
> sofia/internal/+4001 at 192.1
>
> 68.200.100 SET [effective_caller_id_name]=[4001]
>
> EXECUTE sofia/internal/+4001 at 192.168.200.100
> set(effective_caller_id_number=4001
>
> )
>
> 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
> sofia/internal/+4001 at 192.1
>
> 68.200.100 SET [effective_caller_id_number]=[4001]
>
> 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
> sofia/intern
>
> al/+4001 at 192.168.200.100 has executed the last dialplan instruction, hanging
> up.
>
>
>
> 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
> (sofia/internal/+4001 at 1
>
> 92.168.200.100) Callstate Change RINGING -> HANGUP
>
> 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia
>
> /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
> sofia/inter
>
> nal/+4001 at 192.168.200.100 [KILL]
>
> 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/+4001 at 192.168.200.100 [BREAK]
>
> 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State EXECUTE going to sleep
>
> 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
>
> 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State HANGUP
>
> 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
> sofia/internal/+4001@
>
> 192.168.200.100 hanging up, cause: NORMAL_CLEARING
>
> 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE
> with: 48
>
> 0
>
> 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
> sofia/internal
>
> /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>
> 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State HANGUP going to sleep
>
> 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>
> 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/+4001 at 192.168.200.100 [BREAK]
>
> 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
>
> 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State REPORTING
>
> 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
> sofia/internal
>
> /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>
> 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State REPORTING going to sleep
>
> 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>
> 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/+4001 at 192.168.200.100 [BREAK]
>
> 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2
> (sofia/i
>
> nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
>
> 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session 2
> (sofia/
>
> internal/+4001 at 192.168.200.100) Ended
>
> 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close Channel
> sof
>
> ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
>
> 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
> (sofia/intern
>
> al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>
> 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
> (sofia/intern
>
> al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
>
> 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State DESTROY
>
> 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
> sofia/internal/+4001 at 192.168.
>
> 200.100 SOFIA DESTROY
>
> 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
> sofia/internal
>
> /+4001 at 192.168.200.100 Standard DESTROY
>
> 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
> (sofia/intern
>
> al/+4001 at 192.168.200.100) State DESTROY going to sleep
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 moises.silva at gmail.com  Sat May  7 21:23:37 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 7 May 2011 13:23:37 -0400
Subject: [Freeswitch-users] Trillium Error
In-Reply-To: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49C@jehuty.supportkids.com>
References: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49C@jehuty.supportkids.com>
Message-ID: <BANLkTikkmgxH8vJM6C7hthNKPahXuL2HLQ@mail.gmail.com>

On Fri, May 6, 2011 at 11:46 AM, Dennis Young
<Dennis.Young at supportkids.com>wrote:

> All,
>
>
>
> I?m seeing this error message in the command prompt window that is running
> Freeswitch but it?s not showing up in the freeswitch log file or remote
> console.  I running GIT HEAD 5/1/11 on WIN32 system.
>
>
>
> UNTSS: sw error:  ent: 010  inst: 000  proc id: 000
>
>                 file: ..\..\trillium\in\in_bdy4.c line: 483 errcode: 14536
> errcls: ERRCLS_DEBUG
>
>                 errval: 00018  errdesc: inUsrT302S25() failed, timer not
> defined for switch.
>
>
>
>
Hello Dennis,

Can you provide your freetdm.conf and freetdm.conf.xml? (use pastebin)

Also, which version of libsng_isdn are you using?

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
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From avi at avimarcus.net  Sat May  7 21:48:28 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 7 May 2011 20:48:28 +0300
Subject: [Freeswitch-users] many outgoing calls with mod_lcr at the same
 time possible?
In-Reply-To: <E6513A570B0E4F47A248C4E0735EAD99@omni1.local>
References: <E6513A570B0E4F47A248C4E0735EAD99@omni1.local>
Message-ID: <BANLkTim8H82JnYQbs9ox8zgii8iXYdy1hw@mail.gmail.com>

Since LCR registers itself as an endpoint, I suppose you can bridge to both
that way.
It's mentioned here, but the sample is actually for something else:
http://wiki.freeswitch.org/wiki/Mod_lcr#Endpoint
 <http://wiki.freeswitch.org/wiki/Mod_lcr#Endpoint>If you could update the
wiki with your use case, that would be great.
-Avi

On Sat, May 7, 2011 at 12:52 PM, Anestis Mavro <devel at omninet.eu> wrote:

>  Hello Freeswitchers,
>
>
>
> Is it possible to make many outgoing calls with mod_lcr at the same time?
> An example would be to redirect an incoming call to two mobile phones and
> ring them simultaneously.
>
> With a ?normal? bridge we can achieve this, but what about having also LCR
> (billing and limit included) involved?
>
>
>
> Regards
>
> 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 avi at avimarcus.net  Sat May  7 21:52:23 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 7 May 2011 20:52:23 +0300
Subject: [Freeswitch-users] How to manipulate destination number before
 routing to PSTN GW
In-Reply-To: <BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
Message-ID: <BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>

Indeed, inline is a missing ingredient. As well though is the previously
mentioned - you can't (directly) overwrite the destination number. You can
either use a new variable throughout, or transfer to set the new destination
number. Note that transferring causes the dialplan to be re-run from the
start..

-Avi

On Sat, May 7, 2011 at 7:13 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> There is certainly more than one way to skin this cat...
>
> However, to keep with your current method you should read up on
> dialplan hunting vs. execution and "inline" execution:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
> In short you need to add inline="true" to your first extension.
>
> On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> > Gurus,
> >
> > i met an issue for dial plan, it sounds easy but puzzled me a few days
> not
> > fix it yet...i belive gurus here could help me...
> >
> > i want to remove the + of destination number before routing to PSTN GW,
> e.g.
> > when i dial +9123, i would like FS remove +, and then route to the PSTN
> GW,
> > below is the dial plan and log,
> >
> > See the log, FS did remove '+' of '+9123', but then when it checks the
> rule
> > ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then
> failed.
> >
> > Thanks.
> >
> >
> >
> > Dial plan:
> >
> > <extension name="remove_plus_of_dst_num" continue="true">
> >
> >    <condition field="destination_number" expression="^\+(\d+)$">
> >
> >        <action application="set" data="effective_destination_number=$1"/>
> >
> >    </condition>
> >
> > </extension>
> >
> > ?...
> >
> > <extension name="9_to_GW" >
> >
> >    <condition field="destination_number" expression="^(9\d+)$">
> >
> >        <action application="bridge"
> > data="sofia/internal/$1 at 192.168.200.101"/>
> >
> >    </condition>
> >
> > </extension>
> >
> >
> >
> > Log:
> >
> > freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
> > 192.168.200.201
> >
> >  Approved by acl "192.168.0.0/16[]". Access Granted.
> >
> > 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
> > sofia/inter
> >
> > nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
> > sofia/internal/+4001 at 192
> >
> > .168.200.100 entering state [received][100]
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
> >
> > v=0
> >
> > o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
> >
> > s=Phone-Call
> >
> > c=IN IP4 192.168.200.201
> >
> > t=0 0
> >
> > m=audio 6060 RTP/AVP 8 0 96
> >
> > a=rtpmap:8 PCMA/8000
> >
> > a=rtpmap:0 PCMU/8000
> >
> > a=rtpmap:96 telephone-event/8000
> >
> > a=fmtp:96 0-15
> >
> > a=ptime:20
> >
> >
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
> > (sofia/internal/+4001 at 192.168.20
> >
> > 0.100) State Change CS_NEW -> CS_INIT
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_INIT
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State INIT
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
> > sofia/internal/+4001 at 192.168.2
> >
> > 00.100 SOFIA INIT
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
> > (sofia/internal/+4001 at 192.168
> >
> > .200.100) State Change CS_INIT -> CS_ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State INIT going to sleep
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
> > (sofia/internal/+4001 at 1
> >
> > 92.168.200.100) Callstate Change DOWN -> RINGING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
> > sofia/internal/+4001 at 192.168.
> >
> > 200.100 SOFIA ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard ROUTING
> >
> > 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing +4001
> > <+4001
> >
> >>->+9123 in context default
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->unloop]
> > continu
> >
> > e=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
> > ${unroll_lo
> >
> > ops}(true) =~ /^true$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
> > ${sip_loope
> >
> > d_call}() =~ /^true$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> > [default->remove_plus_of_
> >
> > dst_num] continue=true
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
> > [remove_plus_of_dst_
> >
> > num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> > set(destination_number=912
> >
> > 3)
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> > [default->remove_plus_of_
> >
> > src_num] continue=true
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
> > [remove_plus_of_src_
> >
> > num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> > set(effective_caller_id_na
> >
> > me=4001)
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> > set(effective_caller_id_nu
> >
> > mber=4001)
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> > [default->7_8_to_Lync] co
> >
> > ntinue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
> [7_8_to_Lync]
> > destin
> >
> > ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> [default->9_to_GW]
> > contin
> >
> > ue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
> > destinatio
> >
> > n_number(+9123) =~ /^(9\d+)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> [default->1_to_IPP]
> > conti
> >
> > nue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
> > destinati
> >
> > on_number(+9123) =~ /^(1\d{3})$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA]
> > continue=
> >
> > false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
> > destination_n
> >
> > umber(+9123) =~ /^\*(3472)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> [default->Recordings]
> > con
> >
> > tinue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [Recordings]
> > destina
> >
> > tion_number(+9123) =~ /^\*(732673)$/ break=on-false
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State ROUTING going to sleep
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
> > sofia/internal/+4001 at 192.168.
> >
> > 200.100 SOFIA EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
> > sofia/interna
> >
> > l/+4001 at 192.168.200.100 Standard EXECUTE
> >
> > EXECUTE sofia/internal/+4001 at 192.168.200.100set(destination_number=9123)
> >
> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
> > sofia/internal/+4001 at 192.1
> >
> > 68.200.100 SET [destination_number]=[9123]
> >
> > EXECUTE sofia/internal/+4001 at 192.168.200.100
> > set(effective_caller_id_name=4001)
> >
> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
> > sofia/internal/+4001 at 192.1
> >
> > 68.200.100 SET [effective_caller_id_name]=[4001]
> >
> > EXECUTE sofia/internal/+4001 at 192.168.200.100
> > set(effective_caller_id_number=4001
> >
> > )
> >
> > 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
> > sofia/internal/+4001 at 192.1
> >
> > 68.200.100 SET [effective_caller_id_number]=[4001]
> >
> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
> > sofia/intern
> >
> > al/+4001 at 192.168.200.100 has executed the last dialplan instruction,
> hanging
> > up.
> >
> >
> >
> > 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
> > (sofia/internal/+4001 at 1
> >
> > 92.168.200.100) Callstate Change RINGING -> HANGUP
> >
> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191
> Hangup
> > sofia
> >
> > /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
> > sofia/inter
> >
> > nal/+4001 at 192.168.200.100 [KILL]
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State EXECUTE going to sleep
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
> >
> > 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State HANGUP
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
> > sofia/internal/+4001@
> >
> > 192.168.200.100 hanging up, cause: NORMAL_CLEARING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE
> > with: 48
> >
> > 0
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State HANGUP going to sleep
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State REPORTING
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State REPORTING going to sleep
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2
> > (sofia/i
> >
> > nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
> >
> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session 2
> > (sofia/
> >
> > internal/+4001 at 192.168.200.100) Ended
> >
> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close
> Channel
> > sof
> >
> > ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
> > sofia/internal/+4001 at 192.168.
> >
> > 200.100 SOFIA DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State DESTROY going to sleep
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 gavin.henry at gmail.com  Sun May  8 01:19:24 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 7 May 2011 22:19:24 +0100
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTi=F6faH5HYkeRwLi2Eg5vCfazFaUg@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<BANLkTin5dVPK6WN_LgHP6_PfvE2ZyPwRyw@mail.gmail.com>
	<BANLkTinZFi3eEG0P3A+6rz-b+MKAOfU6ww@mail.gmail.com>
	<1304713411.4953.21.camel@raul-laptop>
	<BANLkTi=F6faH5HYkeRwLi2Eg5vCfazFaUg@mail.gmail.com>
Message-ID: <BANLkTim4FviPSRs0xYtQXX0MGnBGhr3XEw@mail.gmail.com>

Same here.
> We have been running CentOS for about 3 years now without any issues.
> CentOS 5.6 has been
> working great as well.
>
> I am not planning on switching any time soon.
>
> Norman Tomlins
> Voice Network Inc.
> http://www.VoiceNetwork.ca
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david.ponzone at ipeva.fr  Sun May  8 04:55:56 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 8 May 2011 02:55:56 +0200
Subject: [Freeswitch-users] mod_limit Error
In-Reply-To: <636535.38388.qm@web34305.mail.mud.yahoo.com>
References: <636535.38388.qm@web34305.mail.mud.yahoo.com>
Message-ID: <48BCDAC7-7579-41AE-8372-58526BF4E320@ipeva.fr>

Fernando,

I suspect you upgraded from a quite old version.
mod_limit is dead, it was moved to core.
You just need mod_hash if you used to do hash limits.
Check the wiki for details.

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

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

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




Le 07/05/2011 ? 05:23, FERNANDO VILLARROEL a ?crit :

> Hi All.
> 
> I rebuild my FS box and now i am getting the following error when i load the module limit:
> 
> 2011-05-06 23:56:11.229308 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_limit.so
> **/usr/local/freeswitch/mod/mod_limit.so: undefined symbol: switch_channel_get_variable**
> 
> How i can solve?
> 
> Fernando.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Sun May  8 05:04:09 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 8 May 2011 03:04:09 +0200
Subject: [Freeswitch-users] Gateways ACL Context
In-Reply-To: <310161.61604.qm@web34307.mail.mud.yahoo.com>
References: <310161.61604.qm@web34307.mail.mud.yahoo.com>
Message-ID: <D0D39988-65C1-4EA5-97A0-FF8E7A255883@ipeva.fr>

in the context of the SIP profile receiving the calls, you need to add some rules, where you put some conditions on network_addr and make the required action.
Basically, the idea is:

  <extension name="from_gatewayfoo">
    <condition field="network_addr" expression="190.XXX.XX.XXX"/>
    <condition field="destination_number" expression="^\(.*)$">
      <action application="transfer" data="$1 XML foo"/>
    </condition>
  </extension>

You can also use this to do some stuff on your calls, like setting accountcode or normalizing your destination_number format before the transfer, in order to clean your gateway context (foo) and increase readability.

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

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

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




Le 07/05/2011 ? 04:00, FERNANDO VILLARROEL a ?crit :

> Dear All.
> 
> How i can change to a private context like "my_context" for inbound traffic from gateways authorized on ACL and how i can assign different private context for each gateway authorized in ACL, like:
> 
> <list name="domains" default="deny">
>      <node type="allow" cidr="190.XXX.XX.XXX/32"/> context foo
>      <node type="allow" cidr="66.X.XXX.XX/32"/>    context foo1
> 
> 
> </list>
> 
> 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 david.ponzone at ipeva.fr  Sun May  8 05:12:51 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 8 May 2011 03:12:51 +0200
Subject: [Freeswitch-users] How to manipulate destination number before
	routing to PSTN GW
In-Reply-To: <BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
	<BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
Message-ID: <C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>

Avi,

are you sure about that ?

I have LUA scripts which rewrite the destination_number with SetVariable and that works fine.
Of course, to do that in the XML dialplan, as said before, inline should be used to set the value.

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

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

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




Le 07/05/2011 ? 19:52, Avi Marcus a ?crit :

> Indeed, inline is a missing ingredient. As well though is the previously mentioned - you can't (directly) overwrite the destination number. You can either use a new variable throughout, or transfer to set the new destination number. Note that transferring causes the dialplan to be re-run from the start..
> 
> -Avi
> 
> On Sat, May 7, 2011 at 7:13 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> There is certainly more than one way to skin this cat...
> 
> However, to keep with your current method you should read up on
> dialplan hunting vs. execution and "inline" execution:
> 
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
> 
> In short you need to add inline="true" to your first extension.
> 
> On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> > Gurus,
> >
> > i met an issue for dial plan, it sounds easy but puzzled me a few days not
> > fix it yet...i belive gurus here could help me...
> >
> > i want to remove the + of destination number before routing to PSTN GW, e.g.
> > when i dial +9123, i would like FS remove +, and then route to the PSTN GW,
> > below is the dial plan and log,
> >
> > See the log, FS did remove '+' of '+9123', but then when it checks the rule
> > ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then failed.
> >
> > Thanks.
> >
> >
> >
> > Dial plan:
> >
> > <extension name="remove_plus_of_dst_num" continue="true">
> >
> >    <condition field="destination_number" expression="^\+(\d+)$">
> >
> >        <action application="set" data="effective_destination_number=$1"/>
> >
> >    </condition>
> >
> > </extension>
> >
> > ?...
> >
> > <extension name="9_to_GW" >
> >
> >    <condition field="destination_number" expression="^(9\d+)$">
> >
> >        <action application="bridge"
> > data="sofia/internal/$1 at 192.168.200.101"/>
> >
> >    </condition>
> >
> > </extension>
> >
> >
> >
> > Log:
> >
> > freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
> > 192.168.200.201
> >
> >  Approved by acl "192.168.0.0/16[]". Access Granted.
> >
> > 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
> > sofia/inter
> >
> > nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
> > sofia/internal/+4001 at 192
> >
> > .168.200.100 entering state [received][100]
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
> >
> > v=0
> >
> > o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
> >
> > s=Phone-Call
> >
> > c=IN IP4 192.168.200.201
> >
> > t=0 0
> >
> > m=audio 6060 RTP/AVP 8 0 96
> >
> > a=rtpmap:8 PCMA/8000
> >
> > a=rtpmap:0 PCMU/8000
> >
> > a=rtpmap:96 telephone-event/8000
> >
> > a=fmtp:96 0-15
> >
> > a=ptime:20
> >
> >
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
> > (sofia/internal/+4001 at 192.168.20
> >
> > 0.100) State Change CS_NEW -> CS_INIT
> >
> > 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_INIT
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State INIT
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
> > sofia/internal/+4001 at 192.168.2
> >
> > 00.100 SOFIA INIT
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
> > (sofia/internal/+4001 at 192.168
> >
> > .200.100) State Change CS_INIT -> CS_ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State INIT going to sleep
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
> > (sofia/internal/+4001 at 1
> >
> > 92.168.200.100) Callstate Change DOWN -> RINGING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
> > sofia/internal/+4001 at 192.168.
> >
> > 200.100 SOFIA ROUTING
> >
> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard ROUTING
> >
> > 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing +4001
> > <+4001
> >
> >>->+9123 in context default
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->unloop]
> > continu
> >
> > e=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
> > ${unroll_lo
> >
> > ops}(true) =~ /^true$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
> > ${sip_loope
> >
> > d_call}() =~ /^true$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> > [default->remove_plus_of_
> >
> > dst_num] continue=true
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
> > [remove_plus_of_dst_
> >
> > num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> > set(destination_number=912
> >
> > 3)
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> > [default->remove_plus_of_
> >
> > src_num] continue=true
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
> > [remove_plus_of_src_
> >
> > num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> > set(effective_caller_id_na
> >
> > me=4001)
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
> > set(effective_caller_id_nu
> >
> > mber=4001)
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
> > [default->7_8_to_Lync] co
> >
> > ntinue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
> > destin
> >
> > ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->9_to_GW]
> > contin
> >
> > ue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
> > destinatio
> >
> > n_number(+9123) =~ /^(9\d+)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->1_to_IPP]
> > conti
> >
> > nue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
> > destinati
> >
> > on_number(+9123) =~ /^(1\d{3})$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA]
> > continue=
> >
> > false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
> > destination_n
> >
> > umber(+9123) =~ /^\*(3472)$/ break=on-false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->Recordings]
> > con
> >
> > tinue=false
> >
> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [Recordings]
> > destina
> >
> > tion_number(+9123) =~ /^\*(732673)$/ break=on-false
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State ROUTING going to sleep
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
> > sofia/internal/+4001 at 192.168.
> >
> > 200.100 SOFIA EXECUTE
> >
> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
> > sofia/interna
> >
> > l/+4001 at 192.168.200.100 Standard EXECUTE
> >
> > EXECUTE sofia/internal/+4001 at 192.168.200.100 set(destination_number=9123)
> >
> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
> > sofia/internal/+4001 at 192.1
> >
> > 68.200.100 SET [destination_number]=[9123]
> >
> > EXECUTE sofia/internal/+4001 at 192.168.200.100
> > set(effective_caller_id_name=4001)
> >
> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
> > sofia/internal/+4001 at 192.1
> >
> > 68.200.100 SET [effective_caller_id_name]=[4001]
> >
> > EXECUTE sofia/internal/+4001 at 192.168.200.100
> > set(effective_caller_id_number=4001
> >
> > )
> >
> > 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
> > sofia/internal/+4001 at 192.1
> >
> > 68.200.100 SET [effective_caller_id_number]=[4001]
> >
> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
> > sofia/intern
> >
> > al/+4001 at 192.168.200.100 has executed the last dialplan instruction, hanging
> > up.
> >
> >
> >
> > 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
> > (sofia/internal/+4001 at 1
> >
> > 92.168.200.100) Callstate Change RINGING -> HANGUP
> >
> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191 Hangup
> > sofia
> >
> > /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
> > sofia/inter
> >
> > nal/+4001 at 192.168.200.100 [KILL]
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State EXECUTE going to sleep
> >
> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
> >
> > 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State HANGUP
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
> > sofia/internal/+4001@
> >
> > 192.168.200.100 hanging up, cause: NORMAL_CLEARING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE
> > with: 48
> >
> > 0
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State HANGUP going to sleep
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
> >
> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State REPORTING
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State REPORTING going to sleep
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send signal
> > sofia/
> >
> > internal/+4001 at 192.168.200.100 [BREAK]
> >
> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2
> > (sofia/i
> >
> > nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
> >
> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session 2
> > (sofia/
> >
> > internal/+4001 at 192.168.200.100) Ended
> >
> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close Channel
> > sof
> >
> > ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
> > sofia/internal/+4001 at 192.168.
> >
> > 200.100 SOFIA DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
> > sofia/internal
> >
> > /+4001 at 192.168.200.100 Standard DESTROY
> >
> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
> > (sofia/intern
> >
> > al/+4001 at 192.168.200.100) State DESTROY going to sleep
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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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> 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 fieldpeak at gmail.com  Sun May  8 05:56:13 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 8 May 2011 09:56:13 +0800
Subject: [Freeswitch-users] How to manipulate destination number before
 routing to PSTN GW
In-Reply-To: <C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
	<BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
	<C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
Message-ID: <BANLkTikU6yZg2vEAC5TW3JiHfbm7H4e84A@mail.gmail.com>

i tested the inline usage, unluckly, it failed. it looks the inline still
can not effect 9 to GW rule...

*dial plan is:*
<extension name="remove_plus_of_dst_num" continue="true">
   <condition field="destination_number" expression="^\+(\d+)$">
       <action inline="true" application="set"
data="destination_number=$1"/>
   </condition>
</extension>

...
<extension name="9_to_GW" >
   <condition field="destination_number" expression="^(9\d+)$">
       <action application="bridge" data="sofia/internal/$1 at 192.168.200.101
"/>
   </condition>
</extension>

*the log is:*

freeswitch at mypc>
2011-05-08 09:35:40.067099 [ERR] switch_xml.c:1311 Couldnt open
C:\FreeSWITCH\conf\
autoload_configs\..\sip_profiles\external/*.xml (No such file or directory)
2011-05-08 09:35:40.543126 [ERR] switch_xml.c:1311 Couldnt open
C:\FreeSWITCH\conf\
dialplan\public/*.xml (No such file or directory)

+OK [Success]

2011-05-08 09:35:41.115159 [INFO] mod_pocketsphinx.c:482 PocketSphinx
Reloaded
2011-05-08 09:35:41.117159 [INFO] switch_time.c:999 Timezone reloaded 530
defini
tions
2011-05-08 09:35:44.882374 [DEBUG] sofia.c:6488 IP 192.168.200.201 Approved
by a
cl "192.168.0.0/16[]". Access Granted.
2011-05-08 09:35:44.883374 [NOTICE] switch_channel.c:812 New Channel
sofia/inter
nal/4001 at 192.168.200.100 [efeb55d7-3712-4a30-9eba-786ad83b2e90]
2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4760 Channel
sofia/internal/4001 at 192.
168.200.100 entering state [received][100]
2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4771 Remote SDP:
v=0
o=AudiocodesGW 2088509637 2088509507 IN IP4 192.168.200.201
s=Phone-Call
c=IN IP4 192.168.200.201
t=0 0
m=audio 6020 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20

2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4908
(sofia/internal/4001 at 192.168.200
.100) State Change CS_NEW -> CS_INIT
2011-05-08 09:35:44.884374 [DEBUG] switch_core_session.c:1116 Send signal
sofia/
internal/4001 at 192.168.200.100 [BREAK]
2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
(sofia/intern
al/4001 at 192.168.200.100) Running State Change CS_INIT
2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
(sofia/intern
al/4001 at 192.168.200.100) State INIT
2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:84
sofia/internal/4001 at 192.168.20
0.100 SOFIA INIT
2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:124
(sofia/internal/4001 at 192.168.
200.100) State Change CS_INIT -> CS_ROUTING
2011-05-08 09:35:44.885374 [DEBUG] switch_core_session.c:1116 Send signal
sofia/
internal/4001 at 192.168.200.100 [BREAK]
2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
(sofia/intern
al/4001 at 192.168.200.100) State INIT going to sleep
2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
(sofia/intern
al/4001 at 192.168.200.100) Running State Change CS_ROUTING
2011-05-08 09:35:44.885374 [DEBUG] switch_channel.c:1668
(sofia/internal/4001 at 19
2.168.200.100) Callstate Change DOWN -> RINGING
2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:364
(sofia/intern
al/4001 at 192.168.200.100) State ROUTING
2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:147
sofia/internal/4001 at 192.168.2
00.100 SOFIA ROUTING
2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:77
sofia/internal
/4001 at 192.168.200.100 Standard ROUTING
2011-05-08 09:35:44.885374 [INFO] mod_dialplan_xml.c:331 Processing 4001
<4001>-
>+9123 in context default
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->unloop]
continue
=false
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS) [unloop]
${unroll_loo
ps}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [unloop]
${sip_looped
_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
[default->remove_plus_of_d
st_num] continue=true
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS)
[remove_plus_of_dst_n
um] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 Action
set(destination_number=9123
) INLINE
EXECUTE sofia/internal/4001 at 192.168.200.100 set(destination_number=9123)
2011-05-08 09:35:44.888374 [DEBUG] mod_dptools.c:1060
sofia/internal/4001 at 192.16
8.200.100 SET [destination_number]=[9123]
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
[default->remove_plus_of_s
rc_num] continue=true
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL)
[remove_plus_of_src_n
um] caller_id_number(4001) =~ /^\+(\d+)$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->7_8_to_Lync]
con
tinue=false
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
destina
tion_number(+9123) =~ /^([78]\d{3})$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->9_to_GW]
continu
e=false
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
destination
_number(+9123) =~ /^(9\d+)$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->1_to_IPP]
contin
ue=false
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
destinatio
n_number(+9123) =~ /^\+{0,1}(1\d{3})$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->DISA]
continue=f
alse
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [DISA]
destination_nu
mber(+9123) =~ /^\*(3472)$/ break=on-false
Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->Recordings]
cont
inue=false
Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [Recordings]
destinat
ion_number(+9123) =~ /^\*(732673)$/ break=on-false
2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:119
(sofia/intern
al/4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
2011-05-08 09:35:44.889374 [DEBUG] switch_core_session.c:1116 Send signal
sofia/
internal/4001 at 192.168.200.100 [BREAK]
2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:364
(sofia/intern
al/4001 at 192.168.200.100) State ROUTING going to sleep
2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:325
(sofia/intern
al/4001 at 192.168.200.100) Running State Change CS_EXECUTE
2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:371
(sofia/intern
al/4001 at 192.168.200.100) State EXECUTE
2011-05-08 09:35:44.889374 [DEBUG] mod_sofia.c:240
sofia/internal/4001 at 192.168.2
00.100 SOFIA EXECUTE
2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:157
sofia/interna
l/4001 at 192.168.200.100 Standard EXECUTE
2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:189
sofia/intern
al/4001 at 192.168.200.100 has executed the last dialplan instruction, hanging
up.
2011-05-08 09:35:44.889374 [DEBUG] switch_channel.c:2563
(sofia/internal/4001 at 19
2.168.200.100) Callstate Change RINGING -> HANGUP
2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia
/internal/4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
2011-05-08 09:35:44.890375 [DEBUG] switch_channel.c:2579 Send signal
sofia/inter
nal/4001 at 192.168.200.100 [KILL]
2011-05-08 09:35:44.890375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/
internal/4001 at 192.168.200.100 [BREAK]
2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:371
(sofia/intern
al/4001 at 192.168.200.100) State EXECUTE going to sleep
2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:325
(sofia/intern
al/4001 at 192.168.200.100) Running State Change CS_HANGUP
2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
(sofia/intern
al/4001 at 192.168.200.100) State HANGUP
2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:457 Channel
sofia/internal/4001 at 1
92.168.200.100 hanging up, cause: NORMAL_CLEARING
2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:519 Responding to INVITE
with: 48
0
2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:46
sofia/internal
/4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
(sofia/intern
al/4001 at 192.168.200.100) State HANGUP going to sleep
2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:356
(sofia/intern
al/4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
2011-05-08 09:35:44.891375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/
internal/4001 at 192.168.200.100 [BREAK]
2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:325
(sofia/intern
al/4001 at 192.168.200.100) Running State Change CS_REPORTING
2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:625
(sofia/intern
al/4001 at 192.168.200.100) State REPORTING
2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:53
sofia/internal
/4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:625
(sofia/intern
al/4001 at 192.168.200.100) State REPORTING going to sleep
2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:350
(sofia/intern
al/4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1116 Send signal
sofia/
internal/4001 at 192.168.200.100 [BREAK]
2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1288 Session 4
(sofia/i
nternal/4001 at 192.168.200.100) Locked, Waiting on external entities
2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1306 Session 4
(sofia/
internal/4001 at 192.168.200.100) Ended
2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1308 Close Channel
sof
ia/internal/4001 at 192.168.200.100 [CS_DESTROY]
2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:454
(sofia/intern
al/4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:457
(sofia/intern
al/4001 at 192.168.200.100) Running State Change CS_DESTROY
2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
(sofia/intern
al/4001 at 192.168.200.100) State DESTROY
2011-05-08 09:35:45.154390 [DEBUG] mod_sofia.c:362
sofia/internal/4001 at 192.168.2
00.100 SOFIA DESTROY
2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:60
sofia/internal
/4001 at 192.168.200.100 Standard DESTROY
2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
(sofia/intern
al/4001 at 192.168.200.100) State DESTROY going to sleep


2011/5/8 David Ponzone <david.ponzone at ipeva.fr>

> Avi,
>
> are you sure about that ?
>
> I have LUA scripts which rewrite the destination_number with SetVariable
> and that works fine.
> Of course, to do that in the XML dialplan, as said before, inline should be
> used to set the value.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/05/2011 ? 19:52, Avi Marcus a ?crit :
>
> Indeed, inline is a missing ingredient. As well though is the previously
> mentioned - you can't (directly) overwrite the destination number. You can
> either use a new variable throughout, or transfer to set the new destination
> number. Note that transferring causes the dialplan to be re-run from the
> start..
>
> -Avi
>
> On Sat, May 7, 2011 at 7:13 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> There is certainly more than one way to skin this cat...
>>
>> However, to keep with your current method you should read up on
>> dialplan hunting vs. execution and "inline" execution:
>>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>
>> In short you need to add inline="true" to your first extension.
>>
>> On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>> > Gurus,
>> >
>> > i met an issue for dial plan, it sounds easy but puzzled me a few days
>> not
>> > fix it yet...i belive gurus here could help me...
>> >
>> > i want to remove the + of destination number before routing to PSTN GW,
>> e.g.
>> > when i dial +9123, i would like FS remove +, and then route to the PSTN
>> GW,
>> > below is the dial plan and log,
>> >
>> > See the log, FS did remove '+' of '+9123', but then when it checks the
>> rule
>> > ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then
>> failed.
>> >
>> > Thanks.
>> >
>> >
>> >
>> > Dial plan:
>> >
>> > <extension name="remove_plus_of_dst_num" continue="true">
>> >
>> >    <condition field="destination_number" expression="^\+(\d+)$">
>> >
>> >        <action application="set"
>> data="effective_destination_number=$1"/>
>> >
>> >    </condition>
>> >
>> > </extension>
>> >
>> > ?...
>> >
>> > <extension name="9_to_GW" >
>> >
>> >    <condition field="destination_number" expression="^(9\d+)$">
>> >
>> >        <action application="bridge"
>> > data="sofia/internal/$1 at 192.168.200.101"/>
>> >
>> >    </condition>
>> >
>> > </extension>
>> >
>> >
>> >
>> > Log:
>> >
>> > freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
>> > 192.168.200.201
>> >
>> >  Approved by acl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>".
>> Access Granted.
>> >
>> > 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
>> > sofia/inter
>> >
>> > nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
>> >
>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
>> > sofia/internal/+4001 at 192
>> >
>> > .168.200.100 entering state [received][100]
>> >
>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
>> >
>> > v=0
>> >
>> > o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
>> >
>> > s=Phone-Call
>> >
>> > c=IN IP4 192.168.200.201
>> >
>> > t=0 0
>> >
>> > m=audio 6060 RTP/AVP 8 0 96
>> >
>> > a=rtpmap:8 PCMA/8000
>> >
>> > a=rtpmap:0 PCMU/8000
>> >
>> > a=rtpmap:96 telephone-event/8000
>> >
>> > a=fmtp:96 0-15
>> >
>> > a=ptime:20
>> >
>> >
>> >
>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
>> > (sofia/internal/+4001 at 192.168.20
>> >
>> > 0.100) State Change CS_NEW -> CS_INIT
>> >
>> > 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send
>> signal
>> > sofia/
>> >
>> > internal/+4001 at 192.168.200.100 [BREAK]
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) Running State Change CS_INIT
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State INIT
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
>> > sofia/internal/+4001 at 192.168.2
>> >
>> > 00.100 SOFIA INIT
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
>> > (sofia/internal/+4001 at 192.168
>> >
>> > .200.100) State Change CS_INIT -> CS_ROUTING
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send
>> signal
>> > sofia/
>> >
>> > internal/+4001 at 192.168.200.100 [BREAK]
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State INIT going to sleep
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
>> > (sofia/internal/+4001 at 1
>> >
>> > 92.168.200.100) Callstate Change DOWN -> RINGING
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State ROUTING
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
>> > sofia/internal/+4001 at 192.168.
>> >
>> > 200.100 SOFIA ROUTING
>> >
>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
>> > sofia/internal
>> >
>> > /+4001 at 192.168.200.100 Standard ROUTING
>> >
>> > 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing
>> +4001
>> > <+4001
>> >
>> >>->+9123 in context default
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> [default->unloop]
>> > continu
>> >
>> > e=false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
>> > ${unroll_lo
>> >
>> > ops}(true) =~ /^true$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
>> > ${sip_loope
>> >
>> > d_call}() =~ /^true$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> > [default->remove_plus_of_
>> >
>> > dst_num] continue=true
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>> > [remove_plus_of_dst_
>> >
>> > num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>> > set(destination_number=912
>> >
>> > 3)
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> > [default->remove_plus_of_
>> >
>> > src_num] continue=true
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>> > [remove_plus_of_src_
>> >
>> > num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>> > set(effective_caller_id_na
>> >
>> > me=4001)
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>> > set(effective_caller_id_nu
>> >
>> > mber=4001)
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> > [default->7_8_to_Lync] co
>> >
>> > ntinue=false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>> [7_8_to_Lync]
>> > destin
>> >
>> > ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> [default->9_to_GW]
>> > contin
>> >
>> > ue=false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
>> > destinatio
>> >
>> > n_number(+9123) =~ /^(9\d+)$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> [default->1_to_IPP]
>> > conti
>> >
>> > nue=false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
>> > destinati
>> >
>> > on_number(+9123) =~ /^(1\d{3})$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA]
>> > continue=
>> >
>> > false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
>> > destination_n
>> >
>> > umber(+9123) =~ /^\*(3472)$/ break=on-false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>> [default->Recordings]
>> > con
>> >
>> > tinue=false
>> >
>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>> [Recordings]
>> > destina
>> >
>> > tion_number(+9123) =~ /^\*(732673)$/ break=on-false
>> >
>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>> >
>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send
>> signal
>> > sofia/
>> >
>> > internal/+4001 at 192.168.200.100 [BREAK]
>> >
>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State ROUTING going to sleep
>> >
>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
>> >
>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State EXECUTE
>> >
>> > 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
>> > sofia/internal/+4001 at 192.168.
>> >
>> > 200.100 SOFIA EXECUTE
>> >
>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
>> > sofia/interna
>> >
>> > l/+4001 at 192.168.200.100 Standard EXECUTE
>> >
>> > EXECUTE sofia/internal/+4001 at 192.168.200.100set(destination_number=9123)
>> >
>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>> > sofia/internal/+4001 at 192.1
>> >
>> > 68.200.100 SET [destination_number]=[9123]
>> >
>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>> > set(effective_caller_id_name=4001)
>> >
>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>> > sofia/internal/+4001 at 192.1
>> >
>> > 68.200.100 SET [effective_caller_id_name]=[4001]
>> >
>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>> > set(effective_caller_id_number=4001
>> >
>> > )
>> >
>> > 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
>> > sofia/internal/+4001 at 192.1
>> >
>> > 68.200.100 SET [effective_caller_id_number]=[4001]
>> >
>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
>> > sofia/intern
>> >
>> > al/+4001 at 192.168.200.100 has executed the last dialplan instruction,
>> hanging
>> > up.
>> >
>> >
>> >
>> > 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
>> > (sofia/internal/+4001 at 1
>> >
>> > 92.168.200.100) Callstate Change RINGING -> HANGUP
>> >
>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191
>> Hangup
>> > sofia
>> >
>> > /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>> >
>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
>> > sofia/inter
>> >
>> > nal/+4001 at 192.168.200.100 [KILL]
>> >
>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send
>> signal
>> > sofia/
>> >
>> > internal/+4001 at 192.168.200.100 [BREAK]
>> >
>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State EXECUTE going to sleep
>> >
>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
>> >
>> > 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State HANGUP
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
>> > sofia/internal/+4001@
>> >
>> > 192.168.200.100 hanging up, cause: NORMAL_CLEARING
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE
>> > with: 48
>> >
>> > 0
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
>> > sofia/internal
>> >
>> > /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State HANGUP going to sleep
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send
>> signal
>> > sofia/
>> >
>> > internal/+4001 at 192.168.200.100 [BREAK]
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
>> >
>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State REPORTING
>> >
>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
>> > sofia/internal
>> >
>> > /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>> >
>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State REPORTING going to sleep
>> >
>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>> >
>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send
>> signal
>> > sofia/
>> >
>> > internal/+4001 at 192.168.200.100 [BREAK]
>> >
>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2
>> > (sofia/i
>> >
>> > nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
>> >
>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session 2
>> > (sofia/
>> >
>> > internal/+4001 at 192.168.200.100) Ended
>> >
>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close
>> Channel
>> > sof
>> >
>> > ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
>> >
>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>> >
>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
>> >
>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State DESTROY
>> >
>> > 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
>> > sofia/internal/+4001 at 192.168.
>> >
>> > 200.100 SOFIA DESTROY
>> >
>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
>> > sofia/internal
>> >
>> > /+4001 at 192.168.200.100 Standard DESTROY
>> >
>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>> > (sofia/intern
>> >
>> > al/+4001 at 192.168.200.100) State DESTROY going to sleep
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun May  8 15:03:52 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 08 May 2011 19:03:52 +0800
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
Message-ID: <4DC67898.2090106@coppice.org>

On 05/07/2011 04:02 AM, Andrew Keil wrote:
> Steve,
>
> Thanks for your response.
>
> Further clarification on my part:
>
> 1) 20,000 to 30,000 pages per day to be sent out.
So, this is fairly small scale. A single E1 will do fine.
> 2) It will be an e-mail to fax style gateway (not fax to e-mail since that would involve inbound faxes)
That covers a few requirements, depending what you expect to be in the 
e-mails - send the whole e-mail as a FAX; extract a PDF, Word document, 
etc. from an e-mail, and turn that into a FAX; and so on.
> 3) The reason I asked about e-mail to PDF is the initial comments from my client requested the ability to send PDFs and WORD documents (I guess from attachments to the original e-mail), I understand the format that gets faxed should be TIFF so I saw on the freeswitch wiki email2pdf mentioned.  Then ImageMagick can help get a PDF to TIFF.
If the incoming e-mails are limited to ones containing PDFs and doc 
files to be extracted and turned into FAXes things you seem to have a 
fairly well defined requirement. OpenOffice can be used to turn the doc 
files into FAXable images, but I am not clear how well the newer docx 
files are handled. Ghostscript can be used to turn PDFs into TIFF files.

Avoid ImageMagick for this kind of work. It uses Ghostscript to do the 
hard work, but it doesn't get the best from it. If you use Ghostscript 
directly you get better control, and you can achieve good results.
> Can I ask some more questions:
>
> Q1) Based on your experience what would be the average time (in seconds) to send a single fax page (TIFF file) via Freeswitch&  Sangoma TDM?  You can quote a TIFF file size to make it more accurate.  From there I should be able to do the math to calculate my Client's requirements better.
The time per page depends a lot on its complexity. I use a torture test 
file with images that take half an hour to send. Typical office work is 
probably 20s per page at 14400bps.
> Q2) Running on CentOS and using mod_spandsp/Freeswitch&  Sangoma TDM what percentage CPU usage would I expect to see if 30 concurrent faxes are being sent at the same time (ie. All channels of my E1 are faxing)?  (The hardware would be a new 1U rack server from a major hardware vendor)
The greater part of the CPU load is likely to be what you didn't list 
there - the processing from e-mail to FAXable TIFF files. A single E1 of 
FAXing is a really low load these days, though.
> Q3) What version of CentOS 5.x would you recommend?  Would the latest version 5.6 be fine?
5.6 is fine.
> Q4) From memory there used to be different fax quality modes on fax machines (STANDARD, FINE&  SUPER FINE or something like that).  Is it possible to set the fax send quality from mod_spandsp (also can you provide an example)?  If this is the case could you also answer question (Q1) based on the different fax send quality modes.
You can't really set the quality in mod_spandsp. The quality follows the 
TIFF files to be sent. You can, however, select the image quality as you 
generate the TIFF files in Ghostscript. STANDARD, FINE and SUPERFINE are 
the right names, although many machines refuse to use SUPERFINE.
> Q5) From exisiting deployments of Freeswitch using mod_spandsp (&  Sagoma TDM cards (although this is not critical)) what is the largest number of concurrent outbound faxes done on a single box that you know of?
I'm not sure of the biggest, but an E1 of FAXing is pretty small volume 
these days. The biggest number of channels should be in the hundreds.
> I appreciate your feedback and experience.  It sounds like this will work fine with the mod_spandsp/Freeswitch&  Sangoma TDM combination on CentOS.
>
> I will most likely go for two servers with at least 2 x E1s in each, that way I future proof it a little and add redundancy.  Plus my Client can start using Freeswitch for Voice related services also.  On my side I can also test faxes going out on the first server and via a cross-over cable I can terminate them on the second server (for testing) - much nicer.
>
> Thanks again,
>
> Andrew
Two servers with one E1 in each sounds like more than enough to meet 
your needs, unless your 30k pages per day occur over a fairly short 
working hours, and the customer demands rapid delivery. Then you might 
need more channels to deal with rush hour.

Steve



From jcasale at activenetwerx.com  Sun May  8 20:16:38 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 8 May 2011 16:16:38 +0000
Subject: [Freeswitch-users] Domain tag in sofia profile
Message-ID: <assp.01095cb6c2.92097A6A775D5147B1078E3F15430B924B993D@prato.activenetwerx.local>

I looked through the wiki but I couldn't find an explanation as to what you accomplish
with the domain tags. When you alias a domain from the directory, what functionality
does this provide?

Thanks,
jlc


From mgg at giagnocavo.net  Sun May  8 20:40:15 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sun, 8 May 2011 12:40:15 -0400
Subject: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin
In-Reply-To: <A4544360CBA2AC4298EE6AA1350E53AEC8A5B5@WNRCOLEX01.wnr.internal>
References: <A4544360CBA2AC4298EE6AA1350E53AEC8A5B5@WNRCOLEX01.wnr.internal>
Message-ID: <03351FCC6082174C8534AB714B8258A5AAFECA84@mse17be1.mse17.exchange.ms>

ILoadNotificationPlugin is probably what you want, as it'll get notified that the assembly is being initialized for FS.

I think you're right about creating new ManagedSession objects, but I don't recall exactly at this moment.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Schenk, Oliver
Sent: Monday, May 02, 2011 10:03 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin

Hi All,

I'm completely new to FreeSWITCH, but in a week I managed to achieve quite a bit. It was quite a learning curve to say the least. This is my current setup:


-          FreeSWITCH 1.0.7 built from tarball.

-          Developing on Windows XP with VS2008 in C#.

-          Configured extension 1024 to connect to a managed dll file, which implements IAppPlugin.

-          Using X-Lite softphone I can connect to FreeSWITCH as user 1001 and dial 1024 and start going through a menu that I built in C#. It successfully retrieves records from an SQL database and so forth. No problems there.

My first step was to create a module that gets called whenever a user dials IN bound. They will hear the menu just described.


My struggle now is relating to OUT bound calls. What I want is a module that is started as soon as FreeSWITCH is started and begins executing on an endless processing loop in the background. This will continuously monitor a database and if certain conditions occur an outbound call should be queued and then made. If multiple calls need to be made I guess they will be queued and processed one by one. I can handle the queuing part.

At this stage I will be testing using extension 1001 as the receiver of the call using my softphone.


Question 1:

I've been trying to use a class that implements IApiPlugin, but how do I get it to start when FreeSWITCH starts? As I said it should simply be a never ending thread as long as FreeSWITCH is running. I can't find any information regarding how to "Execute" a managed module immediately when FreeSWITCH has started.


Question 2:

If IApiPlugin can do this, how do I get the session object? Like this?

ManagedSession session = new ManagedSession();
session.Originate(???);

I can't find any help at all on this.


Thanks very much!



Oliver Schenk


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed.
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From gmaruzz at gmail.com  Sun May  8 21:08:09 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 8 May 2011 19:08:09 +0200
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
In-Reply-To: <1304777571644-6340156.post@n2.nabble.com>
References: <1304777571644-6340156.post@n2.nabble.com>
Message-ID: <BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com>

On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> I wondered if this ?http://developer.skype.com/public/skypekit SkypeKit ?will
> help making mod_skypopen more useable. Has any FS developers here gotten a
> chance to take a look at ?http://developer.skype.com/public/skypekit
> SkypeKit ?and tries to incorporate into FS for mod_skypopen to replace the
> need for a Skype client?

Seems very difficult to have that kit. I (and others) have asked for
it, without success. Btw, I had a deep discussion with another
developer (outside FS community) with access to that kit, and we
concurred it's not usable for our purpose. It's tageted to single
call, for embedded devices usage (we need multiple concurrent calls
for FS).

-giovanni
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at gmail.com  Sun May  8 21:11:36 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 8 May 2011 19:11:36 +0200
Subject: [Freeswitch-users] Skypopen: start_skype_clients script
	permissions
In-Reply-To: <254712.55188.qm@web59401.mail.ac4.yahoo.com>
References: <254712.55188.qm@web59401.mail.ac4.yahoo.com>
Message-ID: <BANLkTi=ZXBwHZTr112KjsiRyCU8dLrRbUg@mail.gmail.com>

On Sat, May 7, 2011 at 1:07 AM, Chris Monkey <ckmonkey158 at yahoo.com> wrote:

> I am curious though, why must skype run as root? I'd prefer that it didn't, but I can't quite get it working when run as a normal user. I've given all rw permissions to /dev/dsp and the skype-clients-configuration-dir, which allows me to load mod_skypeopen and a skype call to ring in, but as soon as I answer, skypopen reports a hangup with normal clearing.
>
> Am I missing a permission somewhere else, or is there a fundamental reason why skype must be root for this to work?


Hi Chris,

no, there is no any reason I know of.

Can you double check if you got all permission right? Also, the
permissions to reach that  configuration directory and the /dev/dsp.

Which OS-distro are you using?

-giovanni


--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From juanito1982 at gmail.com  Sun May  8 21:16:18 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Sun, 8 May 2011 19:16:18 +0200
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <4DC67898.2090106@coppice.org>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
	<4DC67898.2090106@coppice.org>
Message-ID: <BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>

How much reliable FS + mod_spandsp is compared to other solutions (open
source or not)?

2011/5/8 Steve Underwood <steveu at coppice.org>

> On 05/07/2011 04:02 AM, Andrew Keil wrote:
> > Steve,
> >
> > Thanks for your response.
> >
> > Further clarification on my part:
> >
> > 1) 20,000 to 30,000 pages per day to be sent out.
> So, this is fairly small scale. A single E1 will do fine.
> > 2) It will be an e-mail to fax style gateway (not fax to e-mail since
> that would involve inbound faxes)
> That covers a few requirements, depending what you expect to be in the
> e-mails - send the whole e-mail as a FAX; extract a PDF, Word document,
> etc. from an e-mail, and turn that into a FAX; and so on.
> > 3) The reason I asked about e-mail to PDF is the initial comments from my
> client requested the ability to send PDFs and WORD documents (I guess from
> attachments to the original e-mail), I understand the format that gets faxed
> should be TIFF so I saw on the freeswitch wiki email2pdf mentioned.  Then
> ImageMagick can help get a PDF to TIFF.
> If the incoming e-mails are limited to ones containing PDFs and doc
> files to be extracted and turned into FAXes things you seem to have a
> fairly well defined requirement. OpenOffice can be used to turn the doc
> files into FAXable images, but I am not clear how well the newer docx
> files are handled. Ghostscript can be used to turn PDFs into TIFF files.
>
> Avoid ImageMagick for this kind of work. It uses Ghostscript to do the
> hard work, but it doesn't get the best from it. If you use Ghostscript
> directly you get better control, and you can achieve good results.
> > Can I ask some more questions:
> >
> > Q1) Based on your experience what would be the average time (in seconds)
> to send a single fax page (TIFF file) via Freeswitch&  Sangoma TDM?  You can
> quote a TIFF file size to make it more accurate.  From there I should be
> able to do the math to calculate my Client's requirements better.
> The time per page depends a lot on its complexity. I use a torture test
> file with images that take half an hour to send. Typical office work is
> probably 20s per page at 14400bps.
> > Q2) Running on CentOS and using mod_spandsp/Freeswitch&  Sangoma TDM what
> percentage CPU usage would I expect to see if 30 concurrent faxes are being
> sent at the same time (ie. All channels of my E1 are faxing)?  (The hardware
> would be a new 1U rack server from a major hardware vendor)
> The greater part of the CPU load is likely to be what you didn't list
> there - the processing from e-mail to FAXable TIFF files. A single E1 of
> FAXing is a really low load these days, though.
> > Q3) What version of CentOS 5.x would you recommend?  Would the latest
> version 5.6 be fine?
> 5.6 is fine.
> > Q4) From memory there used to be different fax quality modes on fax
> machines (STANDARD, FINE&  SUPER FINE or something like that).  Is it
> possible to set the fax send quality from mod_spandsp (also can you provide
> an example)?  If this is the case could you also answer question (Q1) based
> on the different fax send quality modes.
> You can't really set the quality in mod_spandsp. The quality follows the
> TIFF files to be sent. You can, however, select the image quality as you
> generate the TIFF files in Ghostscript. STANDARD, FINE and SUPERFINE are
> the right names, although many machines refuse to use SUPERFINE.
> > Q5) From exisiting deployments of Freeswitch using mod_spandsp (&  Sagoma
> TDM cards (although this is not critical)) what is the largest number of
> concurrent outbound faxes done on a single box that you know of?
> I'm not sure of the biggest, but an E1 of FAXing is pretty small volume
> these days. The biggest number of channels should be in the hundreds.
> > I appreciate your feedback and experience.  It sounds like this will work
> fine with the mod_spandsp/Freeswitch&  Sangoma TDM combination on CentOS.
> >
> > I will most likely go for two servers with at least 2 x E1s in each, that
> way I future proof it a little and add redundancy.  Plus my Client can start
> using Freeswitch for Voice related services also.  On my side I can also
> test faxes going out on the first server and via a cross-over cable I can
> terminate them on the second server (for testing) - much nicer.
> >
> > Thanks again,
> >
> > Andrew
> Two servers with one E1 in each sounds like more than enough to meet
> your needs, unless your 30k pages per day occur over a fairly short
> working hours, and the customer demands rapid delivery. Then you might
> need more channels to deal with rush hour.
>
> 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 infos at madovsky.org  Sun May  8 21:21:36 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 8 May 2011 13:21:36 -0400
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
References: <1304777571644-6340156.post@n2.nabble.com>
	<BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com>
Message-ID: <970051C748874993B176975D6B1FEC4C@e1705>

Skype sucks,
they are scared ;)

----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, May 08, 2011 1:08 PM
Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?


On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com> 
wrote:
> I wondered if this http://developer.skype.com/public/skypekit SkypeKit 
> will
> help making mod_skypopen more useable. Has any FS developers here gotten a
> chance to take a look at http://developer.skype.com/public/skypekit
> SkypeKit and tries to incorporate into FS for mod_skypopen to replace the
> need for a Skype client?

Seems very difficult to have that kit. I (and others) have asked for
it, without success. Btw, I had a deep discussion with another
developer (outside FS community) with access to that kit, and we
concurred it's not usable for our purpose. It's tageted to single
call, for embedded devices usage (we need multiple concurrent calls
for FS).

-giovanni
-- 
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 



From kris at livecall.com  Sun May  8 22:02:17 2011
From: kris at livecall.com (Kris)
Date: Sun, 8 May 2011 11:02:17 -0700
Subject: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin
References: <A4544360CBA2AC4298EE6AA1350E53AEC8A5B5@WNRCOLEX01.wnr.internal>
	<03351FCC6082174C8534AB714B8258A5AAFECA84@mse17be1.mse17.exchange.ms>
Message-ID: <CA773815B29046979DB0AD193ED8A27B@stor1>

You can launch a separate thread...using delegate void Listener();
I haven't done outcalling yet, but my guess is to create a new Session, do 
the bridge...and transfer the session to the extension(could start the 
bridge here too) that launches the App with the script to play in it's own 
thread.

If you get it going, could you send me some of that code...?

Kris
public class Loading : FreeSWITCH.ILoadNotificationPlugin

{

delegate void Listener();

public Loading()

{

Thread currentThread = Thread.CurrentThread;

string threadName = "Thread Loading ";

currentThread.Name = threadName;

Event loading_event = new Event("CUSTOM", "livematch::maintenance");

loading_event.AddHeader("Action", "LiveMatchLoading");

loading_event.Fire();

Log.WriteLine(LogLevel.Info, "Loading constructor. ");

freeswitch.msleep(200);

}

public bool Load()

{

Log.WriteLine(LogLevel.Info, "Load() called.");

ManagedApplicationBase.LoadSystemSettings(); //from the Database

// EventConsumers EventsLivematch = new EventConsumers(10);//<<< INCREASE to 
add more events

ManagedApplicationBase.EventsLivematch.Add("CUSTOM", 
"livematch::maintenance");

ManagedApplicationBase.EventsLivematch.Add("HEARTBEAT", "");

ManagedApplicationBase.EventsLivematch.Add("CHANNEL_HANGUP", "");

ManagedApplicationBase.EventsLivematch.Add("SHUTDOWN", "");

// ManagedApplicationBase.EventsLivematch.Add("DTMF", "");

new 
Listener(ManagedApplicationBase.LiveMatchMaintenanceEvents).BeginInvoke(null, 
null);

//subscribe to events

// EventConsumers EventsConference = new EventConsumers(4);//<<< INCREASE to 
add more events

ManagedApplicationBase.EventsConference.Add("CUSTOM", 
"conference::maintenance");

ManagedApplicationBase.EventsConference.Add("CUSTOM", 
"livematch::maintenance"); //to catch loading event so we can exit loop

// ManagedApplicationBase.EventsConference.Add("CUSTOM", 
"multicast::event");

ManagedApplicationBase.EventsConference.Add("SHUTDOWN", "");

new 
Listener(ManagedApplicationBase.ConferenceMaintenanceEvents).BeginInvoke(null, 
null);

Log.WriteLine(LogLevel.Info, "Load() ending.");

return true;

}

}



public static void ConferenceMaintenanceEvents()//runs only in the new 
thread

{

Thread.CurrentThread.Name = "Thread ConferenceMaintenanceEvents";

BaseLog.WriteLine(BaseLogLevel.Info, "Starting 
ConferenceMaintenanceEvents");

string EventName;

string SessionUUID;

FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();

string DigitPressedByMember;

string xml_list = fsApi.Execute("conference", "xml_list");

BaseLog.WriteLine(BaseLogLevel.Info, "Conferences:" + xml_list);

while (!ShuttingDown&&!FreeSwitchShutdown && !LoadingNewDLL)

{

//do DB lookups, outcall here

using (Event ev_all = EventsConference.Pop())

{

if (ev_all != null)

{

}

}

}

----- Original Message ----- 
From: "Michael Giagnocavo" <mgg at giagnocavo.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, May 08, 2011 9:40 AM
Subject: Re: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin


ILoadNotificationPlugin is probably what you want, as it'll get notified 
that the assembly is being initialized for FS.

I think you're right about creating new ManagedSession objects, but I don't 
recall exactly at this moment.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org 
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Schenk, 
Oliver
Sent: Monday, May 02, 2011 10:03 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] FreeSWITCH, mod_managed and IApiPlugin

Hi All,

I'm completely new to FreeSWITCH, but in a week I managed to achieve quite a 
bit. It was quite a learning curve to say the least. This is my current 
setup:


-          FreeSWITCH 1.0.7 built from tarball.

-          Developing on Windows XP with VS2008 in C#.

-          Configured extension 1024 to connect to a managed dll file, which 
implements IAppPlugin.

-          Using X-Lite softphone I can connect to FreeSWITCH as user 1001 
and dial 1024 and start going through a menu that I built in C#. It 
successfully retrieves records from an SQL database and so forth. No 
problems there.

My first step was to create a module that gets called whenever a user dials 
IN bound. They will hear the menu just described.


My struggle now is relating to OUT bound calls. What I want is a module that 
is started as soon as FreeSWITCH is started and begins executing on an 
endless processing loop in the background. This will continuously monitor a 
database and if certain conditions occur an outbound call should be queued 
and then made. If multiple calls need to be made I guess they will be queued 
and processed one by one. I can handle the queuing part.

At this stage I will be testing using extension 1001 as the receiver of the 
call using my softphone.


Question 1:

I've been trying to use a class that implements IApiPlugin, but how do I get 
it to start when FreeSWITCH starts? As I said it should simply be a never 
ending thread as long as FreeSWITCH is running. I can't find any information 
regarding how to "Execute" a managed module immediately when FreeSWITCH has 
started.


Question 2:

If IApiPlugin can do this, how do I get the session object? Like this?

ManagedSession session = new ManagedSession();
session.Originate(???);

I can't find any help at all on this.


Thanks very much!



Oliver Schenk


NOTICE - This e-mail and any files transmitted with it are confidential and 
are only for the use of the person to whom they are addressed.
If you are not the intended recipient then you have received this e-mail in 
error; please advise us immediately if this is the case.
Any views expressed in this message are those of the individual sender, 
except where the sender specifically states them to be the views of WestNet 
Rail.



From shamun.toha at gmail.com  Mon May  9 01:14:21 2011
From: shamun.toha at gmail.com (Shamun)
Date: Sun, 8 May 2011 23:14:21 +0200
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
In-Reply-To: <BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com>
References: <1304777571644-6340156.post@n2.nabble.com>
	<BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com>
Message-ID: <BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com>

Giovanni,

Respect guru!!.

Why FreeSWITCH, server is not a embedded server to Skype theory/logic? If we
put FreeSwitch in ATOM or ARM or 64 bit server, its not embedded? What is
there definition of embedded? Skype really is a crazy company, underestimate
FreeSwitch and your knowledge (you did genius job, job well done).

Reg
Shamun



On Sun, May 8, 2011 at 7:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com>
> wrote:
> > I wondered if this  http://developer.skype.com/public/skypekit SkypeKit
>  will
> > help making mod_skypopen more useable. Has any FS developers here gotten
> a
> > chance to take a look at  http://developer.skype.com/public/skypekit
> > SkypeKit  and tries to incorporate into FS for mod_skypopen to replace
> the
> > need for a Skype client?
>
> Seems very difficult to have that kit. I (and others) have asked for
> it, without success. Btw, I had a deep discussion with another
> developer (outside FS community) with access to that kit, and we
> concurred it's not usable for our purpose. It's tageted to single
> call, for embedded devices usage (we need multiple concurrent calls
> for FS).
>
> -giovanni
> --
> 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 admin at blindi.net  Mon May  9 03:37:52 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 9 May 2011 01:37:52 +0200 (CEST)
Subject: [Freeswitch-users] Can.t find a download for astconf2fsconf
In-Reply-To: <C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
	<BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
	<C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
Message-ID: <Pine.LNX.4.64.1105090135330.6782@debian>

Hi all,
a find the url:
http://wiki.freeswitch.org/wiki/Bounty_challenged
I don.t find a downloadlink for this tool
Do your have a downloadlink please?
Thankx


---------------
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 riedinger at sns.eu  Mon May  9 03:39:03 2011
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 09 May 2011 01:39:03 +0200
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
Message-ID: <4DC72997.4090900@sns.eu>

Hi Dirk,

Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein 
Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.

Viele Gr??e
     Jan

Am 06.05.2011 16:25, schrieb Max Clark:
> Hello all,
>
> Recent developments (or absolute lack of) within the CentOS project
> and its perceived long term viability has forced an internal
> discussion to select a successor distribution. The most likely
> candidates at this point are Ubuntu and its LTS releases for servers,
> and Scientific Linux with the new 6.x releases. The pro/con lists for
> each are growing and the issue is complicated.
>
> With CentOS 5.x being the reference distro for FreeSWITCH development
> I'm curious if this conversation has started among the FreeSWITCH
> developers, and if it has, what is the project leaning to for a
> successor distribution?
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From rhuddleston at gmail.com  Mon May  9 04:09:47 2011
From: rhuddleston at gmail.com (rhuddleston at gmail.com)
Date: Sun, 8 May 2011 20:09:47 -0400
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <4DC72997.4090900@sns.eu>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<4DC72997.4090900@sns.eu>
Message-ID: <6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>

Auf English bitte? Mein deutsch ist sehr sclecht und ich kann nicht verstehen

On May 8, 2011, at 7:39 PM, Jan Riedinger <riedinger at sns.eu> wrote:

> Hi Dirk,
> 
> Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein 
> Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.
> 
> Viele Gr??e
>     Jan
> 
> Am 06.05.2011 16:25, schrieb Max Clark:
>> Hello all,
>> 
>> Recent developments (or absolute lack of) within the CentOS project
>> and its perceived long term viability has forced an internal
>> discussion to select a successor distribution. The most likely
>> candidates at this point are Ubuntu and its LTS releases for servers,
>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>> each are growing and the issue is complicated.
>> 
>> With CentOS 5.x being the reference distro for FreeSWITCH development
>> I'm curious if this conversation has started among the FreeSWITCH
>> developers, and if it has, what is the project leaning to for a
>> successor distribution?
>> 
>> Thanks,
>> Max
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
> 
> AG Charlottenburg - HRB 71973
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 May  9 04:18:04 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 09 May 2011 08:18:04 +0800
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>	<4DC40C86.9000503@coppice.org>	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>	<4DC67898.2090106@coppice.org>
	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
Message-ID: <4DC732BC.9070206@coppice.org>

On 05/09/2011 01:16 AM, Juan Antonio Iba?ez Santorum wrote:
> How much reliable FS + mod_spandsp is compared to other solutions 
> (open source or not)?
The open source options are:

     Asterisk + spandsp
     Asterisk + Hylafax + iaxmodem + spandsp
     Freeswitch

These are all tested and proven to give well below 1% failures, even 
with quite a lot of concurrent FAX channels in use, if things are set up 
well. They can give you bad failure rates if things are not set up well. 
I believe that right now you will have less trouble achieving a reliable 
setup with Freeswitch. Going forward, most of my effort goes into making 
the Freeswitch option the most thoroughly implemented one.

The main commercial option is:

     Asterisk + Digium's commercial FAX

Of course, there are numerous other fully commercial FAXing options 
which could be used in conjunction with things like Asterisk or Freeswitch

The Digium FAX module is based on the well known Commetrex FAX engine, 
which is widely deployed, and should be capable of solid results. 
However, the module is more than just the core FAX engine, and some 
people do have serious trouble with the module. I have helped moved 
people off this, and onto Asterisk + spandsp, to improve their 
reliability. In a couple of those cases people were getting quite a lot 
of pages cut short when receiving FAXes with T.38, even though a 
wireshark log showed a perfect exchange, from which I could correctly 
decode these FAXes. The module was not reporting any errors. In a couple 
of cases strange machines were sending weird things the Digium FAX 
didn't cope with very well. I worked with these people to make sure 
spandsp did handle the weird stuff well, and we ended up with a more 
usable solution. These people told me that when they complained to 
Digium they got little help. The best was an offer of a refund. Paying 
to get some support didn't seem to work out too well for these people, 
but I guess if the support you are looking for is mostly in getting 
things configured and working on day one you might get value for money.

All these solutions require reliable signaling and reliable media 
timing, and many people have setups which cannot achieve that. Most 
people don't understand how things work, and will claim a solution 
doesn't function for spurious reasons. For example, a number of people 
say the spandsp module for Asterisk doesn't work, because they keep 
getting a 488 response. That response has nothing to do with the FAX 
engine. It is a negotiation error that occurs outside the FAX engine. If 
they fixed their configuration the error would go away. However, many 
just move on, having "proven" to themselves the solution doesn't work.


> 2011/5/8 Steve Underwood <steveu at coppice.org <mailto:steveu at coppice.org>>
>
>     On 05/07/2011 04:02 AM, Andrew Keil wrote:
>     > Steve,
>     >
>     > Thanks for your response.
>     >
>     > Further clarification on my part:
>     >
>     > 1) 20,000 to 30,000 pages per day to be sent out.
>     So, this is fairly small scale. A single E1 will do fine.
>     > 2) It will be an e-mail to fax style gateway (not fax to e-mail
>     since that would involve inbound faxes)
>     That covers a few requirements, depending what you expect to be in the
>     e-mails - send the whole e-mail as a FAX; extract a PDF, Word
>     document,
>     etc. from an e-mail, and turn that into a FAX; and so on.
>     > 3) The reason I asked about e-mail to PDF is the initial
>     comments from my client requested the ability to send PDFs and
>     WORD documents (I guess from attachments to the original e-mail),
>     I understand the format that gets faxed should be TIFF so I saw on
>     the freeswitch wiki email2pdf mentioned.  Then ImageMagick can
>     help get a PDF to TIFF.
>     If the incoming e-mails are limited to ones containing PDFs and doc
>     files to be extracted and turned into FAXes things you seem to have a
>     fairly well defined requirement. OpenOffice can be used to turn
>     the doc
>     files into FAXable images, but I am not clear how well the newer docx
>     files are handled. Ghostscript can be used to turn PDFs into TIFF
>     files.
>
>     Avoid ImageMagick for this kind of work. It uses Ghostscript to do the
>     hard work, but it doesn't get the best from it. If you use Ghostscript
>     directly you get better control, and you can achieve good results.
>     > Can I ask some more questions:
>     >
>     > Q1) Based on your experience what would be the average time (in
>     seconds) to send a single fax page (TIFF file) via Freeswitch&
>      Sangoma TDM?  You can quote a TIFF file size to make it more
>     accurate.  From there I should be able to do the math to calculate
>     my Client's requirements better.
>     The time per page depends a lot on its complexity. I use a torture
>     test
>     file with images that take half an hour to send. Typical office
>     work is
>     probably 20s per page at 14400bps.
>     > Q2) Running on CentOS and using mod_spandsp/Freeswitch&  Sangoma
>     TDM what percentage CPU usage would I expect to see if 30
>     concurrent faxes are being sent at the same time (ie. All channels
>     of my E1 are faxing)?  (The hardware would be a new 1U rack server
>     from a major hardware vendor)
>     The greater part of the CPU load is likely to be what you didn't list
>     there - the processing from e-mail to FAXable TIFF files. A single
>     E1 of
>     FAXing is a really low load these days, though.
>     > Q3) What version of CentOS 5.x would you recommend?  Would the
>     latest version 5.6 be fine?
>     5.6 is fine.
>     > Q4) From memory there used to be different fax quality modes on
>     fax machines (STANDARD, FINE&  SUPER FINE or something like that).
>      Is it possible to set the fax send quality from mod_spandsp (also
>     can you provide an example)?  If this is the case could you also
>     answer question (Q1) based on the different fax send quality modes.
>     You can't really set the quality in mod_spandsp. The quality
>     follows the
>     TIFF files to be sent. You can, however, select the image quality
>     as you
>     generate the TIFF files in Ghostscript. STANDARD, FINE and
>     SUPERFINE are
>     the right names, although many machines refuse to use SUPERFINE.
>     > Q5) From exisiting deployments of Freeswitch using mod_spandsp
>     (&  Sagoma TDM cards (although this is not critical)) what is the
>     largest number of concurrent outbound faxes done on a single box
>     that you know of?
>     I'm not sure of the biggest, but an E1 of FAXing is pretty small
>     volume
>     these days. The biggest number of channels should be in the hundreds.
>     > I appreciate your feedback and experience.  It sounds like this
>     will work fine with the mod_spandsp/Freeswitch&  Sangoma TDM
>     combination on CentOS.
>     >
>     > I will most likely go for two servers with at least 2 x E1s in
>     each, that way I future proof it a little and add redundancy.
>      Plus my Client can start using Freeswitch for Voice related
>     services also.  On my side I can also test faxes going out on the
>     first server and via a cross-over cable I can terminate them on
>     the second server (for testing) - much nicer.
>     >
>     > Thanks again,
>     >
>     > Andrew
>     Two servers with one E1 in each sounds like more than enough to meet
>     your needs, unless your 30k pages per day occur over a fairly short
>     working hours, and the customer demands rapid delivery. Then you might
>     need more channels to deal with rush hour.
>
>     Steve
>
Steve



From steveu at coppice.org  Mon May  9 04:35:47 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 09 May 2011 08:35:47 +0800
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <4DC732BC.9070206@coppice.org>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>	<4DC40C86.9000503@coppice.org>	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>	<4DC67898.2090106@coppice.org>	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
	<4DC732BC.9070206@coppice.org>
Message-ID: <4DC736E3.2000200@coppice.org>

On 05/09/2011 08:18 AM, Steve Underwood wrote:
> On 05/09/2011 01:16 AM, Juan Antonio Iba?ez Santorum wrote:
>> How much reliable FS + mod_spandsp is compared to other solutions
>> (open source or not)?
> The open source options are:
>
>       Asterisk + spandsp
>       Asterisk + Hylafax + iaxmodem + spandsp
>       Freeswitch
>
> These are all tested and proven to give well below 1% failures, even
> with quite a lot of concurrent FAX channels in use, if things are set up
> well. They can give you bad failure rates if things are not set up well.
> I believe that right now you will have less trouble achieving a reliable
> setup with Freeswitch. Going forward, most of my effort goes into making
> the Freeswitch option the most thoroughly implemented one.
>
> The main commercial option is:
>
>       Asterisk + Digium's commercial FAX
>
> Of course, there are numerous other fully commercial FAXing options
> which could be used in conjunction with things like Asterisk or Freeswitch
>
> The Digium FAX module is based on the well known Commetrex FAX engine,
> which is widely deployed, and should be capable of solid results.
> However, the module is more than just the core FAX engine, and some
> people do have serious trouble with the module. I have helped moved
> people off this, and onto Asterisk + spandsp, to improve their
> reliability. In a couple of those cases people were getting quite a lot
> of pages cut short when receiving FAXes with T.38, even though a
> wireshark log showed a perfect exchange, from which I could correctly
> decode these FAXes. The module was not reporting any errors. In a couple
> of cases strange machines were sending weird things the Digium FAX
> didn't cope with very well. I worked with these people to make sure
> spandsp did handle the weird stuff well, and we ended up with a more
> usable solution. These people told me that when they complained to
> Digium they got little help. The best was an offer of a refund. Paying
> to get some support didn't seem to work out too well for these people,
> but I guess if the support you are looking for is mostly in getting
> things configured and working on day one you might get value for money.
>
> All these solutions require reliable signaling and reliable media
> timing, and many people have setups which cannot achieve that. Most
> people don't understand how things work, and will claim a solution
> doesn't function for spurious reasons. For example, a number of people
> say the spandsp module for Asterisk doesn't work, because they keep
> getting a 488 response. That response has nothing to do with the FAX
> engine. It is a negotiation error that occurs outside the FAX engine. If
> they fixed their configuration the error would go away. However, many
> just move on, having "proven" to themselves the solution doesn't work.
>
I forgot one important point. Right now, the only out of the box 
solution for T.38 gateway is Freeswitch. There are patches in the Digium 
bugtracker to add T.38 gateway functionality, but its not in the 
software distribution. They seem to be seriously working on T.38 gateway 
for a future release of Asterisk. I don't know if there will be a 
commercial T.38 gateway from Digium. It looks like all their current 
work is with spandsp.

Steve



From vivid333 at 163.com  Mon May  9 07:12:13 2011
From: vivid333 at 163.com (vivid)
Date: Mon, 09 May 2011 11:12:13 +0800
Subject: [Freeswitch-users] How to eliminate acoustic echo from server
	side(FreeSwitch)
Message-ID: <4DC75B8D.9090008@163.com>



  In the multi-party conference, because some termianl phone does not 
process echo, which causes the call have echo, so SERVER side needs to 
do echo processing. How to eliminate echo in this scenario?

        Do I need to add hareware device? If I want to implement it 
using software competely, what kind of open source echo library can be used?
        Does SPEEX work in Server Side? So many VIOP termianl use SPEEX, 
how to use it in the server side?

        Processing echo in server side should be different from client 
side, there are some complicated secnarios shuch as network delay and 
package lost, how to deal with these? Any advice will be appriciated.



From curriegrad2004 at gmail.com  Mon May  9 08:01:41 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 8 May 2011 21:01:41 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<4DC72997.4090900@sns.eu>
	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
Message-ID: <BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>

It would be beneficial to the majority of the users if we continued
this discussion in English

On Sun, May 8, 2011 at 5:09 PM,  <rhuddleston at gmail.com> wrote:
> Auf English bitte? Mein deutsch ist sehr sclecht und ich kann nicht verstehen
>
> On May 8, 2011, at 7:39 PM, Jan Riedinger <riedinger at sns.eu> wrote:
>
>> Hi Dirk,
>>
>> Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein
>> Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.
>>
>> Viele Gr??e
>> ? ? Jan
>>
>> Am 06.05.2011 16:25, schrieb Max Clark:
>>> Hello all,
>>>
>>> Recent developments (or absolute lack of) within the CentOS project
>>> and its perceived long term viability has forced an internal
>>> discussion to select a successor distribution. The most likely
>>> candidates at this point are Ubuntu and its LTS releases for servers,
>>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>>> each are growing and the issue is complicated.
>>>
>>> With CentOS 5.x being the reference distro for FreeSWITCH development
>>> I'm curious if this conversation has started among the FreeSWITCH
>>> developers, and if it has, what is the project leaning to for a
>>> successor distribution?
>>>
>>> Thanks,
>>> Max
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger ? ? ? ? ? ? ? ? ? ? ? ? ? Phone : ?+49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director ? ? ? ? ?Fax ? : ?+49-30-39 73 19 64
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? E-Mail: ?riedinger at sns.eu
>> SNS Consult GmbH ? ? ? ? ? ? ? ? ? ? ? ?ICQ ? : ?163-237-041
>> S?dwestkorso 49a ? ? ? ? ? ? ? ? ? ? ? ?MSN ? : ?jan at sns-consult.de
>> 14197 Berlin GERMANY ? ? ? ? ? ? ? ? ? ?Skype : ?Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Mon May  9 08:16:29 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 9 May 2011 00:16:29 -0400
Subject: [Freeswitch-users] Moving on from CentOS
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com><4DC72997.4090900@sns.eu><6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
Message-ID: <610EA61937E646A59C183FC013A88D21@e1705>

bah, deutsch is beautiful language :)
copy and paste in google translate, it makes miracles :D

----- Original Message ----- 
From: "curriegrad2004" <curriegrad2004 at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, May 09, 2011 12:01 AM
Subject: Re: [Freeswitch-users] Moving on from CentOS


> It would be beneficial to the majority of the users if we continued
> this discussion in English
>
> On Sun, May 8, 2011 at 5:09 PM,  <rhuddleston at gmail.com> wrote:
>> Auf English bitte? Mein deutsch ist sehr sclecht und ich kann nicht 
>> verstehen
>>
>> On May 8, 2011, at 7:39 PM, Jan Riedinger <riedinger at sns.eu> wrote:
>>
>>> Hi Dirk,
>>>
>>> Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein
>>> Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.
>>>
>>> Viele Gr??e
>>> Jan
>>>
>>> Am 06.05.2011 16:25, schrieb Max Clark:
>>>> Hello all,
>>>>
>>>> Recent developments (or absolute lack of) within the CentOS project
>>>> and its perceived long term viability has forced an internal
>>>> discussion to select a successor distribution. The most likely
>>>> candidates at this point are Ubuntu and its LTS releases for servers,
>>>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>>>> each are growing and the issue is complicated.
>>>>
>>>> With CentOS 5.x being the reference distro for FreeSWITCH development
>>>> I'm curious if this conversation has started among the FreeSWITCH
>>>> developers, and if it has, what is the project leaning to for a
>>>> successor distribution?
>>>>
>>>> Thanks,
>>>> Max
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> --
>>> Jan Riedinger Phone : +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
>>> E-Mail: riedinger at sns.eu
>>> SNS Consult GmbH ICQ : 163-237-041
>>> S?dwestkorso 49a MSN : jan at sns-consult.de
>>> 14197 Berlin GERMANY Skype : Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From andrew.keil at askinteractive.net  Mon May  9 09:34:39 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Mon, 9 May 2011 15:34:39 +1000
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <4DC732BC.9070206@coppice.org>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
	<4DC67898.2090106@coppice.org>
	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
	<4DC732BC.9070206@coppice.org>
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE658284AF10F@ASKSERVER.asksbs.local>

Steve,

I very much appreciate your detailed reply and comments.  They have enabled me to feel very confident in proposing FREESWITCH (using mod_spandsp & Sangoma E1 cards) to run a FAX OUT service for my Client.

There is only one last question I have.  My Client has requested that the initial installation runs on Windows Server 2008.  Do you see any issues with the Windows version of FreeSWITCH using mod_spandsp (based on the service design I have already sent through)?  

I know the Sangoma A10x series of E1 cards do support Windows (via FreeTDM) so I do not see any issue there.

Once again thanks so much for all your feedback.

Kind Regards,

Andrew Keil


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood
Sent: Monday, 9 May 2011 10:18 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Major deployment of outbound FAX on latest version of Freeswitch question

On 05/09/2011 01:16 AM, Juan Antonio Iba?ez Santorum wrote:
> How much reliable FS + mod_spandsp is compared to other solutions 
> (open source or not)?
The open source options are:

     Asterisk + spandsp
     Asterisk + Hylafax + iaxmodem + spandsp
     Freeswitch

These are all tested and proven to give well below 1% failures, even with quite a lot of concurrent FAX channels in use, if things are set up well. They can give you bad failure rates if things are not set up well. 
I believe that right now you will have less trouble achieving a reliable setup with Freeswitch. Going forward, most of my effort goes into making the Freeswitch option the most thoroughly implemented one.

The main commercial option is:

     Asterisk + Digium's commercial FAX

Of course, there are numerous other fully commercial FAXing options which could be used in conjunction with things like Asterisk or Freeswitch

The Digium FAX module is based on the well known Commetrex FAX engine, which is widely deployed, and should be capable of solid results. 
However, the module is more than just the core FAX engine, and some people do have serious trouble with the module. I have helped moved people off this, and onto Asterisk + spandsp, to improve their reliability. In a couple of those cases people were getting quite a lot of pages cut short when receiving FAXes with T.38, even though a wireshark log showed a perfect exchange, from which I could correctly decode these FAXes. The module was not reporting any errors. In a couple of cases strange machines were sending weird things the Digium FAX didn't cope with very well. I worked with these people to make sure spandsp did handle the weird stuff well, and we ended up with a more usable solution. These people told me that when they complained to Digium they got little help. The best was an offer of a refund. Paying to get some support didn't seem to work out too well for these people, but I guess if the support you are looking for is mostly in getting things configured and working on day one you might get value for money.

All these solutions require reliable signaling and reliable media timing, and many people have setups which cannot achieve that. Most people don't understand how things work, and will claim a solution doesn't function for spurious reasons. For example, a number of people say the spandsp module for Asterisk doesn't work, because they keep getting a 488 response. That response has nothing to do with the FAX engine. It is a negotiation error that occurs outside the FAX engine. If they fixed their configuration the error would go away. However, many just move on, having "proven" to themselves the solution doesn't work.


> 2011/5/8 Steve Underwood <steveu at coppice.org 
> <mailto:steveu at coppice.org>>
>
>     On 05/07/2011 04:02 AM, Andrew Keil wrote:
>     > Steve,
>     >
>     > Thanks for your response.
>     >
>     > Further clarification on my part:
>     >
>     > 1) 20,000 to 30,000 pages per day to be sent out.
>     So, this is fairly small scale. A single E1 will do fine.
>     > 2) It will be an e-mail to fax style gateway (not fax to e-mail
>     since that would involve inbound faxes)
>     That covers a few requirements, depending what you expect to be in the
>     e-mails - send the whole e-mail as a FAX; extract a PDF, Word
>     document,
>     etc. from an e-mail, and turn that into a FAX; and so on.
>     > 3) The reason I asked about e-mail to PDF is the initial
>     comments from my client requested the ability to send PDFs and
>     WORD documents (I guess from attachments to the original e-mail),
>     I understand the format that gets faxed should be TIFF so I saw on
>     the freeswitch wiki email2pdf mentioned.  Then ImageMagick can
>     help get a PDF to TIFF.
>     If the incoming e-mails are limited to ones containing PDFs and doc
>     files to be extracted and turned into FAXes things you seem to have a
>     fairly well defined requirement. OpenOffice can be used to turn
>     the doc
>     files into FAXable images, but I am not clear how well the newer docx
>     files are handled. Ghostscript can be used to turn PDFs into TIFF
>     files.
>
>     Avoid ImageMagick for this kind of work. It uses Ghostscript to do the
>     hard work, but it doesn't get the best from it. If you use Ghostscript
>     directly you get better control, and you can achieve good results.
>     > Can I ask some more questions:
>     >
>     > Q1) Based on your experience what would be the average time (in
>     seconds) to send a single fax page (TIFF file) via Freeswitch&
>      Sangoma TDM?  You can quote a TIFF file size to make it more
>     accurate.  From there I should be able to do the math to calculate
>     my Client's requirements better.
>     The time per page depends a lot on its complexity. I use a torture
>     test
>     file with images that take half an hour to send. Typical office
>     work is
>     probably 20s per page at 14400bps.
>     > Q2) Running on CentOS and using mod_spandsp/Freeswitch&  Sangoma
>     TDM what percentage CPU usage would I expect to see if 30
>     concurrent faxes are being sent at the same time (ie. All channels
>     of my E1 are faxing)?  (The hardware would be a new 1U rack server
>     from a major hardware vendor)
>     The greater part of the CPU load is likely to be what you didn't list
>     there - the processing from e-mail to FAXable TIFF files. A single
>     E1 of
>     FAXing is a really low load these days, though.
>     > Q3) What version of CentOS 5.x would you recommend?  Would the
>     latest version 5.6 be fine?
>     5.6 is fine.
>     > Q4) From memory there used to be different fax quality modes on
>     fax machines (STANDARD, FINE&  SUPER FINE or something like that).
>      Is it possible to set the fax send quality from mod_spandsp (also
>     can you provide an example)?  If this is the case could you also
>     answer question (Q1) based on the different fax send quality modes.
>     You can't really set the quality in mod_spandsp. The quality
>     follows the
>     TIFF files to be sent. You can, however, select the image quality
>     as you
>     generate the TIFF files in Ghostscript. STANDARD, FINE and
>     SUPERFINE are
>     the right names, although many machines refuse to use SUPERFINE.
>     > Q5) From exisiting deployments of Freeswitch using mod_spandsp
>     (&  Sagoma TDM cards (although this is not critical)) what is the
>     largest number of concurrent outbound faxes done on a single box
>     that you know of?
>     I'm not sure of the biggest, but an E1 of FAXing is pretty small
>     volume
>     these days. The biggest number of channels should be in the hundreds.
>     > I appreciate your feedback and experience.  It sounds like this
>     will work fine with the mod_spandsp/Freeswitch&  Sangoma TDM
>     combination on CentOS.
>     >
>     > I will most likely go for two servers with at least 2 x E1s in
>     each, that way I future proof it a little and add redundancy.
>      Plus my Client can start using Freeswitch for Voice related
>     services also.  On my side I can also test faxes going out on the
>     first server and via a cross-over cable I can terminate them on
>     the second server (for testing) - much nicer.
>     >
>     > Thanks again,
>     >
>     > Andrew
>     Two servers with one E1 in each sounds like more than enough to meet
>     your needs, unless your 30k pages per day occur over a fairly short
>     working hours, and the customer demands rapid delivery. Then you might
>     need more channels to deal with rush hour.
>
>     Steve
>
Steve


_______________________________________________
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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
 

__________ Information from ESET NOD32 Antivirus, version of virus signature database 6105 (20110508) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 


From michal.bielicki at seventhsignal.de  Mon May  9 10:07:55 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Mon, 09 May 2011 08:07:55 +0200
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>	<4DC72997.4090900@sns.eu>	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
Message-ID: <4DC784BB.4080907@seventhsignal.de>

I guess Jan just wanted to explain. Sometimes its better to clear up
things for people in their local language, don't you think ?

Am 09.05.2011 06:01, schrieb curriegrad2004:
> It would be beneficial to the majority of the users if we continued
> this discussion in English
>
> On Sun, May 8, 2011 at 5:09 PM,  <rhuddleston at gmail.com> wrote:
>> Auf English bitte? Mein deutsch ist sehr sclecht und ich kann nicht verstehen
>>
>> On May 8, 2011, at 7:39 PM, Jan Riedinger <riedinger at sns.eu> wrote:
>>
>>> Hi Dirk,
>>>
>>> Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein
>>> Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.
>>>
>>> Viele Gr??e
>>>     Jan
>>>
>>> Am 06.05.2011 16:25, schrieb Max Clark:
>>>> Hello all,
>>>>
>>>> Recent developments (or absolute lack of) within the CentOS project
>>>> and its perceived long term viability has forced an internal
>>>> discussion to select a successor distribution. The most likely
>>>> candidates at this point are Ubuntu and its LTS releases for servers,
>>>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>>>> each are growing and the issue is complicated.
>>>>
>>>> With CentOS 5.x being the reference distro for FreeSWITCH development
>>>> I'm curious if this conversation has started among the FreeSWITCH
>>>> developers, and if it has, what is the project leaning to for a
>>>> successor distribution?
>>>>
>>>> Thanks,
>>>> Max
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> --
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                         E-Mail:  riedinger at sns.eu
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From juanito1982 at gmail.com  Mon May  9 11:23:29 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Mon, 9 May 2011 09:23:29 +0200
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <4DC732BC.9070206@coppice.org>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
	<4DC67898.2090106@coppice.org>
	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
	<4DC732BC.9070206@coppice.org>
Message-ID: <BANLkTikS_8t81GnKGgxuDyYhYJBQEKaLKw@mail.gmail.com>

Thank you very much for your answer Steve.

Would be FS + mod_spandsp good both for incoming and outgoing faxeds?

What could be achieved to avoid media timing issues you told?

Regards


2011/5/9 Steve Underwood <steveu at coppice.org>

> On 05/09/2011 01:16 AM, Juan Antonio Iba?ez Santorum wrote:
> > How much reliable FS + mod_spandsp is compared to other solutions
> > (open source or not)?
> The open source options are:
>
>     Asterisk + spandsp
>     Asterisk + Hylafax + iaxmodem + spandsp
>     Freeswitch
>
> These are all tested and proven to give well below 1% failures, even
> with quite a lot of concurrent FAX channels in use, if things are set up
> well. They can give you bad failure rates if things are not set up well.
> I believe that right now you will have less trouble achieving a reliable
> setup with Freeswitch. Going forward, most of my effort goes into making
> the Freeswitch option the most thoroughly implemented one.
>
> The main commercial option is:
>
>     Asterisk + Digium's commercial FAX
>
> Of course, there are numerous other fully commercial FAXing options
> which could be used in conjunction with things like Asterisk or Freeswitch
>
> The Digium FAX module is based on the well known Commetrex FAX engine,
> which is widely deployed, and should be capable of solid results.
> However, the module is more than just the core FAX engine, and some
> people do have serious trouble with the module. I have helped moved
> people off this, and onto Asterisk + spandsp, to improve their
> reliability. In a couple of those cases people were getting quite a lot
> of pages cut short when receiving FAXes with T.38, even though a
> wireshark log showed a perfect exchange, from which I could correctly
> decode these FAXes. The module was not reporting any errors. In a couple
> of cases strange machines were sending weird things the Digium FAX
> didn't cope with very well. I worked with these people to make sure
> spandsp did handle the weird stuff well, and we ended up with a more
> usable solution. These people told me that when they complained to
> Digium they got little help. The best was an offer of a refund. Paying
> to get some support didn't seem to work out too well for these people,
> but I guess if the support you are looking for is mostly in getting
> things configured and working on day one you might get value for money.
>
> All these solutions require reliable signaling and reliable media
> timing, and many people have setups which cannot achieve that. Most
> people don't understand how things work, and will claim a solution
> doesn't function for spurious reasons. For example, a number of people
> say the spandsp module for Asterisk doesn't work, because they keep
> getting a 488 response. That response has nothing to do with the FAX
> engine. It is a negotiation error that occurs outside the FAX engine. If
> they fixed their configuration the error would go away. However, many
> just move on, having "proven" to themselves the solution doesn't work.
>
>
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From anton.vazir at gmail.com  Mon May  9 11:36:40 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 9 May 2011 12:36:40 +0500
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
In-Reply-To: <BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com>
References: <1304777571644-6340156.post@n2.nabble.com>
	<BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com>
	<BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com>
Message-ID: <BANLkTinAqinUyWzYNrWMJL3uzxA4Jhsf1A@mail.gmail.com>

It's a world of money. They want to control how you call, where you
call. If there will be a free software which will implement skype
calls, in large scale, using native methods... it will go out of their
control. They a still bulb, and they want to put as much air into the
bulb as possible... FS pierce holes in their bulb... ;)

2011/5/9 Shamun <shamun.toha at gmail.com>:
> Giovanni,
> Respect guru!!.
> Why FreeSWITCH, server is not a embedded server to Skype theory/logic? If we
> put FreeSwitch in ATOM or ARM or 64 bit server, its not embedded? What is
> there?definition?of embedded??Skype really is a crazy company, underestimate
> FreeSwitch and your knowledge (you did genius job, job well done).
> Reg
> Shamun
>
>
> On Sun, May 8, 2011 at 7:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com>
>> wrote:
>> > I wondered if this ?http://developer.skype.com/public/skypekit SkypeKit
>> > ?will
>> > help making mod_skypopen more useable. Has any FS developers here gotten
>> > a
>> > chance to take a look at ?http://developer.skype.com/public/skypekit
>> > SkypeKit ?and tries to incorporate into FS for mod_skypopen to replace
>> > the
>> > need for a Skype client?
>>
>> Seems very difficult to have that kit. I (and others) have asked for
>> it, without success. Btw, I had a deep discussion with another
>> developer (outside FS community) with access to that kit, and we
>> concurred it's not usable for our purpose. It's tageted to single
>> call, for embedded devices usage (we need multiple concurrent calls
>> for FS).
>>
>> -giovanni
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Mon May  9 12:04:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 9 May 2011 09:04:05 +0100
Subject: [Freeswitch-users] mod_limit Error
In-Reply-To: <48BCDAC7-7579-41AE-8372-58526BF4E320@ipeva.fr>
References: <636535.38388.qm@web34305.mail.mud.yahoo.com>
	<48BCDAC7-7579-41AE-8372-58526BF4E320@ipeva.fr>
Message-ID: <BANLkTimZQ0P5PqYhPJ4P66VtxRGn+T9ZVQ@mail.gmail.com>

There's still a shim that loads mod_hash for you with a warning though. It
shouldn't give errors...

Fernando, it sounds like you've managed to install files from 2 different
versions of FS while upgrading. I suggest you delete (or backup) your
existing install directories, delete the existing git checkout, do a new
checkout with 'git clone' and install from scratch from there.

-Steve


2011/5/8 David Ponzone <david.ponzone at ipeva.fr>

> Fernando,
>
> I suspect you upgraded from a quite old version.
> mod_limit is dead, it was moved to core.
> You just need mod_hash if you used to do hash limits.
> Check the wiki for details.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/05/2011 ? 05:23, FERNANDO VILLARROEL a ?crit :
>
> Hi All.
>
> I rebuild my FS box and now i am getting the following error when i load
> the module limit:
>
> 2011-05-06 23:56:11.229308 [CRIT] switch_loadable_module.c:928 Error
> Loading module /usr/local/freeswitch/mod/mod_limit.so
> **/usr/local/freeswitch/mod/mod_limit.so: undefined symbol:
> switch_channel_get_variable**
>
> How i can solve?
>
> Fernando.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dnotivol at gmail.com  Mon May  9 12:19:25 2011
From: dnotivol at gmail.com (David Notivol)
Date: Mon, 9 May 2011 10:19:25 +0200
Subject: [Freeswitch-users] How to add a new language
Message-ID: <BANLkTin=22R+tFVCB6sz-FcDXjZMUB-3oQ@mail.gmail.com>

Hi all,

I'm trying to make some tests with some custom audio files in English and
Spanish, but I'm not being able to add Spanish as language.
The module mod_say_es is compiled and loaded. I duplicated the structure in
lang/en to lang/es, but without any luck, always getting a "couldn't find es
language" message.

Any advise or suggestion?
Thanks.

-- 
Regards,
David Notivol
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From william at xofap.com  Mon May  9 12:25:19 2011
From: william at xofap.com (William Alianto)
Date: Mon, 09 May 2011 15:25:19 +0700
Subject: [Freeswitch-users] No Audio on Gateway Incoming
In-Reply-To: <mailman.7828.1303231692.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.7828.1303231692.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <4DC7A4EF.8040408@xofap.com>

Here is the debug log from the server :

http://pastebin.freeswitch.org/16257

-- 
Regards,

William



From fieldpeak at gmail.com  Mon May  9 12:40:11 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 9 May 2011 16:40:11 +0800
Subject: [Freeswitch-users] How to manipulate destination number before
 routing to PSTN GW
In-Reply-To: <BANLkTikU6yZg2vEAC5TW3JiHfbm7H4e84A@mail.gmail.com>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
	<BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
	<C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
	<BANLkTikU6yZg2vEAC5TW3JiHfbm7H4e84A@mail.gmail.com>
Message-ID: <BANLkTinWEQcEXdA1ZnDGrD=KTKSHSKBw8Q@mail.gmail.com>

Is there anyone can help below, it is still pending, thanks!

Regards,
Charles

2011/5/8 fieldpeak <fieldpeak at gmail.com>

> i tested the inline usage, unluckly, it failed. it looks the inline still
> can not effect 9 to GW rule...
>
> *dial plan is:*
>
> <extension name="remove_plus_of_dst_num" continue="true">
>    <condition field="destination_number" expression="^\+(\d+)$">
>        <action inline="true" application="set"
> data="destination_number=$1"/>
>
>    </condition>
> </extension>
>
> ...
> <extension name="9_to_GW" >
>    <condition field="destination_number" expression="^(9\d+)$">
>        <action application="bridge" data="sofia/internal/$
> 1 at 192.168.200.101"/>
>    </condition>
> </extension>
>
> *the log is:*
>
> freeswitch at mypc>
> 2011-05-08 09:35:40.067099 [ERR] switch_xml.c:1311 Couldnt open
> C:\FreeSWITCH\conf\
> autoload_configs\..\sip_profiles\external/*.xml (No such file or directory)
> 2011-05-08 09:35:40.543126 [ERR] switch_xml.c:1311 Couldnt open
> C:\FreeSWITCH\conf\
> dialplan\public/*.xml (No such file or directory)
>
> +OK [Success]
>
> 2011-05-08 09:35:41.115159 [INFO] mod_pocketsphinx.c:482 PocketSphinx
> Reloaded
> 2011-05-08 09:35:41.117159 [INFO] switch_time.c:999 Timezone reloaded 530
> defini
> tions
> 2011-05-08 09:35:44.882374 [DEBUG] sofia.c:6488 IP 192.168.200.201 Approved
> by a
>
> cl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>". Access Granted.
> 2011-05-08 09:35:44.883374 [NOTICE] switch_channel.c:812 New Channel
> sofia/inter
> nal/4001 at 192.168.200.100 [efeb55d7-3712-4a30-9eba-786ad83b2e90]
> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4760 Channel
> sofia/internal/4001 at 192.
>
> 168.200.100 entering state [received][100]
> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4771 Remote SDP:
> v=0
> o=AudiocodesGW 2088509637 2088509507 IN IP4 192.168.200.201
>
> s=Phone-Call
> c=IN IP4 192.168.200.201
> t=0 0
> m=audio 6020 RTP/AVP 8 0 96
>
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-15
> a=ptime:20
>
> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4908
> (sofia/internal/4001 at 192.168.200
>
> .100) State Change CS_NEW -> CS_INIT
> 2011-05-08 09:35:44.884374 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_INIT
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
> (sofia/intern
>
> al/4001 at 192.168.200.100) State INIT
> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:84
> sofia/internal/4001 at 192.168.20
> 0.100 SOFIA INIT
> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:124
> (sofia/internal/4001 at 192.168.
>
> 200.100) State Change CS_INIT -> CS_ROUTING
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
> (sofia/intern
>
> al/4001 at 192.168.200.100) State INIT going to sleep
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_ROUTING
> 2011-05-08 09:35:44.885374 [DEBUG] switch_channel.c:1668
> (sofia/internal/4001 at 19
>
> 2.168.200.100) Callstate Change DOWN -> RINGING
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:364
> (sofia/intern
>
> al/4001 at 192.168.200.100) State ROUTING
> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:147
> sofia/internal/4001 at 192.168.2
> 00.100 SOFIA ROUTING
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:77
> sofia/internal
>
> /4001 at 192.168.200.100 Standard ROUTING
> 2011-05-08 09:35:44.885374 [INFO] mod_dialplan_xml.c:331 Processing 4001
> <4001>-
> >+9123 in context default
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->unloop]
> continue
> =false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS) [unloop]
> ${unroll_loo
>
> ps}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [unloop]
> ${sip_looped
> _call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->remove_plus_of_d
> st_num] continue=true
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS)
> [remove_plus_of_dst_n
> um] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Action
> set(destination_number=9123
> ) INLINE
>
> EXECUTE sofia/internal/4001 at 192.168.200.100 set(destination_number=9123)
> 2011-05-08 09:35:44.888374 [DEBUG] mod_dptools.c:1060
> sofia/internal/4001 at 192.16
>
> 8.200.100 SET [destination_number]=[9123]
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->remove_plus_of_s
> rc_num] continue=true
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL)
> [remove_plus_of_src_n
>
> um] caller_id_number(4001) =~ /^\+(\d+)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->7_8_to_Lync] con
> tinue=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
> destina
>
> tion_number(+9123) =~ /^([78]\d{3})$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->9_to_GW]
> continu
> e=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
> destination
>
> _number(+9123) =~ /^(9\d+)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->1_to_IPP]
> contin
> ue=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
> destinatio
> n_number(+9123) =~ /^\+{0,1}(1\d{3})$/ break=on-false
>
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->DISA]
> continue=f
> alse
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [DISA]
> destination_nu
>
> mber(+9123) =~ /^\*(3472)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->Recordings] cont
> inue=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [Recordings]
> destinat
>
> ion_number(+9123) =~ /^\*(732673)$/ break=on-false
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:119
> (sofia/intern
>
> al/4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:364
> (sofia/intern
>
> al/4001 at 192.168.200.100) State ROUTING going to sleep
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:371
> (sofia/intern
>
> al/4001 at 192.168.200.100) State EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] mod_sofia.c:240
> sofia/internal/4001 at 192.168.2
> 00.100 SOFIA EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:157
> sofia/interna
>
> l/4001 at 192.168.200.100 Standard EXECUTE
> 2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:189
> sofia/intern
>
> al/4001 at 192.168.200.100 has executed the last dialplan instruction,
> hanging up.
> 2011-05-08 09:35:44.889374 [DEBUG] switch_channel.c:2563
> (sofia/internal/4001 at 19
>
> 2.168.200.100) Callstate Change RINGING -> HANGUP
> 2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia
>
> /internal/4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-05-08 09:35:44.890375 [DEBUG] switch_channel.c:2579 Send signal
> sofia/inter
>
> nal/4001 at 192.168.200.100 [KILL]
> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:371
> (sofia/intern
>
> al/4001 at 192.168.200.100) State EXECUTE going to sleep
> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_HANGUP
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
> (sofia/intern
>
> al/4001 at 192.168.200.100) State HANGUP
> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:457 Channel
> sofia/internal/4001 at 1
>
> 92.168.200.100 hanging up, cause: NORMAL_CLEARING
> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:519 Responding to INVITE
> with: 48
> 0
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:46
> sofia/internal
>
> /4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
> (sofia/intern
>
> al/4001 at 192.168.200.100) State HANGUP going to sleep
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:356
> (sofia/intern
>
> al/4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_REPORTING
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:625
> (sofia/intern
>
> al/4001 at 192.168.200.100) State REPORTING
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:53
> sofia/internal
>
> /4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:625
> (sofia/intern
>
> al/4001 at 192.168.200.100) State REPORTING going to sleep
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:350
> (sofia/intern
>
> al/4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1288 Session 4
> (sofia/i
>
> nternal/4001 at 192.168.200.100) Locked, Waiting on external entities
> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1306 Session 4
> (sofia/
>
> internal/4001 at 192.168.200.100) Ended
> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1308 Close
> Channel sof
>
> ia/internal/4001 at 192.168.200.100 [CS_DESTROY]
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:454
> (sofia/intern
>
> al/4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:457
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
> (sofia/intern
>
> al/4001 at 192.168.200.100) State DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] mod_sofia.c:362
> sofia/internal/4001 at 192.168.2
> 00.100 SOFIA DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:60
> sofia/internal
>
> /4001 at 192.168.200.100 Standard DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
> (sofia/intern
>
> al/4001 at 192.168.200.100) State DESTROY going to sleep
>
>
> 2011/5/8 David Ponzone <david.ponzone at ipeva.fr>
>
>> Avi,
>>
>> are you sure about that ?
>>
>> I have LUA scripts which rewrite the destination_number with SetVariable
>> and that works fine.
>> Of course, to do that in the XML dialplan, as said before, inline should
>> be used to set the value.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/05/2011 ? 19:52, Avi Marcus a ?crit :
>>
>> Indeed, inline is a missing ingredient. As well though is the previously
>> mentioned - you can't (directly) overwrite the destination number. You can
>> either use a new variable throughout, or transfer to set the new destination
>> number. Note that transferring causes the dialplan to be re-run from the
>> start..
>>
>> -Avi
>>
>> On Sat, May 7, 2011 at 7:13 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>>
>>> There is certainly more than one way to skin this cat...
>>>
>>> However, to keep with your current method you should read up on
>>> dialplan hunting vs. execution and "inline" execution:
>>>
>>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>>
>>> In short you need to add inline="true" to your first extension.
>>>
>>> On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>> > Gurus,
>>> >
>>> > i met an issue for dial plan, it sounds easy but puzzled me a few days
>>> not
>>> > fix it yet...i belive gurus here could help me...
>>> >
>>> > i want to remove the + of destination number before routing to PSTN GW,
>>> e.g.
>>> > when i dial +9123, i would like FS remove +, and then route to the PSTN
>>> GW,
>>> > below is the dial plan and log,
>>> >
>>> > See the log, FS did remove '+' of '+9123', but then when it checks the
>>> rule
>>> > ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then
>>> failed.
>>> >
>>> > Thanks.
>>> >
>>> >
>>> >
>>> > Dial plan:
>>> >
>>> > <extension name="remove_plus_of_dst_num" continue="true">
>>> >
>>> >    <condition field="destination_number" expression="^\+(\d+)$">
>>> >
>>> >        <action application="set"
>>> data="effective_destination_number=$1"/>
>>> >
>>> >    </condition>
>>> >
>>> > </extension>
>>> >
>>> > ?...
>>> >
>>> > <extension name="9_to_GW" >
>>> >
>>> >    <condition field="destination_number" expression="^(9\d+)$">
>>> >
>>> >        <action application="bridge"
>>> > data="sofia/internal/$1 at 192.168.200.101"/>
>>> >
>>> >    </condition>
>>> >
>>> > </extension>
>>> >
>>> >
>>> >
>>> > Log:
>>> >
>>> > freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
>>> > 192.168.200.201
>>> >
>>> >  Approved by acl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>".
>>> Access Granted.
>>> >
>>> > 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
>>> > sofia/inter
>>> >
>>> > nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
>>> > sofia/internal/+4001 at 192
>>> >
>>> > .168.200.100 entering state [received][100]
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
>>> >
>>> > v=0
>>> >
>>> > o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
>>> >
>>> > s=Phone-Call
>>> >
>>> > c=IN IP4 192.168.200.201
>>> >
>>> > t=0 0
>>> >
>>> > m=audio 6060 RTP/AVP 8 0 96
>>> >
>>> > a=rtpmap:8 PCMA/8000
>>> >
>>> > a=rtpmap:0 PCMU/8000
>>> >
>>> > a=rtpmap:96 telephone-event/8000
>>> >
>>> > a=fmtp:96 0-15
>>> >
>>> > a=ptime:20
>>> >
>>> >
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
>>> > (sofia/internal/+4001 at 192.168.20
>>> >
>>> > 0.100) State Change CS_NEW -> CS_INIT
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_INIT
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State INIT
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
>>> > sofia/internal/+4001 at 192.168.2
>>> >
>>> > 00.100 SOFIA INIT
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
>>> > (sofia/internal/+4001 at 192.168
>>> >
>>> > .200.100) State Change CS_INIT -> CS_ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State INIT going to sleep
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
>>> > (sofia/internal/+4001 at 1
>>> >
>>> > 92.168.200.100) Callstate Change DOWN -> RINGING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
>>> > sofia/internal/+4001 at 192.168.
>>> >
>>> > 200.100 SOFIA ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing
>>> +4001
>>> > <+4001
>>> >
>>> >>->+9123 in context default
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->unloop]
>>> > continu
>>> >
>>> > e=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
>>> > ${unroll_lo
>>> >
>>> > ops}(true) =~ /^true$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
>>> > ${sip_loope
>>> >
>>> > d_call}() =~ /^true$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> > [default->remove_plus_of_
>>> >
>>> > dst_num] continue=true
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>> > [remove_plus_of_dst_
>>> >
>>> > num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>> > set(destination_number=912
>>> >
>>> > 3)
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> > [default->remove_plus_of_
>>> >
>>> > src_num] continue=true
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>> > [remove_plus_of_src_
>>> >
>>> > num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>> > set(effective_caller_id_na
>>> >
>>> > me=4001)
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>> > set(effective_caller_id_nu
>>> >
>>> > mber=4001)
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> > [default->7_8_to_Lync] co
>>> >
>>> > ntinue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>> [7_8_to_Lync]
>>> > destin
>>> >
>>> > ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->9_to_GW]
>>> > contin
>>> >
>>> > ue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
>>> > destinatio
>>> >
>>> > n_number(+9123) =~ /^(9\d+)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->1_to_IPP]
>>> > conti
>>> >
>>> > nue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
>>> > destinati
>>> >
>>> > on_number(+9123) =~ /^(1\d{3})$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA]
>>> > continue=
>>> >
>>> > false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
>>> > destination_n
>>> >
>>> > umber(+9123) =~ /^\*(3472)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->Recordings]
>>> > con
>>> >
>>> > tinue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>> [Recordings]
>>> > destina
>>> >
>>> > tion_number(+9123) =~ /^\*(732673)$/ break=on-false
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State ROUTING going to sleep
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
>>> > sofia/internal/+4001 at 192.168.
>>> >
>>> > 200.100 SOFIA EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
>>> > sofia/interna
>>> >
>>> > l/+4001 at 192.168.200.100 Standard EXECUTE
>>> >
>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100set(destination_number=9123)
>>> >
>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>> > sofia/internal/+4001 at 192.1
>>> >
>>> > 68.200.100 SET [destination_number]=[9123]
>>> >
>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>> > set(effective_caller_id_name=4001)
>>> >
>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>> > sofia/internal/+4001 at 192.1
>>> >
>>> > 68.200.100 SET [effective_caller_id_name]=[4001]
>>> >
>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>> > set(effective_caller_id_number=4001
>>> >
>>> > )
>>> >
>>> > 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
>>> > sofia/internal/+4001 at 192.1
>>> >
>>> > 68.200.100 SET [effective_caller_id_number]=[4001]
>>> >
>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
>>> > sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100 has executed the last dialplan instruction,
>>> hanging
>>> > up.
>>> >
>>> >
>>> >
>>> > 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
>>> > (sofia/internal/+4001 at 1
>>> >
>>> > 92.168.200.100) Callstate Change RINGING -> HANGUP
>>> >
>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191
>>> Hangup
>>> > sofia
>>> >
>>> > /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
>>> > sofia/inter
>>> >
>>> > nal/+4001 at 192.168.200.100 [KILL]
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State EXECUTE going to sleep
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
>>> >
>>> > 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State HANGUP
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
>>> > sofia/internal/+4001@
>>> >
>>> > 192.168.200.100 hanging up, cause: NORMAL_CLEARING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE
>>> > with: 48
>>> >
>>> > 0
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State HANGUP going to sleep
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State REPORTING
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State REPORTING going to sleep
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2
>>> > (sofia/i
>>> >
>>> > nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
>>> >
>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session
>>> 2
>>> > (sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100) Ended
>>> >
>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close
>>> Channel
>>> > sof
>>> >
>>> > ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
>>> > sofia/internal/+4001 at 192.168.
>>> >
>>> > 200.100 SOFIA DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State DESTROY going to sleep
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at 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
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From fdelawarde at wirelessmundi.com  Mon May  9 13:24:56 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 09 May 2011 11:24:56 +0200
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <4DC784BB.4080907@seventhsignal.de>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<4DC72997.4090900@sns.eu>	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
	<4DC784BB.4080907@seventhsignal.de>
Message-ID: <1304933096.28727.28.camel@luna.tc.commsmundi.com>

Using debian lenny and squeeze, it's boringly stable. Thinking of
switching to Gentoo, Ubuntu, or LFS for some more fun. What's the worse
distro for freeswitch?

Fran?ois.

On Mon, 2011-05-09 at 08:07 +0200, Michal Bielicki wrote: 
> I guess Jan just wanted to explain. Sometimes its better to clear up
> things for people in their local language, don't you think ?
> 
> Am 09.05.2011 06:01, schrieb curriegrad2004:
> > It would be beneficial to the majority of the users if we continued
> > this discussion in English
> >
> > On Sun, May 8, 2011 at 5:09 PM,  <rhuddleston at gmail.com> wrote:
> >> Auf English bitte? Mein deutsch ist sehr sclecht und ich kann nicht verstehen
> >>
> >> On May 8, 2011, at 7:39 PM, Jan Riedinger <riedinger at sns.eu> wrote:
> >>
> >>> Hi Dirk,
> >>>
> >>> Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein
> >>> Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.
> >>>
> >>> Viele Gr??e
> >>>     Jan
> >>>
> >>> Am 06.05.2011 16:25, schrieb Max Clark:
> >>>> Hello all,
> >>>>
> >>>> Recent developments (or absolute lack of) within the CentOS project
> >>>> and its perceived long term viability has forced an internal
> >>>> discussion to select a successor distribution. The most likely
> >>>> candidates at this point are Ubuntu and its LTS releases for servers,
> >>>> and Scientific Linux with the new 6.x releases. The pro/con lists for
> >>>> each are growing and the issue is complicated.
> >>>>
> >>>> With CentOS 5.x being the reference distro for FreeSWITCH development
> >>>> I'm curious if this conversation has started among the FreeSWITCH
> >>>> developers, and if it has, what is the project leaning to for a
> >>>> successor distribution?
> >>>>
> >>>> Thanks,
> >>>> Max
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>> --
> >>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> >>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
> >>>                                         E-Mail:  riedinger at sns.eu
> >>> SNS Consult GmbH                        ICQ   :  163-237-041
> >>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> >>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
> >>>
> >>> AG Charlottenburg - HRB 71973
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon May  9 14:50:26 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 9 May 2011 13:50:26 +0300
Subject: [Freeswitch-users] How to manipulate destination number before
 routing to PSTN GW
In-Reply-To: <BANLkTikU6yZg2vEAC5TW3JiHfbm7H4e84A@mail.gmail.com>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
	<BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
	<C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
	<BANLkTikU6yZg2vEAC5TW3JiHfbm7H4e84A@mail.gmail.com>
Message-ID: <BANLkTinjHUyRbZO4PP=D=Ds6d9rNNEqLLA@mail.gmail.com>

As mentioned, that's because you can't set destination_number -- it's a
reserved variable.
Either a) transfer to the new number, or b) create a new variable for
processing the number.
e.g. with an anti-action that sets the new variable to be the unchanged
number.
-Avi


On Sun, May 8, 2011 at 4:56 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> i tested the inline usage, unluckly, it failed. it looks the inline still
> can not effect 9 to GW rule...
>
> *dial plan is:*
>
> <extension name="remove_plus_of_dst_num" continue="true">
>    <condition field="destination_number" expression="^\+(\d+)$">
>        <action inline="true" application="set"
> data="destination_number=$1"/>
>
>    </condition>
> </extension>
>
> ...
> <extension name="9_to_GW" >
>    <condition field="destination_number" expression="^(9\d+)$">
>        <action application="bridge" data="sofia/internal/$
> 1 at 192.168.200.101"/>
>    </condition>
> </extension>
>
> *the log is:*
>
> freeswitch at mypc>
> 2011-05-08 09:35:40.067099 [ERR] switch_xml.c:1311 Couldnt open
> C:\FreeSWITCH\conf\
> autoload_configs\..\sip_profiles\external/*.xml (No such file or directory)
> 2011-05-08 09:35:40.543126 [ERR] switch_xml.c:1311 Couldnt open
> C:\FreeSWITCH\conf\
> dialplan\public/*.xml (No such file or directory)
>
> +OK [Success]
>
> 2011-05-08 09:35:41.115159 [INFO] mod_pocketsphinx.c:482 PocketSphinx
> Reloaded
> 2011-05-08 09:35:41.117159 [INFO] switch_time.c:999 Timezone reloaded 530
> defini
> tions
> 2011-05-08 09:35:44.882374 [DEBUG] sofia.c:6488 IP 192.168.200.201 Approved
> by a
>
> cl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>". Access Granted.
>  2011-05-08 09:35:44.883374 [NOTICE] switch_channel.c:812 New Channel
> sofia/inter
> nal/4001 at 192.168.200.100 [efeb55d7-3712-4a30-9eba-786ad83b2e90]
> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4760 Channel
> sofia/internal/4001 at 192.
>
> 168.200.100 entering state [received][100]
> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4771 Remote SDP:
> v=0
> o=AudiocodesGW 2088509637 2088509507 IN IP4 192.168.200.201
>
> s=Phone-Call
> c=IN IP4 192.168.200.201
> t=0 0
> m=audio 6020 RTP/AVP 8 0 96
>
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-15
> a=ptime:20
>
> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4908
> (sofia/internal/4001 at 192.168.200
>
> .100) State Change CS_NEW -> CS_INIT
> 2011-05-08 09:35:44.884374 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_INIT
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
> (sofia/intern
>
> al/4001 at 192.168.200.100) State INIT
> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:84
> sofia/internal/4001 at 192.168.20
> 0.100 SOFIA INIT
> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:124
> (sofia/internal/4001 at 192.168.
>
> 200.100) State Change CS_INIT -> CS_ROUTING
>  2011-05-08 09:35:44.885374 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
> (sofia/intern
>
> al/4001 at 192.168.200.100) State INIT going to sleep
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_ROUTING
>  2011-05-08 09:35:44.885374 [DEBUG] switch_channel.c:1668
> (sofia/internal/4001 at 19
>
> 2.168.200.100) Callstate Change DOWN -> RINGING
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:364
> (sofia/intern
>
> al/4001 at 192.168.200.100) State ROUTING
> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:147
> sofia/internal/4001 at 192.168.2
> 00.100 SOFIA ROUTING
> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:77
> sofia/internal
>
> /4001 at 192.168.200.100 Standard ROUTING
> 2011-05-08 09:35:44.885374 [INFO] mod_dialplan_xml.c:331 Processing 4001
> <4001>-
> >+9123 in context default
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->unloop]
> continue
> =false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS) [unloop]
> ${unroll_loo
>
> ps}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [unloop]
> ${sip_looped
> _call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->remove_plus_of_d
> st_num] continue=true
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS)
> [remove_plus_of_dst_n
> um] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Action
> set(destination_number=9123
>  ) INLINE
>
> EXECUTE sofia/internal/4001 at 192.168.200.100 set(destination_number=9123)
> 2011-05-08 09:35:44.888374 [DEBUG] mod_dptools.c:1060
> sofia/internal/4001 at 192.16
>
> 8.200.100 SET [destination_number]=[9123]
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->remove_plus_of_s
> rc_num] continue=true
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL)
> [remove_plus_of_src_n
>
> um] caller_id_number(4001) =~ /^\+(\d+)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->7_8_to_Lync] con
> tinue=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
> destina
>
> tion_number(+9123) =~ /^([78]\d{3})$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->9_to_GW]
> continu
> e=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
> destination
>
> _number(+9123) =~ /^(9\d+)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->1_to_IPP]
> contin
> ue=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
> destinatio
> n_number(+9123) =~ /^\+{0,1}(1\d{3})$/ break=on-false
>
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->DISA]
> continue=f
> alse
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [DISA]
> destination_nu
>
> mber(+9123) =~ /^\*(3472)$/ break=on-false
> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
> [default->Recordings] cont
> inue=false
> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [Recordings]
> destinat
>
> ion_number(+9123) =~ /^\*(732673)$/ break=on-false
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:119
> (sofia/intern
>
> al/4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:364
> (sofia/intern
>
> al/4001 at 192.168.200.100) State ROUTING going to sleep
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:371
> (sofia/intern
>
> al/4001 at 192.168.200.100) State EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] mod_sofia.c:240
> sofia/internal/4001 at 192.168.2
> 00.100 SOFIA EXECUTE
> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:157
> sofia/interna
>
> l/4001 at 192.168.200.100 Standard EXECUTE
>  2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:189
> sofia/intern
>
> al/4001 at 192.168.200.100 has executed the last dialplan instruction,
> hanging up.
> 2011-05-08 09:35:44.889374 [DEBUG] switch_channel.c:2563
> (sofia/internal/4001 at 19
>
> 2.168.200.100) Callstate Change RINGING -> HANGUP
> 2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia
>
> /internal/4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>  2011-05-08 09:35:44.890375 [DEBUG] switch_channel.c:2579 Send signal
> sofia/inter
>
> nal/4001 at 192.168.200.100 [KILL]
> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:371
> (sofia/intern
>
> al/4001 at 192.168.200.100) State EXECUTE going to sleep
>  2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_HANGUP
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
> (sofia/intern
>
> al/4001 at 192.168.200.100) State HANGUP
> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:457 Channel
> sofia/internal/4001 at 1
>
> 92.168.200.100 hanging up, cause: NORMAL_CLEARING
> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:519 Responding to INVITE
> with: 48
> 0
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:46
> sofia/internal
>
> /4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
> (sofia/intern
>
> al/4001 at 192.168.200.100) State HANGUP going to sleep
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:356
> (sofia/intern
>
> al/4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:325
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_REPORTING
> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:625
> (sofia/intern
>
> al/4001 at 192.168.200.100) State REPORTING
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:53
> sofia/internal
>
> /4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:625
> (sofia/intern
>
> al/4001 at 192.168.200.100) State REPORTING going to sleep
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:350
> (sofia/intern
>
> al/4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/
>
> internal/4001 at 192.168.200.100 [BREAK]
> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1288 Session 4
> (sofia/i
>
> nternal/4001 at 192.168.200.100) Locked, Waiting on external entities
> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1306 Session 4
> (sofia/
>
> internal/4001 at 192.168.200.100) Ended
> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1308 Close
> Channel sof
>
> ia/internal/4001 at 192.168.200.100 [CS_DESTROY]
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:454
> (sofia/intern
>
> al/4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:457
> (sofia/intern
>
> al/4001 at 192.168.200.100) Running State Change CS_DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
> (sofia/intern
>
> al/4001 at 192.168.200.100) State DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] mod_sofia.c:362
> sofia/internal/4001 at 192.168.2
> 00.100 SOFIA DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:60
> sofia/internal
>
> /4001 at 192.168.200.100 Standard DESTROY
> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
> (sofia/intern
>
> al/4001 at 192.168.200.100) State DESTROY going to sleep
>
>
> 2011/5/8 David Ponzone <david.ponzone at ipeva.fr>
>
>> Avi,
>>
>> are you sure about that ?
>>
>> I have LUA scripts which rewrite the destination_number with SetVariable
>> and that works fine.
>> Of course, to do that in the XML dialplan, as said before, inline should
>> be used to set the value.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/05/2011 ? 19:52, Avi Marcus a ?crit :
>>
>> Indeed, inline is a missing ingredient. As well though is the previously
>> mentioned - you can't (directly) overwrite the destination number. You can
>> either use a new variable throughout, or transfer to set the new destination
>> number. Note that transferring causes the dialplan to be re-run from the
>> start..
>>
>> -Avi
>>
>> On Sat, May 7, 2011 at 7:13 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>>
>>> There is certainly more than one way to skin this cat...
>>>
>>> However, to keep with your current method you should read up on
>>> dialplan hunting vs. execution and "inline" execution:
>>>
>>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>>
>>> In short you need to add inline="true" to your first extension.
>>>
>>> On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>> > Gurus,
>>> >
>>> > i met an issue for dial plan, it sounds easy but puzzled me a few days
>>> not
>>> > fix it yet...i belive gurus here could help me...
>>> >
>>> > i want to remove the + of destination number before routing to PSTN GW,
>>> e.g.
>>> > when i dial +9123, i would like FS remove +, and then route to the PSTN
>>> GW,
>>> > below is the dial plan and log,
>>> >
>>> > See the log, FS did remove '+' of '+9123', but then when it checks the
>>> rule
>>> > ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then
>>> failed.
>>> >
>>> > Thanks.
>>> >
>>> >
>>> >
>>> > Dial plan:
>>> >
>>> > <extension name="remove_plus_of_dst_num" continue="true">
>>> >
>>> >    <condition field="destination_number" expression="^\+(\d+)$">
>>> >
>>> >        <action application="set"
>>> data="effective_destination_number=$1"/>
>>> >
>>> >    </condition>
>>> >
>>> > </extension>
>>> >
>>> > ?...
>>> >
>>> > <extension name="9_to_GW" >
>>> >
>>> >    <condition field="destination_number" expression="^(9\d+)$">
>>> >
>>> >        <action application="bridge"
>>> > data="sofia/internal/$1 at 192.168.200.101"/>
>>> >
>>> >    </condition>
>>> >
>>> > </extension>
>>> >
>>> >
>>> >
>>> > Log:
>>> >
>>> > freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
>>> > 192.168.200.201
>>> >
>>> >  Approved by acl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>".
>>> Access Granted.
>>> >
>>> > 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
>>> > sofia/inter
>>> >
>>> > nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
>>> > sofia/internal/+4001 at 192
>>> >
>>> > .168.200.100 entering state [received][100]
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
>>> >
>>> > v=0
>>> >
>>> > o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
>>> >
>>> > s=Phone-Call
>>> >
>>> > c=IN IP4 192.168.200.201
>>> >
>>> > t=0 0
>>> >
>>> > m=audio 6060 RTP/AVP 8 0 96
>>> >
>>> > a=rtpmap:8 PCMA/8000
>>> >
>>> > a=rtpmap:0 PCMU/8000
>>> >
>>> > a=rtpmap:96 telephone-event/8000
>>> >
>>> > a=fmtp:96 0-15
>>> >
>>> > a=ptime:20
>>> >
>>> >
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
>>> > (sofia/internal/+4001 at 192.168.20
>>> >
>>> > 0.100) State Change CS_NEW -> CS_INIT
>>> >
>>> > 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_INIT
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State INIT
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
>>> > sofia/internal/+4001 at 192.168.2
>>> >
>>> > 00.100 SOFIA INIT
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
>>> > (sofia/internal/+4001 at 192.168
>>> >
>>> > .200.100) State Change CS_INIT -> CS_ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State INIT going to sleep
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
>>> > (sofia/internal/+4001 at 1
>>> >
>>> > 92.168.200.100) Callstate Change DOWN -> RINGING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
>>> > sofia/internal/+4001 at 192.168.
>>> >
>>> > 200.100 SOFIA ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard ROUTING
>>> >
>>> > 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing
>>> +4001
>>> > <+4001
>>> >
>>> >>->+9123 in context default
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->unloop]
>>> > continu
>>> >
>>> > e=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
>>> > ${unroll_lo
>>> >
>>> > ops}(true) =~ /^true$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
>>> > ${sip_loope
>>> >
>>> > d_call}() =~ /^true$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> > [default->remove_plus_of_
>>> >
>>> > dst_num] continue=true
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>> > [remove_plus_of_dst_
>>> >
>>> > num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>> > set(destination_number=912
>>> >
>>> > 3)
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> > [default->remove_plus_of_
>>> >
>>> > src_num] continue=true
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>> > [remove_plus_of_src_
>>> >
>>> > num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>> > set(effective_caller_id_na
>>> >
>>> > me=4001)
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>> > set(effective_caller_id_nu
>>> >
>>> > mber=4001)
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> > [default->7_8_to_Lync] co
>>> >
>>> > ntinue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>> [7_8_to_Lync]
>>> > destin
>>> >
>>> > ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->9_to_GW]
>>> > contin
>>> >
>>> > ue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
>>> > destinatio
>>> >
>>> > n_number(+9123) =~ /^(9\d+)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->1_to_IPP]
>>> > conti
>>> >
>>> > nue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
>>> > destinati
>>> >
>>> > on_number(+9123) =~ /^(1\d{3})$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing [default->DISA]
>>> > continue=
>>> >
>>> > false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
>>> > destination_n
>>> >
>>> > umber(+9123) =~ /^\*(3472)$/ break=on-false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>> [default->Recordings]
>>> > con
>>> >
>>> > tinue=false
>>> >
>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>> [Recordings]
>>> > destina
>>> >
>>> > tion_number(+9123) =~ /^\*(732673)$/ break=on-false
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State ROUTING going to sleep
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
>>> > sofia/internal/+4001 at 192.168.
>>> >
>>> > 200.100 SOFIA EXECUTE
>>> >
>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
>>> > sofia/interna
>>> >
>>> > l/+4001 at 192.168.200.100 Standard EXECUTE
>>> >
>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100set(destination_number=9123)
>>> >
>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>> > sofia/internal/+4001 at 192.1
>>> >
>>> > 68.200.100 SET [destination_number]=[9123]
>>> >
>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>> > set(effective_caller_id_name=4001)
>>> >
>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>> > sofia/internal/+4001 at 192.1
>>> >
>>> > 68.200.100 SET [effective_caller_id_name]=[4001]
>>> >
>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>> > set(effective_caller_id_number=4001
>>> >
>>> > )
>>> >
>>> > 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
>>> > sofia/internal/+4001 at 192.1
>>> >
>>> > 68.200.100 SET [effective_caller_id_number]=[4001]
>>> >
>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
>>> > sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100 has executed the last dialplan instruction,
>>> hanging
>>> > up.
>>> >
>>> >
>>> >
>>> > 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
>>> > (sofia/internal/+4001 at 1
>>> >
>>> > 92.168.200.100) Callstate Change RINGING -> HANGUP
>>> >
>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191
>>> Hangup
>>> > sofia
>>> >
>>> > /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
>>> > sofia/inter
>>> >
>>> > nal/+4001 at 192.168.200.100 [KILL]
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State EXECUTE going to sleep
>>> >
>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
>>> >
>>> > 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State HANGUP
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
>>> > sofia/internal/+4001@
>>> >
>>> > 192.168.200.100 hanging up, cause: NORMAL_CLEARING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to INVITE
>>> > with: 48
>>> >
>>> > 0
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State HANGUP going to sleep
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
>>> >
>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State REPORTING
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State REPORTING going to sleep
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send
>>> signal
>>> > sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>> >
>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session 2
>>> > (sofia/i
>>> >
>>> > nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
>>> >
>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session
>>> 2
>>> > (sofia/
>>> >
>>> > internal/+4001 at 192.168.200.100) Ended
>>> >
>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close
>>> Channel
>>> > sof
>>> >
>>> > ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
>>> > sofia/internal/+4001 at 192.168.
>>> >
>>> > 200.100 SOFIA DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
>>> > sofia/internal
>>> >
>>> > /+4001 at 192.168.200.100 Standard DESTROY
>>> >
>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>> > (sofia/intern
>>> >
>>> > al/+4001 at 192.168.200.100) State DESTROY going to sleep
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at 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
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May  9 14:52:56 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 9 May 2011 13:52:56 +0300
Subject: [Freeswitch-users] How to add a new language
In-Reply-To: <BANLkTin=22R+tFVCB6sz-FcDXjZMUB-3oQ@mail.gmail.com>
References: <BANLkTin=22R+tFVCB6sz-FcDXjZMUB-3oQ@mail.gmail.com>
Message-ID: <BANLkTi=nFTzzV3DtDFKVj6POQJo3v9bb0g@mail.gmail.com>

First check:
/usr/local/freeswitch/conf/freeswitch.xml and see if you have a line like:
<X-PRE-PROCESS cmd="include" data="lang/es/*.xml"/>

It should be inside the <section name="phrases" description="Speech Phrase
Management">

-Avi

On Mon, May 9, 2011 at 11:19 AM, David Notivol <dnotivol at gmail.com> wrote:

> Hi all,
>
> I'm trying to make some tests with some custom audio files in English and
> Spanish, but I'm not being able to add Spanish as language.
> The module mod_say_es is compiled and loaded. I duplicated the structure in
> lang/en to lang/es, but without any luck, always getting a "couldn't find es
> language" message.
>
> Any advise or suggestion?
> Thanks.
>
> --
> Regards,
> David Notivol
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May  9 15:58:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 9 May 2011 12:58:29 +0100
Subject: [Freeswitch-users] How to manipulate destination number before
 routing to PSTN GW
In-Reply-To: <BANLkTinjHUyRbZO4PP=D=Ds6d9rNNEqLLA@mail.gmail.com>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
	<BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
	<C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
	<BANLkTikU6yZg2vEAC5TW3JiHfbm7H4e84A@mail.gmail.com>
	<BANLkTinjHUyRbZO4PP=D=Ds6d9rNNEqLLA@mail.gmail.com>
Message-ID: <BANLkTi=SULGgcvytahBcCcOAF5Uzn9Ramg@mail.gmail.com>

fieldpark,

It would be far simpler to just ignore it in the regex of the 2nd extension
- then you wouldn't need the 1st extension at all!

<extension name="9_to_GW">
   <condition field="dest_num" expression="^\+?(9\d+)$">
       <action application="bridge" data="sofia/internal/$1 at 192.168.200.101
"/>
   </condition>
</extension>

The ? after the \+ makes it optional. This extension will match both 9123
and +9123, and bridge both as 123 at 192.168.200.101

The only reason I can think of for stripping it first would be if you have a
very long dialplan, where it might then be simpler than optionally checking
for a + in every regex in the rest of the dialplan. For a simple dialplan
the above extension would be simpler.


Avi's suggestion should work fine and would look like this:

<extension name="strip_plus" continue="true">
   <condition field="destination_number" expression="^\+(\d+)$">
       <action application="set" data="dest_num=$1"/>
       <anti-action application="set"
data="dest_num=${destination_number}"/>
   </condition>
</extension>
<extension name="9_to_GW">
   <condition field="${dest_num}" expression="^(9\d+)$">
       <action application="bridge" data="sofia/internal/$1 at 192.168.200.101
"/>
   </condition>
</extension>


I'd suggest a transfer would be simpler than doing this though because it
this approach changes the field you're accessing in every extension. A
transfer would rewrite the destination_number so you wouldn't need to check
a channel variable each time any more. Yes it does reexecute the dialplan,
but if it's placed as the very first extension that should have a very small
impact on performance (the time taken to run a single regex that fails on
the first character is pretty small, and checking a channel variable on
every extension is slower than checking the destination_number field so it
might actually be faster than the above*).

<extension name="strip_plus">
   <condition field="destination_number" expression="^\+(\d+)$">
       <action application="transfer" data="$1"/>
   </condition>
</extension>
<extension name="9_to_GW">
   <condition field="destination_number" expression="^(9\d+)$">
       <action application="bridge" data="sofia/internal/$1 at 192.168.200.101
"/>
   </condition>
</extension>



-Steve


* Looking for a channel variable means looping through the list of all
channel variables while destination_number is hardcoded to look it up in the
caller profile structure so there's no loop involved.


On 9 May 2011 11:50, Avi Marcus <avi at avimarcus.net> wrote:

> As mentioned, that's because you can't set destination_number -- it's a
> reserved variable.
> Either a) transfer to the new number, or b) create a new variable for
> processing the number.
> e.g. with an anti-action that sets the new variable to be the unchanged
> number.
> -Avi
>
>
> On Sun, May 8, 2011 at 4:56 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> i tested the inline usage, unluckly, it failed. it looks the inline still
>> can not effect 9 to GW rule...
>>
>> *dial plan is:*
>>
>> <extension name="remove_plus_of_dst_num" continue="true">
>>    <condition field="destination_number" expression="^\+(\d+)$">
>>        <action inline="true" application="set"
>> data="destination_number=$1"/>
>>
>>    </condition>
>> </extension>
>>
>> ...
>> <extension name="9_to_GW" >
>>    <condition field="destination_number" expression="^(9\d+)$">
>>        <action application="bridge" data="sofia/internal/$
>> 1 at 192.168.200.101"/>
>>    </condition>
>> </extension>
>>
>> *the log is:*
>>
>> freeswitch at mypc>
>> 2011-05-08 09:35:40.067099 [ERR] switch_xml.c:1311 Couldnt open
>> C:\FreeSWITCH\conf\
>> autoload_configs\..\sip_profiles\external/*.xml (No such file or
>> directory)
>> 2011-05-08 09:35:40.543126 [ERR] switch_xml.c:1311 Couldnt open
>> C:\FreeSWITCH\conf\
>> dialplan\public/*.xml (No such file or directory)
>>
>> +OK [Success]
>>
>> 2011-05-08 09:35:41.115159 [INFO] mod_pocketsphinx.c:482 PocketSphinx
>> Reloaded
>> 2011-05-08 09:35:41.117159 [INFO] switch_time.c:999 Timezone reloaded 530
>> defini
>> tions
>> 2011-05-08 09:35:44.882374 [DEBUG] sofia.c:6488 IP 192.168.200.201
>> Approved by a
>>
>> cl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>". Access Granted.
>>  2011-05-08 09:35:44.883374 [NOTICE] switch_channel.c:812 New Channel
>> sofia/inter
>> nal/4001 at 192.168.200.100 [efeb55d7-3712-4a30-9eba-786ad83b2e90]
>> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4760 Channel
>> sofia/internal/4001 at 192.
>>
>> 168.200.100 entering state [received][100]
>> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4771 Remote SDP:
>> v=0
>> o=AudiocodesGW 2088509637 2088509507 IN IP4 192.168.200.201
>>
>> s=Phone-Call
>> c=IN IP4 192.168.200.201
>> t=0 0
>> m=audio 6020 RTP/AVP 8 0 96
>>
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:96 telephone-event/8000
>> a=fmtp:96 0-15
>> a=ptime:20
>>
>> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4908
>> (sofia/internal/4001 at 192.168.200
>>
>> .100) State Change CS_NEW -> CS_INIT
>> 2011-05-08 09:35:44.884374 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/
>>
>> internal/4001 at 192.168.200.100 [BREAK]
>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) Running State Change CS_INIT
>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State INIT
>> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:84
>> sofia/internal/4001 at 192.168.20
>> 0.100 SOFIA INIT
>> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:124
>> (sofia/internal/4001 at 192.168.
>>
>> 200.100) State Change CS_INIT -> CS_ROUTING
>>  2011-05-08 09:35:44.885374 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/
>>
>> internal/4001 at 192.168.200.100 [BREAK]
>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State INIT going to sleep
>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) Running State Change CS_ROUTING
>>  2011-05-08 09:35:44.885374 [DEBUG] switch_channel.c:1668
>> (sofia/internal/4001 at 19
>>
>> 2.168.200.100) Callstate Change DOWN -> RINGING
>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:364
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State ROUTING
>> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:147
>> sofia/internal/4001 at 192.168.2
>> 00.100 SOFIA ROUTING
>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:77
>> sofia/internal
>>
>> /4001 at 192.168.200.100 Standard ROUTING
>> 2011-05-08 09:35:44.885374 [INFO] mod_dialplan_xml.c:331 Processing 4001
>> <4001>-
>> >+9123 in context default
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->unloop]
>> continue
>> =false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS) [unloop]
>> ${unroll_loo
>>
>> ps}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [unloop]
>> ${sip_looped
>> _call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>> [default->remove_plus_of_d
>> st_num] continue=true
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS)
>> [remove_plus_of_dst_n
>> um] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Action
>> set(destination_number=9123
>>  ) INLINE
>>
>> EXECUTE sofia/internal/4001 at 192.168.200.100 set(destination_number=9123)
>> 2011-05-08 09:35:44.888374 [DEBUG] mod_dptools.c:1060
>> sofia/internal/4001 at 192.16
>>
>> 8.200.100 SET [destination_number]=[9123]
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>> [default->remove_plus_of_s
>> rc_num] continue=true
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL)
>> [remove_plus_of_src_n
>>
>> um] caller_id_number(4001) =~ /^\+(\d+)$/ break=on-false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>> [default->7_8_to_Lync] con
>> tinue=false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
>> destina
>>
>> tion_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->9_to_GW]
>> continu
>> e=false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
>> destination
>>
>> _number(+9123) =~ /^(9\d+)$/ break=on-false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->1_to_IPP]
>> contin
>> ue=false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
>> destinatio
>> n_number(+9123) =~ /^\+{0,1}(1\d{3})$/ break=on-false
>>
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->DISA]
>> continue=f
>> alse
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [DISA]
>> destination_nu
>>
>> mber(+9123) =~ /^\*(3472)$/ break=on-false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>> [default->Recordings] cont
>> inue=false
>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [Recordings]
>> destinat
>>
>> ion_number(+9123) =~ /^\*(732673)$/ break=on-false
>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:119
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/
>>
>> internal/4001 at 192.168.200.100 [BREAK]
>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:364
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State ROUTING going to sleep
>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:325
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) Running State Change CS_EXECUTE
>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:371
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State EXECUTE
>> 2011-05-08 09:35:44.889374 [DEBUG] mod_sofia.c:240
>> sofia/internal/4001 at 192.168.2
>> 00.100 SOFIA EXECUTE
>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:157
>> sofia/interna
>>
>> l/4001 at 192.168.200.100 Standard EXECUTE
>>  2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:189
>> sofia/intern
>>
>> al/4001 at 192.168.200.100 has executed the last dialplan instruction,
>> hanging up.
>> 2011-05-08 09:35:44.889374 [DEBUG] switch_channel.c:2563
>> (sofia/internal/4001 at 19
>>
>> 2.168.200.100) Callstate Change RINGING -> HANGUP
>> 2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:191 Hangup
>> sofia
>>
>> /internal/4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>  2011-05-08 09:35:44.890375 [DEBUG] switch_channel.c:2579 Send signal
>> sofia/inter
>>
>> nal/4001 at 192.168.200.100 [KILL]
>> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/
>>
>> internal/4001 at 192.168.200.100 [BREAK]
>> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:371
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State EXECUTE going to sleep
>>  2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:325
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) Running State Change CS_HANGUP
>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State HANGUP
>> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:457 Channel
>> sofia/internal/4001 at 1
>>
>> 92.168.200.100 hanging up, cause: NORMAL_CLEARING
>> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:519 Responding to INVITE
>> with: 48
>> 0
>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal
>>
>> /4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State HANGUP going to sleep
>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:356
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/
>>
>> internal/4001 at 192.168.200.100 [BREAK]
>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:325
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) Running State Change CS_REPORTING
>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:625
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State REPORTING
>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal
>>
>> /4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:625
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State REPORTING going to sleep
>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:350
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/
>>
>> internal/4001 at 192.168.200.100 [BREAK]
>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1288 Session 4
>> (sofia/i
>>
>> nternal/4001 at 192.168.200.100) Locked, Waiting on external entities
>> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1306 Session 4
>> (sofia/
>>
>> internal/4001 at 192.168.200.100) Ended
>> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1308 Close
>> Channel sof
>>
>> ia/internal/4001 at 192.168.200.100 [CS_DESTROY]
>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:454
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:457
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) Running State Change CS_DESTROY
>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State DESTROY
>> 2011-05-08 09:35:45.154390 [DEBUG] mod_sofia.c:362
>> sofia/internal/4001 at 192.168.2
>> 00.100 SOFIA DESTROY
>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal
>>
>> /4001 at 192.168.200.100 Standard DESTROY
>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
>> (sofia/intern
>>
>> al/4001 at 192.168.200.100) State DESTROY going to sleep
>>
>>
>> 2011/5/8 David Ponzone <david.ponzone at ipeva.fr>
>>
>>> Avi,
>>>
>>> are you sure about that ?
>>>
>>> I have LUA scripts which rewrite the destination_number with SetVariable
>>> and that works fine.
>>> Of course, to do that in the XML dialplan, as said before, inline should
>>> be used to set the value.
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 07/05/2011 ? 19:52, Avi Marcus a ?crit :
>>>
>>> Indeed, inline is a missing ingredient. As well though is the previously
>>> mentioned - you can't (directly) overwrite the destination number. You can
>>> either use a new variable throughout, or transfer to set the new destination
>>> number. Note that transferring causes the dialplan to be re-run from the
>>> start..
>>>
>>> -Avi
>>>
>>> On Sat, May 7, 2011 at 7:13 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>>>
>>>> There is certainly more than one way to skin this cat...
>>>>
>>>> However, to keep with your current method you should read up on
>>>> dialplan hunting vs. execution and "inline" execution:
>>>>
>>>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>>>
>>>> In short you need to add inline="true" to your first extension.
>>>>
>>>> On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>> > Gurus,
>>>> >
>>>> > i met an issue for dial plan, it sounds easy but puzzled me a few days
>>>> not
>>>> > fix it yet...i belive gurus here could help me...
>>>> >
>>>> > i want to remove the + of destination number before routing to PSTN
>>>> GW, e.g.
>>>> > when i dial +9123, i would like FS remove +, and then route to the
>>>> PSTN GW,
>>>> > below is the dial plan and log,
>>>> >
>>>> > See the log, FS did remove '+' of '+9123', but then when it checks the
>>>> rule
>>>> > ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then
>>>> failed.
>>>> >
>>>> > Thanks.
>>>> >
>>>> >
>>>> >
>>>> > Dial plan:
>>>> >
>>>> > <extension name="remove_plus_of_dst_num" continue="true">
>>>> >
>>>> >    <condition field="destination_number" expression="^\+(\d+)$">
>>>> >
>>>> >        <action application="set"
>>>> data="effective_destination_number=$1"/>
>>>> >
>>>> >    </condition>
>>>> >
>>>> > </extension>
>>>> >
>>>> > ?...
>>>> >
>>>> > <extension name="9_to_GW" >
>>>> >
>>>> >    <condition field="destination_number" expression="^(9\d+)$">
>>>> >
>>>> >        <action application="bridge"
>>>> > data="sofia/internal/$1 at 192.168.200.101"/>
>>>> >
>>>> >    </condition>
>>>> >
>>>> > </extension>
>>>> >
>>>> >
>>>> >
>>>> > Log:
>>>> >
>>>> > freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
>>>> > 192.168.200.201
>>>> >
>>>> >  Approved by acl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>".
>>>> Access Granted.
>>>> >
>>>> > 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
>>>> > sofia/inter
>>>> >
>>>> > nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
>>>> >
>>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
>>>> > sofia/internal/+4001 at 192
>>>> >
>>>> > .168.200.100 entering state [received][100]
>>>> >
>>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
>>>> >
>>>> > v=0
>>>> >
>>>> > o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
>>>> >
>>>> > s=Phone-Call
>>>> >
>>>> > c=IN IP4 192.168.200.201
>>>> >
>>>> > t=0 0
>>>> >
>>>> > m=audio 6060 RTP/AVP 8 0 96
>>>> >
>>>> > a=rtpmap:8 PCMA/8000
>>>> >
>>>> > a=rtpmap:0 PCMU/8000
>>>> >
>>>> > a=rtpmap:96 telephone-event/8000
>>>> >
>>>> > a=fmtp:96 0-15
>>>> >
>>>> > a=ptime:20
>>>> >
>>>> >
>>>> >
>>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
>>>> > (sofia/internal/+4001 at 192.168.20
>>>> >
>>>> > 0.100) State Change CS_NEW -> CS_INIT
>>>> >
>>>> > 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> > sofia/
>>>> >
>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) Running State Change CS_INIT
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State INIT
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
>>>> > sofia/internal/+4001 at 192.168.2
>>>> >
>>>> > 00.100 SOFIA INIT
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
>>>> > (sofia/internal/+4001 at 192.168
>>>> >
>>>> > .200.100) State Change CS_INIT -> CS_ROUTING
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> > sofia/
>>>> >
>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State INIT going to sleep
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
>>>> > (sofia/internal/+4001 at 1
>>>> >
>>>> > 92.168.200.100) Callstate Change DOWN -> RINGING
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State ROUTING
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
>>>> > sofia/internal/+4001 at 192.168.
>>>> >
>>>> > 200.100 SOFIA ROUTING
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
>>>> > sofia/internal
>>>> >
>>>> > /+4001 at 192.168.200.100 Standard ROUTING
>>>> >
>>>> > 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing
>>>> +4001
>>>> > <+4001
>>>> >
>>>> >>->+9123 in context default
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> [default->unloop]
>>>> > continu
>>>> >
>>>> > e=false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
>>>> > ${unroll_lo
>>>> >
>>>> > ops}(true) =~ /^true$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
>>>> > ${sip_loope
>>>> >
>>>> > d_call}() =~ /^true$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> > [default->remove_plus_of_
>>>> >
>>>> > dst_num] continue=true
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>>> > [remove_plus_of_dst_
>>>> >
>>>> > num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>>> > set(destination_number=912
>>>> >
>>>> > 3)
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> > [default->remove_plus_of_
>>>> >
>>>> > src_num] continue=true
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>>> > [remove_plus_of_src_
>>>> >
>>>> > num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>>> > set(effective_caller_id_na
>>>> >
>>>> > me=4001)
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>>> > set(effective_caller_id_nu
>>>> >
>>>> > mber=4001)
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> > [default->7_8_to_Lync] co
>>>> >
>>>> > ntinue=false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>>> [7_8_to_Lync]
>>>> > destin
>>>> >
>>>> > ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> [default->9_to_GW]
>>>> > contin
>>>> >
>>>> > ue=false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
>>>> > destinatio
>>>> >
>>>> > n_number(+9123) =~ /^(9\d+)$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> [default->1_to_IPP]
>>>> > conti
>>>> >
>>>> > nue=false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>>> [1_to_IPP]
>>>> > destinati
>>>> >
>>>> > on_number(+9123) =~ /^(1\d{3})$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> [default->DISA]
>>>> > continue=
>>>> >
>>>> > false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
>>>> > destination_n
>>>> >
>>>> > umber(+9123) =~ /^\*(3472)$/ break=on-false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>> [default->Recordings]
>>>> > con
>>>> >
>>>> > tinue=false
>>>> >
>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>>> [Recordings]
>>>> > destina
>>>> >
>>>> > tion_number(+9123) =~ /^\*(732673)$/ break=on-false
>>>> >
>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>>>> >
>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> > sofia/
>>>> >
>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>> >
>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State ROUTING going to sleep
>>>> >
>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
>>>> >
>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State EXECUTE
>>>> >
>>>> > 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
>>>> > sofia/internal/+4001 at 192.168.
>>>> >
>>>> > 200.100 SOFIA EXECUTE
>>>> >
>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
>>>> > sofia/interna
>>>> >
>>>> > l/+4001 at 192.168.200.100 Standard EXECUTE
>>>> >
>>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100set(destination_number=9123)
>>>> >
>>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>>> > sofia/internal/+4001 at 192.1
>>>> >
>>>> > 68.200.100 SET [destination_number]=[9123]
>>>> >
>>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>>> > set(effective_caller_id_name=4001)
>>>> >
>>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>>> > sofia/internal/+4001 at 192.1
>>>> >
>>>> > 68.200.100 SET [effective_caller_id_name]=[4001]
>>>> >
>>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>>> > set(effective_caller_id_number=4001
>>>> >
>>>> > )
>>>> >
>>>> > 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
>>>> > sofia/internal/+4001 at 192.1
>>>> >
>>>> > 68.200.100 SET [effective_caller_id_number]=[4001]
>>>> >
>>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
>>>> > sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100 has executed the last dialplan instruction,
>>>> hanging
>>>> > up.
>>>> >
>>>> >
>>>> >
>>>> > 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
>>>> > (sofia/internal/+4001 at 1
>>>> >
>>>> > 92.168.200.100) Callstate Change RINGING -> HANGUP
>>>> >
>>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191
>>>> Hangup
>>>> > sofia
>>>> >
>>>> > /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>>> >
>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
>>>> > sofia/inter
>>>> >
>>>> > nal/+4001 at 192.168.200.100 [KILL]
>>>> >
>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> > sofia/
>>>> >
>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>> >
>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State EXECUTE going to sleep
>>>> >
>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
>>>> >
>>>> > 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State HANGUP
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
>>>> > sofia/internal/+4001@
>>>> >
>>>> > 192.168.200.100 hanging up, cause: NORMAL_CLEARING
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to
>>>> INVITE
>>>> > with: 48
>>>> >
>>>> > 0
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
>>>> > sofia/internal
>>>> >
>>>> > /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State HANGUP going to sleep
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> > sofia/
>>>> >
>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
>>>> >
>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State REPORTING
>>>> >
>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
>>>> > sofia/internal
>>>> >
>>>> > /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>>>> >
>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State REPORTING going to sleep
>>>> >
>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>>>> >
>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> > sofia/
>>>> >
>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>> >
>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session
>>>> 2
>>>> > (sofia/i
>>>> >
>>>> > nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
>>>> >
>>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306 Session
>>>> 2
>>>> > (sofia/
>>>> >
>>>> > internal/+4001 at 192.168.200.100) Ended
>>>> >
>>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close
>>>> Channel
>>>> > sof
>>>> >
>>>> > ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
>>>> >
>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>>>> >
>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
>>>> >
>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State DESTROY
>>>> >
>>>> > 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
>>>> > sofia/internal/+4001 at 192.168.
>>>> >
>>>> > 200.100 SOFIA DESTROY
>>>> >
>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
>>>> > sofia/internal
>>>> >
>>>> > /+4001 at 192.168.200.100 Standard DESTROY
>>>> >
>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>>> > (sofia/intern
>>>> >
>>>> > al/+4001 at 192.168.200.100) State DESTROY going to sleep
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at 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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>>> 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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>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>
>>>
>>
>> _______________________________________________
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cjbujold at accra.ca  Mon May  9 17:03:21 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Mon, 9 May 2011 10:03:21 -0300
Subject: [Freeswitch-users] Unable to install Centos on AMD Motherboard
	(SB850)
Message-ID: <00f401cc0e49$79497340$6bdc59c0$@accra.ca>

Upgrading our Freeswitch server to a newer PC which has an AMD SB850 chipset
motherboard and Centos is unable to install since it does not have the SATA
drivers support.  First tried finding a driver with no success.  Then tried
seeing when Centos 6 was going to be released but it looks like nobody
really knows.  So now I'm looking at Ubuntu 11.4 as a possible OS.    I like
Centos but if it can't keep up and can only support older equipment  I'm
wondering if it is the right platform for us.  Is anybody using Ubuntu 11.4
and how does Freeswitch work on it?

 

Thanks

 

cjb

 

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From jcasale at activenetwerx.com  Mon May  9 17:13:07 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 9 May 2011 13:13:07 +0000
Subject: [Freeswitch-users] Unable to install Centos on AMD
	Motherboard	(SB850)
In-Reply-To: <00f401cc0e49$79497340$6bdc59c0$@accra.ca>
References: <00f401cc0e49$79497340$6bdc59c0$@accra.ca>
Message-ID: <assp.0110554d82.92097A6A775D5147B1078E3F15430B924C13D3@prato.activenetwerx.local>

>Upgrading our Freeswitch server to a newer PC which has an AMD SB850 chipset
>motherboard and Centos is unable to install since it does not have the SATA drivers support.

Does your BIOS have an AHCI setting, possibly you are in some RAID mode?



From cjbujold at accra.ca  Mon May  9 17:58:45 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Mon, 9 May 2011 10:58:45 -0300
Subject: [Freeswitch-users] Unable to install Centos on
	AMD	Motherboard	(SB850)
In-Reply-To: <assp.0110554d82.92097A6A775D5147B1078E3F15430B924C13D3@prato.activenetwerx.local>
References: <00f401cc0e49$79497340$6bdc59c0$@accra.ca>
	<assp.0110554d82.92097A6A775D5147B1078E3F15430B924C13D3@prato.activenetwerx.local>
Message-ID: <011301cc0e51$36643b20$a32cb160$@accra.ca>

Yes thought about that and set it to sata and still not recognized by Centos

cjb

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joseph
L. Casale
Sent: May-09-11 10:13 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Unable to install Centos on AMD Motherboard
(SB850)

>Upgrading our Freeswitch server to a newer PC which has an AMD SB850 
>chipset motherboard and Centos is unable to install since it does not have
the SATA drivers support.

Does your BIOS have an AHCI setting, possibly you are in some RAID mode?


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



From jcasale at activenetwerx.com  Mon May  9 18:18:31 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 9 May 2011 14:18:31 +0000
Subject: [Freeswitch-users] Unable to install Centos
	on	AMD	Motherboard	(SB850)
In-Reply-To: <011301cc0e51$36643b20$a32cb160$@accra.ca>
References: <00f401cc0e49$79497340$6bdc59c0$@accra.ca>
	<92097A6A775D5147B1078E3F15430B924C13D3@prato.activenetwerx.local>
	<011301cc0e51$36643b20$a32cb160$@accra.ca>
Message-ID: <assp.011011f5ea.92097A6A775D5147B1078E3F15430B924C2665@prato.activenetwerx.local>

>Yes thought about that and set it to sata and still not recognized by Centos

Boot a Fedora 14/15 live cd and do a lspci and boot the CentOS installer, drop
to a shell and do the same, getting OT for this list but it might prove insightful
to see the difference. RHEL makes a ddiskit util to build a dd image for adding
support but seriously, for the cost of a mobo, I'd just get something different myself:)

jlc


From gmaruzz at gmail.com  Mon May  9 18:26:27 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 9 May 2011 16:26:27 +0200
Subject: [Freeswitch-users] Unable to install Centos on AMD Motherboard
	(SB850)
In-Reply-To: <assp.011011f5ea.92097A6A775D5147B1078E3F15430B924C2665@prato.activenetwerx.local>
References: <00f401cc0e49$79497340$6bdc59c0$@accra.ca>
	<92097A6A775D5147B1078E3F15430B924C13D3@prato.activenetwerx.local>
	<011301cc0e51$36643b20$a32cb160$@accra.ca>
	<assp.011011f5ea.92097A6A775D5147B1078E3F15430B924C2665@prato.activenetwerx.local>
Message-ID: <BANLkTimdmkEsWwd4oxHRfdaUSZkcJJeEGw@mail.gmail.com>

If you like CentOS/RedHat, go for Scientific Linux, it's a clone of RHEL 6
If you want to go Ubuntu, stay with 10.04 LTS

On Mon, May 9, 2011 at 4:18 PM, Joseph L. Casale
<jcasale at activenetwerx.com> wrote:
>>Yes thought about that and set it to sata and still not recognized by Centos
>
> Boot a Fedora 14/15 live cd and do a lspci and boot the CentOS installer, drop
> to a shell and do the same, getting OT for this list but it might prove insightful
> to see the difference. RHEL makes a ddiskit util to build a dd image for adding
> support but seriously, for the cost of a mobo, I'd just get something different myself:)
>
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From curriegrad2004 at gmail.com  Mon May  9 18:38:11 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 9 May 2011 07:38:11 -0700
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <1304933096.28727.28.camel@luna.tc.commsmundi.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<4DC72997.4090900@sns.eu>
	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
	<4DC784BB.4080907@seventhsignal.de>
	<1304933096.28727.28.camel@luna.tc.commsmundi.com>
Message-ID: <BANLkTinkY34_SsQHyDWZqjFvHH37Kft1qA@mail.gmail.com>

There is no "worst" distro for freeswitch. It depends on how you are
going to use FreeSwitch for, really.

On Mon, May 9, 2011 at 2:24 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> Using debian lenny and squeeze, it's boringly stable. Thinking of
> switching to Gentoo, Ubuntu, or LFS for some more fun. What's the worse
> distro for freeswitch?
>
> Fran?ois.
>
> On Mon, 2011-05-09 at 08:07 +0200, Michal Bielicki wrote:
>> I guess Jan just wanted to explain. Sometimes its better to clear up
>> things for people in their local language, don't you think ?
>>
>> Am 09.05.2011 06:01, schrieb curriegrad2004:
>> > It would be beneficial to the majority of the users if we continued
>> > this discussion in English
>> >
>> > On Sun, May 8, 2011 at 5:09 PM, ?<rhuddleston at gmail.com> wrote:
>> >> Auf English bitte? Mein deutsch ist sehr sclecht und ich kann nicht verstehen
>> >>
>> >> On May 8, 2011, at 7:39 PM, Jan Riedinger <riedinger at sns.eu> wrote:
>> >>
>> >>> Hi Dirk,
>> >>>
>> >>> Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein
>> >>> Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.
>> >>>
>> >>> Viele Gr??e
>> >>> ? ? Jan
>> >>>
>> >>> Am 06.05.2011 16:25, schrieb Max Clark:
>> >>>> Hello all,
>> >>>>
>> >>>> Recent developments (or absolute lack of) within the CentOS project
>> >>>> and its perceived long term viability has forced an internal
>> >>>> discussion to select a successor distribution. The most likely
>> >>>> candidates at this point are Ubuntu and its LTS releases for servers,
>> >>>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>> >>>> each are growing and the issue is complicated.
>> >>>>
>> >>>> With CentOS 5.x being the reference distro for FreeSWITCH development
>> >>>> I'm curious if this conversation has started among the FreeSWITCH
>> >>>> developers, and if it has, what is the project leaning to for a
>> >>>> successor distribution?
>> >>>>
>> >>>> Thanks,
>> >>>> Max
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>> --
>> >>> Jan Riedinger ? ? ? ? ? ? ? ? ? ? ? ? ? Phone : ?+49-30-39 73 19 66
>> >>> Dipl.-Inf. | Managing Director ? ? ? ? ?Fax ? : ?+49-30-39 73 19 64
>> >>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? E-Mail: ?riedinger at sns.eu
>> >>> SNS Consult GmbH ? ? ? ? ? ? ? ? ? ? ? ?ICQ ? : ?163-237-041
>> >>> S?dwestkorso 49a ? ? ? ? ? ? ? ? ? ? ? ?MSN ? : ?jan at sns-consult.de
>> >>> 14197 Berlin GERMANY ? ? ? ? ? ? ? ? ? ?Skype : ?Jan Riedinger
>> >>>
>> >>> AG Charlottenburg - HRB 71973
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Dennis.Young at supportkids.com  Mon May  9 18:40:18 2011
From: Dennis.Young at supportkids.com (Dennis Young)
Date: Mon, 9 May 2011 09:40:18 -0500
Subject: [Freeswitch-users] Trillium Error
In-Reply-To: <BANLkTikkmgxH8vJM6C7hthNKPahXuL2HLQ@mail.gmail.com>
References: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49C@jehuty.supportkids.com>
	<BANLkTikkmgxH8vJM6C7hthNKPahXuL2HLQ@mail.gmail.com>
Message-ID: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49F@jehuty.supportkids.com>

Moises,

Here are my config files: http://pastebin.freeswitch.org/16258 .

Also note that I have two open bugs reports with freetdm:  OPENZAP-156<http://jira.freeswitch.org/browse/OPENZAP-156>, OPENZAP-142<http://jira.freeswitch.org/browse/OPENZAP-142>

The debug information from OPENZAP-156<http://jira.freeswitch.org/browse/OPENZAP-156> may help.

...dly

From: Moises Silva [mailto:moises.silva at gmail.com]
Sent: Saturday, May 07, 2011 12:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trillium Error

On Fri, May 6, 2011 at 11:46 AM, Dennis Young <Dennis.Young at supportkids.com<mailto:Dennis.Young at supportkids.com>> wrote:
All,

I?m seeing this error message in the command prompt window that is running Freeswitch but it?s not showing up in the freeswitch log file or remote console.  I running GIT HEAD 5/1/11 on WIN32 system.

UNTSS: sw error:  ent: 010  inst: 000  proc id: 000
                file: ..\..\trillium\in\in_bdy4.c line: 483 errcode: 14536 errcls: ERRCLS_DEBUG
                errval: 00018  errdesc: inUsrT302S25() failed, timer not defined for switch.


Hello Dennis,

Can you provide your freetdm.conf and freetdm.conf.xml? (use pastebin)

Also, which version of libsng_isdn are you using?

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<mailto:moy at sangoma.com>

Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law.  Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited.  If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.
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From curriegrad2004 at gmail.com  Mon May  9 18:40:29 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 9 May 2011 07:40:29 -0700
Subject: [Freeswitch-users] Unable to install Centos on AMD Motherboard
	(SB850)
In-Reply-To: <BANLkTimdmkEsWwd4oxHRfdaUSZkcJJeEGw@mail.gmail.com>
References: <00f401cc0e49$79497340$6bdc59c0$@accra.ca>
	<92097A6A775D5147B1078E3F15430B924C13D3@prato.activenetwerx.local>
	<011301cc0e51$36643b20$a32cb160$@accra.ca>
	<assp.011011f5ea.92097A6A775D5147B1078E3F15430B924C2665@prato.activenetwerx.local>
	<BANLkTimdmkEsWwd4oxHRfdaUSZkcJJeEGw@mail.gmail.com>
Message-ID: <BANLkTiku6-BD8uuYvU521=2dzw=tGDHGSA@mail.gmail.com>

You're better off asking the people over at CentOS to see how you are
going to be able to install the OS for your specific board, but as
others have outlined, SL 6 will support the board you're using because
of it being a newer distro.

On Mon, May 9, 2011 at 7:26 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> If you like CentOS/RedHat, go for Scientific Linux, it's a clone of RHEL 6
> If you want to go Ubuntu, stay with 10.04 LTS
>
> On Mon, May 9, 2011 at 4:18 PM, Joseph L. Casale
> <jcasale at activenetwerx.com> wrote:
>>>Yes thought about that and set it to sata and still not recognized by Centos
>>
>> Boot a Fedora 14/15 live cd and do a lspci and boot the CentOS installer, drop
>> to a shell and do the same, getting OT for this list but it might prove insightful
>> to see the difference. RHEL makes a ddiskit util to build a dd image for adding
>> support but seriously, for the cost of a mobo, I'd just get something different myself:)
>>
>> jlc
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
>


From Dennis.Young at supportkids.com  Mon May  9 18:43:13 2011
From: Dennis.Young at supportkids.com (Dennis Young)
Date: Mon, 9 May 2011 09:43:13 -0500
Subject: [Freeswitch-users] Trillium Error
In-Reply-To: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49F@jehuty.supportkids.com>
References: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49C@jehuty.supportkids.com>
	<BANLkTikkmgxH8vJM6C7hthNKPahXuL2HLQ@mail.gmail.com>
	<32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A49F@jehuty.supportkids.com>
Message-ID: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A4A0@jehuty.supportkids.com>

Sorry, you also said you need the libsng_isdn version:  7.1.0-win32

...dly

Dennis Young |CIO |Phone 512.437.3901 | Fax 512.437.7202
Supportkids Services, Inc. | P.O. Box 18988 | Austin | TX 78760 | Phone 512.437.6000 | Fax 512.437.6030

From: Dennis Young
Sent: Monday, May 09, 2011 9:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trillium Error

Moises,

Here are my config files: http://pastebin.freeswitch.org/16258 .

Also note that I have two open bugs reports with freetdm:  OPENZAP-156<http://jira.freeswitch.org/browse/OPENZAP-156>, OPENZAP-142<http://jira.freeswitch.org/browse/OPENZAP-142>

The debug information from OPENZAP-156<http://jira.freeswitch.org/browse/OPENZAP-156> may help.

...dly

From: Moises Silva [mailto:moises.silva at gmail.com]
Sent: Saturday, May 07, 2011 12:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trillium Error

On Fri, May 6, 2011 at 11:46 AM, Dennis Young <Dennis.Young at supportkids.com<mailto:Dennis.Young at supportkids.com>> wrote:
All,

I?m seeing this error message in the command prompt window that is running Freeswitch but it?s not showing up in the freeswitch log file or remote console.  I running GIT HEAD 5/1/11 on WIN32 system.

UNTSS: sw error:  ent: 010  inst: 000  proc id: 000
                file: ..\..\trillium\in\in_bdy4.c line: 483 errcode: 14536 errcls: ERRCLS_DEBUG
                errval: 00018  errdesc: inUsrT302S25() failed, timer not defined for switch.


Hello Dennis,

Can you provide your freetdm.conf and freetdm.conf.xml? (use pastebin)

Also, which version of libsng_isdn are you using?

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<mailto:moy at sangoma.com>

Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law. Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.   ??

Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law.  Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited.  If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.
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From steveu at coppice.org  Mon May  9 18:47:37 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 09 May 2011 22:47:37 +0800
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTinkY34_SsQHyDWZqjFvHH37Kft1qA@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>	<4DC72997.4090900@sns.eu>	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>	<4DC784BB.4080907@seventhsignal.de>	<1304933096.28727.28.camel@luna.tc.commsmundi.com>
	<BANLkTinkY34_SsQHyDWZqjFvHH37Kft1qA@mail.gmail.com>
Message-ID: <4DC7FE89.7040402@coppice.org>

On 05/09/2011 10:38 PM, curriegrad2004 wrote:
> There is no "worst" distro for freeswitch. It depends on how you are
> going to use FreeSwitch for, really.
Nonsense. The distro I will take the longest to learn to install is 
*clearly* the worst distro. :-)

Steve



From dnotivol at gmail.com  Mon May  9 19:16:48 2011
From: dnotivol at gmail.com (David Notivol)
Date: Mon, 9 May 2011 17:16:48 +0200
Subject: [Freeswitch-users] How to add a new language
In-Reply-To: <BANLkTi=nFTzzV3DtDFKVj6POQJo3v9bb0g@mail.gmail.com>
References: <BANLkTin=22R+tFVCB6sz-FcDXjZMUB-3oQ@mail.gmail.com>
	<BANLkTi=nFTzzV3DtDFKVj6POQJo3v9bb0g@mail.gmail.com>
Message-ID: <BANLkTi=J2yFc9S2Z75=n_xTLn+02ha05iQ@mail.gmail.com>

Avi, thanks for your answer.

You were right, I was missing to load the file. Now it's finding the
language and playing the right files.

David.

2011/5/9 Avi Marcus <avi at avimarcus.net>

> First check:
> /usr/local/freeswitch/conf/freeswitch.xml and see if you have a line like:
> <X-PRE-PROCESS cmd="include" data="lang/es/*.xml"/>
>
> It should be inside the <section name="phrases" description="Speech Phrase
> Management">
>
> -Avi
>
> On Mon, May 9, 2011 at 11:19 AM, David Notivol <dnotivol at gmail.com> wrote:
>
>> Hi all,
>>
>> I'm trying to make some tests with some custom audio files in English and
>> Spanish, but I'm not being able to add Spanish as language.
>> The module mod_say_es is compiled and loaded. I duplicated the structure
>> in lang/en to lang/es, but without any luck, always getting a "couldn't find
>> es language" message.
>>
>> Any advise or suggestion?
>> Thanks.
>>
>> --
>> Regards,
>> David Notivol
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Mon May  9 19:17:26 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 9 May 2011 11:17:26 -0400
Subject: [Freeswitch-users] SpanDSP: nocng for rxfax?
Message-ID: <BANLkTinrz80SjBhVSEZ=aapeQ6u1idWk9Q@mail.gmail.com>

Hello all,

I'm using tone_detect (1100Hz) to detect incoming faxes.

It's working pretty well but I'm noticing that there seem to be some
timing issues when using rxfax as rxfax seems to do its own tone
detection.  Is it possible to use "nocng" or something similar to tell
rxfax to skip tone detection (as I'm doing it with tone_detect first
anyway)?  Is this a terrible idea?

Thanks!

-- 
Kristian Kielhofner


From anthony.minessale at gmail.com  Mon May  9 19:52:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 9 May 2011 10:52:52 -0500
Subject: [Freeswitch-users] SpanDSP: nocng for rxfax?
In-Reply-To: <BANLkTinrz80SjBhVSEZ=aapeQ6u1idWk9Q@mail.gmail.com>
References: <BANLkTinrz80SjBhVSEZ=aapeQ6u1idWk9Q@mail.gmail.com>
Message-ID: <BANLkTinrP+yGxrSjo3QFE4avPJsh2jLsVw@mail.gmail.com>

I don't think it ever uses tone detection unless you are using the gateway.
the negotiation still needs to take place once you detect and forward it.



On Mon, May 9, 2011 at 10:17 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Hello all,
>
> I'm using tone_detect (1100Hz) to detect incoming faxes.
>
> It's working pretty well but I'm noticing that there seem to be some
> timing issues when using rxfax as rxfax seems to do its own tone
> detection. ?Is it possible to use "nocng" or something similar to tell
> rxfax to skip tone detection (as I'm doing it with tone_detect first
> anyway)? ?Is this a terrible idea?
>
> Thanks!
>
> --
> 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
>



-- 
Anthony Minessale II

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 May  9 20:06:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 May 2011 09:06:49 -0700
Subject: [Freeswitch-users] Problem originate conference hangup
In-Reply-To: <Pine.LNX.4.64.1105060307050.9252@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
Message-ID: <BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>

Thomas,

What is the ultimate goal of this? I'm just curious what problem you're
trying to solve.

-MC

On Thu, May 5, 2011 at 6:47 PM, Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi all, i have setup a admingroup in a conference.
> My section in conference.conf.xml is:
> <group name="admin">
>  <control action="execute_application" digits="2"
>  data="bridge loopback/427/default"/>
>
> my 427:
> <extension name="DISA for conference dialout">
> <condition field="destination_number" expression="^(427)$">
>  <action application="answer"/>
>  <action application="sleep" data="100"/>
> <action application="read"
> data="2 30 'tone_stream://%(10000,0,3 50,440)' digits 30000 #"/>
> <action application="set"
> data="konfnum=${hash(select/confout/thomas)}"/>
> <action application="export"
> data="konfnum1=${hash(select/confout/thomas)}"/>
>
> <action application="system"
> data="$${base_dir}/scripts/callme.sh ${konfnum}  ${digits}" />
>
> My callme.sh:
> #!/bin/sh
> ##callout for a conference
> /usr/local/freeswitch/bin/fs_cli -x
>  "bgapi originate  {ignore_early_media=true,originate_retries=100,
>  origination_caller_id_name=Callback,originate_retry_sleep_ms=10000,
>  originate_timeout=900}loopback/$2/disa &conference($1 at admin)"
>
> My Adminextension:
>    <extension name="local admin  konferenz">
>      <condition field="destination_number" expression="^(7676)$">
>        <action application="answer"/>
>     <action application="sleep" data="1000"/>
> <action application="hash"
> data="insert/confout/thomas/$1-${domain_name}"/>
> <action application="conference" data="$1-${domain_name}@admin"/>
>  </condition>
>  </extension>
>
> These works fine from my sipadapters.
> I  enter the adminextension, press 2 heare the dialtone, enter the outside
> numer, press # and go back to the conference. This is ok.
>
> The problem: i use the originate command to call me self.
> I enter the admin extension. Then i press 2 and hear the dialtone, i enter
> the phonenumber and press #. Freeswitch hangup my callbackline.
> I like to go back to my conference.
> the same from my sip-adapters.
> Can you help me please?
> I neet a parameter on the originate command or so?
>
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May  9 20:07:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 May 2011 09:07:50 -0700
Subject: [Freeswitch-users] Slow dialplan execution
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A339E98@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A339E98@ITEX1.bc.local>
Message-ID: <BANLkTinGx=7jtGFFpmmgV+vPXP4Twk-BOg@mail.gmail.com>

Have you done a SIP trace to see where the lag is?
-MC

2011/5/7 Josh M. Patten <jpatten at co.brazos.tx.us>

>  I seem to be having a big problem when using the "Mad-Boss" Paging setup
> described in: http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom
>
> To connect 5 phones to the conference bridge takes anywhere from 3 - 4
> seconds which is entirely too long to be useful. Here is the dialplan entry:
>
> <extension name="Test Paging Group">
>       <condition field="destination_number" expression="^(3400)$"
> break="never">
>         <action application="set" data="api_hangup_hook=conference 3400
> kick all"/>
>         <action application="answer"/>
>         <action application="export"
> data="sip_invite_params=intercom=true"/>
>         <action application="export" data="sip_auto_answer=true"/>
>         <action application="set"
> data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
>         <action application="set"
> data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
>         <action application="set"
> data="conference_auto_outcall_timeout=60"/>
>         <action application="set"
> data="conference_auto_outcall_flags=none"/>
>         <action application="conference_set_auto_outcall"
> data="{alert_info=sipXpage}sofia/custom_dialplan/7001 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7002 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7004 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7005 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7006 at sipxpbx.bc.local
> ;sipx-noroute=VoiceMail;sipx-userforward=false"/>
>       </condition>
>       <condition field="destination_number" expression="^(3400)$">
>         <action application="conference" data="3400 at default"/>
>         <action application="conference" data="3400 kick all"/>
>       </condition>
>     </extension>
>
> Here are my questions:
>
> Is it possible to speed up XML dialplan execution to overcome this issue? I
> promise I have enough CPU, RAM, and bandwidth to handle the flurry of SIP
> messages.
>
> If it is not possible to speed up the XML dialplan execution is there a way
> to "thread" the dialplan execution so that all of these
> conference_set_auto_outcall applications can be run simultaneously?
>
> If not, would attempting to use a database based dialplan be worth my time?
>
> Because of the complexity of the networks I'm dealing with I'd rather not
> try to fight with multicast.
>
> 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 steveu at coppice.org  Mon May  9 20:08:02 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 10 May 2011 00:08:02 +0800
Subject: [Freeswitch-users] SpanDSP: nocng for rxfax?
In-Reply-To: <BANLkTinrz80SjBhVSEZ=aapeQ6u1idWk9Q@mail.gmail.com>
References: <BANLkTinrz80SjBhVSEZ=aapeQ6u1idWk9Q@mail.gmail.com>
Message-ID: <4DC81162.7070105@coppice.org>

On 05/09/2011 11:17 PM, Kristian Kielhofner wrote:
> Hello all,
>
> I'm using tone_detect (1100Hz) to detect incoming faxes.
>
> It's working pretty well but I'm noticing that there seem to be some
> timing issues when using rxfax as rxfax seems to do its own tone
> detection.  Is it possible to use "nocng" or something similar to tell
> rxfax to skip tone detection (as I'm doing it with tone_detect first
> anyway)?  Is this a terrible idea?
>
> Thanks!
>
rxfax does not attempt to detect CNG. Actually, no receiving FAX machine 
does. The tone was put there for humans, not machines.

Steve



From fdelawarde at wirelessmundi.com  Mon May  9 20:11:19 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 09 May 2011 18:11:19 +0200
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <4DC7FE89.7040402@coppice.org>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<4DC72997.4090900@sns.eu>	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
	<4DC784BB.4080907@seventhsignal.de>
	<1304933096.28727.28.camel@luna.tc.commsmundi.com>
	<BANLkTinkY34_SsQHyDWZqjFvHH37Kft1qA@mail.gmail.com>
	<4DC7FE89.7040402@coppice.org>
Message-ID: <1304957479.28727.60.camel@luna.tc.commsmundi.com>

On Mon, 2011-05-09 at 22:47 +0800, Steve Underwood wrote: 
> On 05/09/2011 10:38 PM, curriegrad2004 wrote:
> > There is no "worst" distro for freeswitch. It depends on how you are
> > going to use FreeSwitch for, really.
> Nonsense. The distro I will take the longest to learn to install is 
> *clearly* the worst distro. :-)

Come on, give me names... Someone must have some horror story trying to
install FS in some strange distro!

Fran?ois.



From michele.garribba at gmail.com  Sat May  7 18:49:48 2011
From: michele.garribba at gmail.com (Michele Garribba)
Date: Sat, 7 May 2011 16:49:48 +0200
Subject: [Freeswitch-users] freeswitch configure file empty
In-Reply-To: <BANLkTikPeCeu2zocHOc6RaqVZGx8Y9n30A@mail.gmail.com>
References: <BA2E8AE6-F9E1-4CDA-99B1-6967B371F4D8@gmail.com>
	<BANLkTikPeCeu2zocHOc6RaqVZGx8Y9n30A@mail.gmail.com>
Message-ID: <8E1966F6-3D1E-4C14-B121-843BA90A2083@gmail.com>

i made it, configure start, compiled zrtp
ran builzrtp.sh then ./configure as the wiki way
and it says cannot guess build type; you must specity one
and it brekas.

smaikol

> You must first do:
> ./bootsrap.sh
> then configure
> 
> -giovanni
> 
> On 5/6/11, Michele Garribba <michele.garribba at gmail.com> wrote:
>> Hi,
>> 
>> I'm tryin  to setup a system with freeswitch and zrtp.
>> Using ubuntu server 10.10
>> 
>> Everything works during the install process following the wiki
>> but when i got to the ./configure --enable-zrtp command nothing
>> happens. Looking the configure file, it's empty!
>> 
>> I downloaded freeswitch with git, as the guide told.
>> 
>> Any suggetion?
>> My i try with wget and pickup the tar? In the zrtp wiki says to
>> use only git to do this.
>> 
>> Thanks
>> smaikol
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> -- 
> 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







From michele.garribba at gmail.com  Mon May  9 12:34:37 2011
From: michele.garribba at gmail.com (Michele Garribba)
Date: Mon, 9 May 2011 10:34:37 +0200
Subject: [Freeswitch-users] zrtp error
Message-ID: <F803E4AC-C5A6-4F79-B555-263C47081412@gmail.com>

Hi,

I'm trying to compile a FS server with zrtp but i get the following errors

cc1: warnings being treated as errors
src/switch_rtp.c: In function ?zrtp_cache_save_callback?:
src/switch_rtp.c:644: error: implicit declaration of function ?zrtp_def_cache_store?
src/switch_rtp.c: In function ?zrtp_event_callback?:
src/switch_rtp.c:686: error: implicit declaration of function ?zrtp_verified_set?
src/switch_rtp.c:686: error: ?zrtp_session_info_t? has no member named ?sas_is_verified?
src/switch_rtp.c:748: error: ?zrtp_session_info_t? has no member named ?sas_is_verified?
src/switch_rtp.c:761: error: ?zrtp_session_info_t? has no member named ?sas_is_verified?
src/switch_rtp.c:774: error: ?zrtp_session_info_t? has no member named ?sas_is_verified?
src/switch_rtp.c: In function ?switch_rtp_init?:
src/switch_rtp.c:828: error: ?zrtp_config_t? has no member named ?def_cache_path?
src/switch_rtp.c:828: error: ?zrtp_config_t? has no member named ?def_cache_path?
src/switch_rtp.c:828: error: ?zrtp_config_t? has no member named ?def_cache_path?
src/switch_rtp.c: In function ?switch_rtp_zerocopy_read_frame?:
src/switch_rtp.c:3476: error: ?zrtp_session_info_t? has no member named ?sas_is_verified?
src/switch_rtp.c: In function ?switch_rtp_write_frame?:
src/switch_rtp.c:4068: error: ?zrtp_session_info_t? has no member named ?sas_is_verified?
make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1
make: *** [all] Error 2

Following the wiki all goes well up to the make step.
My server is ubuntu 10.10, base system + FS requirements.
libzrtp is libzrtp-0.90.572

Any idea?
Wrong versions?

Thanks
smaikol

From yungwei at resolvity.com  Sun May  8 21:03:26 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Sun, 8 May 2011 13:03:26 -0400
Subject: [Freeswitch-users] Pass UUI through SIP
Message-ID: <33095823FD21DF429B481B5163264B79507A9FFC6A@VMBX102.ihostexchange.net>

Hi,

I want to check if Freeswitch supports passing UUI from a sip client to an IVR via SIP messages in the following scenario.
My google search found a few UUI related threads, which are not helpful.

A custom application tells Freeswitch to make an outbound call to someone using call management commands.
When the call is answered, Freeswitch connects the person to an IVR. (So now the person is talking to the IVR.)
In order to be able to find corresponding calls initiated from the application on the IVR side,
the application attches a UUID to each outbound call and the UUID is expected to pass to the IVR via SIP messages.
Thanks.
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From anthony.minessale at gmail.com  Mon May  9 20:09:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 9 May 2011 11:09:07 -0500
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <4DC7FE89.7040402@coppice.org>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<4DC72997.4090900@sns.eu>
	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
	<4DC784BB.4080907@seventhsignal.de>
	<1304933096.28727.28.camel@luna.tc.commsmundi.com>
	<BANLkTinkY34_SsQHyDWZqjFvHH37Kft1qA@mail.gmail.com>
	<4DC7FE89.7040402@coppice.org>
Message-ID: <BANLkTikjZqM1DObv-UaH5X7VNQwhWtPP9A@mail.gmail.com>

my official comment is:
It will work anywhere it works.  Anyone can do whatever they want but
there is a limit to how much free support we will provide when doing
things outside the parameters of our recommendations.

The good news is that I think the newest kernels are getting over a
dark time when timing was really bad as they tried to support more
virtual environments.  I have seen promising results on even ubuntu
and arch linux on bleeding edge kernels.  Centos 5.2-5.3 I can attest
to being rock solid but a bit outdated.  As time unfolds we will
ultimately become more confident supporting other platforms when we
have experts on those platforms willing to maintain packaging and do
support therein.



On Mon, May 9, 2011 at 9:47 AM, Steve Underwood <steveu at coppice.org> wrote:
> On 05/09/2011 10:38 PM, curriegrad2004 wrote:
>> There is no "worst" distro for freeswitch. It depends on how you are
>> going to use FreeSwitch for, really.
> Nonsense. The distro I will take the longest to learn to install is
> *clearly* the worst distro. :-)
>
> 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
>



-- 
Anthony Minessale II

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 steveayre at gmail.com  Mon May  9 20:21:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 9 May 2011 17:21:23 +0100
Subject: [Freeswitch-users] Pass UUI through SIP
In-Reply-To: <33095823FD21DF429B481B5163264B79507A9FFC6A@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B79507A9FFC6A@VMBX102.ihostexchange.net>
Message-ID: <BANLkTinR6mThwuYtprWNW9Lef7jhKemS5A@mail.gmail.com>

Any X- or P- headers in the SIP messages will be set as a variable.

E.g. The value of X-MyUUID will be loaded into the variable sip_X-MyUUID

You can then use that variable within the IVR.

-Steve


On 8 May 2011 18:03, Yungwei Chen <yungwei at resolvity.com> wrote:

>   Hi,
>
> I want to check if Freeswitch supports passing UUI from a sip client to an
> IVR via SIP messages in the following scenario.
> My google search found a few UUI related threads, which are not helpful.
>
> A custom application tells Freeswitch to make an outbound call to someone
> using call management commands.
> When the call is answered, Freeswitch connects the person to an IVR. (So
> now the person is talking to the IVR.)
> In order to be able to find corresponding calls initiated from the
> application on the IVR side,
> the application attches a UUID to each outbound call and the UUID is
> expected to pass to the IVR via SIP messages.
> 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 edpimentl at gmail.com  Mon May  9 20:26:09 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Mon, 9 May 2011 12:26:09 -0400
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <BANLkTikjZqM1DObv-UaH5X7VNQwhWtPP9A@mail.gmail.com>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>
	<4DC72997.4090900@sns.eu>
	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>
	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
	<4DC784BB.4080907@seventhsignal.de>
	<1304933096.28727.28.camel@luna.tc.commsmundi.com>
	<BANLkTinkY34_SsQHyDWZqjFvHH37Kft1qA@mail.gmail.com>
	<4DC7FE89.7040402@coppice.org>
	<BANLkTikjZqM1DObv-UaH5X7VNQwhWtPP9A@mail.gmail.com>
Message-ID: <BANLkTinOXiNyzMa=-t-YJtL-685NtJJW7A@mail.gmail.com>

Here is an excellent example when you may want Pro-Services and
migrate from CentOS.
http://www.nexenta.org/projects/site/wiki/WhyNexenta

Unbuntu distro with Solaris Kernel and full  instant access to ZFS,
DTrace, Containers, Fault Management, 32/64-bits multiarch

Why should I use Nexenta anyway?

As we already learned, Nexenta is based on Ubuntu, with an OpenSolaris
kernel (not unlike Debian GNU/kFreeBSD or Debian GNU/Hurd). So why
should I use Nexenta, as Ubuntu is probably supported/used by more
people, as is maybe more hardware?

You should because it got outstanding, mainline, features. You won't
have to patch your kernel or otherwise use unsupported features to get
instant access to ZFS, DTrace, Containers, Fault Management,
32/64-bits multiarch, ...

Plus, Solaris is known to be one of the best OS out there when it
comes to multi-threading, which is great with today's CPUs with
multiple cores and hardware threading (such as Intel HyperThreading or
Sparc CMT).

Outstanding Features

Let's dig into more details about what Nexenta can bring you:

ZFS: easy dynamic storage management (no SoftRAID + LVM + fs +
mountpoints) of use disk capacities, reliably (COW, RAIDZ), fast, with
snapshots (which also bring you system clones with rollback, usable as
fail-safe upgrade paths, because of integration with your bootloader
GRUB or OpenFirmware), deduplication and compression, Flash log
devices (for best performances)
DTrace: dynamic tracing of your software (want to diagnose about
anything happening on your machine now or in a timely manner?)
Containers: Linux Vserver made easy, simply put: a way to isolate processes
Fault Management: SMF (replaces alltogether initscripts, inetd, and
offers more features) and more
32/64-bits: Solaris is natively 32/64-bits for years, as is the Sparc
architecture, historically its primary target platform
Solaris' excellent multi-threading explains probably why it is
considered in the following applications:

Recommended OS for the Open Source Telephony Project
Behaving exceptionally on high-performances Java multi-threaded applications


On 5/9/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> my official comment is:
> It will work anywhere it works.  Anyone can do whatever they want but
> there is a limit to how much free support we will provide when doing
> things outside the parameters of our recommendations.
>
> The good news is that I think the newest kernels are getting over a
> dark time when timing was really bad as they tried to support more
> virtual environments.  I have seen promising results on even ubuntu
> and arch linux on bleeding edge kernels.  Centos 5.2-5.3 I can attest
> to being rock solid but a bit outdated.  As time unfolds we will
> ultimately become more confident supporting other platforms when we
> have experts on those platforms willing to maintain packaging and do
> support therein.
>
>
>
> On Mon, May 9, 2011 at 9:47 AM, Steve Underwood <steveu at coppice.org> wrote:
>> On 05/09/2011 10:38 PM, curriegrad2004 wrote:
>>> There is no "worst" distro for freeswitch. It depends on how you are
>>> going to use FreeSwitch for, really.
>> Nonsense. The distro I will take the longest to learn to install is
>> *clearly* the worst distro. :-)
>>
>> 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
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
>


-- 
Sincerely,
-E


From ira at connectmevoice.com  Mon May  9 21:07:56 2011
From: ira at connectmevoice.com (Ira Tessler)
Date: Mon, 9 May 2011 13:07:56 -0400
Subject: [Freeswitch-users] Receiving calls from an external IVR
Message-ID: <9b768341594a677779ca0c08b51b7468@mail.gmail.com>

I would like to integrate Freeswitch into my existing IVR platform. For now,
I would like to use it to register VoIP phones, handle outbound calls and
receive inbound calls from my IVR.  We are a service provider, so this would
be a multi-tenant configuration. Each one of our customers would be set up
in their own Freeswitch context. I would set up the same extensions for a
customer in our IVR and for Freeswitch. When someone dials ext 101 on our
IVR, I would like the IVR to make a call to Freeswitch and have ext 101 ring
for the customer?s context. My IVR would pass in the callerid of the inbound
caller and I would like that number to display on the VoIP phone as the
callerid and name.



Furthermore, when someone at ext 101 dials ext 102, I need Freesswitch to
make a call to the IVR.



Is this possible? Are they any examples out there? How can I get started? I
am a newbie to Freeswitch.



Thanks,



Ira Tessler
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From infos at madovsky.org  Mon May  9 21:28:21 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 9 May 2011 13:28:21 -0400
Subject: [Freeswitch-users] Unable to install
	Centoson	AMD	Motherboard	(SB850)
References: <00f401cc0e49$79497340$6bdc59c0$@accra.ca><92097A6A775D5147B1078E3F15430B924C13D3@prato.activenetwerx.local><011301cc0e51$36643b20$a32cb160$@accra.ca>
	<assp.011011f5ea.92097A6A775D5147B1078E3F15430B924C2665@prato.activenetwerx.local>
Message-ID: <2616AEA9836D48C6831CF82B012A8A87@e1705>

why not boot with liveusb from fedora ?

----- Original Message ----- 
From: "Joseph L. Casale" <jcasale at activenetwerx.com>
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, May 09, 2011 10:18 AM
Subject: Re: [Freeswitch-users] Unable to install Centoson AMD Motherboard 
(SB850)


> >Yes thought about that and set it to sata and still not recognized by 
> >Centos
>
> Boot a Fedora 14/15 live cd and do a lspci and boot the CentOS installer, 
> drop
> to a shell and do the same, getting OT for this list but it might prove 
> insightful
> to see the difference. RHEL makes a ddiskit util to build a dd image for 
> adding
> support but seriously, for the cost of a mobo, I'd just get something 
> different myself:)
>
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From avi at avimarcus.net  Mon May  9 21:34:20 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 9 May 2011 20:34:20 +0300
Subject: [Freeswitch-users] Receiving calls from an external IVR
In-Reply-To: <9b768341594a677779ca0c08b51b7468@mail.gmail.com>
References: <9b768341594a677779ca0c08b51b7468@mail.gmail.com>
Message-ID: <BANLkTimgWJGut4YU5bxna+pz2VC0C77ySA@mail.gmail.com>

Hi Ira, Avi Marcus here.

http://wiki.freeswitch.org/wiki/Mod_directory might come in useful here...
to let people search though the extensions for the person to call.
For IVR calling through, on your Dialogic IVR I'd imagine you can tell it to
call a SIP. Simply have 101 in the IVR route to freeswitch's box @ the
appropriate domain.
In FreeSWITCH, you can easily send any number you want to be bridged to
another SIP address, e.g. your current IVR.

-Avi Marcus
FreeSWITCH Consulting
0330-010-5060 (UK)

On Mon, May 9, 2011 at 8:07 PM, Ira Tessler <ira at connectmevoice.com> wrote:

> I would like to integrate Freeswitch into my existing IVR platform. For
> now, I would like to use it to register VoIP phones, handle outbound calls
> and receive inbound calls from my IVR.  We are a service provider, so this
> would be a multi-tenant configuration. Each one of our customers would be
> set up in their own Freeswitch context. I would set up the same extensions
> for a customer in our IVR and for Freeswitch. When someone dials ext 101 on
> our IVR, I would like the IVR to make a call to Freeswitch and have ext 101
> ring for the customer?s context. My IVR would pass in the callerid of the
> inbound caller and I would like that number to display on the VoIP phone as
> the callerid and name.
>
>
>
> Furthermore, when someone at ext 101 dials ext 102, I need Freesswitch to
> make a call to the IVR.
>
>
>
> Is this possible? Are they any examples out there? How can I get started? I
> am a newbie to Freeswitch.
>
>
>
> Thanks,
>
>
>
> Ira Tessler
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Mon May  9 21:37:26 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 9 May 2011 13:37:26 -0400
Subject: [Freeswitch-users] SpanDSP: nocng for rxfax?
In-Reply-To: <4DC81162.7070105@coppice.org>
References: <BANLkTinrz80SjBhVSEZ=aapeQ6u1idWk9Q@mail.gmail.com>
	<4DC81162.7070105@coppice.org>
Message-ID: <BANLkTi=yoREYuL6e=OjXboRds=u9KzfNHw@mail.gmail.com>

Interesting... Good to know, thanks!

On Mon, May 9, 2011 at 12:08 PM, Steve Underwood <steveu at coppice.org> wrote:
> rxfax does not attempt to detect CNG. Actually, no receiving FAX machine
> does. The tone was put there for humans, not machines.
>
> Steve

-- 
Kristian Kielhofner


From infos at madovsky.org  Mon May  9 21:38:25 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 9 May 2011 13:38:25 -0400
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
References: <1304777571644-6340156.post@n2.nabble.com><BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com><BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com>
	<BANLkTinAqinUyWzYNrWMJL3uzxA4Jhsf1A@mail.gmail.com>
Message-ID: <BF7681A4FE1748FBADADA8A6ADE951ED@e1705>

but.... Skype use open source for their APIs, isn't it ?


----- Original Message ----- 
From: "Anton VG" <anton.vazir at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, May 09, 2011 3:36 AM
Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?


It's a world of money. They want to control how you call, where you
call. If there will be a free software which will implement skype
calls, in large scale, using native methods... it will go out of their
control. They a still bulb, and they want to put as much air into the
bulb as possible... FS pierce holes in their bulb... ;)

2011/5/9 Shamun <shamun.toha at gmail.com>:
> Giovanni,
> Respect guru!!.
> Why FreeSWITCH, server is not a embedded server to Skype theory/logic? If 
> we
> put FreeSwitch in ATOM or ARM or 64 bit server, its not embedded? What is
> there definition of embedded? Skype really is a crazy company, 
> underestimate
> FreeSwitch and your knowledge (you did genius job, job well done).
> Reg
> Shamun
>
>
> On Sun, May 8, 2011 at 7:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com>
>> wrote:
>> > I wondered if this http://developer.skype.com/public/skypekit SkypeKit
>> > will
>> > help making mod_skypopen more useable. Has any FS developers here 
>> > gotten
>> > a
>> > chance to take a look at http://developer.skype.com/public/skypekit
>> > SkypeKit and tries to incorporate into FS for mod_skypopen to replace
>> > the
>> > need for a Skype client?
>>
>> Seems very difficult to have that kit. I (and others) have asked for
>> it, without success. Btw, I had a deep discussion with another
>> developer (outside FS community) with access to that kit, and we
>> concurred it's not usable for our purpose. It's tageted to single
>> call, for embedded devices usage (we need multiple concurrent calls
>> for FS).
>>
>> -giovanni
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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



From ira at connectmevoice.com  Mon May  9 21:47:47 2011
From: ira at connectmevoice.com (Ira Tessler)
Date: Mon, 9 May 2011 13:47:47 -0400
Subject: [Freeswitch-users] Receiving calls from an external IVR
In-Reply-To: <BANLkTimgWJGut4YU5bxna+pz2VC0C77ySA@mail.gmail.com>
References: <9b768341594a677779ca0c08b51b7468@mail.gmail.com>
	<BANLkTimgWJGut4YU5bxna+pz2VC0C77ySA@mail.gmail.com>
Message-ID: <f32a908b9dce77e4e36bb3c291e876e5@mail.gmail.com>

So can I set up ?customer a? set up in Freeswitch as
customera.connectmevoice.com domain with ext 101. On my IVR, I can have it
send an invite to 101 at customera.connectmevoice.com? ?Customer b? as
customerb.connectmevoice.com and have the IVR route to
101 at customerb.connectmevoice.com etc?..



Ira Tessler

ConnectMe

(800) 743-1208



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
*Sent:* Monday, May 09, 2011 1:34 PM
*To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] Receiving calls from an external IVR



Hi Ira, Avi Marcus here.



http://wiki.freeswitch.org/wiki/Mod_directory might come in useful here...
to let people search though the extensions for the person to call.

For IVR calling through, on your Dialogic IVR I'd imagine you can tell it to
call a SIP. Simply have 101 in the IVR route to freeswitch's box @ the
appropriate domain.

In FreeSWITCH, you can easily send any number you want to be bridged to
another SIP address, e.g. your current IVR.



-Avi Marcus

FreeSWITCH Consulting

0330-010-5060 (UK)



On Mon, May 9, 2011 at 8:07 PM, Ira Tessler <ira at connectmevoice.com> wrote:

I would like to integrate Freeswitch into my existing IVR platform. For now,
I would like to use it to register VoIP phones, handle outbound calls and
receive inbound calls from my IVR.  We are a service provider, so this would
be a multi-tenant configuration. Each one of our customers would be set up
in their own Freeswitch context. I would set up the same extensions for a
customer in our IVR and for Freeswitch. When someone dials ext 101 on our
IVR, I would like the IVR to make a call to Freeswitch and have ext 101 ring
for the customer?s context. My IVR would pass in the callerid of the inbound
caller and I would like that number to display on the VoIP phone as the
callerid and name.



Furthermore, when someone at ext 101 dials ext 102, I need Freesswitch to
make a call to the IVR.



Is this possible? Are they any examples out there? How can I get started? I
am a newbie to Freeswitch.



Thanks,



Ira Tessler




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From all.eforums at gmail.com  Mon May  9 22:30:14 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 9 May 2011 14:30:14 -0400
Subject: [Freeswitch-users] Rates Normalizer / Importer tool
Message-ID: <BANLkTikn2y_Gchaaa0Dijdms8yAHPvWF9A@mail.gmail.com>

Hi All,

Wondering if anyone has / knows of, a good rate importer tool that can be
used to standardize and normalize the rates obtained from various carriers
so they can be analysed and imported into a DB or be saved as a CSV or
something? Should I have posted this in the Freeswitch-Biz list?

Thanks so much in advance
aeg
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From avi at avimarcus.net  Mon May  9 22:38:51 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 9 May 2011 21:38:51 +0300
Subject: [Freeswitch-users] Rates Normalizer / Importer tool
In-Reply-To: <BANLkTikn2y_Gchaaa0Dijdms8yAHPvWF9A@mail.gmail.com>
References: <BANLkTikn2y_Gchaaa0Dijdms8yAHPvWF9A@mail.gmail.com>
Message-ID: <BANLkTimcpoW3pLxKifoKsPfzS7SfGqiwWg@mail.gmail.com>

I suppose you want to import for mod_lcr?
I have a regex & insert functions written in PHP for a few - xconnect,
grnvoip, VoiceNetwork.ca.. but once you add in USA it becomes kind of
complicated. Also It doesn't really know how to handle when the rate
starts... Hmm, I have an idea for how to do that.

I can perhaps post the code I have and/or help you with the rest of the
carriers.
-Avi Marcus

On Mon, May 9, 2011 at 9:30 PM, A E [Gmail] <all.eforums at gmail.com> wrote:

> Hi All,
>
> Wondering if anyone has / knows of, a good rate importer tool that can be
> used to standardize and normalize the rates obtained from various carriers
> so they can be analysed and imported into a DB or be saved as a CSV or
> something? Should I have posted this in the Freeswitch-Biz list?
>
> Thanks so much in advance
>  aeg
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From marcdecorny at gmail.com  Mon May  9 22:45:34 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 9 May 2011 19:45:34 +0100
Subject: [Freeswitch-users] Fifo - exporting/Importing variables from A to B
	Leg
Message-ID: <BANLkTim=5Le+-qL1Msm=5+-6O32HytWTdg@mail.gmail.com>

Hi all,

I'm back on working on my test freeswitch lab box.

got two questions which on which somebody can maybe shed some light.

1. Is it possible to export variables from the A-leg of the queue to the
B-let of the queue? There are the export and import commands, but not sure
if they are applicable. If I capture calls coming out of the queue with a
loopback I could then act using the variables that have been set from the
A-leg.

2. I'd like to send the call to the Freeswitch to play IVRs etc and then
bridge it without the audio.  I know it can be done on a normal bridge with
bypass_media_after_bridge, but not so sure on a fifo call.

Many thanks for any help
Marc
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From all.eforums at gmail.com  Mon May  9 23:00:15 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 9 May 2011 15:00:15 -0400
Subject: [Freeswitch-users] Rates Normalizer / Importer tool
In-Reply-To: <BANLkTimcpoW3pLxKifoKsPfzS7SfGqiwWg@mail.gmail.com>
References: <BANLkTikn2y_Gchaaa0Dijdms8yAHPvWF9A@mail.gmail.com>
	<BANLkTimcpoW3pLxKifoKsPfzS7SfGqiwWg@mail.gmail.com>
Message-ID: <BANLkTim=PiJqN0eWaFYz03KjkC8wn5TxmQ@mail.gmail.com>

On Mon, May 9, 2011 at 2:38 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I suppose you want to import for mod_lcr?
> I have a regex & insert functions written in PHP for a few - xconnect,
> grnvoip, VoiceNetwork.ca.. but once you add in USA it becomes kind of
> complicated. Also It doesn't really know how to handle when the rate
> starts... Hmm, I have an idea for how to do that.
>
> I can perhaps post the code I have and/or help you with the rest of the
> carriers.
> -Avi Marcus
>
>
> Hi,

Thanks for the response. Not looking specifically for any particular module
or LCR/routing engine. More for a visualization of the rates we obtain from
the carriers, so that we can analyze the rates to quickly vet the carriers'
rates and then further determine costs, pricing structure etc. So more for
BI sort of purposes. If after normalization it then lets us save each rate
deck into a CSV file or some other DB importable format that would be an
extra bonus. We also don't need the US so bad or at least the granular
breakdown for per area code. LATA or market etc.
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From anton.vazir at gmail.com  Mon May  9 23:05:19 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 10 May 2011 00:05:19 +0500
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
In-Reply-To: <BF7681A4FE1748FBADADA8A6ADE951ED@e1705>
References: <1304777571644-6340156.post@n2.nabble.com>
	<BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com>
	<BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com>
	<BANLkTinAqinUyWzYNrWMJL3uzxA4Jhsf1A@mail.gmail.com>
	<BF7681A4FE1748FBADADA8A6ADE951ED@e1705>
Message-ID: <BANLkTinFMmzN7e09QBFA_3_O8M1wAtJ-iw@mail.gmail.com>

A lot of commercial companies use OSS for almost everything. I build a
whole ISP with OSS
including routers/voip/fixed telephony/billing/NASes/BRASes in our
small country. Everything except l2/l3 switches runing OSS.
Just because i have no funds to buy CISCO/HUAWEI/etc, so using OSS in
the core changes nothing, and not makes company releasing everything
OSS.
Whatever Skype writes, what they want free calls for everyone, bla bla bla,
They just looking for a way to enter hardware market deeper than just
usb hand/head sets this time, nothing else.

2011/5/9 Madovsky <infos at madovsky.org>:
> but.... Skype use open source for their APIs, isn't it ?
>
>
> ----- Original Message -----
> From: "Anton VG" <anton.vazir at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, May 09, 2011 3:36 AM
> Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?
>
>
> It's a world of money. They want to control how you call, where you
> call. If there will be a free software which will implement skype
> calls, in large scale, using native methods... it will go out of their
> control. They a still bulb, and they want to put as much air into the
> bulb as possible... FS pierce holes in their bulb... ;)
>
> 2011/5/9 Shamun <shamun.toha at gmail.com>:
>> Giovanni,
>> Respect guru!!.
>> Why FreeSWITCH, server is not a embedded server to Skype theory/logic? If
>> we
>> put FreeSwitch in ATOM or ARM or 64 bit server, its not embedded? What is
>> there definition of embedded? Skype really is a crazy company,
>> underestimate
>> FreeSwitch and your knowledge (you did genius job, job well done).
>> Reg
>> Shamun
>>
>>
>> On Sun, May 8, 2011 at 7:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>>
>>> On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com>
>>> wrote:
>>> > I wondered if this http://developer.skype.com/public/skypekit SkypeKit
>>> > will
>>> > help making mod_skypopen more useable. Has any FS developers here
>>> > gotten
>>> > a
>>> > chance to take a look at http://developer.skype.com/public/skypekit
>>> > SkypeKit and tries to incorporate into FS for mod_skypopen to replace
>>> > the
>>> > need for a Skype client?
>>>
>>> Seems very difficult to have that kit. I (and others) have asked for
>>> it, without success. Btw, I had a deep discussion with another
>>> developer (outside FS community) with access to that kit, and we
>>> concurred it's not usable for our purpose. It's tageted to single
>>> call, for embedded devices usage (we need multiple concurrent calls
>>> for FS).
>>>
>>> -giovanni
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Mon May  9 23:21:31 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 9 May 2011 15:21:31 -0400
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
References: <1304777571644-6340156.post@n2.nabble.com><BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com><BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com><BANLkTinAqinUyWzYNrWMJL3uzxA4Jhsf1A@mail.gmail.com><BF7681A4FE1748FBADADA8A6ADE951ED@e1705>
	<BANLkTinFMmzN7e09QBFA_3_O8M1wAtJ-iw@mail.gmail.com>
Message-ID: <875139A76F694A19B971BE311522A243@e1705>

sure, but I meant that if their protocols are open source,
so they have to share with equity and without any restrictions but the 
licence itself...

----- Original Message ----- 
From: "Anton VG" <anton.vazir at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, May 09, 2011 3:05 PM
Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?


>A lot of commercial companies use OSS for almost everything. I build a
> whole ISP with OSS
> including routers/voip/fixed telephony/billing/NASes/BRASes in our
> small country. Everything except l2/l3 switches runing OSS.
> Just because i have no funds to buy CISCO/HUAWEI/etc, so using OSS in
> the core changes nothing, and not makes company releasing everything
> OSS.
> Whatever Skype writes, what they want free calls for everyone, bla bla 
> bla,
> They just looking for a way to enter hardware market deeper than just
> usb hand/head sets this time, nothing else.
>
> 2011/5/9 Madovsky <infos at madovsky.org>:
>> but.... Skype use open source for their APIs, isn't it ?
>>
>>
>> ----- Original Message -----
>> From: "Anton VG" <anton.vazir at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, May 09, 2011 3:36 AM
>> Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?
>>
>>
>> It's a world of money. They want to control how you call, where you
>> call. If there will be a free software which will implement skype
>> calls, in large scale, using native methods... it will go out of their
>> control. They a still bulb, and they want to put as much air into the
>> bulb as possible... FS pierce holes in their bulb... ;)
>>
>> 2011/5/9 Shamun <shamun.toha at gmail.com>:
>>> Giovanni,
>>> Respect guru!!.
>>> Why FreeSWITCH, server is not a embedded server to Skype theory/logic? 
>>> If
>>> we
>>> put FreeSwitch in ATOM or ARM or 64 bit server, its not embedded? What 
>>> is
>>> there definition of embedded? Skype really is a crazy company,
>>> underestimate
>>> FreeSwitch and your knowledge (you did genius job, job well done).
>>> Reg
>>> Shamun
>>>
>>>
>>> On Sun, May 8, 2011 at 7:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>>
>>>> On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com>
>>>> wrote:
>>>> > I wondered if this http://developer.skype.com/public/skypekit 
>>>> > SkypeKit
>>>> > will
>>>> > help making mod_skypopen more useable. Has any FS developers here
>>>> > gotten
>>>> > a
>>>> > chance to take a look at http://developer.skype.com/public/skypekit
>>>> > SkypeKit and tries to incorporate into FS for mod_skypopen to replace
>>>> > the
>>>> > need for a Skype client?
>>>>
>>>> Seems very difficult to have that kit. I (and others) have asked for
>>>> it, without success. Btw, I had a deep discussion with another
>>>> developer (outside FS community) with access to that kit, and we
>>>> concurred it's not usable for our purpose. It's tageted to single
>>>> call, for embedded devices usage (we need multiple concurrent calls
>>>> for FS).
>>>>
>>>> -giovanni
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From jpatten at co.brazos.tx.us  Mon May  9 23:29:44 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Mon, 9 May 2011 19:29:44 +0000
Subject: [Freeswitch-users] mod_lua play file into conference
Message-ID: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>

OK so I've successfully managed to get a lua script working to enhance the functionality of the "mad boss" intercom via conference example however I'm having trouble getting an audio file to stream into the conference at the beginning. Here is what I'm trying:

Dialplan:

<extension name="outbound-socket">
      <condition field="destination_number" expression="^3402$">
      <action application="lua" data="/usr/local/freeswitch/scripts/test.lua 3402"/>
    </condition>
   </extension>

Snip of test.lua:

session:execute("conference", argv[1] .. "@default")
session:execute("set", "tmp=${conference " .. argv[1] .. " play /usr/local/freeswitch/sounds/tones/norstar.wav}")

I've also tried:

session:execute("conference", argv[1] .. "@default")
api = freeswitch.API()
confplay = api:execute("conference", argv[1] .. " play /usr/local/freeswitch/sounds/tones/norstar.wav ")


In watching fs_cli it appears freeswitch never executes the entry that would set this file.

What am I doing wrong here? I've got the FreeSWITCH 1.0.6 book and I can't seem to find my answer there either.

Oh, playing a sound on participant entry won't work because it will play that sound for EVERY user that enters the conference. When 80 extensions are auto-joined to the conference at the same time it is useless for a while playing entry sounds for all those extensions.

Thanks!

Josh Patten
Brazos County Network Engineer
979.361.4676

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From adam.kelloway at newpace.ca  Mon May  9 23:33:42 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Mon, 09 May 2011 16:33:42 -0300
Subject: [Freeswitch-users] Playing remote prompts
Message-ID: <4DC84196.9050905@newpace.ca>

If I did not wish to store audio files on a FreeSWITCH host, what are my 
options for being able to retrieve a remote file and playing it?
My understanding is that the playback application can only play a local 
file, is that correct?
I did notice that there is a shell_stream module, which you could 
potentially use to retrieve a file (say, via HTTP), and provide the 
audio data as the output bash script output. Is there a way to do this 
or something similar without having to actually load another program/script?

Thanks,

Adam Kelloway


From gmaruzz at celliax.org  Mon May  9 23:45:18 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 9 May 2011 21:45:18 +0200
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
In-Reply-To: <875139A76F694A19B971BE311522A243@e1705>
References: <1304777571644-6340156.post@n2.nabble.com>
	<BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com>
	<BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com>
	<BANLkTinAqinUyWzYNrWMJL3uzxA4Jhsf1A@mail.gmail.com>
	<BF7681A4FE1748FBADADA8A6ADE951ED@e1705>
	<BANLkTinFMmzN7e09QBFA_3_O8M1wAtJ-iw@mail.gmail.com>
	<875139A76F694A19B971BE311522A243@e1705>
Message-ID: <BANLkTimsA-hT4EMMPw7eJisM66mipzeu+w@mail.gmail.com>

Their protocol is not at all open source, it' proprietary and completely close.
They piblished an API to control the skypeclient, thatps all.

-giovanni


On 5/9/11, Madovsky <infos at madovsky.org> wrote:
> sure, but I meant that if their protocols are open source,
> so they have to share with equity and without any restrictions but the
> licence itself...
>
> ----- Original Message -----
> From: "Anton VG" <anton.vazir at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, May 09, 2011 3:05 PM
> Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?
>
>
>>A lot of commercial companies use OSS for almost everything. I build a
>> whole ISP with OSS
>> including routers/voip/fixed telephony/billing/NASes/BRASes in our
>> small country. Everything except l2/l3 switches runing OSS.
>> Just because i have no funds to buy CISCO/HUAWEI/etc, so using OSS in
>> the core changes nothing, and not makes company releasing everything
>> OSS.
>> Whatever Skype writes, what they want free calls for everyone, bla bla
>> bla,
>> They just looking for a way to enter hardware market deeper than just
>> usb hand/head sets this time, nothing else.
>>
>> 2011/5/9 Madovsky <infos at madovsky.org>:
>>> but.... Skype use open source for their APIs, isn't it ?
>>>
>>>
>>> ----- Original Message -----
>>> From: "Anton VG" <anton.vazir at gmail.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, May 09, 2011 3:36 AM
>>> Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?
>>>
>>>
>>> It's a world of money. They want to control how you call, where you
>>> call. If there will be a free software which will implement skype
>>> calls, in large scale, using native methods... it will go out of their
>>> control. They a still bulb, and they want to put as much air into the
>>> bulb as possible... FS pierce holes in their bulb... ;)
>>>
>>> 2011/5/9 Shamun <shamun.toha at gmail.com>:
>>>> Giovanni,
>>>> Respect guru!!.
>>>> Why FreeSWITCH, server is not a embedded server to Skype theory/logic?
>>>> If
>>>> we
>>>> put FreeSwitch in ATOM or ARM or 64 bit server, its not embedded? What
>>>> is
>>>> there definition of embedded? Skype really is a crazy company,
>>>> underestimate
>>>> FreeSwitch and your knowledge (you did genius job, job well done).
>>>> Reg
>>>> Shamun
>>>>
>>>>
>>>> On Sun, May 8, 2011 at 7:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> wrote:
>>>>>
>>>>> On Sat, May 7, 2011 at 4:12 PM, mazilo <Nabble at slickdeals.endjunk.com>
>>>>> wrote:
>>>>> > I wondered if this http://developer.skype.com/public/skypekit
>>>>> > SkypeKit
>>>>> > will
>>>>> > help making mod_skypopen more useable. Has any FS developers here
>>>>> > gotten
>>>>> > a
>>>>> > chance to take a look at http://developer.skype.com/public/skypekit
>>>>> > SkypeKit and tries to incorporate into FS for mod_skypopen to replace
>>>>> > the
>>>>> > need for a Skype client?
>>>>>
>>>>> Seems very difficult to have that kit. I (and others) have asked for
>>>>> it, without success. Btw, I had a deep discussion with another
>>>>> developer (outside FS community) with access to that kit, and we
>>>>> concurred it's not usable for our purpose. It's tageted to single
>>>>> call, for embedded devices usage (we need multiple concurrent calls
>>>>> for FS).
>>>>>
>>>>> -giovanni
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From krice at freeswitch.org  Mon May  9 23:45:26 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 09 May 2011 14:45:26 -0500
Subject: [Freeswitch-users] Rates Normalizer / Importer tool
In-Reply-To: <BANLkTikn2y_Gchaaa0Dijdms8yAHPvWF9A@mail.gmail.com>
Message-ID: <C9EDAE86.18015%krice@freeswitch.org>

Lolol

Thanks funny... No such tool exists... Carriers, (especially US Carriers),
intentionally make their rate decks hard to import so its harder to tell the
the difference in their rates... If its not one thing its another... ie:
LATA/OCN/Tier ratedecks as in level3 to prefix/lrn ratedecks in xls format
with  1 line per cost ammount and 50 prefixes in 1 cell...

Not to mention the ratedeck you get from them this week is in format X and
the one you get next week is Format Y

(ok sure, one could argue they don?t do this on purpose, but it sure could
have fooled me)


K

On 5/9/11 1:30 PM, "A E [Gmail]" <all.eforums at gmail.com> wrote:

> Hi All,
> 
> Wondering if anyone has / knows of, a good rate importer tool that can be used
> to standardize and normalize the rates obtained from various carriers so they
> can be analysed and imported into a DB or be saved as a CSV or something?
> Should I have posted this in the Freeswitch-Biz list?
> 
> Thanks so much in advance
> aeg
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Mon May  9 23:55:30 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 9 May 2011 15:55:30 -0400
Subject: [Freeswitch-users] SkypeKit for mo_skyopen?
References: <1304777571644-6340156.post@n2.nabble.com><BANLkTimPzBVmjBUYX3eTZK7N+Gy9hprBDA@mail.gmail.com><BANLkTimvwE-U9wP=YqJ4Y3osgD3Uxnf8=g@mail.gmail.com><BANLkTinAqinUyWzYNrWMJL3uzxA4Jhsf1A@mail.gmail.com><BF7681A4FE1748FBADADA8A6ADE951ED@e1705><BANLkTinFMmzN7e09QBFA_3_O8M1wAtJ-iw@mail.gmail.com><875139A76F694A19B971BE311522A243@e1705>
	<BANLkTimsA-hT4EMMPw7eJisM66mipzeu+w@mail.gmail.com>
Message-ID: <7F733AED493A4F14AB5ADE3917C2EB93@e1705>

Ah ok , maybe I mix with googletalk
I thought they use xmpp protocol

----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at celliax.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, May 09, 2011 3:45 PM
Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?


> Their protocol is not at all open source, it' proprietary and completely 
> close.
> They piblished an API to control the skypeclient, thatps all.
>
> -giovanni
>
>
> On 5/9/11, Madovsky <infos at madovsky.org> wrote:
>> sure, but I meant that if their protocols are open source,
>> so they have to share with equity and without any restrictions but the
>> licence itself...
>>
>> ----- Original Message -----
>> From: "Anton VG" <anton.vazir at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, May 09, 2011 3:05 PM
>> Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?
>>
>>
>>>A lot of commercial companies use OSS for almost everything. I build a
>>> whole ISP with OSS
>>> including routers/voip/fixed telephony/billing/NASes/BRASes in our
>>> small country. Everything except l2/l3 switches runing OSS.
>>> Just because i have no funds to buy CISCO/HUAWEI/etc, so using OSS in
>>> the core changes nothing, and not makes company releasing everything
>>> OSS.
>>> Whatever Skype writes, what they want free calls for everyone, bla bla
>>> bla,
>>> They just looking for a way to enter hardware market deeper than just
>>> usb hand/head sets this time, nothing else.
>>>
>>> 2011/5/9 Madovsky <infos at madovsky.org>:
>>>> but.... Skype use open source for their APIs, isn't it ?
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Anton VG" <anton.vazir at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, May 09, 2011 3:36 AM
>>>> Subject: Re: [Freeswitch-users] SkypeKit for mo_skyopen?
>>>>
>>>>
>>>> It's a world of money. They want to control how you call, where you
>>>> call. If there will be a free software which will implement skype
>>>> calls, in large scale, using native methods... it will go out of their
>>>> control. They a still bulb, and they want to put as much air into the
>>>> bulb as possible... FS pierce holes in their bulb... ;)
>>>>
>>>> 2011/5/9 Shamun <shamun.toha at gmail.com>:
>>>>> Giovanni,
>>>>> Respect guru!!.
>>>>> Why FreeSWITCH, server is not a embedded server to Skype theory/logic?
>>>>> If
>>>>> we
>>>>> put FreeSwitch in ATOM or ARM or 64 bit server, its not embedded? What
>>>>> is
>>>>> there definition of embedded? Skype really is a crazy company,
>>>>> underestimate
>>>>> FreeSwitch and your knowledge (you did genius job, job well done).
>>>>> Reg
>>>>> Shamun
>>>>>
>>>>>
>>>>> On Sun, May 8, 2011 at 7:08 PM, Giovanni Maruzzelli 
>>>>> <gmaruzz at gmail.com>
>>>>> wrote:
>>>>>>
>>>>>> On Sat, May 7, 2011 at 4:12 PM, mazilo 
>>>>>> <Nabble at slickdeals.endjunk.com>
>>>>>> wrote:
>>>>>> > I wondered if this http://developer.skype.com/public/skypekit
>>>>>> > SkypeKit
>>>>>> > will
>>>>>> > help making mod_skypopen more useable. Has any FS developers here
>>>>>> > gotten
>>>>>> > a
>>>>>> > chance to take a look at http://developer.skype.com/public/skypekit
>>>>>> > SkypeKit and tries to incorporate into FS for mod_skypopen to 
>>>>>> > replace
>>>>>> > the
>>>>>> > need for a Skype client?
>>>>>>
>>>>>> Seems very difficult to have that kit. I (and others) have asked for
>>>>>> it, without success. Btw, I had a deep discussion with another
>>>>>> developer (outside FS community) with access to that kit, and we
>>>>>> concurred it's not usable for our purpose. It's tageted to single
>>>>>> call, for embedded devices usage (we need multiple concurrent calls
>>>>>> for FS).
>>>>>>
>>>>>> -giovanni
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> -- 
> 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 



From avi at avimarcus.net  Mon May  9 23:57:49 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 9 May 2011 22:57:49 +0300
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <4DC84196.9050905@newpace.ca>
References: <4DC84196.9050905@newpace.ca>
Message-ID: <BANLkTimtTJ5QHofCaC0_ExXDexefjj7OqA@mail.gmail.com>

At least one potion is to use mod_shout to stream them from a server.
Do note that this adds another point of failure to your setup...

-Avi

On Mon, May 9, 2011 at 10:33 PM, Adam Kelloway <adam.kelloway at newpace.ca>wrote:

> If I did not wish to store audio files on a FreeSWITCH host, what are my
> options for being able to retrieve a remote file and playing it?
> My understanding is that the playback application can only play a local
> file, is that correct?
> I did notice that there is a shell_stream module, which you could
> potentially use to retrieve a file (say, via HTTP), and provide the
> audio data as the output bash script output. Is there a way to do this
> or something similar without having to actually load another
> program/script?
>
> Thanks,
>
> Adam Kelloway
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From all.eforums at gmail.com  Tue May 10 00:02:55 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 9 May 2011 16:02:55 -0400
Subject: [Freeswitch-users] Rates Normalizer / Importer tool
In-Reply-To: <C9EDAE86.18015%krice@freeswitch.org>
References: <BANLkTikn2y_Gchaaa0Dijdms8yAHPvWF9A@mail.gmail.com>
	<C9EDAE86.18015%krice@freeswitch.org>
Message-ID: <BANLkTi=nZSiQ6Bw4ssJ+mMheiLHhDH=rOg@mail.gmail.com>

On Mon, May 9, 2011 at 3:45 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Lolol
>
> Thanks funny... No such tool exists... Carriers, (especially US Carriers),
> intentionally make their rate decks hard to import so its harder to tell the
> the difference in their rates... If its not one thing its another... ie:
> LATA/OCN/Tier ratedecks as in level3 to prefix/lrn ratedecks in xls format
> with  1 line per cost ammount and 50 prefixes in 1 cell...
>
> Not to mention the ratedeck you get from them this week is in format X and
> the one you get next week is Format Y
>
> (ok sure, one could argue they don?t do this on purpose, but it sure could
> have fooled me)
>
>
> Hi Ken,
Everything you said is true although I have yet to see a case where the same
provider might change the format from week to week. The only time I see
changes from the same provider is the format they give the base/original A-Z
in, and the formats they then deliver rate updates in are usually different.
But yes, the formats vary greatly from provider to provider.

Also, I think/(sort of) know that such tools exist but they exist within
each company/VSP's to work with their own systems and none of them are
particularly keen to share those tools since they take significant time and
resources to develop (but then so does Freeswitch, but that's free ;)). So I
was hoping if anyone on the list had developed this and was willing to share
or even sell (at a reasonable price), we will still consider it. Also, the
idea was to not expect that they would already have solved or created
templates for each carrier out there but simply to be able to create our own
templates for the carriers we choose to then feed to the tool for
normalization.
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From admin at blindi.net  Tue May 10 00:58:37 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 9 May 2011 22:58:37 +0200 (CEST)
Subject: [Freeswitch-users] Problem originate conference hangup
In-Reply-To: <BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105092235440.7827@debian>

Hi michael,
I use a cellphone. And i have a voip-flaterate (unlimited rate) to call 
all cellphones and landlines in germany.
I generate a callback from  the phone. Fs call me back.
Ok i go in my conference with adminfunctions.
  And i like to callout to add more people to the 
conference.
When i setup a callback via the originatecommand, fs hangup the line after 
a party is added to the conference.
I call the conference directlly from the sipadapter fs go back after the 
first party to callout, this is correctlly.

Can i disable the Hangup after the originate command?
I don.t find a variable.
This is my solution.
Thankx.



---------------
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 brad at tritelcomm.com  Tue May 10 01:02:42 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Mon, 9 May 2011 14:02:42 -0700
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <BANLkTimtTJ5QHofCaC0_ExXDexefjj7OqA@mail.gmail.com>
References: <4DC84196.9050905@newpace.ca>
	<BANLkTimtTJ5QHofCaC0_ExXDexefjj7OqA@mail.gmail.com>
Message-ID: <BANLkTi=vPrN84Yjy_BOOooGhc4EB_PMvWQ@mail.gmail.com>

You could setup an internal http server with a directory called 'audio' (
http://server.ip/audio/), then use mod_shout as Avi suggested like this:
<action application="shout" data="http://server.ip/audio/file.mp3"/>


On Mon, May 9, 2011 at 12:57 PM, Avi Marcus <avi at avimarcus.net> wrote:

> At least one potion is to use mod_shout to stream them from a server.
> Do note that this adds another point of failure to your setup...
>
> -Avi
>
>
> On Mon, May 9, 2011 at 10:33 PM, Adam Kelloway <adam.kelloway at newpace.ca>wrote:
>
>> If I did not wish to store audio files on a FreeSWITCH host, what are my
>> options for being able to retrieve a remote file and playing it?
>> My understanding is that the playback application can only play a local
>> file, is that correct?
>> I did notice that there is a shell_stream module, which you could
>> potentially use to retrieve a file (say, via HTTP), and provide the
>> audio data as the output bash script output. Is there a way to do this
>> or something similar without having to actually load another
>> program/script?
>>
>> Thanks,
>>
>> Adam Kelloway
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 10 02:41:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 May 2011 15:41:48 -0700
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
Message-ID: <BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>

On Mon, May 9, 2011 at 12:29 PM, Josh M. Patten <jpatten at co.brazos.tx.us>wrote:

>  OK so I?ve successfully managed to get a lua script working to enhance
> the functionality of the ?mad boss? intercom via conference example however
> I?m having trouble getting an audio file to stream into the conference at
> the beginning.
>

An audio file to stream at the beginning of what, exactly? When the first
person enters the conference? Or when every participant enters? I'm just
trying to understand what problem you're solving. Please describe it in
plain language: when user enters he hears xyz, when next user enters he and
the conference hear abc, etc.

-MC


> Here is what I?m trying:
>
>
>
> Dialplan:
>
>
>
> <extension name="outbound-socket">
>
>       <condition field="destination_number" expression="^3402$">
>
>       <action application="lua"
> data="/usr/local/freeswitch/scripts/test.lua 3402"/>
>
>     </condition>
>
>    </extension>
>
>
>
> Snip of test.lua:
>
>
>
> session:execute("conference", argv[1] .. "@default")
>
> session:execute("set", "tmp=${conference " .. argv[1] .. " play
> /usr/local/freeswitch/sounds/tones/norstar.wav}")
>
>
>
> I?ve also tried:
>
>
>
> session:execute("conference", argv[1] .. "@default")
>
> api = freeswitch.API()
>
> confplay = api:execute("conference", argv[1] .. " play
> /usr/local/freeswitch/sounds/tones/norstar.wav ")
>
>
>
>
>
> In watching fs_cli it appears freeswitch never executes the entry that
> would set this file.
>
>
>
> What am I doing wrong here? I?ve got the FreeSWITCH 1.0.6 book and I can?t
> seem to find my answer there either.
>
>
>
> Oh, playing a sound on participant entry won?t work because it will play
> that sound for EVERY user that enters the conference. When 80 extensions are
> auto-joined to the conference at the same time it is useless for a while
> playing entry sounds for all those extensions.
>
>
>
> Thanks!
>
>
>
> Josh Patten
>
> Brazos County Network Engineer
>
> 979.361.4676
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 10 02:54:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 May 2011 15:54:50 -0700
Subject: [Freeswitch-users] Problem originate conference hangup
In-Reply-To: <Pine.LNX.4.64.1105092235440.7827@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
Message-ID: <BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>

I sort of understand what you're doing. I think you might have an overly
complicated setup. Let me tell you what I think you're trying to do, and
then we'll talk about how to make it happen:

#1 - You call into a conference. The conference moderator has caller
controls that general members do not. (BTW, the way you have it set now,
*everyone* will have the option to press 2. You may not mix conference
profiles in the same conference. In other words, every member of the
conference must have the same caller control group. If you want to have the
moderator have extra commands then I strongly recommend "bind_digit_action")

#2 - The moderator (you) presses 2 and gets dial-tone. You dial someone
else's mobile phone number and then you are returned to the conference room.

#3 - The number you entered in step #2 is called and added to your
conference.

Is that correct? I don't want to spend any more time on this unless I know
the exact steps that you want to have happen.

Thanks,
MC

On Mon, May 9, 2011 at 1:58 PM, Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi michael,
> I use a cellphone. And i have a voip-flaterate (unlimited rate) to call all
> cellphones and landlines in germany.
> I generate a callback from  the phone. Fs call me back.
> Ok i go in my conference with adminfunctions.
>  And i like to callout to add more people to the conference.
> When i setup a callback via the originatecommand, fs hangup the line after
> a party is added to the conference.
> I call the conference directlly from the sipadapter fs go back after the
> first party to callout, this is correctlly.
>
> Can i disable the Hangup after the originate command?
> I don.t find a variable.
> This is my solution.
> Thankx.
>
>
>
>
> ---------------
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jpatten at co.brazos.tx.us  Tue May 10 03:40:23 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Mon, 9 May 2011 23:40:23 +0000
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>,
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
Message-ID: <8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>

I would like to stream the file as soon as the conference is started, no matter who is in there.

What I'm asking isn't really time dependent though. I'm trying to find out why I can't stream an audio file into a conference with lua.

Basically I'm looking to implement a paging system based on the the "Mad Boss" example located here: http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom but implemented in lua using a postgres database to store what extensions to page for what "page group" (I promise to post my code, honest). In most overhead paging environments there is either a short tone or sound before the page is broadcast. I'm trying to replicate that by streaming a short wav file once the conference is established.

Let me know if you need any more info.

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael Collins [msc at freeswitch.org]
Sent: Monday, May 09, 2011 5:41 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference



On Mon, May 9, 2011 at 12:29 PM, Josh M. Patten <jpatten at co.brazos.tx.us<mailto:jpatten at co.brazos.tx.us>> wrote:
OK so I?ve successfully managed to get a lua script working to enhance the functionality of the ?mad boss? intercom via conference example however I?m having trouble getting an audio file to stream into the conference at the beginning.

An audio file to stream at the beginning of what, exactly? When the first person enters the conference? Or when every participant enters? I'm just trying to understand what problem you're solving. Please describe it in plain language: when user enters he hears xyz, when next user enters he and the conference hear abc, etc.

-MC

Here is what I?m trying:

Dialplan:

<extension name="outbound-socket">
      <condition field="destination_number" expression="^3402$">
      <action application="lua" data="/usr/local/freeswitch/scripts/test.lua 3402"/>
    </condition>
   </extension>

Snip of test.lua:

session:execute("conference", argv[1] .. "@default")
session:execute("set", "tmp=${conference " .. argv[1] .. " play /usr/local/freeswitch/sounds/tones/norstar.wav}")

I?ve also tried:

session:execute("conference", argv[1] .. "@default")
api = freeswitch.API()
confplay = api:execute("conference", argv[1] .. " play /usr/local/freeswitch/sounds/tones/norstar.wav ")


In watching fs_cli it appears freeswitch never executes the entry that would set this file.

What am I doing wrong here? I?ve got the FreeSWITCH 1.0.6 book and I can?t seem to find my answer there either.

Oh, playing a sound on participant entry won?t work because it will play that sound for EVERY user that enters the conference. When 80 extensions are auto-joined to the conference at the same time it is useless for a while playing entry sounds for all those extensions.

Thanks!

Josh Patten
Brazos County Network Engineer
979.361.4676


_______________________________________________
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 riedinger at sns.eu  Tue May 10 04:41:02 2011
From: riedinger at sns.eu (Jan Riedinger)
Date: Tue, 10 May 2011 02:41:02 +0200
Subject: [Freeswitch-users] Moving on from CentOS
In-Reply-To: <4DC784BB.4080907@seventhsignal.de>
References: <BANLkTimeszPHcy5OuaugT92Xwz8iUh3y9Q@mail.gmail.com>	<4DC72997.4090900@sns.eu>	<6917277E-9448-46C8-AFF5-CFE62D590BE0@gmail.com>	<BANLkTinf1G53b51q5fpXUkjfEtdvmKKXfw@mail.gmail.com>
	<4DC784BB.4080907@seventhsignal.de>
Message-ID: <4DC8899E.5040304@sns.eu>

Sorry, I didn't want to sent my e-mail to the list, but forward it to a 
friend, who is a little bit distressed about the CentOS release policy 
as well.

BR
    Jan

Am 09.05.2011 08:07, schrieb Michal Bielicki:
> I guess Jan just wanted to explain. Sometimes its better to clear up
> things for people in their local language, don't you think ?
>
> Am 09.05.2011 06:01, schrieb curriegrad2004:
>> It would be beneficial to the majority of the users if we continued
>> this discussion in English
>>
>> On Sun, May 8, 2011 at 5:09 PM,<rhuddleston at gmail.com>  wrote:
>>> Auf English bitte? Mein deutsch ist sehr sclecht und ich kann nicht verstehen
>>>
>>> On May 8, 2011, at 7:39 PM, Jan Riedinger<riedinger at sns.eu>  wrote:
>>>
>>>> Hi Dirk,
>>>>
>>>> Freeswitch wird weiterhin auf CentOS setzen, Max Clark ist lediglich ein
>>>> Anwender. Aber offenbar, st?ren sich noch andere an der Release-Politik.
>>>>
>>>> Viele Gr??e
>>>>      Jan
>>>>
>>>> Am 06.05.2011 16:25, schrieb Max Clark:
>>>>> Hello all,
>>>>>
>>>>> Recent developments (or absolute lack of) within the CentOS project
>>>>> and its perceived long term viability has forced an internal
>>>>> discussion to select a successor distribution. The most likely
>>>>> candidates at this point are Ubuntu and its LTS releases for servers,
>>>>> and Scientific Linux with the new 6.x releases. The pro/con lists for
>>>>> each are growing and the issue is complicated.
>>>>>
>>>>> With CentOS 5.x being the reference distro for FreeSWITCH development
>>>>> I'm curious if this conversation has started among the FreeSWITCH
>>>>> developers, and if it has, what is the project leaning to for a
>>>>> successor distribution?
>>>>>
>>>>> Thanks,
>>>>> Max
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> --
>>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>>                                          E-Mail:  riedinger at sns.eu
>>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>>
>>>> AG Charlottenburg - HRB 71973
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From fieldpeak at gmail.com  Tue May 10 08:49:37 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 10 May 2011 12:49:37 +0800
Subject: [Freeswitch-users] How to manipulate destination number before
 routing to PSTN GW
In-Reply-To: <BANLkTi=SULGgcvytahBcCcOAF5Uzn9Ramg@mail.gmail.com>
References: <BANLkTin2fD=iFFQB8DkdaedQN044CW3tYA@mail.gmail.com>
	<BANLkTinQs--vP16i6PaVBFP35ywa_37gPw@mail.gmail.com>
	<BANLkTinRbn-TUcDbGDGRjCVN25tjcy5ygg@mail.gmail.com>
	<C500EE9F-E9D7-46EC-B24F-A2E2062CD533@ipeva.fr>
	<BANLkTikU6yZg2vEAC5TW3JiHfbm7H4e84A@mail.gmail.com>
	<BANLkTinjHUyRbZO4PP=D=Ds6d9rNNEqLLA@mail.gmail.com>
	<BANLkTi=SULGgcvytahBcCcOAF5Uzn9Ramg@mail.gmail.com>
Message-ID: <BANLkTikwQxicKoDAcSfhmszA_BmKnF+XwA@mail.gmail.com>

Hi Steve & Avi,

It works now with Steve's elabrated info (tested with transfer...will test
new var as well),
Thank you for your great support so much.

Gurus - you deserved!

Regards,
Charles

2011/5/9 Steven Ayre <steveayre at gmail.com>

> fieldpark,
>
> It would be far simpler to just ignore it in the regex of the 2nd extension
> - then you wouldn't need the 1st extension at all!
>
>
> <extension name="9_to_GW">
>    <condition field="dest_num" expression="^\+?(9\d+)$">
>
>        <action application="bridge" data="sofia/internal/$
> 1 at 192.168.200.101"/>
>    </condition>
> </extension>
>
> The ? after the \+ makes it optional. This extension will match both 9123
> and +9123, and bridge both as 123 at 192.168.200.101
>
> The only reason I can think of for stripping it first would be if you have
> a very long dialplan, where it might then be simpler than optionally
> checking for a + in every regex in the rest of the dialplan. For a simple
> dialplan the above extension would be simpler.
>
>
> Avi's suggestion should work fine and would look like this:
>
> <extension name="strip_plus" continue="true">
>
>    <condition field="destination_number" expression="^\+(\d+)$">
>        <action application="set" data="dest_num=$1"/>
>        <anti-action application="set"
> data="dest_num=${destination_number}"/>
>
>    </condition>
> </extension>
> <extension name="9_to_GW">
>    <condition field="${dest_num}" expression="^(9\d+)$">
>
>        <action application="bridge" data="sofia/internal/$
> 1 at 192.168.200.101"/>
>    </condition>
> </extension>
>
>
> I'd suggest a transfer would be simpler than doing this though because it
> this approach changes the field you're accessing in every extension. A
> transfer would rewrite the destination_number so you wouldn't need to check
> a channel variable each time any more. Yes it does reexecute the dialplan,
> but if it's placed as the very first extension that should have a very small
> impact on performance (the time taken to run a single regex that fails on
> the first character is pretty small, and checking a channel variable on
> every extension is slower than checking the destination_number field so it
> might actually be faster than the above*).
>
> <extension name="strip_plus">
>
>    <condition field="destination_number" expression="^\+(\d+)$">
>        <action application="transfer" data="$1"/>
>
>    </condition>
> </extension>
> <extension name="9_to_GW">
>    <condition field="destination_number" expression="^(9\d+)$">
>        <action application="bridge" data="sofia/internal/$
> 1 at 192.168.200.101"/>
>    </condition>
> </extension>
>
>
>
> -Steve
>
>
> * Looking for a channel variable means looping through the list of all
> channel variables while destination_number is hardcoded to look it up in the
> caller profile structure so there's no loop involved.
>
>
>
> On 9 May 2011 11:50, Avi Marcus <avi at avimarcus.net> wrote:
>
>> As mentioned, that's because you can't set destination_number -- it's a
>> reserved variable.
>> Either a) transfer to the new number, or b) create a new variable for
>> processing the number.
>> e.g. with an anti-action that sets the new variable to be the unchanged
>> number.
>> -Avi
>>
>>
>> On Sun, May 8, 2011 at 4:56 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> i tested the inline usage, unluckly, it failed. it looks the inline still
>>> can not effect 9 to GW rule...
>>>
>>> *dial plan is:*
>>>
>>> <extension name="remove_plus_of_dst_num" continue="true">
>>>    <condition field="destination_number" expression="^\+(\d+)$">
>>>        <action inline="true" application="set"
>>> data="destination_number=$1"/>
>>>
>>>    </condition>
>>> </extension>
>>>
>>> ...
>>> <extension name="9_to_GW" >
>>>    <condition field="destination_number" expression="^(9\d+)$">
>>>        <action application="bridge" data="sofia/internal/$
>>> 1 at 192.168.200.101"/>
>>>    </condition>
>>> </extension>
>>>
>>> *the log is:*
>>>
>>> freeswitch at mypc>
>>> 2011-05-08 09:35:40.067099 [ERR] switch_xml.c:1311 Couldnt open
>>> C:\FreeSWITCH\conf\
>>> autoload_configs\..\sip_profiles\external/*.xml (No such file or
>>> directory)
>>> 2011-05-08 09:35:40.543126 [ERR] switch_xml.c:1311 Couldnt open
>>> C:\FreeSWITCH\conf\
>>> dialplan\public/*.xml (No such file or directory)
>>>
>>> +OK [Success]
>>>
>>> 2011-05-08 09:35:41.115159 [INFO] mod_pocketsphinx.c:482 PocketSphinx
>>> Reloaded
>>> 2011-05-08 09:35:41.117159 [INFO] switch_time.c:999 Timezone reloaded 530
>>> defini
>>> tions
>>> 2011-05-08 09:35:44.882374 [DEBUG] sofia.c:6488 IP 192.168.200.201
>>> Approved by a
>>>
>>> cl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>". Access Granted.
>>>  2011-05-08 09:35:44.883374 [NOTICE] switch_channel.c:812 New Channel
>>> sofia/inter
>>> nal/4001 at 192.168.200.100 [efeb55d7-3712-4a30-9eba-786ad83b2e90]
>>> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4760 Channel
>>> sofia/internal/4001 at 192.
>>>
>>> 168.200.100 entering state [received][100]
>>> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4771 Remote SDP:
>>> v=0
>>> o=AudiocodesGW 2088509637 2088509507 IN IP4 192.168.200.201
>>>
>>> s=Phone-Call
>>> c=IN IP4 192.168.200.201
>>> t=0 0
>>> m=audio 6020 RTP/AVP 8 0 96
>>>
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:96 telephone-event/8000
>>> a=fmtp:96 0-15
>>> a=ptime:20
>>>
>>> 2011-05-08 09:35:44.884374 [DEBUG] sofia.c:4908
>>> (sofia/internal/4001 at 192.168.200
>>>
>>> .100) State Change CS_NEW -> CS_INIT
>>> 2011-05-08 09:35:44.884374 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/
>>>
>>> internal/4001 at 192.168.200.100 [BREAK]
>>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) Running State Change CS_INIT
>>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State INIT
>>> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:84
>>> sofia/internal/4001 at 192.168.20
>>> 0.100 SOFIA INIT
>>> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:124
>>> (sofia/internal/4001 at 192.168.
>>>
>>> 200.100) State Change CS_INIT -> CS_ROUTING
>>>  2011-05-08 09:35:44.885374 [DEBUG] switch_core_session.c:1116 Send
>>> signal sofia/
>>>
>>> internal/4001 at 192.168.200.100 [BREAK]
>>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:361
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State INIT going to sleep
>>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) Running State Change CS_ROUTING
>>>  2011-05-08 09:35:44.885374 [DEBUG] switch_channel.c:1668
>>> (sofia/internal/4001 at 19
>>>
>>> 2.168.200.100) Callstate Change DOWN -> RINGING
>>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:364
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State ROUTING
>>> 2011-05-08 09:35:44.885374 [DEBUG] mod_sofia.c:147
>>> sofia/internal/4001 at 192.168.2
>>> 00.100 SOFIA ROUTING
>>> 2011-05-08 09:35:44.885374 [DEBUG] switch_core_state_machine.c:77
>>> sofia/internal
>>>
>>> /4001 at 192.168.200.100 Standard ROUTING
>>> 2011-05-08 09:35:44.885374 [INFO] mod_dialplan_xml.c:331 Processing 4001
>>> <4001>-
>>> >+9123 in context default
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->unloop]
>>> continue
>>> =false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS) [unloop]
>>> ${unroll_loo
>>>
>>> ps}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [unloop]
>>> ${sip_looped
>>> _call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>>> [default->remove_plus_of_d
>>> st_num] continue=true
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (PASS)
>>> [remove_plus_of_dst_n
>>> um] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Action
>>> set(destination_number=9123
>>>  ) INLINE
>>>
>>> EXECUTE sofia/internal/4001 at 192.168.200.100 set(destination_number=9123)
>>> 2011-05-08 09:35:44.888374 [DEBUG] mod_dptools.c:1060
>>> sofia/internal/4001 at 192.16
>>>
>>> 8.200.100 SET [destination_number]=[9123]
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>>> [default->remove_plus_of_s
>>> rc_num] continue=true
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL)
>>> [remove_plus_of_src_n
>>>
>>> um] caller_id_number(4001) =~ /^\+(\d+)$/ break=on-false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>>> [default->7_8_to_Lync] con
>>> tinue=false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [7_8_to_Lync]
>>> destina
>>>
>>> tion_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->9_to_GW]
>>> continu
>>> e=false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [9_to_GW]
>>> destination
>>>
>>> _number(+9123) =~ /^(9\d+)$/ break=on-false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>>> [default->1_to_IPP] contin
>>> ue=false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [1_to_IPP]
>>> destinatio
>>> n_number(+9123) =~ /^\+{0,1}(1\d{3})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing [default->DISA]
>>> continue=f
>>> alse
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [DISA]
>>> destination_nu
>>>
>>> mber(+9123) =~ /^\*(3472)$/ break=on-false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 parsing
>>> [default->Recordings] cont
>>> inue=false
>>> Dialplan: sofia/internal/4001 at 192.168.200.100 Regex (FAIL) [Recordings]
>>> destinat
>>>
>>> ion_number(+9123) =~ /^\*(732673)$/ break=on-false
>>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:119
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/
>>>
>>> internal/4001 at 192.168.200.100 [BREAK]
>>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:364
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State ROUTING going to sleep
>>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) Running State Change CS_EXECUTE
>>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:371
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State EXECUTE
>>> 2011-05-08 09:35:44.889374 [DEBUG] mod_sofia.c:240
>>> sofia/internal/4001 at 192.168.2
>>> 00.100 SOFIA EXECUTE
>>> 2011-05-08 09:35:44.889374 [DEBUG] switch_core_state_machine.c:157
>>> sofia/interna
>>>
>>> l/4001 at 192.168.200.100 Standard EXECUTE
>>>  2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:189
>>> sofia/intern
>>>
>>> al/4001 at 192.168.200.100 has executed the last dialplan instruction,
>>> hanging up.
>>> 2011-05-08 09:35:44.889374 [DEBUG] switch_channel.c:2563
>>> (sofia/internal/4001 at 19
>>>
>>> 2.168.200.100) Callstate Change RINGING -> HANGUP
>>> 2011-05-08 09:35:44.889374 [NOTICE] switch_core_state_machine.c:191
>>> Hangup sofia
>>>
>>> /internal/4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>>  2011-05-08 09:35:44.890375 [DEBUG] switch_channel.c:2579 Send signal
>>> sofia/inter
>>>
>>> nal/4001 at 192.168.200.100 [KILL]
>>> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/
>>>
>>> internal/4001 at 192.168.200.100 [BREAK]
>>> 2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:371
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State EXECUTE going to sleep
>>>  2011-05-08 09:35:44.890375 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) Running State Change CS_HANGUP
>>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State HANGUP
>>> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:457 Channel
>>> sofia/internal/4001 at 1
>>>
>>> 92.168.200.100 hanging up, cause: NORMAL_CLEARING
>>> 2011-05-08 09:35:44.891375 [DEBUG] mod_sofia.c:519 Responding to INVITE
>>> with: 48
>>> 0
>>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:46
>>> sofia/internal
>>>
>>> /4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:565
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State HANGUP going to sleep
>>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:356
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/
>>>
>>> internal/4001 at 192.168.200.100 [BREAK]
>>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) Running State Change CS_REPORTING
>>> 2011-05-08 09:35:44.891375 [DEBUG] switch_core_state_machine.c:625
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State REPORTING
>>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:53
>>> sofia/internal
>>>
>>> /4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:625
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State REPORTING going to sleep
>>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_state_machine.c:350
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/
>>>
>>> internal/4001 at 192.168.200.100 [BREAK]
>>> 2011-05-08 09:35:45.153390 [DEBUG] switch_core_session.c:1288 Session 4
>>> (sofia/i
>>>
>>> nternal/4001 at 192.168.200.100) Locked, Waiting on external entities
>>> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1306 Session 4
>>> (sofia/
>>>
>>> internal/4001 at 192.168.200.100) Ended
>>> 2011-05-08 09:35:45.153390 [NOTICE] switch_core_session.c:1308 Close
>>> Channel sof
>>>
>>> ia/internal/4001 at 192.168.200.100 [CS_DESTROY]
>>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:454
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:457
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) Running State Change CS_DESTROY
>>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State DESTROY
>>> 2011-05-08 09:35:45.154390 [DEBUG] mod_sofia.c:362
>>> sofia/internal/4001 at 192.168.2
>>> 00.100 SOFIA DESTROY
>>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:60
>>> sofia/internal
>>>
>>> /4001 at 192.168.200.100 Standard DESTROY
>>> 2011-05-08 09:35:45.154390 [DEBUG] switch_core_state_machine.c:467
>>> (sofia/intern
>>>
>>> al/4001 at 192.168.200.100) State DESTROY going to sleep
>>>
>>>
>>> 2011/5/8 David Ponzone <david.ponzone at ipeva.fr>
>>>
>>>> Avi,
>>>>
>>>> are you sure about that ?
>>>>
>>>> I have LUA scripts which rewrite the destination_number with SetVariable
>>>> and that works fine.
>>>> Of course, to do that in the XML dialplan, as said before, inline should
>>>> be used to set the value.
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 07/05/2011 ? 19:52, Avi Marcus a ?crit :
>>>>
>>>> Indeed, inline is a missing ingredient. As well though is the previously
>>>> mentioned - you can't (directly) overwrite the destination number. You can
>>>> either use a new variable throughout, or transfer to set the new destination
>>>> number. Note that transferring causes the dialplan to be re-run from the
>>>> start..
>>>>
>>>> -Avi
>>>>
>>>> On Sat, May 7, 2011 at 7:13 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>>>>
>>>>> There is certainly more than one way to skin this cat...
>>>>>
>>>>> However, to keep with your current method you should read up on
>>>>> dialplan hunting vs. execution and "inline" execution:
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>>>>
>>>>> In short you need to add inline="true" to your first extension.
>>>>>
>>>>> On Sat, May 7, 2011 at 11:04 AM, fieldpeak <fieldpeak at gmail.com>
>>>>> wrote:
>>>>> > Gurus,
>>>>> >
>>>>> > i met an issue for dial plan, it sounds easy but puzzled me a few
>>>>> days not
>>>>> > fix it yet...i belive gurus here could help me...
>>>>> >
>>>>> > i want to remove the + of destination number before routing to PSTN
>>>>> GW, e.g.
>>>>> > when i dial +9123, i would like FS remove +, and then route to the
>>>>> PSTN GW,
>>>>> > below is the dial plan and log,
>>>>> >
>>>>> > See the log, FS did remove '+' of '+9123', but then when it checks
>>>>> the rule
>>>>> > ?9_to_GW?, it still checks '+9123' (I expect it is '9123'), and then
>>>>> failed.
>>>>> >
>>>>> > Thanks.
>>>>> >
>>>>> >
>>>>> >
>>>>> > Dial plan:
>>>>> >
>>>>> > <extension name="remove_plus_of_dst_num" continue="true">
>>>>> >
>>>>> >    <condition field="destination_number" expression="^\+(\d+)$">
>>>>> >
>>>>> >        <action application="set"
>>>>> data="effective_destination_number=$1"/>
>>>>> >
>>>>> >    </condition>
>>>>> >
>>>>> > </extension>
>>>>> >
>>>>> > ?...
>>>>> >
>>>>> > <extension name="9_to_GW" >
>>>>> >
>>>>> >    <condition field="destination_number" expression="^(9\d+)$">
>>>>> >
>>>>> >        <action application="bridge"
>>>>> > data="sofia/internal/$1 at 192.168.200.101"/>
>>>>> >
>>>>> >    </condition>
>>>>> >
>>>>> > </extension>
>>>>> >
>>>>> >
>>>>> >
>>>>> > Log:
>>>>> >
>>>>> > freeswitch at mypc> 2011-05-07 22:48:05.717537 [DEBUG] sofia.c:6488 IP
>>>>> > 192.168.200.201
>>>>> >
>>>>> >  Approved by acl "192.168.0.0/16[] <http://192.168.0.0/16%5B%5D>".
>>>>> Access Granted.
>>>>> >
>>>>> > 2011-05-07 22:48:05.717537 [NOTICE] switch_channel.c:812 New Channel
>>>>> > sofia/inter
>>>>> >
>>>>> > nal/+4001 at 192.168.200.100 [92568c57-ff7b-4dbc-b322-a9babf893e62]
>>>>> >
>>>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4760 Channel
>>>>> > sofia/internal/+4001 at 192
>>>>> >
>>>>> > .168.200.100 entering state [received][100]
>>>>> >
>>>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4771 Remote SDP:
>>>>> >
>>>>> > v=0
>>>>> >
>>>>> > o=AudiocodesGW 1062502113 1062501981 IN IP4 192.168.200.201
>>>>> >
>>>>> > s=Phone-Call
>>>>> >
>>>>> > c=IN IP4 192.168.200.201
>>>>> >
>>>>> > t=0 0
>>>>> >
>>>>> > m=audio 6060 RTP/AVP 8 0 96
>>>>> >
>>>>> > a=rtpmap:8 PCMA/8000
>>>>> >
>>>>> > a=rtpmap:0 PCMU/8000
>>>>> >
>>>>> > a=rtpmap:96 telephone-event/8000
>>>>> >
>>>>> > a=fmtp:96 0-15
>>>>> >
>>>>> > a=ptime:20
>>>>> >
>>>>> >
>>>>> >
>>>>> > 2011-05-07 22:48:05.719537 [DEBUG] sofia.c:4908
>>>>> > (sofia/internal/+4001 at 192.168.20
>>>>> >
>>>>> > 0.100) State Change CS_NEW -> CS_INIT
>>>>> >
>>>>> > 2011-05-07 22:48:05.719537 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> > sofia/
>>>>> >
>>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) Running State Change CS_INIT
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State INIT
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:84
>>>>> > sofia/internal/+4001 at 192.168.2
>>>>> >
>>>>> > 00.100 SOFIA INIT
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:124
>>>>> > (sofia/internal/+4001 at 192.168
>>>>> >
>>>>> > .200.100) State Change CS_INIT -> CS_ROUTING
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> > sofia/
>>>>> >
>>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:361
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State INIT going to sleep
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:325
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) Running State Change CS_ROUTING
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_channel.c:1668
>>>>> > (sofia/internal/+4001 at 1
>>>>> >
>>>>> > 92.168.200.100) Callstate Change DOWN -> RINGING
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:364
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State ROUTING
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] mod_sofia.c:147
>>>>> > sofia/internal/+4001 at 192.168.
>>>>> >
>>>>> > 200.100 SOFIA ROUTING
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [DEBUG] switch_core_state_machine.c:77
>>>>> > sofia/internal
>>>>> >
>>>>> > /+4001 at 192.168.200.100 Standard ROUTING
>>>>> >
>>>>> > 2011-05-07 22:48:05.722537 [INFO] mod_dialplan_xml.c:331 Processing
>>>>> +4001
>>>>> > <+4001
>>>>> >
>>>>> >>->+9123 in context default
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> [default->unloop]
>>>>> > continu
>>>>> >
>>>>> > e=false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS) [unloop]
>>>>> > ${unroll_lo
>>>>> >
>>>>> > ops}(true) =~ /^true$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [unloop]
>>>>> > ${sip_loope
>>>>> >
>>>>> > d_call}() =~ /^true$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> > [default->remove_plus_of_
>>>>> >
>>>>> > dst_num] continue=true
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>>>> > [remove_plus_of_dst_
>>>>> >
>>>>> > num] destination_number(+9123) =~ /^\+(\d+)$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>>>> > set(destination_number=912
>>>>> >
>>>>> > 3)
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> > [default->remove_plus_of_
>>>>> >
>>>>> > src_num] continue=true
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (PASS)
>>>>> > [remove_plus_of_src_
>>>>> >
>>>>> > num] caller_id_number(+4001) =~ /^\+(\d+)$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>>>> > set(effective_caller_id_na
>>>>> >
>>>>> > me=4001)
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Action
>>>>> > set(effective_caller_id_nu
>>>>> >
>>>>> > mber=4001)
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> > [default->7_8_to_Lync] co
>>>>> >
>>>>> > ntinue=false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>>>> [7_8_to_Lync]
>>>>> > destin
>>>>> >
>>>>> > ation_number(+9123) =~ /^([78]\d{3})$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> [default->9_to_GW]
>>>>> > contin
>>>>> >
>>>>> > ue=false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>>>> [9_to_GW]
>>>>> > destinatio
>>>>> >
>>>>> > n_number(+9123) =~ /^(9\d+)$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> [default->1_to_IPP]
>>>>> > conti
>>>>> >
>>>>> > nue=false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>>>> [1_to_IPP]
>>>>> > destinati
>>>>> >
>>>>> > on_number(+9123) =~ /^(1\d{3})$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> [default->DISA]
>>>>> > continue=
>>>>> >
>>>>> > false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL) [DISA]
>>>>> > destination_n
>>>>> >
>>>>> > umber(+9123) =~ /^\*(3472)$/ break=on-false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 parsing
>>>>> [default->Recordings]
>>>>> > con
>>>>> >
>>>>> > tinue=false
>>>>> >
>>>>> > Dialplan: sofia/internal/+4001 at 192.168.200.100 Regex (FAIL)
>>>>> [Recordings]
>>>>> > destina
>>>>> >
>>>>> > tion_number(+9123) =~ /^\*(732673)$/ break=on-false
>>>>> >
>>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:119
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
>>>>> >
>>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> > sofia/
>>>>> >
>>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>>> >
>>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:364
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State ROUTING going to sleep
>>>>> >
>>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:325
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) Running State Change CS_EXECUTE
>>>>> >
>>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:371
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State EXECUTE
>>>>> >
>>>>> > 2011-05-07 22:48:05.727537 [DEBUG] mod_sofia.c:240
>>>>> > sofia/internal/+4001 at 192.168.
>>>>> >
>>>>> > 200.100 SOFIA EXECUTE
>>>>> >
>>>>> > 2011-05-07 22:48:05.727537 [DEBUG] switch_core_state_machine.c:157
>>>>> > sofia/interna
>>>>> >
>>>>> > l/+4001 at 192.168.200.100 Standard EXECUTE
>>>>> >
>>>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100set(destination_number=9123)
>>>>> >
>>>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>>>> > sofia/internal/+4001 at 192.1
>>>>> >
>>>>> > 68.200.100 SET [destination_number]=[9123]
>>>>> >
>>>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>>>> > set(effective_caller_id_name=4001)
>>>>> >
>>>>> > 2011-05-07 22:48:05.728537 [DEBUG] mod_dptools.c:1060
>>>>> > sofia/internal/+4001 at 192.1
>>>>> >
>>>>> > 68.200.100 SET [effective_caller_id_name]=[4001]
>>>>> >
>>>>> > EXECUTE sofia/internal/+4001 at 192.168.200.100
>>>>> > set(effective_caller_id_number=4001
>>>>> >
>>>>> > )
>>>>> >
>>>>> > 2011-05-07 22:48:05.729537 [DEBUG] mod_dptools.c:1060
>>>>> > sofia/internal/+4001 at 192.1
>>>>> >
>>>>> > 68.200.100 SET [effective_caller_id_number]=[4001]
>>>>> >
>>>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:189
>>>>> > sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100 has executed the last dialplan instruction,
>>>>> hanging
>>>>> > up.
>>>>> >
>>>>> >
>>>>> >
>>>>> > 2011-05-07 22:48:05.729537 [DEBUG] switch_channel.c:2563
>>>>> > (sofia/internal/+4001 at 1
>>>>> >
>>>>> > 92.168.200.100) Callstate Change RINGING -> HANGUP
>>>>> >
>>>>> > 2011-05-07 22:48:05.729537 [NOTICE] switch_core_state_machine.c:191
>>>>> Hangup
>>>>> > sofia
>>>>> >
>>>>> > /internal/+4001 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>> >
>>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_channel.c:2579 Send signal
>>>>> > sofia/inter
>>>>> >
>>>>> > nal/+4001 at 192.168.200.100 [KILL]
>>>>> >
>>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> > sofia/
>>>>> >
>>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>>> >
>>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:371
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State EXECUTE going to sleep
>>>>> >
>>>>> > 2011-05-07 22:48:05.730538 [DEBUG] switch_core_state_machine.c:325
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) Running State Change CS_HANGUP
>>>>> >
>>>>> > 2011-05-07 22:48:05.731538 [DEBUG] switch_core_state_machine.c:565
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State HANGUP
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:457 Channel
>>>>> > sofia/internal/+4001@
>>>>> >
>>>>> > 192.168.200.100 hanging up, cause: NORMAL_CLEARING
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] mod_sofia.c:519 Responding to
>>>>> INVITE
>>>>> > with: 48
>>>>> >
>>>>> > 0
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:46
>>>>> > sofia/internal
>>>>> >
>>>>> > /+4001 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:565
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State HANGUP going to sleep
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:356
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> > sofia/
>>>>> >
>>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:325
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) Running State Change CS_REPORTING
>>>>> >
>>>>> > 2011-05-07 22:48:05.732538 [DEBUG] switch_core_state_machine.c:625
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State REPORTING
>>>>> >
>>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:53
>>>>> > sofia/internal
>>>>> >
>>>>> > /+4001 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
>>>>> >
>>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:625
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State REPORTING going to sleep
>>>>> >
>>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_state_machine.c:350
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
>>>>> >
>>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> > sofia/
>>>>> >
>>>>> > internal/+4001 at 192.168.200.100 [BREAK]
>>>>> >
>>>>> > 2011-05-07 22:48:06.017554 [DEBUG] switch_core_session.c:1288 Session
>>>>> 2
>>>>> > (sofia/i
>>>>> >
>>>>> > nternal/+4001 at 192.168.200.100) Locked, Waiting on external entities
>>>>> >
>>>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1306
>>>>> Session 2
>>>>> > (sofia/
>>>>> >
>>>>> > internal/+4001 at 192.168.200.100) Ended
>>>>> >
>>>>> > 2011-05-07 22:48:06.017554 [NOTICE] switch_core_session.c:1308 Close
>>>>> Channel
>>>>> > sof
>>>>> >
>>>>> > ia/internal/+4001 at 192.168.200.100 [CS_DESTROY]
>>>>> >
>>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:454
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) Callstate Change HANGUP -> DOWN
>>>>> >
>>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:457
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) Running State Change CS_DESTROY
>>>>> >
>>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State DESTROY
>>>>> >
>>>>> > 2011-05-07 22:48:06.018554 [DEBUG] mod_sofia.c:362
>>>>> > sofia/internal/+4001 at 192.168.
>>>>> >
>>>>> > 200.100 SOFIA DESTROY
>>>>> >
>>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:60
>>>>> > sofia/internal
>>>>> >
>>>>> > /+4001 at 192.168.200.100 Standard DESTROY
>>>>> >
>>>>> > 2011-05-07 22:48:06.018554 [DEBUG] switch_core_state_machine.c:467
>>>>> > (sofia/intern
>>>>> >
>>>>> > al/+4001 at 192.168.200.100) State DESTROY going to sleep
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing 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 mel0torme at gmail.com  Tue May 10 09:08:08 2011
From: mel0torme at gmail.com (Tom C)
Date: Mon, 9 May 2011 22:08:08 -0700
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>
Message-ID: <BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>

I was thinking about this a few days ago.  Slightly different scenario: I
want to stream audio into a conference every ten minutes or so.

My ideas was to do it by connecting another extension to the conference.
This extra extension would (theoretically) run a Lua script to stream the
audio, and then sleep, and every few seconds it would look to see if it was
the only extension still connected to the conference.

In your case, if you're doing the "mad boss" and inviting all the
participants to the conference, it should be easy to add one more extension
to the list of invites.  Just have the extension's script wait 10 seconds or
so before streaming the audio, and then it can just hang up.

Something like this:

streamwelcomefile.lua
<code>

if ( session:ready() ) then
  session:answer();
  session:sleep(10000)  -- wait for 10 seconds for other people to join
conference.
  session:streamFile("ivr/8000/ivr-welcome.wav");
  session:streamFile("ivr/8000/ivr-thank_you_for_calling.wav");
  session:hangup("NORMAL_CLEARING");
end

</code>

And the dialplan might look like this:
<code>

<extension name="StreamWelcome">
        <condition field="destination_number" expression="^000111$">
                <action application="lua" data="streamwelcomefile.lua" />
        </condition>
 </extension>

</code>


Disclaimer:  I haven't actually tested this in a conference.  Good luck!
:-)
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From jpatten at co.brazos.tx.us  Tue May 10 10:21:09 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Tue, 10 May 2011 06:21:09 +0000
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>,
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>
Message-ID: <8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>

Only problem is session:streamFile doesn't make it into conference.

The only way I've found to do this so far is to conference_set_auto_outcall a loopback extension to a non-user extension that simply streams a file in a context:
session:execute("conference_set_auto_outcall","loopback/3041/custom/")

but if I want to mute the conference so only the originator is heard (a standard "overhead" style page) using:
session:execute("set", "conference_auto_outcall_flags=mute") then this is never heard because this loopback "participant" is muted as well.

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Tom C [mel0torme at gmail.com]
Sent: Tuesday, May 10, 2011 12:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference

I was thinking about this a few days ago.  Slightly different scenario: I want to stream audio into a conference every ten minutes or so.

My ideas was to do it by connecting another extension to the conference.  This extra extension would (theoretically) run a Lua script to stream the audio, and then sleep, and every few seconds it would look to see if it was the only extension still connected to the conference.

In your case, if you're doing the "mad boss" and inviting all the participants to the conference, it should be easy to add one more extension to the list of invites.  Just have the extension's script wait 10 seconds or so before streaming the audio, and then it can just hang up.

Something like this:

streamwelcomefile.lua
<code>

if ( session:ready() ) then
  session:answer();
  session:sleep(10000)  -- wait for 10 seconds for other people to join conference.
  session:streamFile("ivr/8000/ivr-welcome.wav");
  session:streamFile("ivr/8000/ivr-thank_you_for_calling.wav");
  session:hangup("NORMAL_CLEARING");
end

</code>

And the dialplan might look like this:
<code>

<extension name="StreamWelcome">
        <condition field="destination_number" expression="^000111$">
                <action application="lua" data="streamwelcomefile.lua" />
        </condition>
 </extension>

</code>


Disclaimer:  I haven't actually tested this in a conference.  Good luck!  :-)
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From jpatten at co.brazos.tx.us  Tue May 10 10:34:43 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Tue, 10 May 2011 06:34:43 +0000
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>,
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>,
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
Message-ID: <8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>

It seems the way this could be accomplished is to allow asynchronous execution capabilities in lua. However I don't think this exists.

Is there a better way to "execute and forget" the conference app so the script can move on to the next bit of code?
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten [jpatten at co.brazos.tx.us]
Sent: Tuesday, May 10, 2011 1:21 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference

Only problem is session:streamFile doesn't make it into conference.

The only way I've found to do this so far is to conference_set_auto_outcall a loopback extension to a non-user extension that simply streams a file in a context:
session:execute("conference_set_auto_outcall","loopback/3041/custom/")

but if I want to mute the conference so only the originator is heard (a standard "overhead" style page) using:
session:execute("set", "conference_auto_outcall_flags=mute") then this is never heard because this loopback "participant" is muted as well.

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Tom C [mel0torme at gmail.com]
Sent: Tuesday, May 10, 2011 12:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference

I was thinking about this a few days ago.  Slightly different scenario: I want to stream audio into a conference every ten minutes or so.

My ideas was to do it by connecting another extension to the conference.  This extra extension would (theoretically) run a Lua script to stream the audio, and then sleep, and every few seconds it would look to see if it was the only extension still connected to the conference.

In your case, if you're doing the "mad boss" and inviting all the participants to the conference, it should be easy to add one more extension to the list of invites.  Just have the extension's script wait 10 seconds or so before streaming the audio, and then it can just hang up.

Something like this:

streamwelcomefile.lua
<code>

if ( session:ready() ) then
  session:answer();
  session:sleep(10000)  -- wait for 10 seconds for other people to join conference.
  session:streamFile("ivr/8000/ivr-welcome.wav");
  session:streamFile("ivr/8000/ivr-thank_you_for_calling.wav");
  session:hangup("NORMAL_CLEARING");
end

</code>

And the dialplan might look like this:
<code>

<extension name="StreamWelcome">
        <condition field="destination_number" expression="^000111$">
                <action application="lua" data="streamwelcomefile.lua" />
        </condition>
 </extension>

</code>


Disclaimer:  I haven't actually tested this in a conference.  Good luck!  :-)
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From devel at omninet.eu  Tue May 10 11:27:53 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Tue, 10 May 2011 10:27:53 +0300
Subject: [Freeswitch-users] many outgoing calls with mod_lcr at the same
	time possible?
In-Reply-To: <BANLkTim8H82JnYQbs9ox8zgii8iXYdy1hw@mail.gmail.com>
References: <E6513A570B0E4F47A248C4E0735EAD99@omni1.local>
	<BANLkTim8H82JnYQbs9ox8zgii8iXYdy1hw@mail.gmail.com>
Message-ID: <C4F1CC5503B9428BB0B1D2E1023E6C36@omni1.local>

Hello Avi,

 

Thanks for the hint.

 

I've tried it but unfortunately without success.

 

I've done these two tests:

 

1) Try to set two variables for the two destinations and bridge

Run application LCR with the called number

Set a variable phoneA (inline=true) with the first result of
${lcr_auto_route}

Run again application LCR with the second number to call

Set another variable phoneB (inline=true) with the second result

Bridge to phoneA,phoneB

 

The problem is that they don't get the value of ${lcr_auto_route}

 

This is the line that should do this:

<action application="set" inline="true" data="phoneA=${lcr_auto_route}"/>

 

 

2) Try to have the endpoint directly in the bridge, as you suggested :-)

 

<action application="bridge" data="lcr/XXXXX"/>

(where XXXXX is the number)

 

This time the phone gets the call, it is ringing, but the a-leg gets dropped
before the ringing. This a-leg stays in the channel list and you can not
kill it with uuid_kill

 

(not to forget to mention: CentOS, latest Git)

 

Any idea?

 

Regards

Anestis

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Saturday, May 07, 2011 8:48 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] many outgoing calls with mod_lcr at the same
time possible?

 

Since LCR registers itself as an endpoint, I suppose you can bridge to both
that way.

It's mentioned here, but the sample is actually for something else:
http://wiki.freeswitch.org/wiki/Mod_lcr#Endpoint

 <http://wiki.freeswitch.org/wiki/Mod_lcr#Endpoint> If you could update the
wiki with your use case, that would be great.

-Avi

On Sat, May 7, 2011 at 12:52 PM, Anestis Mavro <devel at omninet.eu> wrote:

Hello Freeswitchers,

 

Is it possible to make many outgoing calls with mod_lcr at the same time? An
example would be to redirect an incoming call to two mobile phones and ring
them simultaneously.

With a "normal" bridge we can achieve this, but what about having also LCR
(billing and limit included) involved?

 

Regards

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

 

 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

 

The message was checked by ESET NOD32 Antivirus.

 

http://www.eset.com

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From peter.olsson at visionutveckling.se  Tue May 10 11:44:56 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 10 May 2011 09:44:56 +0200
Subject: [Freeswitch-users] many outgoing calls with mod_lcr at the
	same	time possible?
In-Reply-To: <C4F1CC5503B9428BB0B1D2E1023E6C36@omni1.local>
References: <E6513A570B0E4F47A248C4E0735EAD99@omni1.local>
	<BANLkTim8H82JnYQbs9ox8zgii8iXYdy1hw@mail.gmail.com>
	<C4F1CC5503B9428BB0B1D2E1023E6C36@omni1.local>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F7575CCB@cooper>

There is a Jira reported for when using lcr as an endpoint - you probably hit the same bug?

Check out Jira FS-3109.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anestis Mavro
Skickat: den 10 maj 2011 09:28
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] many outgoing calls with mod_lcr at the same time possible?

Hello Avi,

Thanks for the hint.

I've tried it but unfortunately without success.

I've done these two tests:

1) Try to set two variables for the two destinations and bridge
Run application LCR with the called number
Set a variable phoneA (inline=true) with the first result of ${lcr_auto_route}
Run again application LCR with the second number to call
Set another variable phoneB (inline=true) with the second result
Bridge to phoneA,phoneB

The problem is that they don't get the value of ${lcr_auto_route}

This is the line that should do this:
<action application="set" inline="true" data="phoneA=${lcr_auto_route}"/>


2) Try to have the endpoint directly in the bridge, as you suggested :)

<action application="bridge" data="lcr/XXXXX"/>
(where XXXXX is the number)

This time the phone gets the call, it is ringing, but the a-leg gets dropped before the ringing. This a-leg stays in the channel list and you can not kill it with uuid_kill

(not to forget to mention: CentOS, latest Git)

Any idea?

Regards
Anestis

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: Saturday, May 07, 2011 8:48 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] many outgoing calls with mod_lcr at the same time possible?

Since LCR registers itself as an endpoint, I suppose you can bridge to both that way.
It's mentioned here, but the sample is actually for something else: http://wiki.freeswitch.org/wiki/Mod_lcr#Endpoint
If you could update the wiki with your use case, that would be great.
-Avi
On Sat, May 7, 2011 at 12:52 PM, Anestis Mavro <devel at omninet.eu<mailto:devel at omninet.eu>> wrote:
Hello Freeswitchers,

Is it possible to make many outgoing calls with mod_lcr at the same time? An example would be to redirect an incoming call to two mobile phones and ring them simultaneously.
With a "normal" bridge we can achieve this, but what about having also LCR (billing and limit included) involved?

Regards
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<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



__________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________


The message was checked by ESET NOD32 Antivirus.


http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
!DSPAM:4dc8ea5f32761708520604!
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From devel at omninet.eu  Tue May 10 13:08:47 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Tue, 10 May 2011 12:08:47 +0300
Subject: [Freeswitch-users] many outgoing calls with mod_lcr at
	thesame	time possible?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58F7575CCB@cooper>
References: <E6513A570B0E4F47A248C4E0735EAD99@omni1.local><BANLkTim8H82JnYQbs9ox8zgii8iXYdy1hw@mail.gmail.com><C4F1CC5503B9428BB0B1D2E1023E6C36@omni1.local>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F7575CCB@cooper>
Message-ID: <7F5AA9DC4A374F5F890A977912CBBB40@omni1.local>

Yes, it is the bug


 

Thanks

 

Anestis

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: Tuesday, May 10, 2011 10:45 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] many outgoing calls with mod_lcr at thesame
time possible?

 

There is a Jira reported for when using lcr as an endpoint ? you probably
hit the same bug?

 

Check out Jira FS-3109.

 

/Peter

 

Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anestis Mavro
Skickat: den 10 maj 2011 09:28
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] many outgoing calls with mod_lcr at the same
time possible?

 

Hello Avi,

 

Thanks for the hint.

 

I?ve tried it but unfortunately without success.

 

I?ve done these two tests:

 

1) Try to set two variables for the two destinations and bridge

Run application LCR with the called number

Set a variable phoneA (inline=true) with the first result of
${lcr_auto_route}

Run again application LCR with the second number to call

Set another variable phoneB (inline=true) with the second result

Bridge to phoneA,phoneB

 

The problem is that they don?t get the value of ${lcr_auto_route}

 

This is the line that should do this:

<action application=?set? inline=?true? data=?phoneA=${lcr_auto_route}?/>

 

 

2) Try to have the endpoint directly in the bridge, as you suggested :-)

 

<action application=?bridge? data=?lcr/XXXXX?/>

(where XXXXX is the number)

 

This time the phone gets the call, it is ringing, but the a-leg gets dropped
before the ringing. This a-leg stays in the channel list and you can not
kill it with uuid_kill

 

(not to forget to mention: CentOS, latest Git)

 

Any idea?

 

Regards

Anestis

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Saturday, May 07, 2011 8:48 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] many outgoing calls with mod_lcr at the same
time possible?

 

Since LCR registers itself as an endpoint, I suppose you can bridge to both
that way.

It's mentioned here, but the sample is actually for something else:
http://wiki.freeswitch.org/wiki/Mod_lcr#Endpoint

If you could update the wiki with your use case, that would be great.

-Avi

On Sat, May 7, 2011 at 12:52 PM, Anestis Mavro <devel at omninet.eu> wrote:

Hello Freeswitchers,

 

Is it possible to make many outgoing calls with mod_lcr at the same time? An
example would be to redirect an incoming call to two mobile phones and ring
them simultaneously.

With a ?normal? bridge we can achieve this, but what about having also LCR
(billing and limit included) involved?

 

Regards

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

 

 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

 

The message was checked by ESET NOD32 Antivirus.

 

http://www.eset.com



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
!DSPAM:4dc8ea5f32761708520604! 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From marcdecorny at gmail.com  Tue May 10 14:48:44 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Tue, 10 May 2011 11:48:44 +0100
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <BANLkTi=vPrN84Yjy_BOOooGhc4EB_PMvWQ@mail.gmail.com>
References: <4DC84196.9050905@newpace.ca>
	<BANLkTimtTJ5QHofCaC0_ExXDexefjj7OqA@mail.gmail.com>
	<BANLkTi=vPrN84Yjy_BOOooGhc4EB_PMvWQ@mail.gmail.com>
Message-ID: <BANLkTimP1iPV552J66TecEXTen75sQLLOg@mail.gmail.com>

will the fact that the file is on a remote server increase delay in starting
the replay of the audio files?

I setup a remote directory on remote server and played them that way. any
reason why this would be a bad idea?


On Mon, May 9, 2011 at 10:02 PM, Brad Mina <brad at tritelcomm.com> wrote:

> You could setup an internal http server with a directory called 'audio' (
> http://server.ip/audio/), then use mod_shout as Avi suggested like this:
> <action application="shout" data="http://server.ip/audio/file.mp3"/>
>
>
> On Mon, May 9, 2011 at 12:57 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> At least one potion is to use mod_shout to stream them from a server.
>> Do note that this adds another point of failure to your setup...
>>
>> -Avi
>>
>>
>> On Mon, May 9, 2011 at 10:33 PM, Adam Kelloway <adam.kelloway at newpace.ca>wrote:
>>
>>> If I did not wish to store audio files on a FreeSWITCH host, what are my
>>> options for being able to retrieve a remote file and playing it?
>>> My understanding is that the playback application can only play a local
>>> file, is that correct?
>>> I did notice that there is a shell_stream module, which you could
>>> potentially use to retrieve a file (say, via HTTP), and provide the
>>> audio data as the output bash script output. Is there a way to do this
>>> or something similar without having to actually load another
>>> program/script?
>>>
>>> Thanks,
>>>
>>> Adam Kelloway
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jgallartm at gmail.com  Tue May 10 15:09:50 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 10 May 2011 13:09:50 +0200
Subject: [Freeswitch-users] invalid ELF header error in lua startup script
Message-ID: <BANLkTim+bhpk-bcUNqdVRkrUGuTcqSB+FA@mail.gmail.com>

Hello,

we have a lua script that we want to use as startup script in order to start
listening for events as soon as FS starts. This is the configuration:

<param name="module-directory" value="/usr/lib/lua/5.1/?.so"/>
   <param name="module-directory" value="/usr/lib/lua/5.1/?.lua"/>
    <param name="module-directory" value="/usr/lib/lua/5.1/socket/?.lua"/>
    <param name="module-directory" value="/usr/lib/lua/5.1/socket/?.so"/>
    <param name="module-directory" value="/usr/local/share/lua/5.1/?.so"/>
    <param name="module-directory" value="/usr/local/share/lua/5.1/?.lua"/>
    <param name="module-directory"
value="/usr/local/share/lua/5.1/xmlrpc/?.lua"/>
<param name="startup-script" value="script.lua"/>

The script first line is:

require("socket")

The file  /usr/local/share/lua/5.1/socket.lua exists. Actually when we
execute "lua script.lua" it runs properly. But this is the freeswitch log at
startup:
2011-05-10 07:00:31.281411 [ERR] mod_lua.cpp:191 error loading module
'socket' from file '/usr/local/share/lua/5.1/socket.lua':
/usr/local/share/lua/5.1/socket.lua: invalid ELF header
stack traceback:
[C]: ?
[C]: in function 'require'
/usr/local/freeswitch/scripts/script.lua:1: in main chunk

Has anyone gone over the same issue? We tried with the latest git.

Thanks in advance

Javi
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From t.mahe at telemaque.fr  Tue May 10 16:30:48 2011
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Tue, 10 May 2011 14:30:48 +0200
Subject: [Freeswitch-users] invalid ELF header error in lua startup
	script
In-Reply-To: <BANLkTim+bhpk-bcUNqdVRkrUGuTcqSB+FA@mail.gmail.com>
References: <BANLkTim+bhpk-bcUNqdVRkrUGuTcqSB+FA@mail.gmail.com>
Message-ID: <4DC92FF8.7010006@telemaque.fr>

32bit module on a 64bit system or the opposite ?

Le 10/05/2011 13:09, Javier Gallart a ?crit :
> Hello,
>
> we have a lua script that we want to use as startup script in order to 
> start listening for events as soon as FS starts. This is the 
> configuration:
>
> <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/>
> <param name="module-directory" value="/usr/lib/lua/5.1/?.lua"/>
> <param name="module-directory" value="/usr/lib/lua/5.1/socket/?.lua"/>
> <param name="module-directory" value="/usr/lib/lua/5.1/socket/?.so"/>
> <param name="module-directory" value="/usr/local/share/lua/5.1/?.so"/>
> <param name="module-directory" value="/usr/local/share/lua/5.1/?.lua"/>
> <param name="module-directory" 
> value="/usr/local/share/lua/5.1/xmlrpc/?.lua"/>
> <param name="startup-script" value="script.lua"/>
>
> The script first line is:
>
> require("socket")
>
> The file  /usr/local/share/lua/5.1/socket.lua exists. Actually when we 
> execute "lua script.lua" it runs properly. But this is the freeswitch 
> log at startup:
> 2011-05-10 07:00:31.281411 [ERR] mod_lua.cpp:191 error loading module 
> 'socket' from file '/usr/local/share/lua/5.1/socket.lua':
> /usr/local/share/lua/5.1/socket.lua: invalid ELF header
> stack traceback:
> [C]: ?
> [C]: in function 'require'
> /usr/local/freeswitch/scripts/script.lua:1: in main chunk
>
> Has anyone gone over the same issue? We tried with the latest git.
>
> Thanks in advance
>
> Javi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From lakindia89 at gmail.com  Tue May 10 16:34:53 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 10 May 2011 18:04:53 +0530
Subject: [Freeswitch-users] Primary leg hangs up when other leg rejects the
	call
Message-ID: <BANLkTinZ0LO3csoV0jFuRTdzXanaayuQOw@mail.gmail.com>

Dear all,
I'm using Event Inbound Socket. I want to originate a call to the user and
bridge him with different user.
So I used the following commands from the telnet.


api originate {exec_after_bridge_app=park,ignore_early_media=true}user/1010
&park()

Content-Type: api/response
Content-Length: 41

+OK cf9a78a2-7aff-11e0-b732-4f9b0298a6ed

sendmsg cf9a78a2-7aff-11e0-b732-4f9b0298a6ed
call-command: execute
execute-app-name: bridge
execute-app-arg:user/1005

Content-Type: command/reply
Reply-Text: +OK

The user 1005 rejects the call and I got the response as CALL_REJECTED.
I expect that primary leg will not hangup and go into park mode. But it got
hangup.

Since by default hangup_after_bridge will be false, I'm not setting it
explicitly here.

FreeSwitch logs are here,
http://pastebin.freeswitch.org/16264

Can someone please tell me what is the problem I've done and how to make the
primary leg again in park() ??
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From rupa at rupa.com  Tue May 10 17:07:40 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 10 May 2011 08:07:40 -0500
Subject: [Freeswitch-users] many outgoing calls with mod_lcr at thesame
 time possible?
In-Reply-To: <7F5AA9DC4A374F5F890A977912CBBB40@omni1.local>
References: <E6513A570B0E4F47A248C4E0735EAD99@omni1.local>
	<BANLkTim8H82JnYQbs9ox8zgii8iXYdy1hw@mail.gmail.com>
	<C4F1CC5503B9428BB0B1D2E1023E6C36@omni1.local>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F7575CCB@cooper>
	<7F5AA9DC4A374F5F890A977912CBBB40@omni1.local>
Message-ID: <BANLkTinPUQA2XwKs5V_QkE645m7qsEAP8Q@mail.gmail.com>

I'll need to investigate the bug some more, I don't see it here.  I
routinely do forked lcr calls.  The dialstring ends up looking
something like:

 <bridge_early_media=true,origination_caller_id_number=214XXXYYYY,forked_call=true>lcr/1214XXXYYYY:_:lcr/1214XXXYYYY:_:user/1 at voip.rupa.com,user/2 at voip.rupa.com,user/3 at voip.rupa.com,user/4 at voip.rupa.com,user/5 at voip.rupa.com,user/foo at voip.rupa.com

hm.. I'm using enterprise originate rather than regular originate.
This is an artifact of when lcr didn't work as an endpoint and I had
to loop through loopback.

As for my regular dialplan, I use lcr as a dialplan:

        <action application="execute_extension" data="$1$2 lcr default"/>

$1$2 is the number, lcr is the dialplan module (lcr), default is the
lcr configuration.

I try to use lcr in the less-than-common ways to ensure I don't break
it when I make changes...


On Tue, May 10, 2011 at 4:08 AM, Anestis Mavro <devel at omninet.eu> wrote:
> Yes, it is the bug?
>
>
>
> Thanks
>
>
>
> Anestis
>
>
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
> Olsson
> Sent: Tuesday, May 10, 2011 10:45 AM
>
> To: 'FreeSWITCH Users Help'
> Subject: Re: [Freeswitch-users] many outgoing calls with mod_lcr at thesame
> time possible?
>
>
>
> There is a Jira reported for when using lcr as an endpoint ? you probably
> hit the same bug?
>
>
>
> Check out Jira FS-3109.
>
>
>
> /Peter
>
>
>
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anestis Mavro
> Skickat: den 10 maj 2011 09:28
> Till: 'FreeSWITCH Users Help'
> ?mne: Re: [Freeswitch-users] many outgoing calls with mod_lcr at the same
> time possible?
>
>
>
> Hello Avi,
>
>
>
> Thanks for the hint.
>
>
>
> I?ve tried it but unfortunately without success.
>
>
>
> I?ve done these two tests:
>
>
>
> 1) Try to set two variables for the two destinations and bridge
>
> Run application LCR with the called number
>
> Set a variable phoneA (inline=true) with the first result of
> ${lcr_auto_route}
>
> Run again application LCR with the second number to call
>
> Set another variable phoneB (inline=true) with the second result
>
> Bridge to phoneA,phoneB
>
>
>
> The problem is that they don?t get the value of ${lcr_auto_route}
>
>
>
> This is the line that should do this:
>
> <action application=?set? inline=?true? data=?phoneA=${lcr_auto_route}?/>
>
>
>
>
>
> 2) Try to have the endpoint directly in the bridge, as you suggested J
>
>
>
> <action application=?bridge? data=?lcr/XXXXX?/>
>
> (where XXXXX is the number)
>
>
>
> This time the phone gets the call, it is ringing, but the a-leg gets dropped
> before the ringing. This a-leg stays in the channel list and you can not
> kill it with uuid_kill
>
>
>
> (not to forget to mention: CentOS, latest Git)
>
>
>
> Any idea?
>
>
>
> Regards
>
> Anestis
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
> Marcus
> Sent: Saturday, May 07, 2011 8:48 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] many outgoing calls with mod_lcr at the same
> time possible?
>
>
>
> Since LCR registers itself as an endpoint, I suppose you can bridge to both
> that way.
>
> It's mentioned here, but the sample is actually for something
> else:?http://wiki.freeswitch.org/wiki/Mod_lcr#Endpoint
>
> If you could update the wiki with your use case, that would be great.
>
> -Avi
>
> On Sat, May 7, 2011 at 12:52 PM, Anestis Mavro <devel at omninet.eu> wrote:
>
> Hello Freeswitchers,
>
>
>
> Is it possible to make many outgoing calls with mod_lcr at the same time? An
> example would be to redirect an incoming call to two mobile phones and ring
> them simultaneously.
>
> With a ?normal? bridge we can achieve this, but what about having also LCR
> (billing and limit included) involved?
>
>
>
> Regards
>
> 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
>
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
> database 5054 (20100423) __________
>
>
>
> The message was checked by ESET NOD32 Antivirus.
>
>
>
> http://www.eset.com
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
> database 5054 (20100423) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> !DSPAM:4dc8ea5f32761708520604!
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
> database 5054 (20100423) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> __________ 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
>
>



-- 
-Rupa


From rupa at rupa.com  Tue May 10 17:15:52 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 10 May 2011 08:15:52 -0500
Subject: [Freeswitch-users] mod_nibblebill updating 3 times
In-Reply-To: <93DC24C0-3D80-454C-945F-AC8EE790FAC1@gmail.com>
References: <C8A89431-F467-43B0-8D77-480DD2A1F4B4@gmail.com>
	<BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>
	<93DC24C0-3D80-454C-945F-AC8EE790FAC1@gmail.com>
Message-ID: <BANLkTi=-eJWQbaaGdngPzWaA4xYxnx8xDg@mail.gmail.com>

I've removed the on_reporting hook:

http://jira.freeswitch.org/browse/FS-2890

this should at least get rid of the final update and should remove the
small difference between nibblebill numbers and cdrs.

On Mon, Apr 25, 2011 at 11:24 PM, Rogelio Perez <rogelio.perez at gmail.com> wrote:
> Hello, I'm using mod_nibblebill to bill my calls only for leg b and
> heartbeats set to 'off'.
> After the hangup?I see 3 different MySQL updates from nibblebill instead of
> just one.
> This is my dial plan:
> <extension name="TEST">
> ?? ? ? ? <condition field="destination_number" expression="^(1\d+)$">
> ?? ? ? ? ? ? ? ? <action application="set" data="dialed_extension=$1"/>
> ?? ? ? ? ? ? ? ? ? ? ? ?<action application="export"
> data="dialed_extension=$1"/>
> ?? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
> data="hangup_after_bridge=true"/>
> ?? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
> data="continue_on_fail=true"/>
> <action application="bridge"
> data="{nibble_rate=5,nibble_account=13}sofia/gateway/xxxxxxx/$1"/>
> ?? ? ? ? ? ? ? ? ? ? ? ? ?<action application="hangup"/>
> </condition>
> </extension>
> ...this is the MySQL log:
> 68080 Query select 1
> 68080 Query UPDATE web_account SET balance=balance-0.343372 WHERE id='13'
> 68080 Query select 1
> 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13'
> 68080 Query select 1
> 68080 Query UPDATE web_account SET balance=balance-0.001177 WHERE id='13'
> 68080 Query select 1
> 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13'
> 68080 Query select 1
> 68080 Query UPDATE web_account SET balance=balance-0.001197 WHERE id='13'
> 68080 Query select 1
> 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13'
>
> ...and here's the FS log:?http://pastebin.freeswitch.org/16175
> The xml_cdr is saved after the second update, so the?nibble_total_billed is
> different than the real total value debited from the account balance.
> Is this normal?
> Shouldn't nibblebill just do one update after the hangup?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sid.kshatriya at gmail.com  Tue May 10 17:19:38 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Tue, 10 May 2011 18:49:38 +0530
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
Message-ID: <BANLkTi=dyU4WHz1edncYCu43v0+jQbZEXw@mail.gmail.com>

Have you tried didforsale.com ? Their pricing seems to be very competitive.
They seem to make the most sense to me, but their website is a little
outdated...

Any more inbount DID providers? Anybody? (I need multi channel support -- at
least 5 channels)

On Wed, May 4, 2011 at 4:33 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Is this just for inbound? The benefit of the freepbx trunks is that it
> includes outgoing for that price.. and therefore costs a lot more.
>
> 1) Have a look at one of FS's sponsors, VoiceNetwork.ca's pricing:
> http://www.voicenetwork.ca/voip.php?page=4
>  <http://www.voicenetwork.ca/voip.php?page=4>
> USA DID Origination
> Unlimited Incoming USA DID's for $3.95 MRC (2 Channels)
> Per-Minute Incoming USA DID's for $0.99 MRC @ $0.011 per minute (10
> Channels)
>
> 2) Or try http://www.didforsale.com for multi-channel incoming.
>
> -Avi
>
>
>
> On Wed, May 4, 2011 at 1:46 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:
>
>> Have a look  http://www.dslreports.com/gbu here .
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Can-anybody-recommend-some-good-sip-trunk-providers-tp6330281p6330365.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
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From rupa at rupa.com  Tue May 10 17:24:18 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 10 May 2011 08:24:18 -0500
Subject: [Freeswitch-users] mod_crd_sqlite entry limit and sqlite
 segfaults on triggers
In-Reply-To: <4DB874FF.5060901@yx.cl>
References: <4DB5E0F1.4050602@yx.cl> <4DB809B4.5060601@yx.cl>
	<1303921474677-6310155.post@n2.nabble.com> <4DB874FF.5060901@yx.cl>
Message-ID: <BANLkTimb7GqKX2kT3Y2oAwRJfnhuHPK9Hw@mail.gmail.com>

I would argue that you do not want to check the row count on every
single insert.  Have a cron job that runs periodically and does the
deletes.  Run that job once a day.  Doing the maintenance on each
insert is going to kill performance *and* concurrency.

On Wed, Apr 27, 2011 at 2:56 PM, Neven Boric <nboric at yx.cl> wrote:
> Will do. What about the limit on the number of cdrs? Should I report
> that too?



-- 
-Rupa


From jaybinks at gmail.com  Tue May 10 17:53:40 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 10 May 2011 23:53:40 +1000
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
Message-ID: <BANLkTimZy_E10_PM1bWt3hkez8sCy1LJDg@mail.gmail.com>

for US & Canada DID's use
http://www.voicenetwork.ca/ <http://www.voicenetwork.ca/voip.php?page=4>

NormT is the guy behind that and is very active on IRC and mailing list.
and is a FS sponsor.

so using him kinda in a way supports the community :)

aside from that , Norm is a great bloke..
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From rupa at rupa.com  Tue May 10 18:10:50 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 10 May 2011 09:10:50 -0500
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTi=dyU4WHz1edncYCu43v0+jQbZEXw@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
	<BANLkTi=dyU4WHz1edncYCu43v0+jQbZEXw@mail.gmail.com>
Message-ID: <BANLkTin+m1gPcuJ4a-JPVyyLW-fvWDj8Rg@mail.gmail.com>

I use didforsale for one of my trunks where I needed a cheep
conference bridge.  I've had issues with some people getting a reorder
even though the trunk isn't full (25 concurrent is what I'm set up
for).  Most of those issues were > 6mo ago, so maybe they've gotten
better.  Usually a call retry would be fine.  I didn't have enough
people on enough different carriers to see if this was an issue with a
single provider (say ATT or Cox or whatever).

didforsale *is* cheap for multi-channel support -- at least for the
first "chunk".  They also have a formula for channel sharing if you
purchase a large # of DIDs from them where the per-channel cost gets
even lower.

On Tue, May 10, 2011 at 8:19 AM, Sidharth Kshatriya
<sid.kshatriya at gmail.com> wrote:
> Have you tried didforsale.com ? Their pricing seems to be very competitive.
> They seem to make the most sense to me, but their website is a little
> outdated...
> Any more inbount DID providers? Anybody? (I need multi channel support -- at
> least 5 channels)

-- 
-Rupa


From jonas_e at swipnet.se  Tue May 10 15:55:36 2011
From: jonas_e at swipnet.se (Jonas Ericsson)
Date: Tue, 10 May 2011 13:55:36 +0200
Subject: [Freeswitch-users] Problems with multiple IP-addresses
Message-ID: <web-58862251@mailbe04.swip.net>

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From qiqi7036 at 163.com  Tue May 10 11:40:55 2011
From: qiqi7036 at 163.com (qiqi7036)
Date: Tue, 10 May 2011 15:40:55 +0800
Subject: [Freeswitch-users] =?gb2312?b?tPC4tDogRnJlZVNXSVRDSC11c2VycyBE?=
	=?gb2312?b?aWdlc3QsIFZvbCA1OSwgSXNzdWUgNDg=?=
In-Reply-To: <mailman.8772.1304971426.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.8772.1304971426.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <00c501cc0ee5$98dde3c0$ca99ab40$@com>

Hi,
  I think mod_shout can play remote prompts, please refer this link:
http://wiki.freeswitch.org/wiki/Mod_shout


dennis.deng

-----????-----
???: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] ??
freeswitch-users-request at lists.freeswitch.org
????: 2011?5?10? 4:04
???: freeswitch-users at lists.freeswitch.org
??: FreeSWITCH-users Digest, Vol 59, Issue 48

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
	freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific than
"Re: Contents of FreeSWITCH-users digest..."




From mike.tesliuk at ultra.net.br  Tue May 10 18:54:57 2011
From: mike.tesliuk at ultra.net.br (Mike Tesliuk)
Date: Tue, 10 May 2011 11:54:57 -0300
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTin+m1gPcuJ4a-JPVyyLW-fvWDj8Rg@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
	<BANLkTi=dyU4WHz1edncYCu43v0+jQbZEXw@mail.gmail.com>
	<BANLkTin+m1gPcuJ4a-JPVyyLW-fvWDj8Rg@mail.gmail.com>
Message-ID: <BANLkTike33jWuNX72kkhzhbHiKnvjoqfTg@mail.gmail.com>

you can try grnvoip.com or rnktel.com , for united states and others country
is good, for brazil is not.

2011/5/10 Rupa Schomaker <rupa at rupa.com>

> I use didforsale for one of my trunks where I needed a cheep
> conference bridge.  I've had issues with some people getting a reorder
> even though the trunk isn't full (25 concurrent is what I'm set up
> for).  Most of those issues were > 6mo ago, so maybe they've gotten
> better.  Usually a call retry would be fine.  I didn't have enough
> people on enough different carriers to see if this was an issue with a
> single provider (say ATT or Cox or whatever).
>
> didforsale *is* cheap for multi-channel support -- at least for the
> first "chunk".  They also have a formula for channel sharing if you
> purchase a large # of DIDs from them where the per-channel cost gets
> even lower.
>
> On Tue, May 10, 2011 at 8:19 AM, Sidharth Kshatriya
> <sid.kshatriya at gmail.com> wrote:
> > Have you tried didforsale.com ? Their pricing seems to be very
> competitive.
> > They seem to make the most sense to me, but their website is a little
> > outdated...
> > Any more inbount DID providers? Anybody? (I need multi channel support --
> at
> > least 5 channels)
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mayamatakeshi at gmail.com  Tue May 10 19:11:43 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 11 May 2011 00:11:43 +0900
Subject: [Freeswitch-users] Does FS supports speex with ptime=40?
In-Reply-To: <484AE924-7A82-437F-B8E1-AA38183F4ECA@freeswitch.org>
References: <BANLkTinRv5swkHVTkKPqnaT_fCtiUqP3yg@mail.gmail.com>
	<484AE924-7A82-437F-B8E1-AA38183F4ECA@freeswitch.org>
Message-ID: <BANLkTin31dND2cZG9HNcSSOpLYPFMksdbQ@mail.gmail.com>

On Thu, Apr 28, 2011 at 11:32 PM, Brian West <brian at freeswitch.org> wrote:

> Nope.
>

Brian, thanks.
But could you elaborate a little?
Is this an issue with the underlying speex lib? It is not possible or hard
to make it work with ptime=40?
If it is possible but not implemented due to lack of users, we could offer a
bounty.
(the issue with x-lite not advertising ptime=40 I think I can solve with
opensips/kamailio in front of FS, by correctlng the SDP).

regards,
takeshi


> On Apr 28, 2011, at 8:18 AM, mayamatakeshi wrote:
>
> > I am testing making calls with x-lite 4. Although it doesn't send a ptime
> in
> > its INVITE/SDP, i can see it is sending RTP using ptime=40 as I can count
> 25
> > UDP packets per second.
> > When I try to bridge this to PSTN which responds with PCMU, at the PSTN
> side
> > I hear trembling/metallic/robotic audio.
> >
> > I was suggested to set FS to use ptime=40, so I've tried setting codecs
> with
> > speex at 8000h@40i
> >
> > But it doesn't work. I get: "488 Not Acceptable Here" (Reason:
> > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION)
> >
> > So it seems only ptime=20 is supported as listed here:
> > http://wiki.freeswitch.org/wiki/Codecs#Transcodable_codecs
> >
> > Could mod_speex be changed to work with ptime=40 too?
> > Or is there anything else that i can try?
> >
> > regards,
> > takeshi
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jpatten at co.brazos.tx.us  Tue May 10 19:08:56 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Tue, 10 May 2011 15:08:56 +0000
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>,
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>,
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>,
	<8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
Message-ID: <8C8A3D4965236A42BDFF1758727F049A33BEC0@ITEX1.bc.local>

For now I'll have to use a combination of lua and PHP to accomplish this as fs_ivrd is async capable. I'll post the code somewhere when I'm done.
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten [jpatten at co.brazos.tx.us]
Sent: Tuesday, May 10, 2011 1:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference

It seems the way this could be accomplished is to allow asynchronous execution capabilities in lua. However I don't think this exists.

Is there a better way to "execute and forget" the conference app so the script can move on to the next bit of code?
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten [jpatten at co.brazos.tx.us]
Sent: Tuesday, May 10, 2011 1:21 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference

Only problem is session:streamFile doesn't make it into conference.

The only way I've found to do this so far is to conference_set_auto_outcall a loopback extension to a non-user extension that simply streams a file in a context:
session:execute("conference_set_auto_outcall","loopback/3041/custom/")

but if I want to mute the conference so only the originator is heard (a standard "overhead" style page) using:
session:execute("set", "conference_auto_outcall_flags=mute") then this is never heard because this loopback "participant" is muted as well.

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Tom C [mel0torme at gmail.com]
Sent: Tuesday, May 10, 2011 12:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference

I was thinking about this a few days ago.  Slightly different scenario: I want to stream audio into a conference every ten minutes or so.

My ideas was to do it by connecting another extension to the conference.  This extra extension would (theoretically) run a Lua script to stream the audio, and then sleep, and every few seconds it would look to see if it was the only extension still connected to the conference.

In your case, if you're doing the "mad boss" and inviting all the participants to the conference, it should be easy to add one more extension to the list of invites.  Just have the extension's script wait 10 seconds or so before streaming the audio, and then it can just hang up.

Something like this:

streamwelcomefile.lua
<code>

if ( session:ready() ) then
  session:answer();
  session:sleep(10000)  -- wait for 10 seconds for other people to join conference.
  session:streamFile("ivr/8000/ivr-welcome.wav");
  session:streamFile("ivr/8000/ivr-thank_you_for_calling.wav");
  session:hangup("NORMAL_CLEARING");
end

</code>

And the dialplan might look like this:
<code>

<extension name="StreamWelcome">
        <condition field="destination_number" expression="^000111$">
                <action application="lua" data="streamwelcomefile.lua" />
        </condition>
 </extension>

</code>


Disclaimer:  I haven't actually tested this in a conference.  Good luck!  :-)
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From jgallartm at gmail.com  Tue May 10 19:36:23 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 10 May 2011 17:36:23 +0200
Subject: [Freeswitch-users] invalid ELF header error in lua startup
	script
Message-ID: <BANLkTi=g3RUohuVFehf4RL32VbjHPWdDHg@mail.gmail.com>

Hello Tristan

CentOS 5, 64  bits

Thanks!

Javi



> ---------- Forwarded message ----------
> From: "Tristan Mah?" <t.mahe at telemaque.fr>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 10 May 2011 14:30:48 +0200
> Subject: Re: [Freeswitch-users] invalid ELF header error in lua startup
> script
>  32bit module on a 64bit system or the opposite ?
>
> Le 10/05/2011 13:09, Javier Gallart a ?crit :
>
> Hello,
>
>  we have a lua script that we want to use as startup script in order to
> start listening for events as soon as FS starts. This is the configuration:
>
>  <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/>
>    <param name="module-directory" value="/usr/lib/lua/5.1/?.lua"/>
>     <param name="module-directory" value="/usr/lib/lua/5.1/socket/?.lua"/>
>     <param name="module-directory" value="/usr/lib/lua/5.1/socket/?.so"/>
>     <param name="module-directory" value="/usr/local/share/lua/5.1/?.so"/>
>     <param name="module-directory" value="/usr/local/share/lua/5.1/?.lua"/>
>     <param name="module-directory"
> value="/usr/local/share/lua/5.1/xmlrpc/?.lua"/>
>  <param name="startup-script" value="script.lua"/>
>
>  The script first line is:
>
>  require("socket")
>
>  The file  /usr/local/share/lua/5.1/socket.lua exists. Actually when we
> execute "lua script.lua" it runs properly. But this is the freeswitch log at
> startup:
>  2011-05-10 07:00:31.281411 [ERR] mod_lua.cpp:191 error loading module
> 'socket' from file '/usr/local/share/lua/5.1/socket.lua':
>  /usr/local/share/lua/5.1/socket.lua: invalid ELF header
> stack traceback:
>  [C]: ?
>  [C]: in function 'require'
>  /usr/local/freeswitch/scripts/script.lua:1: in main chunk
>
>  Has anyone gone over the same issue? We tried with the latest git.
>
>  Thanks in advance
>
>  Javi
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
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From jcasale at activenetwerx.com  Tue May 10 19:39:38 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 10 May 2011 15:39:38 +0000
Subject: [Freeswitch-users] Problems with multiple IP-addresses
In-Reply-To: <web-58862251@mailbe04.swip.net>
References: <web-58862251@mailbe04.swip.net>
Message-ID: <assp.01119b1983.92097A6A775D5147B1078E3F15430B924C96F5@prato.activenetwerx.local>

>Is there any way to configure FS to stop trying to detect a ip-change, or a way
>to make mod_sofia ignore the ip-change event ?

I think that was just answered a few days ago, In sofia.conf.xml set:
<param name="auto-restart" value="false"/>

jlc

From anthony.minessale at gmail.com  Tue May 10 19:43:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 10 May 2011 10:43:07 -0500
Subject: [Freeswitch-users] mod_crd_sqlite entry limit and sqlite
 segfaults on triggers
In-Reply-To: <BANLkTimb7GqKX2kT3Y2oAwRJfnhuHPK9Hw@mail.gmail.com>
References: <4DB5E0F1.4050602@yx.cl> <4DB809B4.5060601@yx.cl>
	<1303921474677-6310155.post@n2.nabble.com> <4DB874FF.5060901@yx.cl>
	<BANLkTimb7GqKX2kT3Y2oAwRJfnhuHPK9Hw@mail.gmail.com>
Message-ID: <BANLkTin+t8J=3aWXwsJs4uM21rWAoJS+Qg@mail.gmail.com>

you're working in circles

our version of sqliteStrDup does check for NULL the same way the other
function does

#define sqliteStrDup(x)          (x?strdup(x):NULL)

see......?     x ? strdup(x) : NULL

if x strdup it else return NULL

sqlite3StrDup does the same thing
if( z==0 ) return 0;


The only difference is that sqlite devs use their own memory
management system that does not work for us under heavy loads so we
stick with good old malloc/free.

besides, tolerating NULL is a fail safe. the real question is what is
even trying to pass NULL to be duped, and, its actually doing exactly
what it's asked it dups NULL to NULL.





On Tue, May 10, 2011 at 8:24 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> I would argue that you do not want to check the row count on every
> single insert. ?Have a cron job that runs periodically and does the
> deletes. ?Run that job once a day. ?Doing the maintenance on each
> insert is going to kill performance *and* concurrency.
>
> On Wed, Apr 27, 2011 at 2:56 PM, Neven Boric <nboric at yx.cl> wrote:
>> Will do. What about the limit on the number of cdrs? Should I report
>> that too?
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 benkokakao at gmail.com  Tue May 10 20:56:54 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 10 May 2011 18:56:54 +0200
Subject: [Freeswitch-users] BRI-ISDN-Card Alternatives to Sangoma A500 - Do
 you have a working EuroISDN-Setup with a different card? Please comment!
Message-ID: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>

Hello!

I'm having a hard time finding a working ISDN-Card for FreeSWITCH.
I've succesfully tested the Sangoma A500 and it would be a great card
- but the card is just a litte bit too long for the mini-ITX-Hardware
we want to use. Therefore i'm desperately looking for alternatives:

- I've tested an OpenVox B100P, without success so far(See
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071872.html)
- Today i've installed an Eicon Diva BRI-2 card but so far i was not
able to get a working setup, right now i have no clue how to continue
marrying FreeTDM with the official CAPI-drivers

So if anyone has a working setup with EuroISDN in PTP- and TE-mode and
a card other than Sangoma A500, please let me know! Or if you know
there's nothing working BUT Sangoma A500, i'm happy too(Then i have a
final answer at least).

In case anyone knows how to get a Eicon/Dialogic Diva BRI-2 card to
work with FreeTDM i'd also be very thankful for a comment. I've
installed the drivers provided by Dialogic(Diva4Linux 9.50111-98) and
in an blind attempt tried to configure FreeSWITCH with libpri, so far
the card has not been recognized.
Unfortunately i have a limited knowledge about ISDN beyond vanilla
installation of drivers and getting the drivers/devices/modules to
show up in Linux(i.e. i don't really know what's going on between ftdm
and the diverse ISDN-drivers).

Best regards
Christian


From jmesquita at freeswitch.org  Tue May 10 21:18:31 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 10 May 2011 14:18:31 -0300
Subject: [Freeswitch-users] BRI-ISDN-Card Alternatives to Sangoma A500 -
 Do you have a working EuroISDN-Setup with a different card? Please comment!
In-Reply-To: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
References: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
Message-ID: <BANLkTim6xjiybXS23jwco6nTg7kt3h8ugw@mail.gmail.com>

Have you tried this:
http://www.khomp.com.br/?menu=produto&content=produtos&type=SPX&base=42

They officially support FreeSWITCH and they come from the Dialogic-like
world where all the protocols are run on the card itself so you don't need
libpri or similar. Same thing applies to CAS R1/R2, EL7, E1LC and QSIG as
the spec sheet tells us. I use this card on a daily basis as well as 80% of
the Brazilian market, so I guess it must work, right? ;-)

Contact me off list if you're interested.

Regards,
Jo?o Mesquita



On Tue, May 10, 2011 at 1:56 PM, Christian Benke <benkokakao at gmail.com>wrote:

> Hello!
>
> I'm having a hard time finding a working ISDN-Card for FreeSWITCH.
> I've succesfully tested the Sangoma A500 and it would be a great card
> - but the card is just a litte bit too long for the mini-ITX-Hardware
> we want to use. Therefore i'm desperately looking for alternatives:
>
> - I've tested an OpenVox B100P, without success so far(See
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071872.html
> )
> - Today i've installed an Eicon Diva BRI-2 card but so far i was not
> able to get a working setup, right now i have no clue how to continue
> marrying FreeTDM with the official CAPI-drivers
>
> So if anyone has a working setup with EuroISDN in PTP- and TE-mode and
> a card other than Sangoma A500, please let me know! Or if you know
> there's nothing working BUT Sangoma A500, i'm happy too(Then i have a
> final answer at least).
>
> In case anyone knows how to get a Eicon/Dialogic Diva BRI-2 card to
> work with FreeTDM i'd also be very thankful for a comment. I've
> installed the drivers provided by Dialogic(Diva4Linux 9.50111-98) and
> in an blind attempt tried to configure FreeSWITCH with libpri, so far
> the card has not been recognized.
> Unfortunately i have a limited knowledge about ISDN beyond vanilla
> installation of drivers and getting the drivers/devices/modules to
> show up in Linux(i.e. i don't really know what's going on between ftdm
> and the diverse ISDN-drivers).
>
> Best regards
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mel0torme at gmail.com  Tue May 10 21:21:30 2011
From: mel0torme at gmail.com (Tom C)
Date: Tue, 10 May 2011 10:21:30 -0700
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
Message-ID: <BANLkTi=_8_qmfRGMo6phPhkO2YcGizRSeg@mail.gmail.com>

Setting conference_set_auto_outcall_flags=mute only sets the initial state
to muted.  The attendees can still press 0 to unmute (if the key mappings
are left at their default values).  So you can use send_dtmf to unmute the
loopback extension, and then stream the audio.

E.g., expanding my code from earlier:

streamwelcomefile.lua
<code>

if ( session:ready() ) then
  session:answer();
  -- Must wait a couple seconds after connecting before sending DTMF,
  -- or it will get lost:
  session:sleep(2000)
   -- Unmute this attendee:
  session:execute("send_dtmf","0 at 200")
  session:sleep(1000)
  session:streamFile("ivr/8000/ivr-welcome.wav");
  session:streamFile("ivr/8000/ivr-thank_you_for_calling.wav");
  session:hangup("NORMAL_CLEARING");
end

</code>
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From stkn at freeswitch.org  Tue May 10 22:07:30 2011
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Tue, 10 May 2011 20:07:30 +0200
Subject: [Freeswitch-users] BRI-ISDN-Card Alternatives to Sangoma A500 -
	Do you have a working EuroISDN-Setup with a different card?
	Please comment!
In-Reply-To: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
References: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
Message-ID: <201105102007.34785.stkn@freeswitch.org>

Am Tuesday 10 May 2011 schrieb Christian Benke:
> - I've tested an OpenVox B100P, without success so far(See
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071872.html)

That was a configuration problem (missing "node"/"mode" parameter), 
mixed with missing parameter validation in libpri and not setting sane defaults
in ftmod_libpri (which is now fixed in git: http://oss.axsentis.de/gitweb/?p=freeswitch.git;a=commitdiff;h=2ac7a9de4f69 ).

> - Today i've installed an Eicon Diva BRI-2 card but so far i was not
> able to get a working setup, right now i have no clue how to continue
> marrying FreeTDM with the official CAPI-drivers

Not possible, FreeTDM has no support for CAPI.


stkn
-- 
-------------------------------------------------------------------------------
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
-------------------------------------------------------------------------------
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Email: stkn at freeswitch.org
IRC:   #freeswitch-de @ irc.freenode.net
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From avi at avimarcus.net  Tue May 10 22:42:01 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 10 May 2011 21:42:01 +0300
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTike33jWuNX72kkhzhbHiKnvjoqfTg@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
	<BANLkTi=dyU4WHz1edncYCu43v0+jQbZEXw@mail.gmail.com>
	<BANLkTin+m1gPcuJ4a-JPVyyLW-fvWDj8Rg@mail.gmail.com>
	<BANLkTike33jWuNX72kkhzhbHiKnvjoqfTg@mail.gmail.com>
Message-ID: <BANLkTin_Xb_uEm7UB3EipjPhi1xn3wm6Mg@mail.gmail.com>

DIDforsale has $3 extra channels (most places charge $10+ per channel) and
automatic 20%? free channel sharing between DIDs.
grnvoip doesn't do DIDs yet..

-Avi

On Tue, May 10, 2011 at 5:54 PM, Mike Tesliuk <mike.tesliuk at ultra.net.br>wrote:

> you can try grnvoip.com or rnktel.com , for united states and others
> country is good, for brazil is not.
>
>
> 2011/5/10 Rupa Schomaker <rupa at rupa.com>
>
>> I use didforsale for one of my trunks where I needed a cheep
>> conference bridge.  I've had issues with some people getting a reorder
>> even though the trunk isn't full (25 concurrent is what I'm set up
>> for).  Most of those issues were > 6mo ago, so maybe they've gotten
>> better.  Usually a call retry would be fine.  I didn't have enough
>> people on enough different carriers to see if this was an issue with a
>> single provider (say ATT or Cox or whatever).
>>
>> didforsale *is* cheap for multi-channel support -- at least for the
>> first "chunk".  They also have a formula for channel sharing if you
>> purchase a large # of DIDs from them where the per-channel cost gets
>> even lower.
>>
>> On Tue, May 10, 2011 at 8:19 AM, Sidharth Kshatriya
>> <sid.kshatriya at gmail.com> wrote:
>> > Have you tried didforsale.com ? Their pricing seems to be very
>> competitive.
>> > They seem to make the most sense to me, but their website is a little
>> > outdated...
>> > Any more inbount DID providers? Anybody? (I need multi channel support
>> -- at
>> > least 5 channels)
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jpatten at co.brazos.tx.us  Wed May 11 00:21:18 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Tue, 10 May 2011 20:21:18 +0000
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33BEC0@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>,
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>,
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>,
	<8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
	<8C8A3D4965236A42BDFF1758727F049A33BEC0@ITEX1.bc.local>
Message-ID: <8C8A3D4965236A42BDFF1758727F049A33C101@ITEX1.bc.local>

http://wiki.sipfoundry.org/display/sipXecs/Potential+paging+system+replacement I'm not exactly a coder...but this works pretty well. It allows users that are in a page group to set a temporary "time out" so they don't get paged for a little while if they're in a meeting or otherwise don't wish to be disturbed.

Josh Patten
Brazos County Network Engineer
979.361.4676

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh M. Patten
Sent: Tuesday, May 10, 2011 10:09 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference

For now I'll have to use a combination of lua and PHP to accomplish this as fs_ivrd is async capable. I'll post the code somewhere when I'm done.
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten [jpatten at co.brazos.tx.us]
Sent: Tuesday, May 10, 2011 1:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference
It seems the way this could be accomplished is to allow asynchronous execution capabilities in lua. However I don't think this exists.

Is there a better way to "execute and forget" the conference app so the script can move on to the next bit of code?
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten [jpatten at co.brazos.tx.us]
Sent: Tuesday, May 10, 2011 1:21 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference
Only problem is session:streamFile doesn't make it into conference.

The only way I've found to do this so far is to conference_set_auto_outcall a loopback extension to a non-user extension that simply streams a file in a context:
session:execute("conference_set_auto_outcall","loopback/3041/custom/")

but if I want to mute the conference so only the originator is heard (a standard "overhead" style page) using:
session:execute("set", "conference_auto_outcall_flags=mute") then this is never heard because this loopback "participant" is muted as well.
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Tom C [mel0torme at gmail.com]
Sent: Tuesday, May 10, 2011 12:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference
I was thinking about this a few days ago.  Slightly different scenario: I want to stream audio into a conference every ten minutes or so.

My ideas was to do it by connecting another extension to the conference.  This extra extension would (theoretically) run a Lua script to stream the audio, and then sleep, and every few seconds it would look to see if it was the only extension still connected to the conference.

In your case, if you're doing the "mad boss" and inviting all the participants to the conference, it should be easy to add one more extension to the list of invites.  Just have the extension's script wait 10 seconds or so before streaming the audio, and then it can just hang up.

Something like this:

streamwelcomefile.lua
<code>

if ( session:ready() ) then
  session:answer();
  session:sleep(10000)  -- wait for 10 seconds for other people to join conference.
  session:streamFile("ivr/8000/ivr-welcome.wav");
  session:streamFile("ivr/8000/ivr-thank_you_for_calling.wav");
  session:hangup("NORMAL_CLEARING");
end

</code>

And the dialplan might look like this:
<code>

<extension name="StreamWelcome">
        <condition field="destination_number" expression="^000111$">
                <action application="lua" data="streamwelcomefile.lua" />
        </condition>
 </extension>

</code>


Disclaimer:  I haven't actually tested this in a conference.  Good luck!  :-)
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From admin at blindi.net  Wed May 11 02:27:03 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 11 May 2011 00:27:03 +0200 (CEST)
Subject: [Freeswitch-users] Problem originate conference hangup
In-Reply-To: <BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105110023470.8977@debian>

Hi Michael, i have fixed the problem as follows:
I have append a action to go back in the room after to execute the script.

It.s work fine now.

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 admin at blindi.net  Wed May 11 02:45:34 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 11 May 2011 00:45:34 +0200 (CEST)
Subject: [Freeswitch-users] Play_and_get_digits Menuselection not working
In-Reply-To: <BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105110027460.8977@debian>

Hi all,
i have create a menuselection via play_and_get_digits.
Problem, this application ignore the digits 1 and 2.
This is my dialplanextestnion:

  <extension name="callback_only_menuselection">
  <condition field="destination_number" expression="^callback_only_sel$"/>
     <condition field="${ivrlevel}" expression="^$" break="never">
       <action application="play_and_get_digits"
       data="0 1 1 5000 #
      $${base_dir}/sounds/callback/confirm-callback-press1.alaw
     $${base_dir}/sounds/callback/invalid.alaw  nr1 [1-2]"/>

     </condition>
     <condition field="${ivrlevel}-${nr1}" expression="^1$" >
  <action application="playback"
    data="$${base_dir}/sounds/callback/callback-send.alaw"/>
  <action application="system" data="$${base_dir}/scripts/callback.sh"/>
     </condition>
     <condition field="${ivrlevel}-${nr1}" expression="^2$">
  <action application="ivr" data="callback_only_ivr"/>
     </condition>
   </extension>

I press 1 and 2 play_and_get_digits don.t execute these actions.
What is wrong?
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 david.ponzone at ipeva.fr  Wed May 11 04:27:05 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 11 May 2011 02:27:05 +0200
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
	version of Freeswitch question
In-Reply-To: <BANLkTikS_8t81GnKGgxuDyYhYJBQEKaLKw@mail.gmail.com>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
	<4DC67898.2090106@coppice.org>
	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
	<4DC732BC.9070206@coppice.org>
	<BANLkTikS_8t81GnKGgxuDyYhYJBQEKaLKw@mail.gmail.com>
Message-ID: <62A73319-BB90-4EB5-91F5-D95CED42A2A8@ipeva.fr>

Juan,

I suppose a decent server with a hw clock running at 1000Hz and a recommended OS (that would be CentOS).
Also, if you do T38 to send the faxes to an external GW, the smallest possible jitter between your FreeSWITCH and the gateway.
Also, if you do T38 for inbound faxes, check that the gateway supports T38 redundancy.

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

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

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




Le 09/05/2011 ? 09:23, Juan Antonio Iba?ez Santorum a ?crit :

> Thank you very much for your answer Steve.
> 
> Would be FS + mod_spandsp good both for incoming and outgoing faxeds?
> 
> What could be achieved to avoid media timing issues you told? 
> 
> Regards
> 
> 
> 2011/5/9 Steve Underwood <steveu at coppice.org>
> On 05/09/2011 01:16 AM, Juan Antonio Iba?ez Santorum wrote:
> > How much reliable FS + mod_spandsp is compared to other solutions
> > (open source or not)?
> The open source options are:
> 
>     Asterisk + spandsp
>     Asterisk + Hylafax + iaxmodem + spandsp
>     Freeswitch
> 
> These are all tested and proven to give well below 1% failures, even
> with quite a lot of concurrent FAX channels in use, if things are set up
> well. They can give you bad failure rates if things are not set up well.
> I believe that right now you will have less trouble achieving a reliable
> setup with Freeswitch. Going forward, most of my effort goes into making
> the Freeswitch option the most thoroughly implemented one.
> 
> The main commercial option is:
> 
>     Asterisk + Digium's commercial FAX
> 
> Of course, there are numerous other fully commercial FAXing options
> which could be used in conjunction with things like Asterisk or Freeswitch
> 
> The Digium FAX module is based on the well known Commetrex FAX engine,
> which is widely deployed, and should be capable of solid results.
> However, the module is more than just the core FAX engine, and some
> people do have serious trouble with the module. I have helped moved
> people off this, and onto Asterisk + spandsp, to improve their
> reliability. In a couple of those cases people were getting quite a lot
> of pages cut short when receiving FAXes with T.38, even though a
> wireshark log showed a perfect exchange, from which I could correctly
> decode these FAXes. The module was not reporting any errors. In a couple
> of cases strange machines were sending weird things the Digium FAX
> didn't cope with very well. I worked with these people to make sure
> spandsp did handle the weird stuff well, and we ended up with a more
> usable solution. These people told me that when they complained to
> Digium they got little help. The best was an offer of a refund. Paying
> to get some support didn't seem to work out too well for these people,
> but I guess if the support you are looking for is mostly in getting
> things configured and working on day one you might get value for money.
> 
> All these solutions require reliable signaling and reliable media
> timing, and many people have setups which cannot achieve that. Most
> people don't understand how things work, and will claim a solution
> doesn't function for spurious reasons. For example, a number of people
> say the spandsp module for Asterisk doesn't work, because they keep
> getting a 488 response. That response has nothing to do with the FAX
> engine. It is a negotiation error that occurs outside the FAX engine. If
> they fixed their configuration the error would go away. However, many
> just move on, having "proven" to themselves the solution doesn't work.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed May 11 05:48:12 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 10 May 2011 18:48:12 -0700 (PDT)
Subject: [Freeswitch-users] Major deployment of outbound FAX on latest
 version of Freeswitch question
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE658284AF10F@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
	<4DC67898.2090106@coppice.org>
	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
	<4DC732BC.9070206@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF10F@ASKSERVER.asksbs.local>
Message-ID: <1305078492903-6350327.post@n2.nabble.com>

Hi Andrew,

Probably the only thing I can say would be that the Windows port of SpanDsp
does not receive the same amount of testing as the Linux build so keep that
in mind but there should be no particular reason not to expect the same
performance(as far as I know) as other builds.

Please update the list here with any results that you experience.

Thanks
Jeff

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Major-deployment-of-outbound-FAX-on-latest-version-of-Freeswitch-question-tp6336513p6350327.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Wed May 11 05:54:42 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 10 May 2011 18:54:42 -0700 (PDT)
Subject: [Freeswitch-users] Play_and_get_digits Menuselection not working
In-Reply-To: <Pine.LNX.4.64.1105110027460.8977@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
Message-ID: <1305078882613-6350342.post@n2.nabble.com>

Please dont reply to an existing message and change the subject line as this
hijacks the thread and makes it more difficult for others to track the
thread by its subject line.

When starting a new thread always start with a new mail message!

Thanks

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-read-RTP-package-tp6007317p6350342.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sid.kshatriya at gmail.com  Wed May 11 09:32:36 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 11 May 2011 11:02:36 +0530
Subject: [Freeswitch-users] Can anybody recommend some good sip trunk
	providers?
In-Reply-To: <BANLkTin_Xb_uEm7UB3EipjPhi1xn3wm6Mg@mail.gmail.com>
References: <BANLkTi=_o53b8LMJf-9n0q_i9gR3es8xWw@mail.gmail.com>
	<1304505994974-6330365.post@n2.nabble.com>
	<BANLkTikYGv8u+fAZqrLdamQ2PVaWdHSaKA@mail.gmail.com>
	<BANLkTi=dyU4WHz1edncYCu43v0+jQbZEXw@mail.gmail.com>
	<BANLkTin+m1gPcuJ4a-JPVyyLW-fvWDj8Rg@mail.gmail.com>
	<BANLkTike33jWuNX72kkhzhbHiKnvjoqfTg@mail.gmail.com>
	<BANLkTin_Xb_uEm7UB3EipjPhi1xn3wm6Mg@mail.gmail.com>
Message-ID: <BANLkTimyw1ppOJsD4m7hg_e2ojijT7cjjg@mail.gmail.com>

Thanks for sharing that avi -- I'm greatly interested in knowing whether you
have used DIDforsale in the past and if you did, was it a good experience?

On Wed, May 11, 2011 at 12:12 AM, Avi Marcus <avi at avimarcus.net> wrote:

> DIDforsale has $3 extra channels (most places charge $10+ per channel) and
> automatic 20%? free channel sharing between DIDs.
> grnvoip doesn't do DIDs yet..
>
> -Avi
>
>
> On Tue, May 10, 2011 at 5:54 PM, Mike Tesliuk <mike.tesliuk at ultra.net.br>wrote:
>
>> you can try grnvoip.com or rnktel.com , for united states and others
>> country is good, for brazil is not.
>>
>>
>> 2011/5/10 Rupa Schomaker <rupa at rupa.com>
>>
>>> I use didforsale for one of my trunks where I needed a cheep
>>> conference bridge.  I've had issues with some people getting a reorder
>>> even though the trunk isn't full (25 concurrent is what I'm set up
>>> for).  Most of those issues were > 6mo ago, so maybe they've gotten
>>> better.  Usually a call retry would be fine.  I didn't have enough
>>> people on enough different carriers to see if this was an issue with a
>>> single provider (say ATT or Cox or whatever).
>>>
>>> didforsale *is* cheap for multi-channel support -- at least for the
>>> first "chunk".  They also have a formula for channel sharing if you
>>> purchase a large # of DIDs from them where the per-channel cost gets
>>> even lower.
>>>
>>> On Tue, May 10, 2011 at 8:19 AM, Sidharth Kshatriya
>>> <sid.kshatriya at gmail.com> wrote:
>>> > Have you tried didforsale.com ? Their pricing seems to be very
>>> competitive.
>>> > They seem to make the most sense to me, but their website is a little
>>> > outdated...
>>> > Any more inbount DID providers? Anybody? (I need multi channel support
>>> -- at
>>> > least 5 channels)
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From msc at freeswitch.org  Wed May 11 09:48:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 May 2011 22:48:07 -0700
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
	<8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
Message-ID: <BANLkTinY_te4g4CV=7capn_PTKcmLdnhfw@mail.gmail.com>

On Mon, May 9, 2011 at 11:34 PM, Josh M. Patten <jpatten at co.brazos.tx.us>wrote:

>  It seems the way this could be accomplished is to allow asynchronous
> execution capabilities in lua. However I don't think this exists.
>
> Is there a better way to "execute and forget" the conference app so the
> script can move on to the next bit of code?
>

Try using an API instead of messing with session:xxx and all that stuff:

api = freeswitch.API();
res = api:execute("conference",conf_name .. " play /path/to/file.wav";

conf_name is the name of the conference you are using. If you're not sure
what name you have then just start up a conference and then type "conference
list" at fs_cli and you'll see the name. You can use channel variables, etc.
if the name changes depending on the destination number.

-MC
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From sid.kshatriya at gmail.com  Wed May 11 10:00:36 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 11 May 2011 11:30:36 +0530
Subject: [Freeswitch-users] Freeswitch Code compromised? Hacked?
Message-ID: <BANLkTik0aaaKmLmYMOBH-tV+eJ1G1L7XYQ@mail.gmail.com>

OK. So maybe I'm being paranoid. I downloaded Freeswitch 1.0.7 from
http://latest.freeswitch.org/

<http://latest.freeswitch.org/>I get the following startup message:

FreeSWITCH Version 1.0.7 (hacked-20110509T092215Z) Started.

Now in common parlance "hacked" could mean "built". But I still* *just* *want
to know if the code wasn't compromised in any way and any cheeky hacker left
a message in "plain sight" like osama :-)

Is this startup message okay? Can it be changed to something a little bit
nicer / less alarming le.g. built-20110509T092215Z ?

Now another question: I downloaded this on May 8 from
http://latest.freeswitch.org/. There seems to be another download dated May
10. Has 1.0.7 not been released yet? Is it still being finalized?

-- 
Sidharth Kshatriya
www.sidk.info
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From steveayre at gmail.com  Wed May 11 10:45:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 11 May 2011 07:45:47 +0100
Subject: [Freeswitch-users] Freeswitch Code compromised? Hacked?
In-Reply-To: <BANLkTik0aaaKmLmYMOBH-tV+eJ1G1L7XYQ@mail.gmail.com>
References: <BANLkTik0aaaKmLmYMOBH-tV+eJ1G1L7XYQ@mail.gmail.com>
Message-ID: <97627200-0857-4BC2-9889-5B3BF884C312@gmail.com>

1.0.7 is the nightly build, more of a release candidate for 1.0.8

Any reason you don't want to build from git head? That's all 1.0.7 is, delayed by a few hours.

Steve on iPhone

On 11 May 2011, at 07:00, Sidharth Kshatriya <sid.kshatriya at gmail.com> wrote:

> OK. So maybe I'm being paranoid. I downloaded Freeswitch 1.0.7 from http://latest.freeswitch.org/
> 
> I get the following startup message:
> 
> FreeSWITCH Version 1.0.7 (hacked-20110509T092215Z) Started.
> 
> Now in common parlance "hacked" could mean "built". But I still just want to know if the code wasn't compromised in any way and any cheeky hacker left a message in "plain sight" like osama :-)
> 
> Is this startup message okay? Can it be changed to something a little bit nicer / less alarming le.g. built-20110509T092215Z ?
> 
> Now another question: I downloaded this on May 8 from http://latest.freeswitch.org/. There seems to be another download dated May 10. Has 1.0.7 not been released yet? Is it still being finalized?
> 
> -- 
> Sidharth Kshatriya
> www.sidk.info
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sadhika at gmail.com  Wed May 11 13:01:54 2011
From: sadhika at gmail.com (Sadhika Sharma)
Date: Wed, 11 May 2011 14:31:54 +0530
Subject: [Freeswitch-users] How to report bad HDLC frames?
Message-ID: <BANLkTikow9ut9=NtPR46e+Z6gF1B7PJggw@mail.gmail.com>

In freeswitch logs I can see the received data but only for those packets
which are received accurately (HDLC framing is OK.)
How can I see bad HDLC frames (if there are any) in the logs?

-- 
Sadhika
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From benkokakao at gmail.com  Wed May 11 13:51:13 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 11 May 2011 11:51:13 +0200
Subject: [Freeswitch-users] BRI-ISDN-Card Alternatives to Sangoma A500 -
 Do you have a working EuroISDN-Setup with a different card? Please comment!
In-Reply-To: <201105102007.34785.stkn@freeswitch.org>
References: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
	<201105102007.34785.stkn@freeswitch.org>
Message-ID: <BANLkTimSfJny2axaq1uD7FYBBU6fNbqUaw@mail.gmail.com>

On 10 May 2011 20:07, Stefan Knoblich <stkn at freeswitch.org> wrote:
> Am Tuesday 10 May 2011 schrieb Christian Benke:
>> - I've tested an OpenVox B100P, without success so far(See
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071872.html)
>
> That was a configuration problem (missing "node"/"mode" parameter),
> mixed with missing parameter validation in libpri and not setting sane defaults
> in ftmod_libpri (which is now fixed in git: http://oss.axsentis.de/gitweb/?p=freeswitch.git;a=commitdiff;h=2ac7a9de4f69 ).

Holy moly! Thanks Stefan, that's fantastic! Tested and works with the
OpenVox B100P! In case you ever come to Vienna, let me invite you to a
beer or a Mate at metalab(http://metalab.at) as a little gesture of
appreciation(You have no idea on how many levels this fix helps me)!

Cheers,
Christian


From lakindia89 at gmail.com  Wed May 11 15:45:53 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 11 May 2011 17:15:53 +0530
Subject: [Freeswitch-users] Primary leg hangs up when other leg rejects
	the call
In-Reply-To: <BANLkTinZ0LO3csoV0jFuRTdzXanaayuQOw@mail.gmail.com>
References: <BANLkTinZ0LO3csoV0jFuRTdzXanaayuQOw@mail.gmail.com>
Message-ID: <BANLkTimgBan-Y4YHPcGYvE0qr8aVAo5jpw@mail.gmail.com>

Hi all,
I've also check it against the latest GIT as of today and there is no change
in the behavior. Can someone please tell me why the primary leg is also got
hangup?? and how can I avoid that??


On Tue, May 10, 2011 at 6:04 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Dear all,
> I'm using Event Inbound Socket. I want to originate a call to the user and
> bridge him with different user.
> So I used the following commands from the telnet.
>
>
> api originate {exec_after_bridge_app=park,ignore_early_media=true}user/1010
> &park()
>
> Content-Type: api/response
> Content-Length: 41
>
> +OK cf9a78a2-7aff-11e0-b732-4f9b0298a6ed
>
> sendmsg cf9a78a2-7aff-11e0-b732-4f9b0298a6ed
> call-command: execute
> execute-app-name: bridge
> execute-app-arg:user/1005
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> The user 1005 rejects the call and I got the response as CALL_REJECTED.
> I expect that primary leg will not hangup and go into park mode. But it got
> hangup.
>
> Since by default hangup_after_bridge will be false, I'm not setting it
> explicitly here.
>
> FreeSwitch logs are here,
> http://pastebin.freeswitch.org/16264
>
> Can someone please tell me what is the problem I've done and how to make
> the primary leg again in park() ??
>
>
>
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From sid.kshatriya at gmail.com  Wed May 11 16:01:11 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 11 May 2011 17:31:11 +0530
Subject: [Freeswitch-users] Freeswitch 1.0.8 likely release date?
Message-ID: <BANLkTikjvj-o4pdNdRHKc6f-Znvgyw3HHQ@mail.gmail.com>

On the freeswitch users list way back in Jan 2011, it was mentioned that
1.0.8 can be expected in a month or so. Its May now :-)

Can anyone tell us approximately when 1.0.8 is likely to be released... even
a ball park?

Thanks,

Sidharth

-- 
Sidharth Kshatriya
www.sidk.info
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From jeroeneeuwes at gmail.com  Wed May 11 16:09:55 2011
From: jeroeneeuwes at gmail.com (Jeroen Eeuwes)
Date: Wed, 11 May 2011 14:09:55 +0200
Subject: [Freeswitch-users] Primary leg hangs up when other leg rejects
 the call
In-Reply-To: <BANLkTimgBan-Y4YHPcGYvE0qr8aVAo5jpw@mail.gmail.com>
References: <BANLkTinZ0LO3csoV0jFuRTdzXanaayuQOw@mail.gmail.com>
	<BANLkTimgBan-Y4YHPcGYvE0qr8aVAo5jpw@mail.gmail.com>
Message-ID: <BANLkTinJnb3An_P4CwVx_BGQvbV7manyTw@mail.gmail.com>

Hi lakshmanan

> I've also check it against the latest GIT as of today and there is no change
> in the behavior. Can someone please tell me why the primary leg is also got
> hangup?? and how can I avoid that??

I think it does a hangup because the bridge was not yet active.

Have you tried with continue on fail? See
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

Best regards,
Jeroen Eeuwes


From leonardo.bidinoto at voicetechnology.com.br  Wed May 11 17:27:51 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Wed, 11 May 2011 10:27:51 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
Message-ID: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>

Hi everyone!

I having some troubles with my FS box(CentOS  5.3 64-bit,  Xeon(R) CPU
2.66GHz with 8 cores , 8GbRAM, (git-bc19d28 2011-04-25 15-53-54 -0400) ).
Sometimes, when a user calls to my box and hangs up, the user's channel
remains on FS when i execute a "show channels" command. with "uuid_exists"
command, im receiving "false".
looking at freeswitch log, the last message for the channels is:
0eeb975f-2ae9-4f7d-9888-1478faf87d2d 2011-05-11 00:05:57.555142 [DEBUG]
switch_core_session.c:1286 Session 38 (sofia/external/2562687580 at 10.0.70.6)
Locked, Waiting on external entities
Im not receiving "Ended" State after the previous message.

did someone have seen this issue? or know a way to remove from freeswitch?

Thanks all.

-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From jpatten at co.brazos.tx.us  Wed May 11 18:23:57 2011
From: jpatten at co.brazos.tx.us (Josh M. Patten)
Date: Wed, 11 May 2011 14:23:57 +0000
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <BANLkTinY_te4g4CV=7capn_PTKcmLdnhfw@mail.gmail.com>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
	<8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
	<BANLkTinY_te4g4CV=7capn_PTKcmLdnhfw@mail.gmail.com>
Message-ID: <8C8A3D4965236A42BDFF1758727F049A33C7F1@ITEX1.bc.local>

If you look at my original post you'll see I tried to do that with the same results:

session:execute("conference", argv[1] .. "@default")
api = freeswitch.API()
confplay = api:execute("conference", argv[1] .. " play /usr/local/freeswitch/sounds/tones/norstar.wav ")

What is happening in Lua is because it's not a threadable language you can't do an "execute and forget" on a particular command. So when the conference is run Lua will stop executing at that point.

Josh Patten
Brazos County Network Engineer
979.361.4676

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, May 11, 2011 12:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lua play file into conference


On Mon, May 9, 2011 at 11:34 PM, Josh M. Patten <jpatten at co.brazos.tx.us<mailto:jpatten at co.brazos.tx.us>> wrote:
It seems the way this could be accomplished is to allow asynchronous execution capabilities in lua. However I don't think this exists.

Is there a better way to "execute and forget" the conference app so the script can move on to the next bit of code?

Try using an API instead of messing with session:xxx and all that stuff:

api = freeswitch.API();
res = api:execute("conference",conf_name .. " play /path/to/file.wav";

conf_name is the name of the conference you are using. If you're not sure what name you have then just start up a conference and then type "conference list" at fs_cli and you'll see the name. You can use channel variables, etc. if the name changes depending on the destination number.

-MC
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From testa at voicetechnology.com.br  Wed May 11 18:51:11 2011
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Wed, 11 May 2011 11:51:11 -0300
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A33C7F1@ITEX1.bc.local>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
	<8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
	<BANLkTinY_te4g4CV=7capn_PTKcmLdnhfw@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33C7F1@ITEX1.bc.local>
Message-ID: <BANLkTinqpaMwr7adfj4KUMTrKZBDXZzWOQ@mail.gmail.com>

When you api:execute('conference'...) the function blocks. Then you have to
stream from outside, possibly using an inbound socket application.
Maybe alternatively you can call
http://wiki.freeswitch.org/wiki/Mod_commands#sched_api just before put
someone on the conference.

On Wed, May 11, 2011 at 11:23 AM, Josh M. Patten <jpatten at co.brazos.tx.us>wrote:

>  If you look at my original post you?ll see I tried to do that with the
> same results:
>
>
>
> session:execute("conference", argv[1] .. "@default")
>
> api = freeswitch.API()
>
> confplay = api:execute("conference", argv[1] .. " play
> /usr/local/freeswitch/sounds/tones/norstar.wav ")
>
>
>
> What is happening in Lua is because it?s not a threadable language you
> can?t do an ?execute and forget? on a particular command. So when the
> conference is run Lua will stop executing at that point.
>
>
>
> Josh Patten
>
> Brazos County Network Engineer
>
> 979.361.4676
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, May 11, 2011 12:48 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_lua play file into conference
>
>
>
>
>
> On Mon, May 9, 2011 at 11:34 PM, Josh M. Patten <jpatten at co.brazos.tx.us>
> wrote:
>
> It seems the way this could be accomplished is to allow asynchronous
> execution capabilities in lua. However I don't think this exists.
>
> Is there a better way to "execute and forget" the conference app so the
> script can move on to the next bit of code?
>
>
>
> Try using an API instead of messing with session:xxx and all that stuff:
>
>
>
> api = freeswitch.API();
>
> res = api:execute("conference",conf_name .. " play /path/to/file.wav";
>
>
>
> conf_name is the name of the conference you are using. If you're not sure
> what name you have then just start up a conference and then type "conference
> list" at fs_cli and you'll see the name. You can use channel variables, etc.
> if the name changes depending on the destination number.
>
>
>
> -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 msc at freeswitch.org  Wed May 11 19:14:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 11 May 2011 08:14:26 -0700
Subject: [Freeswitch-users] mod_lua play file into conference
In-Reply-To: <BANLkTinqpaMwr7adfj4KUMTrKZBDXZzWOQ@mail.gmail.com>
References: <8C8A3D4965236A42BDFF1758727F049A33B6E0@ITEX1.bc.local>
	<BANLkTimEcUDURZv7gA3rhMQYEsAx+Kv-Tg@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33B967@ITEX1.bc.local>
	<BANLkTi=SwNkBq3fX91OfOXYXVDfs3dgpQQ@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33BBB7@ITEX1.bc.local>
	<8C8A3D4965236A42BDFF1758727F049A33BBF9@ITEX1.bc.local>
	<BANLkTinY_te4g4CV=7capn_PTKcmLdnhfw@mail.gmail.com>
	<8C8A3D4965236A42BDFF1758727F049A33C7F1@ITEX1.bc.local>
	<BANLkTinqpaMwr7adfj4KUMTrKZBDXZzWOQ@mail.gmail.com>
Message-ID: <BANLkTi=_k+6yj34wYq1h2V4OkfVkd=AsQQ@mail.gmail.com>

On Wed, May 11, 2011 at 7:51 AM, Fernando Testa <
testa at voicetechnology.com.br> wrote:

> When you api:execute('conference'...) the function blocks. Then you have to
> stream from outside, possibly using an inbound socket application.
> Maybe alternatively you can call
> http://wiki.freeswitch.org/wiki/Mod_commands#sched_api just before put
> someone on the conference.
>
Try this:

api = freeswitch.API()

confplay = api:execute("bgapi", "conference " .. argv[1] .. " play
/usr/local/freeswitch/sounds/tones/norstar.wav ")

Let us know if that works...

-MC
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From msc at freeswitch.org  Wed May 11 19:17:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 11 May 2011 08:17:15 -0700
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <BANLkTimP1iPV552J66TecEXTen75sQLLOg@mail.gmail.com>
References: <4DC84196.9050905@newpace.ca>
	<BANLkTimtTJ5QHofCaC0_ExXDexefjj7OqA@mail.gmail.com>
	<BANLkTi=vPrN84Yjy_BOOooGhc4EB_PMvWQ@mail.gmail.com>
	<BANLkTimP1iPV552J66TecEXTen75sQLLOg@mail.gmail.com>
Message-ID: <BANLkTi=hXEACMYSdw604HBSeHoGf0hqn8Q@mail.gmail.com>

You might experience a small delay depending on your network resources.
However, this is essentially the way MRCP works, so I doubt anyone would
necessarily call it a bad idea. Best thing to do would be to write a proof
of concept test.

-MC

On Tue, May 10, 2011 at 3:48 AM, Marc de Corny <marcdecorny at gmail.com>wrote:

> will the fact that the file is on a remote server increase delay in
> starting the replay of the audio files?
>
> I setup a remote directory on remote server and played them that way. any
> reason why this would be a bad idea?
>
>
> On Mon, May 9, 2011 at 10:02 PM, Brad Mina <brad at tritelcomm.com> wrote:
>
>> You could setup an internal http server with a directory called 'audio' (
>> http://server.ip/audio/), then use mod_shout as Avi suggested like this:
>> <action application="shout" data="http://server.ip/audio/file.mp3"/>
>>
>>
>> On Mon, May 9, 2011 at 12:57 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> At least one potion is to use mod_shout to stream them from a server.
>>> Do note that this adds another point of failure to your setup...
>>>
>>> -Avi
>>>
>>>
>>> On Mon, May 9, 2011 at 10:33 PM, Adam Kelloway <adam.kelloway at newpace.ca
>>> > wrote:
>>>
>>>> If I did not wish to store audio files on a FreeSWITCH host, what are my
>>>> options for being able to retrieve a remote file and playing it?
>>>> My understanding is that the playback application can only play a local
>>>> file, is that correct?
>>>> I did notice that there is a shell_stream module, which you could
>>>> potentially use to retrieve a file (say, via HTTP), and provide the
>>>> audio data as the output bash script output. Is there a way to do this
>>>> or something similar without having to actually load another
>>>> program/script?
>>>>
>>>> Thanks,
>>>>
>>>> Adam Kelloway
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed May 11 19:21:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 11 May 2011 08:21:02 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTik2xP+7h56TJZYV5oFD8wx6wk+cJg@mail.gmail.com>

Hello gang!

We have a light agenda again today, however I still want everyone to call in
so that we can discuss a few topics. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_11

Be sure to add any items that you want to discuss today.

-Michael
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From yehavi.bourvine at gmail.com  Wed May 11 20:00:49 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 11 May 2011 19:00:49 +0300
Subject: [Freeswitch-users] What debugs to enable in order to trace
	tone_detect problems
Message-ID: <BANLkTimcwvTewFNecpqmYDyUd1R6TRWZBA@mail.gmail.com>

Hello,

  I am trying a new AudioCodes mediant-1000 PRI gateway (instead of a
cisco gateway). I have a problem with tone_detect: I enable it with
execute_on_answer and it works only if the call gets to voicemail. If I
answer I hear the first beep and that's all. The call is left connected, but
it does not activate the application set by tone_detect.

what debugs shall I enable in order to see what tone_detect is doing?

                    Thanks! __Yehavi:
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From testa at voicetechnology.com.br  Wed May 11 21:05:27 2011
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Wed, 11 May 2011 14:05:27 -0300
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <4DC84196.9050905@newpace.ca>
References: <4DC84196.9050905@newpace.ca>
Message-ID: <BANLkTim3-V-hc+fn6m-WMg2jUWbxp=Yjog@mail.gmail.com>

If your files are on a remote server you can mount it locally using NFS or
something else.

On Mon, May 9, 2011 at 4:33 PM, Adam Kelloway <adam.kelloway at newpace.ca>wrote:

> If I did not wish to store audio files on a FreeSWITCH host, what are my
> options for being able to retrieve a remote file and playing it?
> My understanding is that the playback application can only play a local
> file, is that correct?
> I did notice that there is a shell_stream module, which you could
> potentially use to retrieve a file (say, via HTTP), and provide the
> audio data as the output bash script output. Is there a way to do this
> or something similar without having to actually load another
> program/script?
>
> Thanks,
>
> Adam Kelloway
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From manavid at gmail.com  Wed May 11 21:23:24 2011
From: manavid at gmail.com (Moe Navid)
Date: Wed, 11 May 2011 10:23:24 -0700
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <BANLkTim3-V-hc+fn6m-WMg2jUWbxp=Yjog@mail.gmail.com>
References: <4DC84196.9050905@newpace.ca>
	<BANLkTim3-V-hc+fn6m-WMg2jUWbxp=Yjog@mail.gmail.com>
Message-ID: <78C4A3C1-0CDD-4BA5-BC9E-165AE305C7AF@gmail.com>

We had some performance issues with NFS, we switched to MooseFS for voice prompts as well as recordings with very good results.

On May 11, 2011, at 10:05 AM, Fernando Testa wrote:

> If your files are on a remote server you can mount it locally using NFS or something else.
> 
> On Mon, May 9, 2011 at 4:33 PM, Adam Kelloway <adam.kelloway at newpace.ca> wrote:
> If I did not wish to store audio files on a FreeSWITCH host, what are my
> options for being able to retrieve a remote file and playing it?
> My understanding is that the playback application can only play a local
> file, is that correct?
> I did notice that there is a shell_stream module, which you could
> potentially use to retrieve a file (say, via HTTP), and provide the
> audio data as the output bash script output. Is there a way to do this
> or something similar without having to actually load another program/script?
> 
> Thanks,
> 
> Adam Kelloway
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ira at connectmevoice.com  Wed May 11 22:25:59 2011
From: ira at connectmevoice.com (Ira Tessler)
Date: Wed, 11 May 2011 14:25:59 -0400
Subject: [Freeswitch-users] Receiving calls from an external IVR
In-Reply-To: <BANLkTimgWJGut4YU5bxna+pz2VC0C77ySA@mail.gmail.com>
References: <9b768341594a677779ca0c08b51b7468@mail.gmail.com>
	<BANLkTimgWJGut4YU5bxna+pz2VC0C77ySA@mail.gmail.com>
Message-ID: <c729f07145d58daaa7da6bb270bf21d9@mail.gmail.com>

Can you provide me with a simple example of a FS dialplan that will receive
a call from 101 at domain and route the call to a context?



Ira Tessler

ConnectMe

(800) 743-1208



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
*Sent:* Monday, May 09, 2011 1:34 PM
*To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] Receiving calls from an external IVR



Hi Ira, Avi Marcus here.



http://wiki.freeswitch.org/wiki/Mod_directory might come in useful here...
to let people search though the extensions for the person to call.

For IVR calling through, on your Dialogic IVR I'd imagine you can tell it to
call a SIP. Simply have 101 in the IVR route to freeswitch's box @ the
appropriate domain.

In FreeSWITCH, you can easily send any number you want to be bridged to
another SIP address, e.g. your current IVR.



-Avi Marcus

FreeSWITCH Consulting

0330-010-5060 (UK)



On Mon, May 9, 2011 at 8:07 PM, Ira Tessler <ira at connectmevoice.com> wrote:

I would like to integrate Freeswitch into my existing IVR platform. For now,
I would like to use it to register VoIP phones, handle outbound calls and
receive inbound calls from my IVR.  We are a service provider, so this would
be a multi-tenant configuration. Each one of our customers would be set up
in their own Freeswitch context. I would set up the same extensions for a
customer in our IVR and for Freeswitch. When someone dials ext 101 on our
IVR, I would like the IVR to make a call to Freeswitch and have ext 101 ring
for the customer?s context. My IVR would pass in the callerid of the inbound
caller and I would like that number to display on the VoIP phone as the
callerid and name.



Furthermore, when someone at ext 101 dials ext 102, I need Freesswitch to
make a call to the IVR.



Is this possible? Are they any examples out there? How can I get started? I
am a newbie to Freeswitch.



Thanks,



Ira Tessler




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From jonas_e at swipnet.se  Wed May 11 00:28:28 2011
From: jonas_e at swipnet.se (Jonas Ericsson)
Date: Tue, 10 May 2011 22:28:28 +0200
Subject: [Freeswitch-users] Problems with multiple IP-addresses
In-Reply-To: <assp.01119b1983.92097A6A775D5147B1078E3F15430B924C96F5@prato.activenetwerx.local>
References: <web-58862251@mailbe04.swip.net>
	<assp.01119b1983.92097A6A775D5147B1078E3F15430B924C96F5@prato.activenetwerx.local>
Message-ID: <4DC99FEC.8060403@swipnet.se>

Joseph, thanks for the reply!

I did read about the  auto-restart parameter but missed that it should 
be globally set in sofia.conf.xml. I put it the profile file and this 
also created a log entry like this which made me think it was taking effect:
2011-05-10 16:48:20.859167 [DEBUG] sofia.c:3044 auto-restart [false]

But after setting it in the right place everything is working again!

Thanks again!
/Jonas

>
> I think that was just answered a few days ago, In sofia.conf.xml set:
> <param name="auto-restart" value="false"/>
>
> jlc
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From michel.freiha at splendor.net  Wed May 11 23:29:05 2011
From: michel.freiha at splendor.net (Michel Freiha)
Date: Wed, 11 May 2011 22:29:05 +0300
Subject: [Freeswitch-users] database authentication
Message-ID: <BANLkTikHZD0WDk9bv6Bfn4KNRxYFCAb4TQ@mail.gmail.com>

Dear All,

I have a question that I'm not be able to find an answer or I'm a bit
confused about listed details regarding this topic...As you know, OpenSips
or asterisk has real time authentication which mean all endpoints
credentials are defind into a database and when a user with a specific
extension tries to register to registrar server, this server send an
authentication to database in order to accept registration or not..

Is this scenario doable with freeswitch or not?

Regards
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From admin at blindi.net  Thu May 12 01:31:17 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 11 May 2011 23:31:17 +0200 (CEST)
Subject: [Freeswitch-users] Problem menuselection not working with
	Play_and_get_digits Menuselection
In-Reply-To: <Pine.LNX.4.64.1105110027460.8977@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
Message-ID: <Pine.LNX.4.64.1105112328540.5984@debian>

  Hi all,
  i have create a menuselection via play_and_get_digits.
  Problem, this application ignore the digits 1 and 2.
  This is my dialplanextestnion:

  <extension name="callback_only_menuselection">
  <condition field="destination_number" expression="^callback_only_sel$"/>
     <condition field="${ivrlevel}" expression="^$" break="never">
       <action application="play_and_get_digits"
       data="0 1 1 5000 #
      $${base_dir}/sounds/callback/confirm-callback-press1.alaw
     $${base_dir}/sounds/callback/invalid.alaw  nr1 [1-2]"/>

     </condition>
     <condition field="${ivrlevel}-${nr1}" expression="^1$" >
  <action application="playback"
    data="$${base_dir}/sounds/callback/callback-send.alaw"/>
  <action application="system" data="$${base_dir}/scripts/callback.sh"/>
     </condition>
     <condition field="${ivrlevel}-${nr1}" expression="^2$">
  <action application="ivr" data="callback_only_ivr"/>
     </condition>
   </extension>

  I press 1 and 2 play_and_get_digits don.t execute these actions.
  What is wrong?
  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 shamun.toha at gmail.com  Thu May 12 01:39:24 2011
From: shamun.toha at gmail.com (Shamun)
Date: Wed, 11 May 2011 23:39:24 +0200
Subject: [Freeswitch-users] FreeSwitch video is impossible to do MCU and
 H.239 in H.323, with SIP still there is no BFCP
Message-ID: <BANLkTinGV4goi+V2A-+u0vF4H7jEiqJc3A@mail.gmail.com>

Hello Team,

FreeSWITCH does not work for Video. To do Video with FreeSWITCH, Its simply
a great nightmare. None of our equipment is compatible with FreeSWITCH we
have tried mod_h323/mod_opal and regular SIP from FreeSWITCH (none of them
works with h.323).

- Is it possible to do FreeSWITCH IVVR (Flash or MPEG4 or JPEG) ?
- Will FreeSWITCH support H.323 (and its related family such as H.460,
H.239, etc etc), there are lot of staff just does gets compatible with SIP
(specially our hardwares, i believe same with others).
- Will FreeSWITCH allow following equipment supports + more other equipments
who really depends on mod_h323/mod_opal (SIP has less hope with hardware's
compatibility)

Tandberg Edge 85 MXP
Tandberg C90
VCON vPoint HD 10.0
Radvision


Thanks & Regards
Shamun
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From admin at blindi.net  Thu May 12 01:45:50 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 11 May 2011 23:45:50 +0200 (CEST)
Subject: [Freeswitch-users] Excluding phonenumbers from existing matches?
In-Reply-To: <Pine.LNX.4.64.1105110027460.8977@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
Message-ID: <Pine.LNX.4.64.1105112331320.5984@debian>

Hi all, i have create a inbound dialingrestriction for the german 
landline numbers
my expression:
  ^(0(?:2\d{3}|3[0-1,3-9]\d{2}|[4-6]\d{3}|7[1-9]\d{2}|70[1-9]\d{1}|
  8\d{3}|9[1-9]\d{2}|90[1-9]\d{1})\d*)$

The problem: i cant.t exclude (blacklist) a number for examle:
03066664
0800330x
or so on.
I have many calls from callcenters. i must block numbers and ranges.
I don.t find a reluar expression for allow al and deny limited numbers.
Can your help 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 devel at omninet.eu  Thu May 12 01:57:56 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Thu, 12 May 2011 00:57:56 +0300
Subject: [Freeswitch-users] database authentication
In-Reply-To: <BANLkTikHZD0WDk9bv6Bfn4KNRxYFCAb4TQ@mail.gmail.com>
References: <BANLkTikHZD0WDk9bv6Bfn4KNRxYFCAb4TQ@mail.gmail.com>
Message-ID: <D7977563DB4E41538F41181D5E4B40B6@omni1.local>

Hello Michel,

 

I would say this is one of the main features that I would expect from this
kind of system.

You might be able to find different approaches with FS.

 

Look here: http://wiki.freeswitch.org/wiki/Mod_xml_curl

Basically you need a database and an http server (like Apache). FS posts a
request to the url that you set in xml_curl.conf.xml whenever it needs to
authenticate a client.

Your webpage should return the results in XML after querying the database.
It works fine and don't forget that you are able to "play" with the database
during these requests as you like.

 

It is not as difficult at it sounds.

 

Regards

Anestis

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michel
Freiha
Sent: Wednesday, May 11, 2011 10:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] database authentication

 

Dear All,

I have a question that I'm not be able to find an answer or I'm a bit
confused about listed details regarding this topic...As you know, OpenSips
or asterisk has real time authentication which mean all endpoints
credentials are defind into a database and when a user with a specific
extension tries to register to registrar server, this server send an
authentication to database in order to accept registration or not..

Is this scenario doable with freeswitch or not?

Regards

 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

 

The message was checked by ESET NOD32 Antivirus.

 

http://www.eset.com

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From brad at tritelcomm.com  Thu May 12 02:20:15 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Wed, 11 May 2011 15:20:15 -0700
Subject: [Freeswitch-users] Receiving calls from an external IVR
In-Reply-To: <c729f07145d58daaa7da6bb270bf21d9@mail.gmail.com>
References: <9b768341594a677779ca0c08b51b7468@mail.gmail.com>
	<BANLkTimgWJGut4YU5bxna+pz2VC0C77ySA@mail.gmail.com>
	<c729f07145d58daaa7da6bb270bf21d9@mail.gmail.com>
Message-ID: <BANLkTim7H__7i3WcKX6E3Bxgc=HLE_hE2w@mail.gmail.com>

Ira, what you're looking for is multi-tenancy. All of which can be explained
further here:
http://wiki.freeswitch.org/wiki/Multi-tenant

On Wed, May 11, 2011 at 11:25 AM, Ira Tessler <ira at connectmevoice.com>wrote:

> Can you provide me with a simple example of a FS dialplan that will receive
> a call from 101 at domain and route the call to a context?
>
>
>
> Ira Tessler
>
> ConnectMe
>
> (800) 743-1208
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Monday, May 09, 2011 1:34 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Receiving calls from an external IVR
>
>
>
> Hi Ira, Avi Marcus here.
>
>
>
> http://wiki.freeswitch.org/wiki/Mod_directory might come in useful here...
> to let people search though the extensions for the person to call.
>
> For IVR calling through, on your Dialogic IVR I'd imagine you can tell it
> to call a SIP. Simply have 101 in the IVR route to freeswitch's box @ the
> appropriate domain.
>
> In FreeSWITCH, you can easily send any number you want to be bridged to
> another SIP address, e.g. your current IVR.
>
>
>
> -Avi Marcus
>
> FreeSWITCH Consulting
>
> 0330-010-5060 (UK)
>
>
>
> On Mon, May 9, 2011 at 8:07 PM, Ira Tessler <ira at connectmevoice.com>
> wrote:
>
> I would like to integrate Freeswitch into my existing IVR platform. For
> now, I would like to use it to register VoIP phones, handle outbound calls
> and receive inbound calls from my IVR.  We are a service provider, so this
> would be a multi-tenant configuration. Each one of our customers would be
> set up in their own Freeswitch context. I would set up the same extensions
> for a customer in our IVR and for Freeswitch. When someone dials ext 101 on
> our IVR, I would like the IVR to make a call to Freeswitch and have ext 101
> ring for the customer?s context. My IVR would pass in the callerid of the
> inbound caller and I would like that number to display on the VoIP phone as
> the callerid and name.
>
>
>
> Furthermore, when someone at ext 101 dials ext 102, I need Freesswitch to
> make a call to the IVR.
>
>
>
> Is this possible? Are they any examples out there? How can I get started? I
> am a newbie to Freeswitch.
>
>
>
> Thanks,
>
>
>
> Ira Tessler
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grsingh750 at gmail.com  Thu May 12 03:18:25 2011
From: grsingh750 at gmail.com (guru singh)
Date: Thu, 12 May 2011 04:48:25 +0530
Subject: [Freeswitch-users] wanpipe fails to compile on arch64
Message-ID: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>

Hi,

I am trying to setup an arch box for FS. I cant install wanpipe.
make freetdm fails with this error http://pastebin.freeswitch.org/16273
uname -r 2.6.38-ARCH
gcc version 4.6.0 20110429 (prerelease) (GCC)

Could with a working arch box suggest a solution to this

Thanks
guru


From ckmonkey158 at yahoo.com  Thu May 12 05:54:28 2011
From: ckmonkey158 at yahoo.com (Chris Monkey)
Date: Wed, 11 May 2011 18:54:28 -0700 (PDT)
Subject: [Freeswitch-users] Skypopen: start_skype_clients script
	permissions
In-Reply-To: <BANLkTi=ZXBwHZTr112KjsiRyCU8dLrRbUg@mail.gmail.com>
Message-ID: <750685.42888.qm@web59408.mail.ac4.yahoo.com>

--- On Sun, 5/8/11, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
Can you double check if you got all permission right? Also, the
permissions to reach that? configuration directory and the /dev/dsp.

Which OS-distro are you using?
Thanks Giovanni,



Yup, I lost track of permissions on /dev/dsp somewhere along the way and
 didn't actually have rw access for my skype user. Made a udev rule to 
keep that straight.



Thanks for your help though,



Chris

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From fieldpeak at gmail.com  Thu May 12 07:11:41 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 12 May 2011 11:11:41 +0800
Subject: [Freeswitch-users] call somebody into conference after conference
	established
Message-ID: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>

Hi Gurus,

i dial 666 and enter a conference, then i need call some body's phone number
to join him into this conference...
is there anyone can advise how can realize this scenario?

thanks.

Regards,
Charles
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From admin at blindi.net  Thu May 12 07:58:01 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 12 May 2011 05:58:01 +0200 (CEST)
Subject: [Freeswitch-users] question how to cut a string from a variable in
	fs?
In-Reply-To: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
Message-ID: <Pine.LNX.4.64.1105120530140.9184@debian>

Hi all,
I don.t find a funtion to cut a variable string in fs.
Support Fs a funtion for example:
cut varname, delimiter, fieldspec?
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 admin at blindi.net  Thu May 12 08:43:32 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 12 May 2011 06:43:32 +0200 (CEST)
Subject: [Freeswitch-users] question does freeswitch support multipe group
	confirm keys?
In-Reply-To: <Pine.LNX.4.64.1105120530140.9184@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
Message-ID: <Pine.LNX.4.64.1105120636470.9752@debian>

Hi all,
i found in the fs-wiki only confirmation key, only one action.
Can i make a multiple selection? for example:
Press 1 to accept the call. Press 2 to forward to voicemail. Press 3 to 
transfer to cellphone, and so on?
This is very nice for a callscreeningconstruct
for examle: press 1 to eccept or 2 to blacklist the caller.

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 lakindia89 at gmail.com  Thu May 12 08:46:40 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 12 May 2011 10:16:40 +0530
Subject: [Freeswitch-users] Primary leg hangs up when other leg rejects
 the call
In-Reply-To: <BANLkTinJnb3An_P4CwVx_BGQvbV7manyTw@mail.gmail.com>
References: <BANLkTinZ0LO3csoV0jFuRTdzXanaayuQOw@mail.gmail.com>
	<BANLkTimgBan-Y4YHPcGYvE0qr8aVAo5jpw@mail.gmail.com>
	<BANLkTinJnb3An_P4CwVx_BGQvbV7manyTw@mail.gmail.com>
Message-ID: <BANLkTin7tUa99+=C3C1asTJvCFcd3WU=Cw@mail.gmail.com>

Hi Jeroen Eeuwes,

huh... I forget to try this variable...

Thanks for your reply. Once I used "continue_on_fail=true" on the originate
dial string, it works.

thank u very much..


On Wed, May 11, 2011 at 5:39 PM, Jeroen Eeuwes <jeroeneeuwes at gmail.com>wrote:

> Hi lakshmanan
>
> > I've also check it against the latest GIT as of today and there is no
> change
> > in the behavior. Can someone please tell me why the primary leg is also
> got
> > hangup?? and how can I avoid that??
>
> I think it does a hangup because the bridge was not yet active.
>
> Have you tried with continue on fail? See
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
> Best regards,
> Jeroen Eeuwes
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lakindia89 at gmail.com  Thu May 12 08:54:06 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 12 May 2011 10:24:06 +0530
Subject: [Freeswitch-users] question does freeswitch support multipe
 group confirm keys?
In-Reply-To: <Pine.LNX.4.64.1105120636470.9752@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<Pine.LNX.4.64.1105120636470.9752@debian>
Message-ID: <BANLkTin3DVFTLkEgfG+3ccPhZ-irZhFoLA@mail.gmail.com>

Hi,

As far as I know, you can use group_confirm_key=exec and
group_confirm_file=perl <your script> and do what ever you want in that
script. I'm using it to do answer confirmation and also to play some
messages before bridging the call.




On Thu, May 12, 2011 at 10:13 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi all,
> i found in the fs-wiki only confirmation key, only one action.
> Can i make a multiple selection? for example:
> Press 1 to accept the call. Press 2 to forward to voicemail. Press 3 to
> transfer to cellphone, and so on?
> This is very nice for a callscreeningconstruct
> for examle: press 1 to eccept or 2 to blacklist the caller.
>
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Thu May 12 10:56:11 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 12 May 2011 08:56:11 +0200
Subject: [Freeswitch-users] Excluding phonenumbers from existing matches?
In-Reply-To: <Pine.LNX.4.64.1105112331320.5984@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
	<Pine.LNX.4.64.1105112331320.5984@debian>
Message-ID: <3C27A8D0-EF7E-485C-B3EB-892F1ACB3A99@ipeva.fr>

Thomas,

Sure, you can't find a regexp to match a list of random numbers, as the whole point of a regexp is to match numbers which have some common part(s).
And also, the RegExp implementation in FreeSWITCH is not as complete as in the lib (named placeholders for example would be nice).

I think in your situation, you should go for the simplest/easiest.
Just add a condition before the one with your regexp, with break on-true:

<condition field="caller_id_number" expression="^03066664|0800330\d$" break="on-true">
 <action application="hangup"/>
</condition>
<condition field="caller_id_number" expression="^your regexp from hell$">
etc...

If you need to manage this blacklist from a DB, I would do something different:

<condition field="caller_id_number" expression="^your regexp from hell$">
 <action application="lua" data="validate-callerid.lua"/>
 etc....
</condition>

validate-callerid.lua is a rather easy to write LUA script, where you check the caller_id_number variable against your DB (one SQL SELECT) and if it matches, you can hangup or redirect the call to a message saying "Due to the past activity from your number, your calls to our system have been filtered".
If it doesn't match, you do nothing, and the dialplan will go to the next action, where your "bridge" is, probably.

You may rather call this LUA from a previous extension, quite early in the dialplan, if you want to drop the call ASAP and globally for all your platform.

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

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

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




Le 11/05/2011 ? 23:45, Thomas Hoellriegel a ?crit :

> Hi all, i have create a inbound dialingrestriction for the german landline numbers
> my expression:
> ^(0(?:2\d{3}|3[0-1,3-9]\d{2}|[4-6]\d{3}|7[1-9]\d{2}|70[1-9]\d{1}|
> 8\d{3}|9[1-9]\d{2}|90[1-9]\d{1})\d*)$
> 
> The problem: i cant.t exclude (blacklist) a number for examle:
> 03066664
> 0800330x
> or so on.
> I have many calls from callcenters. i must block numbers and ranges.
> I don.t find a reluar expression for allow al and deny limited numbers.
> Can your help 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
> _______________________________________________
> 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 anton.vazir at gmail.com  Thu May 12 12:36:09 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 12 May 2011 13:36:09 +0500
Subject: [Freeswitch-users] mod_say in different codecs
In-Reply-To: <AANLkTineyBkZ-e_d9vMzCnOHoy2E5aMWU3KeT9Rdb=8b@mail.gmail.com>
References: <AANLkTikb_8mv+L1PSqRk-d2QMi1sNSJN-6_5WFmWDrKW@mail.gmail.com>
	<375B3BB7-DC39-4C80-A32B-DF1941904291@gmail.com>
	<AANLkTik4b1-iHrdA8cOtzvb6797ZpoYWRA6CyThru6vz@mail.gmail.com>
	<AANLkTineyBkZ-e_d9vMzCnOHoy2E5aMWU3KeT9Rdb=8b@mail.gmail.com>
Message-ID: <BANLkTim4WfKJAQqZ6-BYP5tKNC4NjCvbFw@mail.gmail.com>

Looks there is no support for native encoded files yet...

2011/3/10 David Notivol <dnotivol at gmail.com>:
> Hi all,
> Does anyone know how to have mod_say working using mod_native_file for the
> pronounced numbers? It seems mod_say has the .wav extension included in the
> source code...
>
> Thanks.
> --
> David.
>
> 2011/3/3 David Notivol <dnotivol at gmail.com>
>>
>> Thanks Steve for your prompt reply.
>> Yes, mod_native_file does suit my needs for regular audio files; but I
>> think it doesn't for numbers. Actually, I'm using say to pronounce numbers,
>> but it seems mod_say forces FS to use the wav files.
>> I was having a look at the source code for the mod_say_en, and I removed
>> some of the ".wav" strings for the digits and recompiled it; and ?then the
>> native_file was triggered and played the .PCMU file (or any other needed
>> codec) instead of the .wav file.
>> But I'm not sure this is the way to proceed, or if that can get to other
>> problems...
>> Is there any way of having mod_say not forcing to use always .wav files
>> and relying on mod_native_file ?
>> --
>> David
>>
>> 2011/3/3 Steven Ayre <steveayre at gmail.com>
>>>
>>> Look at mod_nativefile - does that suit your needs?
>>>
>>> Steve on iPhone
>>>
>>> On 3 Mar 2011, at 12:19, David Notivol <dnotivol at gmail.com> wrote:
>>>
>>> > Hi all,
>>> >
>>> > I'm trying to setup an IVR server, and I'm using the session:say
>>> > function from a LUA script.
>>> >
>>> > My question is if there's any way of having mod_say playing audios
>>> > different than audio files; I mean audios encoded in G729, G711ulaw,
>>> > G711alaw, etc. to avoid having FS making transcoding every time I run a say
>>> > command.
>>> >
>>> > Checking the folders tree for the sounds, I can see the place for the
>>> > different languages, voices, and wav qualities (8k, 16k...) is clearly
>>> > specified; but is it a way to place files encoded in different codecs?
>>> >
>>> > Thanks in advance.
>>> >
>>> > Regards,
>>> > David Notivol
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From admin at blindi.net  Thu May 12 16:41:36 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 12 May 2011 14:41:36 +0200 (CEST)
Subject: [Freeswitch-users] question does freeswitch support multipe
 group confirm keys?
In-Reply-To: <BANLkTin3DVFTLkEgfG+3ccPhZ-irZhFoLA@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<Pine.LNX.4.64.1105120636470.9752@debian>
	<BANLkTin3DVFTLkEgfG+3ccPhZ-irZhFoLA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105121434510.2165@debian>

Hi lakshmaan,

Ok you execute a script.
U Don.t unserstand: how can you defined different digits to execute from 
script?
Do you have a example please?
My  solution: i like to search a callscreening, to press1 to accept, press 
2 for the blacklist.
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 adam.kelloway at newpace.ca  Thu May 12 16:44:58 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Thu, 12 May 2011 09:44:58 -0300
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <78C4A3C1-0CDD-4BA5-BC9E-165AE305C7AF@gmail.com>
References: <4DC84196.9050905@newpace.ca>	<BANLkTim3-V-hc+fn6m-WMg2jUWbxp=Yjog@mail.gmail.com>
	<78C4A3C1-0CDD-4BA5-BC9E-165AE305C7AF@gmail.com>
Message-ID: <4DCBD64A.6010105@newpace.ca>

I was thinking of NFS, but was also concerned about its reliability. 
Were your performance issues related to file access times? Also, if the 
server was ever unavailable, it would be useful to play a default prompt 
that was guaranteed to be on the local disk. Is there any existing 
fail-over mechanism built into FreeSWITCH?

I hadn't heard of MooseFS, so I will look into that as well, thanks.


On 3:59 PM, Moe Navid wrote:
> We had some performance issues with NFS, we switched to MooseFS for 
> voice prompts as well as recordings with very good results.
>
> On May 11, 2011, at 10:05 AM, Fernando Testa wrote:
>
>> If your files are on a remote server you can mount it locally using 
>> NFS or something else.
>>
>> On Mon, May 9, 2011 at 4:33 PM, Adam Kelloway 
>> <adam.kelloway at newpace.ca <mailto:adam.kelloway at newpace.ca>> wrote:
>>
>>     If I did not wish to store audio files on a FreeSWITCH host, what
>>     are my
>>     options for being able to retrieve a remote file and playing it?
>>     My understanding is that the playback application can only play a
>>     local
>>     file, is that correct?
>>     I did notice that there is a shell_stream module, which you could
>>     potentially use to retrieve a file (say, via HTTP), and provide the
>>     audio data as the output bash script output. Is there a way to do
>>     this
>>     or something similar without having to actually load another
>>     program/script?
>>
>>     Thanks,
>>
>>     Adam Kelloway
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>


-- 
Adam Kelloway
Software Engineer
NewPace Technology Development Inc.
adam.kelloway at newpace.ca
+1 902-406-8375 x1031
www.newpace.ca

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From yehavi.bourvine at gmail.com  Thu May 12 16:48:11 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 12 May 2011 15:48:11 +0300
Subject: [Freeswitch-users] execute_on_answer_n and execute_on_answer
Message-ID: <BANLkTi=XsNR00=TXpxk8HkrOcQu36P_2Aw@mail.gmail.com>

Hello,
 According to the wiki, starting from April there is an additional syntax
for execute_on_answer which is execute_on_answer_1 (and 2 and on).


 If I want to use the new syntax, how shall I do it? Use only
execute_on_answer_1,2,... or first use execute_on_answer and the following
statements will use the numbered format? What happens if I mix the two? What
happens if I miss a number (i.e. use _1 and then _3)?

                                Thanks! __Yehavi:
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From steveu at coppice.org  Thu May 12 17:59:48 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 12 May 2011 21:59:48 +0800
Subject: [Freeswitch-users] Fwd: Major deployment of outbound FAX on
 latest version of Freeswitch question
In-Reply-To: <BANLkTiktkLdRSHWZFVv9JGz5hzP7cAay=w@mail.gmail.com>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>	<4DC40C86.9000503@coppice.org>	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>	<4DC67898.2090106@coppice.org>	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>	<4DC732BC.9070206@coppice.org>
	<BANLkTiktkLdRSHWZFVv9JGz5hzP7cAay=w@mail.gmail.com>
Message-ID: <4DCBE7D4.1070100@coppice.org>

On 05/12/2011 05:21 PM, Juan Antonio Iba?ez Santorum wrote:
> Hello Steve!
>
>    I would like to ask you one question about fax system. You told 
> that using FS+mod_spandsp could be gotten successful faxing rates over 
> 99%. Looking at your experience, is 99% nearer to 100% or to 99%? 
> Which errors do you find usually on that 1%?
>
You can run tests all day in a lab without failure. The real test is how 
well things work against a wide diversity of machines in the real world. 
This http://www.soft-switch.org/spandsp-soft-fax-performance.html says 
something about our testing, and what is achievable. Its quite a lot of 
work to get real reliability numbers. You get a lot of call failures on 
a typical public FAX server due to things like wrong numbers, voice 
calls into a FAX port, people purposefully dropping half completed FAX 
calls, and buggy FAX machines doing weird things. Don't ignore the last 
one. There are famous make machines with serious bugs. You need to 
manually check the failed calls one by one to get real reliability 
figures. In the end there are always a few calls where you are never 
sure who's fault the failure is. All I can say with hand on heart is the 
number of unexplainable failures is well below 1%

Regards,
Steve



From juanito1982 at gmail.com  Thu May 12 19:06:44 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 12 May 2011 17:06:44 +0200
Subject: [Freeswitch-users] Fwd: Major deployment of outbound FAX on
 latest version of Freeswitch question
In-Reply-To: <4DCBE7D4.1070100@coppice.org>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
	<4DC67898.2090106@coppice.org>
	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
	<4DC732BC.9070206@coppice.org>
	<BANLkTiktkLdRSHWZFVv9JGz5hzP7cAay=w@mail.gmail.com>
	<4DCBE7D4.1070100@coppice.org>
Message-ID: <BANLkTim-U86W=Yas7_rhFcQOL0Va4M5V2w@mail.gmail.com>

Thank you again Steve. As you said, best test is a production enviroment. I
will try to set up one machine to start testing a fax engine.

Regards

2011/5/12 Steve Underwood <steveu at coppice.org>

> On 05/12/2011 05:21 PM, Juan Antonio Iba?ez Santorum wrote:
> > Hello Steve!
> >
> >    I would like to ask you one question about fax system. You told
> > that using FS+mod_spandsp could be gotten successful faxing rates over
> > 99%. Looking at your experience, is 99% nearer to 100% or to 99%?
> > Which errors do you find usually on that 1%?
> >
> You can run tests all day in a lab without failure. The real test is how
> well things work against a wide diversity of machines in the real world.
> This http://www.soft-switch.org/spandsp-soft-fax-performance.html says
> something about our testing, and what is achievable. Its quite a lot of
> work to get real reliability numbers. You get a lot of call failures on
> a typical public FAX server due to things like wrong numbers, voice
> calls into a FAX port, people purposefully dropping half completed FAX
> calls, and buggy FAX machines doing weird things. Don't ignore the last
> one. There are famous make machines with serious bugs. You need to
> manually check the failed calls one by one to get real reliability
> figures. In the end there are always a few calls where you are never
> sure who's fault the failure is. All I can say with hand on heart is the
> number of unexplainable failures is well below 1%
>
> Regards,
> 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 msc at freeswitch.org  Thu May 12 19:17:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 08:17:43 -0700
Subject: [Freeswitch-users] question how to cut a string from a variable
 in fs?
In-Reply-To: <Pine.LNX.4.64.1105120530140.9184@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
Message-ID: <BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>

For trimming, etc. use this:
http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables

If you want to do serious variable manipulation then just use regex. It's
"more powerful than you can possibly imagine." ;)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex

-MC

2011/5/11 Thomas Hoellriegel <admin at blindi.net>

> Hi all,
> I don.t find a funtion to cut a variable string in fs.
> Support Fs a funtion for example:
> cut varname, delimiter, fieldspec?
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu May 12 19:30:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 08:30:26 -0700
Subject: [Freeswitch-users] Problem menuselection not working with
 Play_and_get_digits Menuselection
In-Reply-To: <Pine.LNX.4.64.1105112328540.5984@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
	<Pine.LNX.4.64.1105112328540.5984@debian>
Message-ID: <BANLkTikqTYP1z_G_6xjvGC=CybQMdpficg@mail.gmail.com>

This is a classic case of dialplan parsing vs. executing. I'll let you read
chapters 5 and 8 of the FreeSWITCH book to get a better understanding of the
difference between those two. In the meantime I would like to suggest that
you choose an alternate method of building your IVR. There are better tools
available to you. Look at chapter 6 of the book to get an idea of how to use
the XML IVR building system in FreeSWITCH. Look at chapter 7 to see how to
do it in Lua.

That being said, you *can* do this in the dialplan, but it will be ugly.
First off, I recommend that you put all your ivr "menus" into separate
extensions instead of just separate conditions. Second, put all those
extensions into their own dialplan context. Lastly, use the transfer app
after you collect a digit with PAGD:

<action application="play_and_get_digits"
     data="0 1 1 5000 #
    $${base_dir}/sounds/callback/confirm-callback-press1.alaw
   $${base_dir}/sounds/callback/invalid.alaw  nr1 [1-2]"/>
<action application="transfer" data="${nr1} XML your_new_context"/>

Since you're already using the XML IVR for "callback only" I think you
should probably just use XML for the whole thing. That's my $0.02...

-MC

On Wed, May 11, 2011 at 2:31 PM, Thomas Hoellriegel <admin at blindi.net>wrote:

>  Hi all,
>  i have create a menuselection via play_and_get_digits.
>  Problem, this application ignore the digits 1 and 2.
>  This is my dialplanextestnion:
>
>  <extension name="callback_only_menuselection">
>  <condition field="destination_number" expression="^callback_only_sel$"/>
>    <condition field="${ivrlevel}" expression="^$" break="never">
>      <action application="play_and_get_digits"
>      data="0 1 1 5000 #
>     $${base_dir}/sounds/callback/confirm-callback-press1.alaw
>    $${base_dir}/sounds/callback/invalid.alaw  nr1 [1-2]"/>
>
>    </condition>
>    <condition field="${ivrlevel}-${nr1}" expression="^1$" >
>  <action application="playback"
>   data="$${base_dir}/sounds/callback/callback-send.alaw"/>
>  <action application="system" data="$${base_dir}/scripts/callback.sh"/>
>    </condition>
>    <condition field="${ivrlevel}-${nr1}" expression="^2$">
>  <action application="ivr" data="callback_only_ivr"/>
>    </condition>
>  </extension>
>
>  I press 1 and 2 play_and_get_digits don.t execute these actions.
>  What is wrong?
>  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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu May 12 19:33:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 08:33:29 -0700
Subject: [Freeswitch-users] question does freeswitch support multipe
 group confirm keys?
In-Reply-To: <Pine.LNX.4.64.1105121434510.2165@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<Pine.LNX.4.64.1105120636470.9752@debian>
	<BANLkTin3DVFTLkEgfG+3ccPhZ-irZhFoLA@mail.gmail.com>
	<Pine.LNX.4.64.1105121434510.2165@debian>
Message-ID: <BANLkTim5DipSfYuJDhESA2PjJEOwvTZ3mg@mail.gmail.com>

This is a feature request and will require some coding in the module. You
can open a jira feature request and maybe offer a bounty if it is a serious
need.

-MC

On Thu, May 12, 2011 at 5:41 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi lakshmaan,
>
> Ok you execute a script.
> U Don.t unserstand: how can you defined different digits to execute from
> script?
> Do you have a example please?
> My  solution: i like to search a callscreening, to press1 to accept, press
> 2 for the blacklist.
>
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu May 12 20:26:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 09:26:17 -0700
Subject: [Freeswitch-users] execute_on_answer_n and execute_on_answer
In-Reply-To: <BANLkTi=XsNR00=TXpxk8HkrOcQu36P_2Aw@mail.gmail.com>
References: <BANLkTi=XsNR00=TXpxk8HkrOcQu36P_2Aw@mail.gmail.com>
Message-ID: <BANLkTikF=RNRdBfxHF_su1YjZfuZ40VwiQ@mail.gmail.com>

On Thu, May 12, 2011 at 5:48 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello,
>  According to the wiki, starting from April there is an additional syntax
> for execute_on_answer which is execute_on_answer_1 (and 2 and on).
>
>
>  If I want to use the new syntax, how shall I do it? Use only
> execute_on_answer_1,2,... or first use execute_on_answer and the following
> statements will use the numbered format? What happens if I mix the two? What
> happens if I miss a number (i.e. use _1 and then _3)?
>

You can mix all you want. I have not tried execute_on_answer and
execute_on_answer_x but you can try it and see what happens. Also, there is
no need to restrict your use to numbers:

execute_on_answer_1
execute_on_answer_foo
execute_on_answer_whatever

The order of execution is the order in which you set or pass the variables.

-MC
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From msc at freeswitch.org  Thu May 12 20:28:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 09:28:07 -0700
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
Message-ID: <BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>

This sounds like it might be an issue with the CDR reporting stage. How are
you doing CDRs on these calls?
-MC

On Wed, May 11, 2011 at 6:27 AM, Leonardo P. Bidinoto <
leonardo.bidinoto at voicetechnology.com.br> wrote:

> Hi everyone!
>
> I having some troubles with my FS box(CentOS  5.3 64-bit,  Xeon(R) CPU
> 2.66GHz with 8 cores , 8GbRAM, (git-bc19d28 2011-04-25 15-53-54 -0400) ).
> Sometimes, when a user calls to my box and hangs up, the user's channel
> remains on FS when i execute a "show channels" command. with "uuid_exists"
> command, im receiving "false".
> looking at freeswitch log, the last message for the channels is:
> 0eeb975f-2ae9-4f7d-9888-1478faf87d2d 2011-05-11 00:05:57.555142 [DEBUG]
> switch_core_session.c:1286 Session 38 (sofia/external/2562687580 at 10.0.70.6)
> Locked, Waiting on external entities
> Im not receiving "Ended" State after the previous message.
>
> did someone have seen this issue? or know a way to remove from freeswitch?
>
> Thanks all.
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 12 20:29:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 12 May 2011 11:29:12 -0500
Subject: [Freeswitch-users] question does freeswitch support multipe
 group confirm keys?
In-Reply-To: <Pine.LNX.4.64.1105121434510.2165@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<Pine.LNX.4.64.1105120636470.9752@debian>
	<BANLkTin3DVFTLkEgfG+3ccPhZ-irZhFoLA@mail.gmail.com>
	<Pine.LNX.4.64.1105121434510.2165@debian>
Message-ID: <BANLkTinOEk+YgA+PuMaqKPEmsn3qG8d8OA@mail.gmail.com>

When you exec a script such as a lua script then you do whatever you
want in the script
play a file ask for input etc just like any other time.  The script
takes the place of the default behavior.


The first channel to be done with the script that is not hungup is
considered the winner.


On Thu, May 12, 2011 at 7:41 AM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Hi lakshmaan,
>
> Ok you execute a script.
> U Don.t unserstand: how can you defined different digits to execute from
> script?
> Do you have a example please?
> My ?solution: i like to search a callscreening, to press1 to accept, press 2
> for the blacklist.
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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  Thu May 12 20:32:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 09:32:58 -0700
Subject: [Freeswitch-users] call somebody into conference after
	conference established
In-Reply-To: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
Message-ID: <BANLkTikAzr-tUqiQYmRUU44bsWyN_y4LRg@mail.gmail.com>

You can add a caller from fs_cli with the "conference" command:

conference <conf_name> dial sofia/external/foo at bar <caller id num> <caller
id name>

-MC

On Wed, May 11, 2011 at 8:11 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Gurus,
>
> i dial 666 and enter a conference, then i need call some body's phone
> number to join him into this conference...
> is there anyone can advise how can realize this scenario?
>
> thanks.
>
> 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 admin at blindi.net  Thu May 12 22:27:20 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 12 May 2011 20:27:20 +0200 (CEST)
Subject: [Freeswitch-users] Excluding phonenumbers from existing matches?
In-Reply-To: <3C27A8D0-EF7E-485C-B3EB-892F1ACB3A99@ipeva.fr>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
	<Pine.LNX.4.64.1105112331320.5984@debian>
	<3C27A8D0-EF7E-485C-B3EB-892F1ACB3A99@ipeva.fr>
Message-ID: <Pine.LNX.4.64.1105122020110.5073@debian>

Hi David, thank you for your nice help!!!
You have right, the way is very easy.
I coming from asterisk and i like to replace my asterisk to fs.
I think fs is very flexible.
Fs is very stable, and by asterisk i must reboot the server every day.
Fs is not hanging the Server in the datacenter.
Your have programmed the best pbx of the world!!-))
I.m so  very  glad!!




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

From Nabble at slickdeals.endjunk.com  Thu May 12 22:30:47 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 12 May 2011 11:30:47 -0700 (PDT)
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
Message-ID: <1305225047195-6356793.post@n2.nabble.com>

I just did a fresh untar with a git pull this morning to
dc2208e3fe0ec8bdd0eeb63e1db1111f6ac8f982 revision and now FS cross
compilation crashes on compiling libs/curl as shown below:

OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-gcc
-DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
-fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
-I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr-util/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr-util/xml/expat/lib
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/stfu
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/sqlite
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/pcre
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/speex/include
-Ilibs/speex/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/srtp/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/srtp/crypto/include
-Ilibs/srtp/crypto/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/spandsp/src
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/tiff-3.8.2/libtiff
-DCORE_USE_CURL -DENABLE_SRTP -DSWITCH_HAVE_ODBC
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libedit/src
-DSWITCH_HAVE_LIBEDIT
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
-fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
-DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic
-Wdeclaration-after-statement -I. -I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-Os -pipe -march=armv5te -mtune=xscale -fno-caller-saves -fhonour-copts
-msoft-float -DLUA_USE_LINUX -fpic -std=gnu99 -Wno-format -MT
libfreeswitch_la-bit_operations.lo -MD -MP -MF
.deps/libfreeswitch_la-bit_operations.Tpo -c
libs/spandsp/src/bit_operations.c -o libfreeswitch_la-bit_operations.o
>/dev/null 2>&1
OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
-DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
-fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
-I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-I. -I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
.deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
.libs/switch_cpp.o
OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
-DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
-fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
-I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-I. -I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
.deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
OpenWrt-quiet_libtool: link: cannot find the library
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/curl/lib/libcurl.la'
or unhandled argument
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/curl/lib/libcurl.la'
make[4]: *** [libfreeswitch.la] Error 1
make[4]: Leaving directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[3]: *** [all] Error 2
make[3]: Leaving directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[2]: ***
[/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/.built]
Error 2
make[2]: Leaving directory `/opt/OpenWRT/feeds/packages/net/freeswitch_git'
make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
make[1]: Leaving directory `/opt/openwrt-svn-trunk'
make: *** [package/freeswitch_git/compile] Error 2


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
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View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation-on-latest-git-tp6356793p6356793.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Thu May 12 22:40:25 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 12 May 2011 11:40:25 -0700
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <1305225047195-6356793.post@n2.nabble.com>
References: <1305225047195-6356793.post@n2.nabble.com>
Message-ID: <BANLkTik4CUHsxUr6uKMAq+H_7FOTpgdiAg@mail.gmail.com>

reported this to JIRA yet?

On Thu, May 12, 2011 at 11:30 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> I just did a fresh untar with a git pull this morning to
> dc2208e3fe0ec8bdd0eeb63e1db1111f6ac8f982 revision and now FS cross
> compilation crashes on compiling libs/curl as shown below:
>
> OpenWrt-quiet_libtool: compile: ?arm-openwrt-linux-uclibcgnueabi-gcc
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr-util/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr-util/xml/expat/lib
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/stfu
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/sqlite
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/pcre
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/speex/include
> -Ilibs/speex/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/srtp/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/srtp/crypto/include
> -Ilibs/srtp/crypto/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/spandsp/src
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/tiff-3.8.2/libtiff
> -DCORE_USE_CURL -DENABLE_SRTP -DSWITCH_HAVE_ODBC
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libedit/src
> -DSWITCH_HAVE_LIBEDIT
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
> -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic
> -Wdeclaration-after-statement -I. -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -Os -pipe -march=armv5te -mtune=xscale -fno-caller-saves -fhonour-copts
> -msoft-float -DLUA_USE_LINUX -fpic -std=gnu99 -Wno-format -MT
> libfreeswitch_la-bit_operations.lo -MD -MP -MF
> .deps/libfreeswitch_la-bit_operations.Tpo -c
> libs/spandsp/src/bit_operations.c -o libfreeswitch_la-bit_operations.o
>>/dev/null 2>&1
> OpenWrt-quiet_libtool: compile: ?arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp ?-fPIC -DPIC -o
> .libs/switch_cpp.o
> OpenWrt-quiet_libtool: compile: ?arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
> OpenWrt-quiet_libtool: link: cannot find the library
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/curl/lib/libcurl.la'
> or unhandled argument
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/curl/lib/libcurl.la'
> make[4]: *** [libfreeswitch.la] Error 1
> make[4]: Leaving directory
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[3]: *** [all] Error 2
> make[3]: Leaving directory
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[2]: ***
> [/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/.built]
> Error 2
> make[2]: Leaving directory `/opt/OpenWRT/feeds/packages/net/freeswitch_git'
> make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
> make[1]: Leaving directory `/opt/openwrt-svn-trunk'
> make: *** [package/freeswitch_git/compile] Error 2
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation-on-latest-git-tp6356793p6356793.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 leonardo.bidinoto at voicetechnology.com.br  Thu May 12 23:27:42 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Thu, 12 May 2011 16:27:42 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
Message-ID: <BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>

Hi Michael,

Im not using to any cdr module.

2011/5/12 Michael Collins <msc at freeswitch.org>

> This sounds like it might be an issue with the CDR reporting stage. How are
> you doing CDRs on these calls?
> -MC
>
> On Wed, May 11, 2011 at 6:27 AM, Leonardo P. Bidinoto <
> leonardo.bidinoto at voicetechnology.com.br> wrote:
>
>> Hi everyone!
>>
>> I having some troubles with my FS box(CentOS  5.3 64-bit,  Xeon(R) CPU
>> 2.66GHz with 8 cores , 8GbRAM, (git-bc19d28 2011-04-25 15-53-54 -0400) ).
>> Sometimes, when a user calls to my box and hangs up, the user's channel
>> remains on FS when i execute a "show channels" command. with "uuid_exists"
>> command, im receiving "false".
>> looking at freeswitch log, the last message for the channels is:
>> 0eeb975f-2ae9-4f7d-9888-1478faf87d2d 2011-05-11 00:05:57.555142 [DEBUG]
>> switch_core_session.c:1286 Session 38 (sofia/external/
>> 2562687580 at 10.0.70.6) Locked, Waiting on external entities
>> Im not receiving "Ended" State after the previous message.
>>
>> did someone have seen this issue? or know a way to remove from freeswitch?
>>
>> Thanks all.
>>
>> --
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From brian at freeswitch.org  Thu May 12 23:42:37 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 12 May 2011 14:42:37 -0500
Subject: [Freeswitch-users] FreeSwitch video is impossible to do MCU and
	H.239 in H.323, with SIP still there is no BFCP
In-Reply-To: <BANLkTinGV4goi+V2A-+u0vF4H7jEiqJc3A@mail.gmail.com>
References: <BANLkTinGV4goi+V2A-+u0vF4H7jEiqJc3A@mail.gmail.com>
Message-ID: <2E7D68B6-02C5-4574-B6F1-849CF14D2702@freeswitch.org>

I don't think any of the h323 stuff supports video in FreeSWITCH but I can tell you that it does work on SIP.   I have had video calls up between Polycom VVX1500's and Eyebeam and between two Lifesize units and it all works fine.  Are you not on the latest software?


/b

On May 11, 2011, at 4:39 PM, Shamun wrote:

> Hello Team,
> 
> FreeSWITCH does not work for Video. To do Video with FreeSWITCH, Its simply
> a great nightmare. None of our equipment is compatible with FreeSWITCH we
> have tried mod_h323/mod_opal and regular SIP from FreeSWITCH (none of them
> works with h.323).
> 
> - Is it possible to do FreeSWITCH IVVR (Flash or MPEG4 or JPEG) ?
> - Will FreeSWITCH support H.323 (and its related family such as H.460,
> H.239, etc etc), there are lot of staff just does gets compatible with SIP
> (specially our hardwares, i believe same with others).
> - Will FreeSWITCH allow following equipment supports + more other equipments
> who really depends on mod_h323/mod_opal (SIP has less hope with hardware's
> compatibility)
> 
> Tandberg Edge 85 MXP
> Tandberg C90
> VCON vPoint HD 10.0
> Radvision
> 
> 
> Thanks & Regards
> Shamun



From brian at freeswitch.org  Thu May 12 23:44:21 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 12 May 2011 14:44:21 -0500
Subject: [Freeswitch-users] Freeswitch 1.0.8 likely release date?
In-Reply-To: <BANLkTikjvj-o4pdNdRHKc6f-Znvgyw3HHQ@mail.gmail.com>
References: <BANLkTikjvj-o4pdNdRHKc6f-Znvgyw3HHQ@mail.gmail.com>
Message-ID: <033D369C-1FE9-44CA-B9D8-464294294C8A@freeswitch.org>

Use Git Head... the release is all based on 1. finishing all the outstanding issues. 2. getting people to test it.  If nobody tests then we don't release.

/b

On May 11, 2011, at 7:01 AM, Sidharth Kshatriya wrote:

> Can anyone tell us approximately when 1.0.8 is likely to be released... even
> a ball park?



From msc at freeswitch.org  Thu May 12 23:56:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 12:56:58 -0700
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
Message-ID: <BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>

On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
leonardo.bidinoto at voicetechnology.com.br> wrote:

> Hi Michael,
>
> Im not using to any cdr module.


I would recommend that you do several things:

#1 - update to latest git
#2 - rotate logs
#3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
#4 - reproduce the symptom with a single call (if possible)
#5 - pastebin the log for the uuid in question and link to it in this thread

>From there hopefully we'll get a clue as to what is happening.
-MC
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From david.villasmil.work at gmail.com  Thu May 12 23:49:33 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 May 2011 21:49:33 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
Message-ID: <BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>

Hello Guys,

Well, except for a few things, I think this is about-ready to go out to you
guys.

Problem is, I have NO IDEA about Gira or anything like that. I developed
this on my own free time and making no money out of it at all.

So, I would appreciate it if someone can guide me to publish this somehow ;)

Thanks all

David

On Wed, Apr 27, 2011 at 12:30 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You are open to losing money though if they can make more than one call at
> once - each would be scheduled as if they were the only call, allowing the
> callers to drop below 0.
>
> Steve on iPhone
>
> On 27 Apr 2011, at 11:17, Nicolas Brenner <nico at clickfono.com> wrote:
>
> If you know the rate and the balance for the call, you may calculate the
> maximum time for the call as max_time = balance/rate. Then you may schedule
> a hangup for that call max_time in the future, something like this:
>
> sched_hangup +" + max_time + " "+ call_uuid +" 'ALLOTED_TIMEOUT'
>
> That way the balance shouldn't go below 0, since you are hanging up the
> call before it does, with hangup cause ALLOTED_TIMEOUT.
>
>
>
> On Fri, Mar 4, 2011 at 2:44 PM, David Villasmil <<david.villasmil.work at gmail.com>
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>>
>> I do the route selecting with a lua script. overflow and load distribution
>> with mod_distributor loaded via curl.
>>
>> the prepaid side is done with nibble, yes. But i don't like it too much, i
>> might just deduct the balance when the call disconnects and let the
>> authorization block new calls, so the balance might go under 0 a little...
>>
>>
>>
>>
>> On Sun, Feb 27, 2011 at 9:06 PM, Saeed Ahmed < <saeedahmad1981 at gmail.com>
>> saeedahmad1981 at gmail.com> wrote:
>>
>>> press sent too quick..
>>>
>>> what did you use for routing? curl? esl?
>>>
>>> did you use nibble bill for prepaid app?
>>>
>>>
>>> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <<saeedahmad1981 at gmail.com>
>>> saeedahmad1981 at gmail.com> wrote:
>>>
>>>> Great!
>>>>
>>>> want to see it soon.
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri May 13 00:09:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 13:09:47 -0700
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
Message-ID: <BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>

On Thu, May 12, 2011 at 12:49 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> Well, except for a few things, I think this is about-ready to go out to you
> guys.
>
> Problem is, I have NO IDEA about Gira or anything like that. I developed
> this on my own free time and making no money out of it at all.
>
> So, I would appreciate it if someone can guide me to publish this somehow
> ;)
>

Well, everyone is pretty anxious to see this. First question: what's
involved in the installation? What are the dependencies? Secondly, do you
have a sub-folder on the freeswitch-contrib repo? If not then Raymond can
create one for you. You could also publish it to github and then link from
github to your freeswitch-contrib folder at a later date.

-MC
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From msc at freeswitch.org  Fri May 13 00:12:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 13:12:26 -0700
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <1305225047195-6356793.post@n2.nabble.com>
References: <1305225047195-6356793.post@n2.nabble.com>
Message-ID: <BANLkTi=_mLadKL4OCioQ=tGaSGa8rGkG-A@mail.gmail.com>

On Thu, May 12, 2011 at 11:30 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

> I just did a fresh untar with a git pull this morning to
> dc2208e3fe0ec8bdd0eeb63e1db1111f6ac8f982 revision and now FS cross
> compilation crashes on compiling libs/curl as shown below:
>
*SMACK*

Naughty boy! We've told you to report these to jira.freeswitch.org and not
on the mailing list. Time for you to go sit in the naughty corner! :)
-MC
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From mike at jerris.com  Fri May 13 00:42:21 2011
From: mike at jerris.com (Michael Jerris)
Date: Thu, 12 May 2011 16:42:21 -0400
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <1305225047195-6356793.post@n2.nabble.com>
References: <1305225047195-6356793.post@n2.nabble.com>
Message-ID: <266B2D96-0E13-4091-93D0-14F69E07909F@jerris.com>

commit 314713fce14c6daa01dea0f3e57be2e1b0152366
Author: Michael Jerris <mike at jerris.com>
Date:   Thu May 12 16:38:18 2011 -0400

    FS-2936: attempt to fix wrt build

try that maybe?

On May 12, 2011, at 2:30 PM, mazilo wrote:

> I just did a fresh untar with a git pull this morning to
> dc2208e3fe0ec8bdd0eeb63e1db1111f6ac8f982 revision and now FS cross
> compilation crashes on compiling libs/curl as shown below:
> 
> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-gcc
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr-util/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/apr-util/xml/expat/lib
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/stfu
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/sqlite
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/pcre
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/speex/include
> -Ilibs/speex/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/srtp/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/srtp/crypto/include
> -Ilibs/srtp/crypto/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/spandsp/src
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/tiff-3.8.2/libtiff
> -DCORE_USE_CURL -DENABLE_SRTP -DSWITCH_HAVE_ODBC
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libedit/src
> -DSWITCH_HAVE_LIBEDIT
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
> -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic
> -Wdeclaration-after-statement -I. -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -Os -pipe -march=armv5te -mtune=xscale -fno-caller-saves -fhonour-copts
> -msoft-float -DLUA_USE_LINUX -fpic -std=gnu99 -Wno-format -MT
> libfreeswitch_la-bit_operations.lo -MD -MP -MF
> .deps/libfreeswitch_la-bit_operations.Tpo -c
> libs/spandsp/src/bit_operations.c -o libfreeswitch_la-bit_operations.o
>> /dev/null 2>&1
> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
> .libs/switch_cpp.o
> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
> OpenWrt-quiet_libtool: link: cannot find the library
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/curl/lib/libcurl.la'
> or unhandled argument
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/curl/lib/libcurl.la'
> make[4]: *** [libfreeswitch.la] Error 1
> make[4]: Leaving directory
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[3]: *** [all] Error 2
> make[3]: Leaving directory
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[2]: ***
> [/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/.built]
> Error 2
> make[2]: Leaving directory `/opt/OpenWRT/feeds/packages/net/freeswitch_git'
> make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
> make[1]: Leaving directory `/opt/openwrt-svn-trunk'
> make: *** [package/freeswitch_git/compile] Error 2
> 
> 
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation-on-latest-git-tp6356793p6356793.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 all.eforums at gmail.com  Fri May 13 00:49:45 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Thu, 12 May 2011 16:49:45 -0400
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <BANLkTi=vPrN84Yjy_BOOooGhc4EB_PMvWQ@mail.gmail.com>
References: <4DC84196.9050905@newpace.ca>
	<BANLkTimtTJ5QHofCaC0_ExXDexefjj7OqA@mail.gmail.com>
	<BANLkTi=vPrN84Yjy_BOOooGhc4EB_PMvWQ@mail.gmail.com>
Message-ID: <BANLkTikdW1PJnLqxukSv+eD9tVyx9fUXxw@mail.gmail.com>

On Mon, May 9, 2011 at 5:02 PM, Brad Mina <brad at tritelcomm.com> wrote:

> You could setup an internal http server with a directory called 'audio' (
> http://server.ip/audio/), then use mod_shout as Avi suggested like this:
> <action application="shout" data="http://server.ip/audio/file.mp3"/>
>
>
> +1 on "Interest" in this discussion. Additional point of failure argument
aside, I'm very interested to know if anyone else has any experience to
share re: streaming audio files/prompts etc from a remote FS via shout/Ice
type of protocols/techniques rather than using a fileshare mount etc? Is
there a preference of one over the other (SHOUT Vs ICE) or should I be
reading the difference between a SHOUTcast and ICEcast before opening my
mouth? :)

Thx
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From krice at freeswitch.org  Fri May 13 00:56:11 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 12 May 2011 15:56:11 -0500
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <266B2D96-0E13-4091-93D0-14F69E07909F@jerris.com>
Message-ID: <C9F1B39B.181D7%krice@freeswitch.org>

Hey that partially fixed the build on ubuntu... There is also this too

src/switch_curl.c:34:28: error: openssl/crypto.h: No such file or directory
src/switch_curl.c: In function 'switch_curl_ssl_lock_callback':
src/switch_curl.c:41: error: 'CRYPTO_LOCK' undeclared (first use in this
function)
src/switch_curl.c:41: error: (Each undeclared identifier is reported only
once
src/switch_curl.c:41: error: for each function it appears in.)


Probably related to same patch


On 5/12/11 3:42 PM, "Michael Jerris" <mike at jerris.com> wrote:

> commit 314713fce14c6daa01dea0f3e57be2e1b0152366
> Author: Michael Jerris <mike at jerris.com>
> Date:   Thu May 12 16:38:18 2011 -0400
> 
>     FS-2936: attempt to fix wrt build
> 
> try that maybe?
> 
> On May 12, 2011, at 2:30 PM, mazilo wrote:
> 
>> I just did a fresh untar with a git pull this morning to
>> dc2208e3fe0ec8bdd0eeb63e1db1111f6ac8f982 revision and now FS cross
>> compilation crashes on compiling libs/curl as shown below:
>> 
>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-gcc
>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>> -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/apr/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/apr-util/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/apr-util/xml/expat/lib
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/stfu
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/sqlite
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/pcre
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/speex/include
>> -Ilibs/speex/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/srtp/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/srtp/crypto/include
>> -Ilibs/srtp/crypto/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/spandsp/src
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/tiff-3.8.2/libtiff
>> -DCORE_USE_CURL -DENABLE_SRTP -DSWITCH_HAVE_ODBC
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libedit/src
>> -DSWITCH_HAVE_LIBEDIT
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
>> -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic
>> -Wdeclaration-after-statement -I. -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -Os -pipe -march=armv5te -mtune=xscale -fno-caller-saves -fhonour-copts
>> -msoft-float -DLUA_USE_LINUX -fpic -std=gnu99 -Wno-format -MT
>> libfreeswitch_la-bit_operations.lo -MD -MP -MF
>> .deps/libfreeswitch_la-bit_operations.Tpo -c
>> libs/spandsp/src/bit_operations.c -o libfreeswitch_la-bit_operations.o
>>> /dev/null 2>&1
>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>> -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -I. -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
>> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
>> .libs/switch_cpp.o
>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>> -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -I. -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
>> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
>> OpenWrt-quiet_libtool: link: cannot find the library
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git/libs/curl/lib/libcurl.la'
>> or unhandled argument
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git/libs/curl/lib/libcurl.la'
>> make[4]: *** [libfreeswitch.la] Error 1
>> make[4]: Leaving directory
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git'
>> make[3]: *** [all] Error 2
>> make[3]: Leaving directory
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git'
>> make[2]: ***
>> [/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git/.built]
>> Error 2
>> make[2]: Leaving directory `/opt/OpenWRT/feeds/packages/net/freeswitch_git'
>> make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
>> make[1]: Leaving directory `/opt/openwrt-svn-trunk'
>> make: *** [package/freeswitch_git/compile] Error 2
>> 
>> 
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation
>> -on-latest-git-tp6356793p6356793.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
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From krice at freeswitch.org  Fri May 13 01:40:24 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 12 May 2011 16:40:24 -0500
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <266B2D96-0E13-4091-93D0-14F69E07909F@jerris.com>
Message-ID: <C9F1BDF8.181D9%krice@freeswitch.org>

Oh oh that does fix a similar issue on ubuntu 10.04... And to fix the ssl
issue add in libssl-dev on ubuntu to your pre-req packages now...

There are 3 FS modules mod_cidlookup, mod_xml_curl, and mod_xml_cdr that
still have issues... Related to the same bug... Makefile.in/am updates
probably required here looking into it now

K


On 5/12/11 3:42 PM, "Michael Jerris" <mike at jerris.com> wrote:

> commit 314713fce14c6daa01dea0f3e57be2e1b0152366
> Author: Michael Jerris <mike at jerris.com>
> Date:   Thu May 12 16:38:18 2011 -0400
> 
>     FS-2936: attempt to fix wrt build
> 
> try that maybe?
> 
> On May 12, 2011, at 2:30 PM, mazilo wrote:
> 
>> I just did a fresh untar with a git pull this morning to
>> dc2208e3fe0ec8bdd0eeb63e1db1111f6ac8f982 revision and now FS cross
>> compilation crashes on compiling libs/curl as shown below:
>> 
>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-gcc
>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>> -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/apr/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/apr-util/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/apr-util/xml/expat/lib
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/stfu
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/sqlite
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/pcre
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/speex/include
>> -Ilibs/speex/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/srtp/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/srtp/crypto/include
>> -Ilibs/srtp/crypto/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/spandsp/src
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/tiff-3.8.2/libtiff
>> -DCORE_USE_CURL -DENABLE_SRTP -DSWITCH_HAVE_ODBC
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libedit/src
>> -DSWITCH_HAVE_LIBEDIT
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
>> -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic
>> -Wdeclaration-after-statement -I. -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -Os -pipe -march=armv5te -mtune=xscale -fno-caller-saves -fhonour-copts
>> -msoft-float -DLUA_USE_LINUX -fpic -std=gnu99 -Wno-format -MT
>> libfreeswitch_la-bit_operations.lo -MD -MP -MF
>> .deps/libfreeswitch_la-bit_operations.Tpo -c
>> libs/spandsp/src/bit_operations.c -o libfreeswitch_la-bit_operations.o
>>> /dev/null 2>&1
>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>> -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -I. -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
>> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
>> .libs/switch_cpp.o
>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/src/include
>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>> tch_git/libs/libteletone/src
>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>> -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -I. -I./lua
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>> nclude
>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>> de
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/usr/include
>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>> .32_eabi/include
>> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
>> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
>> OpenWrt-quiet_libtool: link: cannot find the library
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git/libs/curl/lib/libcurl.la'
>> or unhandled argument
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git/libs/curl/lib/libcurl.la'
>> make[4]: *** [libfreeswitch.la] Error 1
>> make[4]: Leaving directory
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git'
>> make[3]: *** [all] Error 2
>> make[3]: Leaving directory
>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git'
>> make[2]: ***
>> [/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>> ch_git/.built]
>> Error 2
>> make[2]: Leaving directory `/opt/OpenWRT/feeds/packages/net/freeswitch_git'
>> make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
>> make[1]: Leaving directory `/opt/openwrt-svn-trunk'
>> make: *** [package/freeswitch_git/compile] Error 2
>> 
>> 
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation
>> -on-latest-git-tp6356793p6356793.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 covici at ccs.covici.com  Fri May 13 02:22:26 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 12 May 2011 18:22:26 -0400
Subject: [Freeswitch-users] call waiting problem
Message-ID: <6364.1305238946@ccs.covici.com>

Hi.  I am using a Digium card for a phone extension.  Now when someone
calls that extension through my IVR they hear a nice ring after
selecting the menu option.  However, if I am on another call and they do
this, they get silence.  I do hear the call waiting beep, but why don't
they hear the ringback or transfer_ringback?  I tried it myself and
indeed get silence.

Thanks in advance for any suggestions.

I do have a log if that would help.

-- 
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 mike at jerris.com  Fri May 13 02:22:23 2011
From: mike at jerris.com (Michael Jerris)
Date: Thu, 12 May 2011 18:22:23 -0400
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <C9F1BDF8.181D9%krice@freeswitch.org>
References: <C9F1BDF8.181D9%krice@freeswitch.org>
Message-ID: <8D844C3F-6541-4531-B272-A0935F717A3C@jerris.com>

commit a8da1758cb2ec60b6f1b49d685caf13244c7eec5
Author: Michael Jerris <mike at jerris.com>
Date:   Thu May 12 18:21:34 2011 -0400

    FS-2936: attempt to fix the platform that I'll never have to fix

On May 12, 2011, at 5:40 PM, Ken Rice wrote:

> Oh oh that does fix a similar issue on ubuntu 10.04... And to fix the ssl
> issue add in libssl-dev on ubuntu to your pre-req packages now...
> 
> There are 3 FS modules mod_cidlookup, mod_xml_curl, and mod_xml_cdr that
> still have issues... Related to the same bug... Makefile.in/am updates
> probably required here looking into it now
> 
> K
> 
> 
> On 5/12/11 3:42 PM, "Michael Jerris" <mike at jerris.com> wrote:
> 
>> commit 314713fce14c6daa01dea0f3e57be2e1b0152366
>> Author: Michael Jerris <mike at jerris.com>
>> Date:   Thu May 12 16:38:18 2011 -0400
>> 
>>    FS-2936: attempt to fix wrt build
>> 
>> try that maybe?
>> 
>> On May 12, 2011, at 2:30 PM, mazilo wrote:
>> 
>>> I just did a fresh untar with a git pull this morning to
>>> dc2208e3fe0ec8bdd0eeb63e1db1111f6ac8f982 revision and now FS cross
>>> compilation crashes on compiling libs/curl as shown below:
>>> 
>>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-gcc
>>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/libteletone/src
>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>>> -I./lua
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>>> nclude
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>>> de
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/usr/include
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/include
>>> -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/apr/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/apr-util/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/apr-util/xml/expat/lib
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/stfu
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/sqlite
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/pcre
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/speex/include
>>> -Ilibs/speex/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/srtp/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/srtp/crypto/include
>>> -Ilibs/srtp/crypto/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/spandsp/src
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/tiff-3.8.2/libtiff
>>> -DCORE_USE_CURL -DENABLE_SRTP -DSWITCH_HAVE_ODBC
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>>> nclude
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/libedit/src
>>> -DSWITCH_HAVE_LIBEDIT
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/libteletone/src
>>> -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1
>>> -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic
>>> -Wdeclaration-after-statement -I. -I./lua
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>>> nclude
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>>> de
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/usr/include
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/include
>>> -Os -pipe -march=armv5te -mtune=xscale -fno-caller-saves -fhonour-copts
>>> -msoft-float -DLUA_USE_LINUX -fpic -std=gnu99 -Wno-format -MT
>>> libfreeswitch_la-bit_operations.lo -MD -MP -MF
>>> .deps/libfreeswitch_la-bit_operations.Tpo -c
>>> libs/spandsp/src/bit_operations.c -o libfreeswitch_la-bit_operations.o
>>>> /dev/null 2>&1
>>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
>>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/libteletone/src
>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>>> -I./lua
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>>> nclude
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>>> de
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/usr/include
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/include
>>> -I. -I./lua
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>>> nclude
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>>> de
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/usr/include
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/include
>>> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
>>> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
>>> .libs/switch_cpp.o
>>> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
>>> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/src/include
>>> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswi
>>> tch_git/libs/libteletone/src
>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
>>> -I./lua
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>>> nclude
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>>> de
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/usr/include
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/include
>>> -I. -I./lua
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/i
>>> nclude
>>> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/inclu
>>> de
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/usr/include
>>> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9
>>> .32_eabi/include
>>> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
>>> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
>>> OpenWrt-quiet_libtool: link: cannot find the library
>>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>>> ch_git/libs/curl/lib/libcurl.la'
>>> or unhandled argument
>>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>>> ch_git/libs/curl/lib/libcurl.la'
>>> make[4]: *** [libfreeswitch.la] Error 1
>>> make[4]: Leaving directory
>>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>>> ch_git'
>>> make[3]: *** [all] Error 2
>>> make[3]: Leaving directory
>>> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>>> ch_git'
>>> make[2]: ***
>>> [/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswit
>>> ch_git/.built]
>>> Error 2
>>> make[2]: Leaving directory `/opt/OpenWRT/feeds/packages/net/freeswitch_git'
>>> make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
>>> make[1]: Leaving directory `/opt/openwrt-svn-trunk'
>>> make: *** [package/freeswitch_git/compile] Error 2
>>> 
>>> 
>>> -----
>>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation
>>> -on-latest-git-tp6356793p6356793.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
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 May 13 02:24:05 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 12 May 2011 18:24:05 -0400
Subject: [Freeswitch-users] Playing remote prompts
In-Reply-To: <BANLkTikdW1PJnLqxukSv+eD9tVyx9fUXxw@mail.gmail.com>
References: <4DC84196.9050905@newpace.ca>
	<BANLkTimtTJ5QHofCaC0_ExXDexefjj7OqA@mail.gmail.com>
	<BANLkTi=vPrN84Yjy_BOOooGhc4EB_PMvWQ@mail.gmail.com>
	<BANLkTikdW1PJnLqxukSv+eD9tVyx9fUXxw@mail.gmail.com>
Message-ID: <BANLkTikqGjd2jEmLApuPSUGBazsEU1btiQ@mail.gmail.com>

I wrote a custom module to pull remote prompts over HTTP and cache them on
my FS server.  Given the type of apps I run, I have a 99% hit rate.  I wrote
this using libcurl, but you could probably do something similar with a shell
script as was initially mentioned.

On Thu, May 12, 2011 at 4:49 PM, A E [Gmail] <all.eforums at gmail.com> wrote:

> On Mon, May 9, 2011 at 5:02 PM, Brad Mina <brad at tritelcomm.com> wrote:
>
>> You could setup an internal http server with a directory called 'audio' (
>> http://server.ip/audio/), then use mod_shout as Avi suggested like this:
>> <action application="shout" data="http://server.ip/audio/file.mp3"/>
>>
>>
>> +1 on "Interest" in this discussion. Additional point of failure argument
> aside, I'm very interested to know if anyone else has any experience to
> share re: streaming audio files/prompts etc from a remote FS via shout/Ice
> type of protocols/techniques rather than using a fileshare mount etc? Is
> there a preference of one over the other (SHOUT Vs ICE) or should I be
> reading the difference between a SHOUTcast and ICEcast before opening my
> mouth? :)
>
> Thx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 13 05:36:45 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 13 May 2011 09:36:45 +0800
Subject: [Freeswitch-users] call somebody into conference after
	conference established
In-Reply-To: <BANLkTikAzr-tUqiQYmRUU44bsWyN_y4LRg@mail.gmail.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<BANLkTikAzr-tUqiQYmRUU44bsWyN_y4LRg@mail.gmail.com>
Message-ID: <BANLkTik4Lqnk-gXqpNB40QSOBguFde-u6Q@mail.gmail.com>

Hi Michael,

Thanks for your reply .
Is it possible that when I press the phone number inside the conference the
FS call the cmd api as you wrote below . Thanks.

Regards,
Charles

? 2011-5-13 ??12:33?"Michael Collins" <msc at freeswitch.org>???
> You can add a caller from fs_cli with the "conference" command:
>
> conference <conf_name> dial sofia/external/foo at bar <caller id num> <caller
> id name>
>
> -MC
>
> On Wed, May 11, 2011 at 8:11 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Gurus,
>>
>> i dial 666 and enter a conference, then i need call some body's phone
>> number to join him into this conference...
>> is there anyone can advise how can realize this scenario?
>>
>> thanks.
>>
>> 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 msc at freeswitch.org  Fri May 13 11:31:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 00:31:21 -0700
Subject: [Freeswitch-users] call somebody into conference after
	conference established
In-Reply-To: <BANLkTik4Lqnk-gXqpNB40QSOBguFde-u6Q@mail.gmail.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<BANLkTikAzr-tUqiQYmRUU44bsWyN_y4LRg@mail.gmail.com>
	<BANLkTik4Lqnk-gXqpNB40QSOBguFde-u6Q@mail.gmail.com>
Message-ID: <BANLkTinfD=2Qznwz0Waw7V1EwptzusZLWw@mail.gmail.com>

2011/5/12 fieldpeak <fieldpeak at gmail.com>

> Hi Michael,
>
> Thanks for your reply .
> Is it possible that when I press the phone number inside the conference the
> FS call the cmd api as you wrote below . Thanks.
>

Yes, but I highly recommend that you use the "bind_digit_application" app
instead of the caller_controls in the conference profile. This is because if
you use the caller_controls then *anyone* in the conference can do the magic
dial-out thing if they know the DTMF sequence you've assigned. By applying
bind_digit_action (BDA) items to *your* call into the conference you are the
only one who has the ability to use the magic key sequence.

Here are some tips:
#1 - look up the BDA examples on the wiki so you get an idea of how they
work
#2 - remove at least one key from the caller controls in your conference
profile. The caller control keys will hijack any BDA assignments you've
made. Personally, I like to remove the * binding in the caller controls and
then use *1, *2, etc. as the key sequences in my BDA assignments
#3 - You'll need some sort of dialog to capture the digits used to dial out.
Let's say you assign *1 as the magic key combo. When you dial *1 your BDA
should do something like execute_extension or a lua script that lets you key
in the phone number to dial out. The script (or extension) will need to
execute the conference API to perform the dial out.

You'll need to tinker around with all these pieces in order to make them
into a working app. If you run into trouble email back and maybe in my spare
time (haha) I'll write up a sample Lua script & corresponding dialplan
entry.

-MC
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From admin at blindi.net  Fri May 13 17:46:56 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 13 May 2011 15:46:56 +0200 (CEST)
Subject: [Freeswitch-users] Problem menuselection not working with
 Play_and_get_digits Menuselection
In-Reply-To: <BANLkTikqTYP1z_G_6xjvGC=CybQMdpficg@mail.gmail.com>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
	<Pine.LNX.4.64.1105112328540.5984@debian>
	<BANLkTikqTYP1z_G_6xjvGC=CybQMdpficg@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105131533320.2636@debian>

Hi Michael,
> This is a classic case of dialplan parsing vs. executing. I'll let you read
> chapters 5 and 8 of the FreeSWITCH book to get a better understanding of the
> difference between those two.

My Problem: I working under Linux textbased only. I can.t read a pdf 
format under this console.
A blind user have manny barrieres, to accessability programs. Accrobat 
readers working only under a graffic gui, and can.t be use in textmode.
I have a brailledisplay to output only asciicharacters.

I don.t unserstand a not readable book for me.
I nice help is a html-version in text only. Picturs are not readable.

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 leonardo.bidinoto at voicetechnology.com.br  Fri May 13 18:32:04 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Fri, 13 May 2011 11:32:04 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
Message-ID: <BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>

Hi Michael,

Just succeeded to reproduce the problem.

The condition is: when a channel inside a conference is using a ESL
connection(lets call it "A") through socket application and another ESL
connection(lets call it "B") executes a command with this channel, the "B"
ESL connection will wait the "A" ESL connection close to execute its
command.
If the channel hangs up before the "A" ESL connection is closed, then "B"
ESL command will never be executed and the stucked channel will still be
there, into sofia and the conference too.
To verify that, just do "show channels" and "conference list". with
"uuid_exists" command, return "false".

Here are the actions done by the channel before get stucked:

16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321 [NOTICE]
switch_channel.c:816 New Channel
sofia/external/1000123402 at 192.168.0.154[16e09413-9cb0-4011-a635-f91933a35c0f]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state
[received][100]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia.c:4772 Remote SDP:
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia_glue.c:4656 Audio Codec Compare
[BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia_glue.c:4656 Audio Codec Compare
[BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia_glue.c:4656 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia_glue.c:2788 Set Codec
sofia/external/1000123402 at 192.168.0.154PCMU/8000 20 ms 160 samples
64000 bits
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change CS_NEW
-> CS_INIT
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_session.c:1114 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
Running State Change CS_INIT
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
State INIT
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change
CS_INIT -> CS_ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_session.c:1114 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
State INIT going to sleep
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
Running State Change CS_ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154) Callstate
Change DOWN -> RINGING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
State ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
switch_core_state_machine.c:77
sofia/external/1000123402 at 192.168.0.154Standard ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO]
mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
public
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true) =~
/^true$/ break=on-false
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~
/^true$/ break=on-false
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 parsing [public->public_extensions] continue=false
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
destination_number(1234567890) =~ /^(\d*)$/ break=on-false
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
State Change CS_ROUTING -> CS_EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
switch_core_session.c:1114 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
State ROUTING going to sleep
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
Running State Change CS_EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
State EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_core_state_machine.c:157
sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 transfer(1234567890 XML default)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State Change
CS_EXECUTE -> CS_ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_core_session.c:1114 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_core_session.c:707 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [NOTICE]
switch_ivr.c:1603 Transfer sofia/external/1000123402 at 192.168.0.154 to
XML[1234567890 at default]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
State EXECUTE going to sleep
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
Running State Change CS_ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
State ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
switch_core_state_machine.c:77
sofia/external/1000123402 at 192.168.0.154Standard ROUTING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO]
mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
default
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 parsing [default->flex] continue=false
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Regex (PASS) [flex] destination_number(1234567890)
=~ /^(\d+)$/ break=on-false
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Action log(INFO VOICE received dest=1234567890)
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Action set(playback_terminators=#)
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall we?)
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Action
set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
State Change CS_ROUTING -> CS_EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
switch_core_session.c:1114 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
State ROUTING going to sleep
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
Running State Change CS_EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
State EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
switch_core_state_machine.c:157
sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO]
mod_dptools.c:1184 VOICE received dest=1234567890
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 set(playback_terminators=#)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
[playback_terminators]=[#]
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO]
mod_dptools.c:1184 Let's do some ivrd, shall we?
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
[ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 socket(localhost:8084 full)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
answer()
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 answer()
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154]
192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
sofia_glue.c:3284 Set 2833 dtmf send payload to 101
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
switch_core_session.c:707 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154) Callstate
Change RINGING -> ACTIVE
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [NOTICE]
mod_dptools.c:930 Channel [sofia/external/1000123402 at 192.168.0.154] has been
answered
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG]
sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state
[completed][200]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG]
sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state
[ready][200]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav
flex_digits 5000 )
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 read(1 1
/usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF 1:1120
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
[flex_digits]=[UNDEF]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
read(1 1
/usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
flex_digits 5000 )
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 read(1 1
/usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
flex_digits 5000 )
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF 8:640
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
[flex_digits]=[UNDEF]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
flex_digits 5000 #,*)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 read(1 11
/usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
#,*)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF #:960
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
[flex_digits]=[UNDEF]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF #:800
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
mod_dptools.c:1664 Digit #
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
conference(15646 at teste+flags{waste})
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec
L16:70
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
switch_core_session.c:707 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF *:960
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
previous codec PCMU:0.
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
flex_digits 2000 )
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 read(1 1
/usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF 1:1120
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
[flex_digits]=[UNDEF]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink
session from object
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec
L16:70
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF *:800
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
previous codec PCMU:0.
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
====================================================================================================================================================
While Inside this connection, a "conference 15646 kick [member_id of this
channels]" command is executed by a fs_cli console and get stuck while
waiting response.
====================================================================================================================================================
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
flex_digits 2000 )
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 read(1 1
/usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG]
switch_ivr_play_say.c:1649 done playing file
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG]
switch_rtp.c:3280 RTP RECV DTMF 1:960
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
1000123402 at 192.168.0.154 set(flex_digits)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
[flex_digits]=[UNDEF]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154) Callstate
Change ACTIVE -> HANGUP
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [NOTICE]
sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154 [CS_EXECUTE]
[NORMAL_CLEARING]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
switch_channel.c:2576 Send signal sofia/external/1000123402 at 192.168.0.154[KILL]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
switch_core_session.c:1114 Send signal sofia/external/
1000123402 at 192.168.0.154 [BREAK]
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink
session from object
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec
L16:70
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG]
mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip receive
message [UNBRIDGE] (channel is hungup already)

I hope this info helps.

2011/5/12 Michael Collins <msc at freeswitch.org>

>
>
> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
> leonardo.bidinoto at voicetechnology.com.br> wrote:
>
>> Hi Michael,
>>
>> Im not using to any cdr module.
>
>
> I would recommend that you do several things:
>
> #1 - update to latest git
> #2 - rotate logs
> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
> #4 - reproduce the symptom with a single call (if possible)
> #5 - pastebin the log for the uuid in question and link to it in this
> thread
>
> From there hopefully we'll get a clue as to what is happening.
> -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
>
>


-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From msc at freeswitch.org  Fri May 13 19:08:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 08:08:53 -0700
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
Message-ID: <BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>

Pastebin this info and select "FreeSWITCH Log" as the syntax highlighting. I
need the colorized output to read logs. (I'm getting older and it's hard for
me to ready black and white in an email.)

-MC

On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <
leonardo.bidinoto at voicetechnology.com.br> wrote:

> Hi Michael,
>
> Just succeeded to reproduce the problem.
>
> The condition is: when a channel inside a conference is using a ESL
> connection(lets call it "A") through socket application and another ESL
> connection(lets call it "B") executes a command with this channel, the "B"
> ESL connection will wait the "A" ESL connection close to execute its
> command.
> If the channel hangs up before the "A" ESL connection is closed, then "B"
> ESL command will never be executed and the stucked channel will still be
> there, into sofia and the conference too.
> To verify that, just do "show channels" and "conference list". with
> "uuid_exists" command, return "false".
>
> Here are the actions done by the channel before get stucked:
>
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321 [NOTICE]
> switch_channel.c:816 New Channel sofia/external/1000123402 at 192.168.0.154[16e09413-9cb0-4011-a635-f91933a35c0f]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
> state [received][100]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia.c:4772 Remote SDP:
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia_glue.c:4656 Audio Codec Compare
> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia_glue.c:4656 Audio Codec Compare
> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia_glue.c:4656 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia_glue.c:2788 Set Codec sofia/external/1000123402 at 192.168.0.154PCMU/8000 20 ms 160 samples 64000 bits
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change CS_NEW
> -> CS_INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_session.c:1114 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
> Running State Change CS_INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
> State INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change
> CS_INIT -> CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_session.c:1114 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
> State INIT going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
> Running State Change CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154) Callstate
> Change DOWN -> RINGING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
> State ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO]
> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
> public
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true) =~
> /^true$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~
> /^true$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 parsing [public->public_extensions]
> continue=false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
> destination_number(1234567890) =~ /^(\d*)$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
> switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
> State Change CS_ROUTING -> CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
> switch_core_session.c:1114 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
> State ROUTING going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
> Running State Change CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
> State EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 transfer(1234567890 XML default)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State Change
> CS_EXECUTE -> CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_core_session.c:1114 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_core_session.c:707 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [NOTICE]
> switch_ivr.c:1603 Transfer sofia/external/1000123402 at 192.168.0.154 to
> XML[1234567890 at default]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
> State EXECUTE going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
> Running State Change CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
> State ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO]
> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
> default
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 parsing [default->flex] continue=false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Regex (PASS) [flex]
> destination_number(1234567890) =~ /^(\d+)$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Action log(INFO VOICE received dest=1234567890)
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Action set(playback_terminators=#)
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall we?)
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Action
> set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
> 1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
> switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
> State Change CS_ROUTING -> CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
> switch_core_session.c:1114 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
> State ROUTING going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
> Running State Change CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
> State EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO]
> mod_dptools.c:1184 VOICE received dest=1234567890
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 set(playback_terminators=#)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
> [playback_terminators]=[#]
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO]
> mod_dptools.c:1184 Let's do some ivrd, shall we?
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
> [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 socket(localhost:8084 full)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> answer()
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 answer()
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
> sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154]
> 192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
> switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
> sofia_glue.c:3284 Set 2833 dtmf send payload to 101
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
> sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
> mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
> switch_core_session.c:707 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
> switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154) Callstate
> Change RINGING -> ACTIVE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [NOTICE]
> mod_dptools.c:930 Channel [sofia/external/1000123402 at 192.168.0.154] has
> been answered
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG]
> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
> state [completed][200]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG]
> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
> state [ready][200]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav
> flex_digits 5000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 read(1 1
> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
> )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF 1:1120
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
> [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> read(1 1
> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
> flex_digits 5000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 read(1 1
> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
> flex_digits 5000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF 8:640
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
> [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
> flex_digits 5000 #,*)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 read(1 11
> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
> #,*)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF #:960
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
> [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF #:800
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
> mod_dptools.c:1664 Digit #
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> conference(15646 at teste+flags{waste})
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
> mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1 channel
> 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
> mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1 channel
> 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec
> L16:70
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
> switch_core_session.c:707 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
> mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF *:960
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
> previous codec PCMU:0.
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
> flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 read(1 1
> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF 1:1120
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
> [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink
> session from object
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec
> L16:70
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF *:800
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
> previous codec PCMU:0.
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>
> ====================================================================================================================================================
> While Inside this connection, a "conference 15646 kick [member_id of this
> channels]" command is executed by a fs_cli console and get stuck while
> waiting response.
>
> ====================================================================================================================================================
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
> flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 read(1 1
> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG]
> switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG]
> switch_rtp.c:3280 RTP RECV DTMF 1:960
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
> set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
> 1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
> [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
> switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154) Callstate
> Change ACTIVE -> HANGUP
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [NOTICE]
> sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154 [CS_EXECUTE]
> [NORMAL_CLEARING]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
> switch_channel.c:2576 Send signal sofia/external/1000123402 at 192.168.0.154[KILL]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
> switch_core_session.c:1114 Send signal sofia/external/
> 1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink
> session from object
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec
> L16:70
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
> mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG]
> mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip receive
> message [UNBRIDGE] (channel is hungup already)
>
> I hope this info helps.
>
> 2011/5/12 Michael Collins <msc at freeswitch.org>
>
>>
>>
>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>
>>> Hi Michael,
>>>
>>> Im not using to any cdr module.
>>
>>
>> I would recommend that you do several things:
>>
>> #1 - update to latest git
>> #2 - rotate logs
>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>> #4 - reproduce the symptom with a single call (if possible)
>> #5 - pastebin the log for the uuid in question and link to it in this
>> thread
>>
>> From there hopefully we'll get a clue as to what is happening.
>> -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
>>
>>
>
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri May 13 19:15:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 08:15:18 -0700
Subject: [Freeswitch-users] Problem menuselection not working with
 Play_and_get_digits Menuselection
In-Reply-To: <Pine.LNX.4.64.1105131533320.2636@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
	<Pine.LNX.4.64.1105112328540.5984@debian>
	<BANLkTikqTYP1z_G_6xjvGC=CybQMdpficg@mail.gmail.com>
	<Pine.LNX.4.64.1105131533320.2636@debian>
Message-ID: <BANLkTikUJ6+_Wnz37AKmkM390rGA0en4wA@mail.gmail.com>

Thomas,

Thank you for telling us that you are blind and use screen readers and other
accessibility programs. Unfortunately Packt Publishing does not suport
anything other than PDF. You may wish to talk to Tayeb Meftah about what he
uses to read PDFs. (I know for a fact that he has read the FreeSWITCH book
because he has given me feedback about it.)

In the meantime, let us know if you get your conference and
bind_digit_action stuff working.

-MC

On Fri, May 13, 2011 at 6:46 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi Michael,
>
>  This is a classic case of dialplan parsing vs. executing. I'll let you
>> read
>> chapters 5 and 8 of the FreeSWITCH book to get a better understanding of
>> the
>> difference between those two.
>>
>
> My Problem: I working under Linux textbased only. I can.t read a pdf format
> under this console.
> A blind user have manny barrieres, to accessability programs. Accrobat
> readers working only under a graffic gui, and can.t be use in textmode.
> I have a brailledisplay to output only asciicharacters.
>
>
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From infos at madovsky.org  Fri May 13 19:25:27 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 13 May 2011 11:25:27 -0400
Subject: [Freeswitch-users] Problem menuselection not working with
	Play_and_get_digits Menuselection
References: <807770.34642.qm@web30502.mail.mud.yahoo.com><E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr><Pine.LNX.4.64.1105060307050.9252@debian><BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com><Pine.LNX.4.64.1105092235440.7827@debian><BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com><Pine.LNX.4.64.1105110027460.8977@debian><Pine.LNX.4.64.1105112328540.5984@debian><BANLkTikqTYP1z_G_6xjvGC=CybQMdpficg@mail.gmail.com><Pine.LNX.4.64.1105131533320.2636@debian>
	<BANLkTikUJ6+_Wnz37AKmkM390rGA0en4wA@mail.gmail.com>
Message-ID: <CCCAD6A8384543E58672CED49F82E066@e1705>

you can convert PDF to TEXT on line here http://www.convertpdftotext.net/
but I'm sure there are plenty other websites and tools that do it 
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, May 13, 2011 11:15 AM
  Subject: Re: [Freeswitch-users] Problem menuselection not working with Play_and_get_digits Menuselection


  Thomas,


  Thank you for telling us that you are blind and use screen readers and other accessibility programs. Unfortunately Packt Publishing does not suport anything other than PDF. You may wish to talk to Tayeb Meftah about what he uses to read PDFs. (I know for a fact that he has read the FreeSWITCH book because he has given me feedback about it.)


  In the meantime, let us know if you get your conference and bind_digit_action stuff working.


  -MC


  On Fri, May 13, 2011 at 6:46 AM, Thomas Hoellriegel <admin at blindi.net> wrote:

    Hi Michael,


      This is a classic case of dialplan parsing vs. executing. I'll let you read
      chapters 5 and 8 of the FreeSWITCH book to get a better understanding of the
      difference between those two.



    My Problem: I working under Linux textbased only. I can.t read a pdf format under this console.
    A blind user have manny barrieres, to accessability programs. Accrobat readers working only under a graffic gui, and can.t be use in textmode.
    I have a brailledisplay to output only asciicharacters.






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From bracken_dave at yahoo.com  Fri May 13 20:12:29 2011
From: bracken_dave at yahoo.com (Dave Bracken)
Date: Fri, 13 May 2011 09:12:29 -0700 (PDT)
Subject: [Freeswitch-users] SIP version of TBCT. Please help.
Message-ID: <891004.13083.qm@web114508.mail.gq1.yahoo.com>

does anyone know what the SIP equivalent is to "2 b channel transfer". taking an 
inbound and an outbound, drop operator out, and pass the call up to the telco?

Thanks,
Dave
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From Nabble at slickdeals.endjunk.com  Fri May 13 20:14:49 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 13 May 2011 09:14:49 -0700 (PDT)
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <8D844C3F-6541-4531-B272-A0935F717A3C@jerris.com>
References: <1305225047195-6356793.post@n2.nabble.com>
	<266B2D96-0E13-4091-93D0-14F69E07909F@jerris.com>
	<C9F1BDF8.181D9%krice@freeswitch.org>
	<8D844C3F-6541-4531-B272-A0935F717A3C@jerris.com>
Message-ID: <1305303289916-6359965.post@n2.nabble.com>


Michael Jerris wrote:
> 
> commit a8da1758cb2ec60b6f1b49d685caf13244c7eec5
> Author: Michael Jerris &lt;mike at jerris.com&gt;
> Date:   Thu May 12 18:21:34 2011 -0400
> 
>     FS-2936: attempt to fix the platform that I'll never have to fix
That git commit causes the compilation in an endless loop (I had to press a
ctrl-c to break the compilation) as shown below:
OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
-DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
-fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
-I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-I. -I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
.deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
.libs/switch_cpp.o
OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
-DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
-I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
-fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
-I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-I. -I./lua
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
-I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
-DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
.deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
make[6]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[7]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[8]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[9]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[10]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[11]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[12]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[13]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[14]: Entering directory
`/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
^Cmake[1]: *** [package/feeds/local/freeswitch_git/compile] Error 130
make: *** [package/freeswitch_git/compile] Interrupt


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation-on-latest-git-tp6356793p6359965.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri May 13 20:17:57 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 13 May 2011 17:17:57 +0100
Subject: [Freeswitch-users] SIP version of TBCT. Please help.
In-Reply-To: <891004.13083.qm@web114508.mail.gq1.yahoo.com>
References: <891004.13083.qm@web114508.mail.gq1.yahoo.com>
Message-ID: <BANLkTik8Ex49bzk0NivXRd9F0P8YYi7mRw@mail.gmail.com>

It means doing a reINVITE that pulls you out of the signalling/media path.
FS can do it:

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_simplify
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_auto_simplify

Not sure if there's a term for it in SIP, but it sounds like this is the
same thing.

If they're actually on the call to them then I guess you can unbridge,
bridge them to the other parked party, then do the simplify.

-Steve


On 13 May 2011 17:12, Dave Bracken <bracken_dave at yahoo.com> wrote:

> does anyone know what the SIP equivalent is to "2 b channel transfer".
> taking an inbound and an outbound, drop operator out, and pass the call up
> to the telco?
>
> Thanks,
> Dave
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri May 13 20:18:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 13 May 2011 17:18:35 +0100
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <1305303289916-6359965.post@n2.nabble.com>
References: <1305225047195-6356793.post@n2.nabble.com>
	<266B2D96-0E13-4091-93D0-14F69E07909F@jerris.com>
	<C9F1BDF8.181D9%krice@freeswitch.org>
	<8D844C3F-6541-4531-B272-A0935F717A3C@jerris.com>
	<1305303289916-6359965.post@n2.nabble.com>
Message-ID: <BANLkTike+Drw4rCB8Jgtu81kVAytpBe_FA@mail.gmail.com>

Jira.

On 13 May 2011 17:14, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Michael Jerris wrote:
> >
> > commit a8da1758cb2ec60b6f1b49d685caf13244c7eec5
> > Author: Michael Jerris &lt;mike at jerris.com&gt;
> > Date:   Thu May 12 18:21:34 2011 -0400
> >
> >     FS-2936: attempt to fix the platform that I'll never have to fix
> That git commit causes the compilation in an endless loop (I had to press a
> ctrl-c to break the compilation) as shown below:
> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
> .libs/switch_cpp.o
> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
> make[6]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[7]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[8]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[9]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[10]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[11]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[12]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[13]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[14]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> ^Cmake[1]: *** [package/feeds/local/freeswitch_git/compile] Error 130
> make: *** [package/freeswitch_git/compile] Interrupt
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation-on-latest-git-tp6356793p6359965.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 steveayre at gmail.com  Fri May 13 20:20:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 13 May 2011 17:20:38 +0100
Subject: [Freeswitch-users] Problem menuselection not working with
 Play_and_get_digits Menuselection
In-Reply-To: <CCCAD6A8384543E58672CED49F82E066@e1705>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
	<Pine.LNX.4.64.1105112328540.5984@debian>
	<BANLkTikqTYP1z_G_6xjvGC=CybQMdpficg@mail.gmail.com>
	<Pine.LNX.4.64.1105131533320.2636@debian>
	<BANLkTikUJ6+_Wnz37AKmkM390rGA0en4wA@mail.gmail.com>
	<CCCAD6A8384543E58672CED49F82E066@e1705>
Message-ID: <BANLkTimm12pnej+RyiQPqL9AYmC6bnkmaA@mail.gmail.com>

I believe that Ghostscript can also convert PDF to Text and can be run from
the commandline.

-Steve


On 13 May 2011 16:25, Madovsky <infos at madovsky.org> wrote:

>  you can convert PDF to TEXT on line here http://www.convertpdftotext.net/
> but I'm sure there are plenty other websites and tools that do it
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, May 13, 2011 11:15 AM
> *Subject:* Re: [Freeswitch-users] Problem menuselection not working with
> Play_and_get_digits Menuselection
>
> Thomas,
>
> Thank you for telling us that you are blind and use screen readers and
> other accessibility programs. Unfortunately Packt Publishing does not suport
> anything other than PDF. You may wish to talk to Tayeb Meftah about what he
> uses to read PDFs. (I know for a fact that he has read the FreeSWITCH book
> because he has given me feedback about it.)
>
> In the meantime, let us know if you get your conference and
> bind_digit_action stuff working.
>
> -MC
>
> On Fri, May 13, 2011 at 6:46 AM, Thomas Hoellriegel <admin at blindi.net>wrote:
>
>> Hi Michael,
>>
>> This is a classic case of dialplan parsing vs. executing. I'll let you
>>> read
>>> chapters 5 and 8 of the FreeSWITCH book to get a better understanding of
>>> the
>>> difference between those two.
>>>
>>
>> My Problem: I working under Linux textbased only. I can.t read a pdf
>> format under this console.
>> A blind user have manny barrieres, to accessability programs. Accrobat
>> readers working only under a graffic gui, and can.t be use in textmode.
>> I have a brailledisplay to output only asciicharacters.
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu May 12 11:32:10 2011
From: liuyp2 at asiainfo-linkage.com (=?utf-8?B?bGl1eXAy?=)
Date: Thu, 12 May 2011 15:32:10 +0800
Subject: [Freeswitch-users]
	=?utf-8?q?call_somebody_into_conference_after_?=
	=?utf-8?q?conferenceestablished?=
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
Message-ID: <201105121532017963789@asiainfo-linkage.com>

conference Your-Conf-Name dial user/1002




liuyp2
2011-05-12



???? fieldpeak
????? 2011-05-12 11:11:41
???? FreeSWITCH-users
??? 
??? [Freeswitch-users] call somebody into conference after conferenceestablished

Hi Gurus,

i dial 666 and enter a conference, then i need call some body's phone number to join him into this conference...
is there anyone can advise how can realize this scenario? 

thanks.

Regards,
Charles
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From alfred.stainer at gmail.com  Fri May 13 19:50:47 2011
From: alfred.stainer at gmail.com (Alfred Stainer)
Date: Fri, 13 May 2011 17:50:47 +0200
Subject: [Freeswitch-users] Need help on a calling card service
Message-ID: <BANLkTi=nrJMHcJ+eqeYAAhvmTbWtNZ7pmQ@mail.gmail.com>

Hi all, we are developing a calling card service based on Freeswitch and we
have the necessity to implement the following features but we have not found
a way to do that:
we want that the first 10 seconds of the call are free of charge for the
caller.
This is not only a billing issue because most of our customers don't call a
free number so we want that the connect to the caller is sent only after 10
seconds we receive the connect from called.
We bridge the inbound leg with the outbound leg but we are not capable to
delay the propagation of the connect.
The perfect solution will be that when we receive the connect from outgoing
leg we send the preanswer on incoming leg and 10 second after that we send
the connect.
There is a way to do that? Any advice?

Alfred
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From msc at freeswitch.org  Fri May 13 20:58:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 09:58:50 -0700
Subject: [Freeswitch-users] libcurl fails cross compilation on latest git
In-Reply-To: <1305303289916-6359965.post@n2.nabble.com>
References: <1305225047195-6356793.post@n2.nabble.com>
	<266B2D96-0E13-4091-93D0-14F69E07909F@jerris.com>
	<C9F1BDF8.181D9%krice@freeswitch.org>
	<8D844C3F-6541-4531-B272-A0935F717A3C@jerris.com>
	<1305303289916-6359965.post@n2.nabble.com>
Message-ID: <BANLkTimchAw-9z6NkdFyBSHzLsyvkLPs5A@mail.gmail.com>

User "mazilo" has been moderated. Let this be a warning: I *will* be
moderating people who post bugs and bug discussions to the mailing list.

-MC

On Fri, May 13, 2011 at 9:14 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> Michael Jerris wrote:
> >
> > commit a8da1758cb2ec60b6f1b49d685caf13244c7eec5
> > Author: Michael Jerris &lt;mike at jerris.com&gt;
> > Date:   Thu May 12 18:21:34 2011 -0400
> >
> >     FS-2936: attempt to fix the platform that I'll never have to fix
> That git commit causes the compilation in an endless loop (I had to press a
> ctrl-c to break the compilation) as shown below:
> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp  -fPIC -DPIC -o
> .libs/switch_cpp.o
> OpenWrt-quiet_libtool: compile:  arm-openwrt-linux-uclibcgnueabi-g++
> -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/include
>
> -I/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/libs/libteletone/src
> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I.
> -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -I. -I./lua
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/target-arm_v5te_uClibc-0.9.32_eabi/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/usr/include
>
> -I/opt/openwrt-svn-trunk/staging_dir/toolchain-arm_v5te_gcc-linaro_uClibc-0.9.32_eabi/include
> -DLUA_USE_LINUX -fpic -Wno-format -MT switch_cpp.lo -MD -MP -MF
> .deps/switch_cpp.Tpo -c src/switch_cpp.cpp -o switch_cpp.o >/dev/null 2>&1
> make[6]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[7]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[8]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[9]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[10]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[11]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[12]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[13]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[14]: Entering directory
>
> `/opt/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> ^Cmake[1]: *** [package/feeds/local/freeswitch_git/compile] Error 130
> make: *** [package/freeswitch_git/compile] Interrupt
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/libcurl-fails-cross-compilation-on-latest-git-tp6356793p6359965.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 eagle.antonio at gmail.com  Fri May 13 21:01:51 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Fri, 13 May 2011 17:01:51 +0000
Subject: [Freeswitch-users] ESL Command Not Found
Message-ID: <BANLkTi=q0-sUZyxb0AsJsyhonjE8Z-tuoQ@mail.gmail.com>

Hello

I'm using ESL to Fork a call inside another , this is the commom example of
User (A) pressing 1 and calling another party (B).

I'm forking the call by opening an inbound socket and issuing the command

api('originate',"{step=1,origination_caller_id_number=XXXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s}sofia/external/%s at 2XXXXXXX&socket(
192.168.0.12:8040 sync full)


All works great ( i get the call) and as you can see the call to B is
redirected to the IVR Server

The problem is if i do on the channel B ( By esl)
uuid_getvar step XXXXXXXXXXXXXXXXXXXXXXXXXX

I get
Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: -ERR%20command%20not%20found

Has anyone faced this ??
BTW using the python ESL:

Best Regards
Antonio Teixeira
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From msc at freeswitch.org  Fri May 13 21:02:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 10:02:40 -0700
Subject: [Freeswitch-users] SIP version of TBCT. Please help.
In-Reply-To: <891004.13083.qm@web114508.mail.gq1.yahoo.com>
References: <891004.13083.qm@web114508.mail.gq1.yahoo.com>
Message-ID: <BANLkTimU9Y0mSN-ShnQsJcpXTO504byUcQ@mail.gmail.com>

Get the SIP trace we discussed in IRC. Put it into
pastebin.freeswitch.orgso we can all check it out. Once we know how
your current software is doing
it we can see if FS can do the same thing.
-MC

On Fri, May 13, 2011 at 9:12 AM, Dave Bracken <bracken_dave at yahoo.com>wrote:

> does anyone know what the SIP equivalent is to "2 b channel transfer".
> taking an inbound and an outbound, drop operator out, and pass the call up
> to the telco?
>
> Thanks,
> Dave
>
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From msc at freeswitch.org  Fri May 13 21:04:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 10:04:15 -0700
Subject: [Freeswitch-users] ESL Command Not Found
In-Reply-To: <BANLkTi=q0-sUZyxb0AsJsyhonjE8Z-tuoQ@mail.gmail.com>
References: <BANLkTi=q0-sUZyxb0AsJsyhonjE8Z-tuoQ@mail.gmail.com>
Message-ID: <BANLkTi=O8SGiY=4Y_eUX3pH6KaMo0L3dOA@mail.gmail.com>

haha

uuid_getvar <uuid> <var_name>

-MC

On Fri, May 13, 2011 at 10:01 AM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Hello
>
> I'm using ESL to Fork a call inside another , this is the commom example of
> User (A) pressing 1 and calling another party (B).
>
> I'm forking the call by opening an inbound socket and issuing the command
>
>
> api('originate',"{step=1,origination_caller_id_number=XXXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s}sofia/external/%s at 2XXXXXXX&socket(
> 192.168.0.12:8040 sync full)
>
>
> All works great ( i get the call) and as you can see the call to B is
> redirected to the IVR Server
>
> The problem is if i do on the channel B ( By esl)
> uuid_getvar step XXXXXXXXXXXXXXXXXXXXXXXXXX
>
> I get
> Event-Name: SOCKET_DATA
> Content-Type: command/reply
> Reply-Text: -ERR%20command%20not%20found
>
> Has anyone faced this ??
> BTW using the python ESL:
>
> Best Regards
> Antonio Teixeira
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Fri May 13 21:39:53 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 13 May 2011 10:39:53 -0700
Subject: [Freeswitch-users] ESL Command Not Found
In-Reply-To: <BANLkTi=O8SGiY=4Y_eUX3pH6KaMo0L3dOA@mail.gmail.com>
References: <BANLkTi=q0-sUZyxb0AsJsyhonjE8Z-tuoQ@mail.gmail.com>
	<BANLkTi=O8SGiY=4Y_eUX3pH6KaMo0L3dOA@mail.gmail.com>
Message-ID: <BANLkTi=-Ftkf42wv-aMogFhWM-=-kOW4sg@mail.gmail.com>

Ah yes, the dreaded cup of coffee is needed...

On Fri, May 13, 2011 at 10:04 AM, Michael Collins <msc at freeswitch.org> wrote:
> haha
> uuid_getvar <uuid> <var_name>
> -MC
>
> On Fri, May 13, 2011 at 10:01 AM, Antonio Teixeira <eagle.antonio at gmail.com>
> wrote:
>>
>> Hello
>>
>> I'm using ESL to Fork a call inside another , this is the commom example
>> of User (A) pressing 1 and calling another party (B).
>>
>> I'm forking the call by opening an inbound socket and issuing the command
>>
>>
>> api('originate',"{step=1,origination_caller_id_number=XXXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s}sofia/external/%s at 2XXXXXXX
>> &socket(192.168.0.12:8040 sync full)
>>
>>
>> All works great ( i get the call) and as you can see the call to B is
>> redirected to the IVR Server
>>
>> The problem is if i do on the channel B ( By esl)
>> uuid_getvar step XXXXXXXXXXXXXXXXXXXXXXXXXX
>>
>> I get
>> Event-Name: SOCKET_DATA
>> Content-Type: command/reply
>> Reply-Text: -ERR%20command%20not%20found
>>
>> Has anyone faced this ??
>> BTW using the python ESL:
>>
>> Best Regards
>> Antonio Teixeira
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anton.vazir at gmail.com  Fri May 13 21:50:59 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 13 May 2011 22:50:59 +0500
Subject: [Freeswitch-users] Need help on a calling card service
In-Reply-To: <BANLkTi=nrJMHcJ+eqeYAAhvmTbWtNZ7pmQ@mail.gmail.com>
References: <BANLkTi=nrJMHcJ+eqeYAAhvmTbWtNZ7pmQ@mail.gmail.com>
Message-ID: <BANLkTinngoagb_hUw74RrKL_=bZAwZuOGA@mail.gmail.com>

I do that in my billing.

2011/5/13 Alfred Stainer <alfred.stainer at gmail.com>:
> Hi all, we are developing a calling card service based on Freeswitch and we
> have the necessity to implement the following features but we have not found
> a way to do that:
> we want that the first 10 seconds of the call are free of charge for the
> caller.
> This is not only a billing issue because most of our customers don't call a
> free number so we want that the connect to the caller is sent only after 10
> seconds we receive the connect from called.
> We bridge the inbound leg with the outbound leg but we are not capable to
> delay the propagation of the connect.
> The perfect solution will be that when we receive the connect from outgoing
> leg we send the preanswer on incoming leg and 10 second after that we send
> the connect.
> There is a way to do that? Any advice?
> Alfred
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From covici at ccs.covici.com  Fri May 13 22:22:00 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 13 May 2011 14:22:00 -0400
Subject: [Freeswitch-users] Problem menuselection not working with
	Play_and_get_digits Menuselection
In-Reply-To: <Pine.LNX.4.64.1105131533320.2636@debian>
References: <807770.34642.qm@web30502.mail.mud.yahoo.com>
	<E6C5B092-52B1-40F2-B327-ADEE95C90906@ipeva.fr>
	<Pine.LNX.4.64.1105060307050.9252@debian>
	<BANLkTinxBPe2iVD__fMVLZ-JeoL-5h0QqQ@mail.gmail.com>
	<Pine.LNX.4.64.1105092235440.7827@debian>
	<BANLkTi=n8p9n5b6grzJ6kOhPeQRbbkFemQ@mail.gmail.com>
	<Pine.LNX.4.64.1105110027460.8977@debian>
	<Pine.LNX.4.64.1105112328540.5984@debian>
	<BANLkTikqTYP1z_G_6xjvGC=CybQMdpficg@mail.gmail.com>
	<Pine.LNX.4.64.1105131533320.2636@debian>
Message-ID: <8814.1305310920@ccs.covici.com>

Emacs will read the pdf, or there are some conversion utilities to
convert it to text.  If you buy the book, a text copy can be sent.

Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi Michael,
> > This is a classic case of dialplan parsing vs. executing. I'll let you read
> > chapters 5 and 8 of the FreeSWITCH book to get a better understanding of the
> > difference between those two.
> 
> My Problem: I working under Linux textbased only. I can.t read a pdf
> format under this console.
> A blind user have manny barrieres, to accessability programs. Accrobat
> readers working only under a graffic gui, and can.t be use in
> textmode.
> I have a brailledisplay to output only asciicharacters.
> 
> I don.t unserstand a not readable book for me.
> I nice help is a html-version in text only. Picturs are not readable.
> 
> 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
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 admin at blindi.net  Fri May 13 22:50:36 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 13 May 2011 20:50:36 +0200 (CEST)
Subject: [Freeswitch-users] question how to cut a string from a variable
 in fs?
In-Reply-To: <BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105132046460.5405@debian>

Hi Michael,
Sorry, i don.t find a command to cut digits for example:
I like to remove a sipheader in a variable. cut all strings afer the @ 
character, und remove the @ character.
I search on:
http://wiki.freeswitch.org/wiki/Regular_Expression
A description can.t be fine.
Do you have a string please? thank you


---------------
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 infos at madovsky.org  Fri May 13 22:57:54 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 13 May 2011 14:57:54 -0400
Subject: [Freeswitch-users] question how to cut a string from a variable
	in fs?
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com><Pine.LNX.4.64.1105120530140.9184@debian><BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
Message-ID: <51E0B1392ADF44069FCB8537C2AE75B1@e1705>

if you do
expresion="^(\whatever)@(whatever)$"
$1 will be the id
@ will be ignored
$2 the ip/domain/hostname

hope this helps

----- Original Message ----- 
From: "Thomas Hoellriegel" <admin at blindi.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 13, 2011 2:50 PM
Subject: Re: [Freeswitch-users] question how to cut a string from a variable 
in fs?


Hi Michael,
Sorry, i don.t find a command to cut digits for example:
I like to remove a sipheader in a variable. cut all strings afer the @
character, und remove the @ character.
I search on:
http://wiki.freeswitch.org/wiki/Regular_Expression
A description can.t be fine.
Do you have a string please? thank you


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



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


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



From steveayre at gmail.com  Fri May 13 23:05:51 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 13 May 2011 20:05:51 +0100
Subject: [Freeswitch-users] question how to cut a string from a variable
 in fs?
In-Reply-To: <51E0B1392ADF44069FCB8537C2AE75B1@e1705>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
Message-ID: <BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>

Or:
<action application="set" data="after=${regex(${before}|^.*?@|%1"/>

before = abc at def.com
gives:
after = abc

-Steve

On 13 May 2011 19:57, Madovsky <infos at madovsky.org> wrote:
>
> if you do
> expresion="^(\whatever)@(whatever)$"
> $1 will be the id
> @ will be ignored
> $2 the ip/domain/hostname
>
> hope this helps
>
> ----- Original Message -----
> From: "Thomas Hoellriegel" <admin at blindi.net>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, May 13, 2011 2:50 PM
> Subject: Re: [Freeswitch-users] question how to cut a string from a variable
> in fs?
>
>
> Hi Michael,
> Sorry, i don.t find a command to cut digits for example:
> I like to remove a sipheader in a variable. cut all strings afer the @
> character, und remove the @ character.
> I search on:
> http://wiki.freeswitch.org/wiki/Regular_Expression
> A description can.t be fine.
> Do you have a string please? thank you
>
>
> ---------------
> 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
>
>
>
> --------------------------------------------------------------------------------
>
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Fri May 13 23:06:22 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 13 May 2011 20:06:22 +0100
Subject: [Freeswitch-users] question how to cut a string from a variable
 in fs?
In-Reply-To: <BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
Message-ID: <BANLkTikQ0-uRS0P3P4P87h21xnR=LFn97Q@mail.gmail.com>

Sorry, that should be:
<action application="set" data="after=${regex(${before}|^(.*?)@|%1"/>

-Steve


On 13 May 2011 20:05, Steven Ayre <steveayre at gmail.com> wrote:
> Or:
> <action application="set" data="after=${regex(${before}|^.*?@|%1"/>
>
> before = abc at def.com
> gives:
> after = abc
>
> -Steve
>
> On 13 May 2011 19:57, Madovsky <infos at madovsky.org> wrote:
>>
>> if you do
>> expresion="^(\whatever)@(whatever)$"
>> $1 will be the id
>> @ will be ignored
>> $2 the ip/domain/hostname
>>
>> hope this helps
>>
>> ----- Original Message -----
>> From: "Thomas Hoellriegel" <admin at blindi.net>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, May 13, 2011 2:50 PM
>> Subject: Re: [Freeswitch-users] question how to cut a string from a variable
>> in fs?
>>
>>
>> Hi Michael,
>> Sorry, i don.t find a command to cut digits for example:
>> I like to remove a sipheader in a variable. cut all strings afer the @
>> character, und remove the @ character.
>> I search on:
>> http://wiki.freeswitch.org/wiki/Regular_Expression
>> A description can.t be fine.
>> Do you have a string please? thank you
>>
>>
>> ---------------
>> 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
>>
>>
>>
>> --------------------------------------------------------------------------------
>>
>>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>


From grsingh750 at gmail.com  Sat May 14 00:37:48 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 14 May 2011 02:07:48 +0530
Subject: [Freeswitch-users] mod_freetdm not compiling on archlinux
	2.6.32-lts kernel x86_64
Message-ID: <BANLkTikyTkQsiCMRs8FVoO-yBCtO-5JS-g@mail.gmail.com>

Hi,

Trying to install FS with mod_freetdm.
Fails to compile. I have wanpipe installed
Error : http://pastebin.freeswitch.org/16296

Can anybody with a working setup on this platform comment on this please?

Thanks
guru


From m.sobkow at marketelsystems.com  Sat May 14 01:07:18 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 13 May 2011 15:07:18 -0600
Subject: [Freeswitch-users] Can UUID's change?
Message-ID: <4DCD9D86.3050302@marketelsystems.com>

Here's our scenario:

The operator logs in to our system, and has their UUID parked.

The system calls the customer when requested, which creates a UUID.

The operator and customer UUIDs are bridged.

We watch for events from the customer UUID to determine when the
customer has hung up their end of the call, which causes the operator to
be parked again.

However, I'm seeing some unrecognized UUIDs after bridging the call
(i.e. that weren't returned by the place call code.)

Are there any events which might indicate that a UUID is being changed
by the system, and that I should be watching for a new UUID instead?

-- 

Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/




From admin at blindi.net  Sat May 14 01:08:50 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 13 May 2011 23:08:50 +0200 (CEST)
Subject: [Freeswitch-users] problem exporting variables from shellsctipt not
	working
In-Reply-To: <BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105132228380.6270@debian>

Hi all, i have written a schellscript.
I can.t export variables from this script to Fs.
Is a special syntax needed?
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 banhi.dutra at gmail.com  Sat May 14 01:50:25 2011
From: banhi.dutra at gmail.com (Banhi Dutra)
Date: Fri, 13 May 2011 23:50:25 +0200
Subject: [Freeswitch-users] Different SAS with ZRTP
Message-ID: <BANLkTinsdQvkvgtuV8SEYOY7pP46TYYeow@mail.gmail.com>

Hi all,


I installed FS in test environment with ZRTP enabled. I have two softphone
(I tested with Acrobits and Jitsi) registered and all works fine, except for
SAS exchange, which are different. I'm using libzrtp 0.81.514 and
 FreeSWITCH Version 1.0.head (git-23d8658 2011-05-07 00-27-20 -0400). I read
older post about a similar problem. Could be a version problem ?


Any help will be appreciated


Thank you

Banhi
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From gabe at gundy.org  Sat May 14 01:56:38 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 15:56:38 -0600
Subject: [Freeswitch-users] Can UUID's change?
In-Reply-To: <4DCD9D86.3050302@marketelsystems.com>
References: <4DCD9D86.3050302@marketelsystems.com>
Message-ID: <BANLkTinM4iLdW8JSJsch8shPJ8Zzp8cXRA@mail.gmail.com>

On Fri, May 13, 2011 at 3:07 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> However, I'm seeing some unrecognized UUIDs after bridging the call
> (i.e. that weren't returned by the place call code.)
>
> Are there any events which might indicate that a UUID is being changed
> by the system, and that I should be watching for a new UUID instead?

The UUID of the channel never changes.  If you're using a UUID to
originate a call, then you'll have that UUID to work with, but FS will
still pick it's own UUID for the channel --and it will not change.

Calls get UUIDs, conferences get UUIDs, channels get UUIDs, so it's
just a matter of finding the right one and not worrying too much about
the other UUIDs that might show up.  Hope that helps.

Best,
Gabe


From gabe at gundy.org  Sat May 14 01:57:49 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 15:57:49 -0600
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <Pine.LNX.4.64.1105132228380.6270@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
	<Pine.LNX.4.64.1105132228380.6270@debian>
Message-ID: <BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>

2011/5/13 Thomas Hoellriegel <admin at blindi.net>:
> Hi all, i have written a schellscript.
> I can.t export variables from this script to Fs.
> Is a special syntax needed?

You're not giving us very much to work with.  Can you tell us how
you're doing it and what you're seeing?  An example would be best.


Regards,
Gabe


From gabe at gundy.org  Sat May 14 02:08:58 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 16:08:58 -0600
Subject: [Freeswitch-users] call waiting problem
In-Reply-To: <6364.1305238946@ccs.covici.com>
References: <6364.1305238946@ccs.covici.com>
Message-ID: <BANLkTi=gS8AiqmqYtRWHAem=Pd9ZE590HA@mail.gmail.com>

On Thu, May 12, 2011 at 4:22 PM,  <covici at ccs.covici.com> wrote:
> Hi. ?I am using a Digium card for a phone extension. ?Now when someone
> calls that extension through my IVR they hear a nice ring after
> selecting the menu option. ?However, if I am on another call and they do
> this, they get silence. ?I do hear the call waiting beep, but why don't
> they hear the ringback or transfer_ringback? ?I tried it myself and
> indeed get silence.

Did you make any progress on this?  If not, it would be helpful to see
some logs.  What do you have set as the "transfer_ringback"?

Best,
Gabe


From admin at blindi.net  Sat May 14 02:28:09 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 14 May 2011 00:28:09 +0200 (CEST)
Subject: [Freeswitch-users] question how to cut a string from a variable
 in fs?
In-Reply-To: <BANLkTikQ0-uRS0P3P4P87h21xnR=LFn97Q@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
	<BANLkTikQ0-uRS0P3P4P87h21xnR=LFn97Q@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105140017110.8087@debian>


Hi Steven, thank you for you help. I.ts very going fine.
My problem: I have troubles to search its self. i enter in google a 
searchstring and become manny results. I can.t select the right result.
My brailledisplay reads letter for letter.
This is very slowly.
I working as systemadministrator under unix and unix like systems freebsd 
solaris und oter linuxsystem.
I manage all systems from the console remotly with openssh.
Fs is very nice to handle without guis. I edit all configurationfile 
manually.
I have tested yate asterisk callweaver.
why is these profucs so very instabil?
Fs have a nice voicequality, is stable my server run 1 week on the uptime.


---------------
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 m.sobkow at marketelsystems.com  Sat May 14 02:28:16 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 13 May 2011 16:28:16 -0600
Subject: [Freeswitch-users] MEDIA_BUG_START
Message-ID: <4DCDB080.6050402@marketelsystems.com>

I'm getting a MEDIA_BUG_START event.  I think there's a correlation
between this event and some dropped calls we're experiencing.

I haven't found any way to force the event to occur, so I haven't been
able to capture a log file of it happening in fs_cli.

Our FreeSwitch box is configured to use a SIP Trunk provided by our
production Asterisk box, with T1 lines from thereon out.  For the most
part it works, but one user in particular has a serious problem with
dropped calls and I'm trying to figure it out.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From admin at blindi.net  Sat May 14 02:40:17 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 14 May 2011 00:40:17 +0200 (CEST)
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
	<Pine.LNX.4.64.1105132228380.6270@debian>
	<BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105140030150.8087@debian>

Hi gabriel,
I have written a datecheck.  I like to compare the inputdate to the 
systmdate.

This is my script:
#!/bin/bash
date1=`date -d $1 +%s`
if test $? -ne 0
then
dateerror="yes"
exit 0
fi
exit 0
I can.t export the variable dateerror.
Is the given date wrong, give a yes return.
fs ignore action.
Can you see please what is wrong?
thanks



---------------
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From gabe at gundy.org  Sat May 14 02:46:29 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 16:46:29 -0600
Subject: [Freeswitch-users] MEDIA_BUG_START
In-Reply-To: <4DCDB080.6050402@marketelsystems.com>
References: <4DCDB080.6050402@marketelsystems.com>
Message-ID: <BANLkTi=L4gk+JmxHP_OpeHJF5SwvYaqV5g@mail.gmail.com>

On Fri, May 13, 2011 at 4:28 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> I'm getting a MEDIA_BUG_START event. ?I think there's a correlation
> between this event and some dropped calls we're experiencing.
>
> I haven't found any way to force the event to occur, so I haven't been
> able to capture a log file of it happening in fs_cli.

You should be getting this event anytime a media bug is attached to a
channel.  There are several things that will result in a media bug
being created.  uuid_audio, uuid_displace and others attach a media
bug to the channel with the given UUID.

It should be pretty easy to re-create the event.


> Our FreeSwitch box is configured to use a SIP Trunk provided by our
> production Asterisk box, with T1 lines from thereon out. ?For the most
> part it works, but one user in particular has a serious problem with
> dropped calls and I'm trying to figure it out.

I don't think it matters much how this is connected to other systems
(not just yet anyway).

When you say *one* user has a problem, do the users have different
settings that would affect how the call is treated?  For example, if
you try to play audio located at /some/dir/USER_ONE/filename.wav, and
that file was not found on the server, FreeSWITCH would kill the
channel (and maybe the call if it was bridged).

Could this be what you're seeing?  Let us know what you find. Oh, BTW,
use uuid_buglist to show you what bugs are attached to that channel.
Good luck debugging (sorry... bad, bad pun).


Best,
Gabe


From steveayre at gmail.com  Sat May 14 03:01:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 14 May 2011 00:01:20 +0100
Subject: [Freeswitch-users] MEDIA_BUG_START
In-Reply-To: <4DCDB080.6050402@marketelsystems.com>
References: <4DCDB080.6050402@marketelsystems.com>
Message-ID: <29295C79-BA9A-4AFA-A380-3E0DFD7DCB19@gmail.com>

What codec are you using? Attaching a media bug to a passthrough-only codec such as mod_g729 will cause a dropped call.

A media bug is used by anything that needs the raw audio to detect something inband, eg start_dtmf, tone_detect and many others

Steve on iPhone

On 13 May 2011, at 23:28, Mark Sobkow <m.sobkow at marketelsystems.com> wrote:

> I'm getting a MEDIA_BUG_START event.  I think there's a correlation
> between this event and some dropped calls we're experiencing.
> 
> I haven't found any way to force the event to occur, so I haven't been
> able to capture a log file of it happening in fs_cli.
> 
> Our FreeSwitch box is configured to use a SIP Trunk provided by our
> production Asterisk box, with T1 lines from thereon out.  For the most
> part it works, but one user in particular has a serious problem with
> dropped calls and I'm trying to figure it out.
> 
> -- 
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
> 
> Visit our Blog for industry related information.
> http://marketel-systems.blogspot.com/
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gabe at gundy.org  Sat May 14 03:15:51 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 17:15:51 -0600
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <Pine.LNX.4.64.1105140030150.8087@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
	<Pine.LNX.4.64.1105132228380.6270@debian>
	<BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>
	<Pine.LNX.4.64.1105140030150.8087@debian>
Message-ID: <BANLkTikBvmqMSrqYSWsrU5pD5vKga82+YA@mail.gmail.com>

On Fri, May 13, 2011 at 4:40 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
> I have written a datecheck. ?I like to compare the inputdate to the
> systmdate.

<bash code here>

> I can.t export the variable dateerror.
> Is the given date wrong, give a yes return.
> fs ignore action.

I'm not able to see how you're working with FreeSWITCH in this case.

Is this a bash script that that will call fs_cli -x "some_command"?

Are you trying to set environmental variables in the shell before
starting freeswitch so they will be available to it?

Is this in a bash script that's called from the dialplan?  If so,
consider using the date-time related conditions (year, yday, mon,
mday, week, mweek, wday, hour, minute, minute-of-day, time-of-day,
date-time):

http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Condition

Lastly, there are also time real dialplan tools that might help:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_strepoch
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_strftime
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_strftime_tz

I don't think we'll be able to offer any more help without better
information.  However, we're happy to help if you can get us more
details.


Best,
Gabe


From gabe at gundy.org  Sat May 14 03:30:59 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 17:30:59 -0600
Subject: [Freeswitch-users] Fifo - exporting/Importing variables from A
 to B Leg
In-Reply-To: <BANLkTim=5Le+-qL1Msm=5+-6O32HytWTdg@mail.gmail.com>
References: <BANLkTim=5Le+-qL1Msm=5+-6O32HytWTdg@mail.gmail.com>
Message-ID: <BANLkTimKgeHXRtu3b7_wSit9ZsHerg_xZg@mail.gmail.com>

On Mon, May 9, 2011 at 12:45 PM, Marc de Corny <marcdecorny at gmail.com> wrote:
> 1. Is it possible to export variables from the A-leg of the queue to the
> B-let of the queue? There are the export and import commands, but not sure
> if they are applicable. If I capture calls coming out of the queue with a
> loopback I could then act using the variables that have been set from the
> A-leg.

I don't know why it wouldn't be applicable, export_vars would work
whenever there is a bridge.  Also, are you sure you would need to use
a loopback?  What about transfer_after_bridge?

http://wiki.freeswitch.org/wiki/Variable_transfer_after_bridge

Seems like it might be cleaner.


> 2. I'd like to send the call to the Freeswitch to play IVRs etc and then
> bridge it without the audio.??I know it can be done on a normal bridge with
> bypass_media_after_bridge, but not so sure on a fifo call.

I haven't used the fifo in a while, but again, I think it's just a
tool that eventually ends up with 2 channels being bridged.

Good luck and let us know what you find.

Best,
Gabe


From gabe at gundy.org  Sat May 14 03:35:52 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 17:35:52 -0600
Subject: [Freeswitch-users] No Audio on Gateway Incoming
In-Reply-To: <4DC7A4EF.8040408@xofap.com>
References: <mailman.7828.1303231692.24555.freeswitch-users@lists.freeswitch.org>
	<4DC7A4EF.8040408@xofap.com>
Message-ID: <BANLkTik2yii=2G6V7Ca_MOoB6CONj9KeRg@mail.gmail.com>

On Mon, May 9, 2011 at 2:25 AM, William Alianto <william at xofap.com> wrote:
> Here is the debug log from the server :
>
> http://pastebin.freeswitch.org/16257

William,

We're here to help, but you'll find better results if you take the
time to summarize the 10k lines of logs that you posted.  Try to help
us understand your problem and we'll try to help you find the answer.

Best,
Gabe


From gmaruzz at gmail.com  Sat May 14 03:41:20 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 14 May 2011 01:41:20 +0200
Subject: [Freeswitch-users] Can UUID's change?
In-Reply-To: <BANLkTinM4iLdW8JSJsch8shPJ8Zzp8cXRA@mail.gmail.com>
References: <4DCD9D86.3050302@marketelsystems.com>
	<BANLkTinM4iLdW8JSJsch8shPJ8Zzp8cXRA@mail.gmail.com>
Message-ID: <BANLkTi=v75SCbQYgXXE1uTwF_2uzCSKJ5Q@mail.gmail.com>

On Origination you may want to set the variable "origination-uuid" to
an uuid you previously created by the API command "create-uuid". This
way you don't have to worry about which uuid is which.
PS: the variable name and the API command are quoted by memory, they
probably have slightly different names, check in the wiki.


On 5/13/11, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Fri, May 13, 2011 at 3:07 PM, Mark Sobkow
> <m.sobkow at marketelsystems.com> wrote:
>> However, I'm seeing some unrecognized UUIDs after bridging the call
>> (i.e. that weren't returned by the place call code.)
>>
>> Are there any events which might indicate that a UUID is being changed
>> by the system, and that I should be watching for a new UUID instead?
>
> The UUID of the channel never changes.  If you're using a UUID to
> originate a call, then you'll have that UUID to work with, but FS will
> still pick it's own UUID for the channel --and it will not change.
>
> Calls get UUIDs, conferences get UUIDs, channels get UUIDs, so it's
> just a matter of finding the right one and not worrying too much about
> the other UUIDs that might show up.  Hope that helps.
>
> Best,
> 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
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gabe at gundy.org  Sat May 14 03:42:31 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 17:42:31 -0600
Subject: [Freeswitch-users] How to eliminate acoustic echo from server
	side(FreeSwitch)
In-Reply-To: <4DC75B8D.9090008@163.com>
References: <4DC75B8D.9090008@163.com>
Message-ID: <BANLkTi=iUWOARjWRdGp98nd0igtwap0Z8g@mail.gmail.com>

On Sun, May 8, 2011 at 9:12 PM, vivid <vivid333 at 163.com> wrote:
> In the multi-party conference, because some termianl phone does not
> process echo, which causes the call have echo, so SERVER side needs to
> do echo processing. How to eliminate echo in this scenario?

Before you can really address echo issues, you need to have a good
understanding of the source.  We really don't have enough information
to work with here.

As you must already know, echo can be tricky.  I'd recommend you try
to 1) find the source and eliminate it if you can. 2) Try to solve it
with the right hardware offering. 3) If all else fails, you'll have to
do a lot of reading and get your head wrapped around the problem.


Good luck,
Gabe


From gabe at gundy.org  Sat May 14 04:22:17 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 18:22:17 -0600
Subject: [Freeswitch-users] Domain tag in sofia profile
In-Reply-To: <assp.01095cb6c2.92097A6A775D5147B1078E3F15430B924B993D@prato.activenetwerx.local>
References: <AcwNmD7jVFWFCWTPSrGfoQKT9fi7EA==>
	<assp.01095cb6c2.92097A6A775D5147B1078E3F15430B924B993D@prato.activenetwerx.local>
Message-ID: <BANLkTikW71RCcXC_eMYCuLrG_AwBFdLczA@mail.gmail.com>

On Sun, May 8, 2011 at 10:16 AM, Joseph L. Casale
<jcasale at activenetwerx.com> wrote:
> I looked through the wiki but I couldn't find an explanation as to what you accomplish
> with the domain tags. When you alias a domain from the directory, what functionality
> does this provide?

The domain in the directory tells FS what domain the user belongs to.
This becomes more important as you run multiple domains on the same
server.

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Domains_.26_Users_Parameters

Given this xml:

<domain name="izeni.com">
  <user id="gabe">
    <params>
      <param name="password" value="blaa_blaa"/>
    </params>
  </user>
</domain>


You'll find the following:

freeswitch> domain_exists example.com
false

freeswitch> domain_exists izeni.com
true



freeswitch> user_exists id gabe example.com
false

freeswitch> user_exists id gabe izeni.com
true

freeswitch> user_exists id bill izeni.com
false

The XML could be extended to include other domains:

<domain name="izeni.com">
  <user id="gabe">
    <params>
      <param name="password" value="blaa_blaa"/>
    </params>
  </user>
</domain>
<domain name="gundy.org">
  <user id="gabe">
    <params>
      <param name="password" value="blaa_blaa"/>
    </params>
  </user>
</domain>

Being able to service unlimited domains is just another thing that
makes FreeSWITCH so sweet :)

Hope that helps.


Best,
Gabe


From gabe at gundy.org  Sat May 14 04:34:39 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 18:34:39 -0600
Subject: [Freeswitch-users] can I overwrite channel var with response
 headers ?
In-Reply-To: <BANLkTi=+duWX7PMsgs0jxLd3_A4mfp-7zQ@mail.gmail.com>
References: <BANLkTi=+duWX7PMsgs0jxLd3_A4mfp-7zQ@mail.gmail.com>
Message-ID: <BANLkTikpVxyKBuFN7F9jezjdt2MRP1PB+Q@mail.gmail.com>

On Sat, May 7, 2011 at 2:26 AM, jay binks <jaybinks at gmail.com> wrote:
> Is there a way to tell FS to overwrite channel var "blah" with these sip
> responses ?

I don't know that you would want them to automatically overwrite the
existing variables as a default behavior. It sounds like you might be
left to your own devices to make that happen.  Have you learned
anything more about this since posting?

Best,
Gabe


From gabe at gundy.org  Sat May 14 04:42:52 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 18:42:52 -0600
Subject: [Freeswitch-users] problem with transferring calls when using
	ivr
In-Reply-To: <op.vuoxye19agcvls@master>
References: <op.vuoxye19agcvls@master>
Message-ID: <BANLkTinMOWy2HfFdnoNVqMkqYBjH9X8LHw@mail.gmail.com>

On Fri, Apr 29, 2011 at 2:31 AM,  <bogusmaster at o2.pl> wrote:
> Currently I'm having a problem with transferring calls that went thru ivr
> menu.

<SNIP/>

> Any hints would be greatly appreciated.


For simple IVRs, it's hard to beat the ease of the built-in IVR application:

http://wiki.freeswitch.org/wiki/Demo_ivr.xml

I'd expect the learning curve might be a lot easier for a newcomer
than building one in js.  Now, at some point, you might need the extra
power.

Good luck and welcome to the world of FreeSWITCH!


Best,
Gabe


From gabe at gundy.org  Sat May 14 04:44:43 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 18:44:43 -0600
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
Message-ID: <BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>

On Mon, May 2, 2011 at 4:27 PM, Avi Marcus <avi at avimarcus.net> wrote:
> I've tried this cool formula for streaming TTS via google:
> http://wiki.freeswitch.org/wiki/TTS
> and while the link produces a pretty darn nice sounding MP3, I get:
> 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at
> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123: Invalid
> mpg123 handle. (code 10)
> Is anyone else using this reliably?

Yes, I too can confirm that others are using this reliably.

What more can you tell us about your problem?

Best,
Gabe


From msc at freeswitch.org  Sat May 14 05:03:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 18:03:51 -0700
Subject: [Freeswitch-users] Different SAS with ZRTP
In-Reply-To: <BANLkTinsdQvkvgtuV8SEYOY7pP46TYYeow@mail.gmail.com>
References: <BANLkTinsdQvkvgtuV8SEYOY7pP46TYYeow@mail.gmail.com>
Message-ID: <BANLkTi=bv703hO6Xjq2Z=+1F9vnEJJ3q_w@mail.gmail.com>

Is FS a trusted man in the middle in this scenario?
-MC

On Fri, May 13, 2011 at 2:50 PM, Banhi Dutra <banhi.dutra at gmail.com> wrote:

> Hi all,
>
>
> I installed FS in test environment with ZRTP enabled. I have two softphone
> (I tested with Acrobits and Jitsi) registered and all works fine, except for
> SAS exchange, which are different. I'm using libzrtp 0.81.514 and
>  FreeSWITCH Version 1.0.head (git-23d8658 2011-05-07 00-27-20 -0400). I read
> older post about a similar problem. Could be a version problem ?
>
>
> Any help will be appreciated
>
>
> Thank you
>
> Banhi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From admin at blindi.net  Sat May 14 05:49:29 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 14 May 2011 03:49:29 +0200 (CEST)
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <BANLkTikBvmqMSrqYSWsrU5pD5vKga82+YA@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
	<Pine.LNX.4.64.1105132228380.6270@debian>
	<BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>
	<Pine.LNX.4.64.1105140030150.8087@debian>
	<BANLkTikBvmqMSrqYSWsrU5pD5vKga82+YA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105140315200.9833@debian>

Am 13.05.11 um 17:15 schrieb Gabriel Gunderson:

This is a bashscript. I call this script from dialplan i compate with 
play_and_get_digits::

   <extension name="datechecking">
     <condition field="destination_number" expression="^4411$"/>
     <condition field="${dateerror}" expression="^$" break="never">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="play_and_get_digits"
       data="0 8 1 1000 #
       $${base_dir}/sounds/date/enter_correct_date.wav
        $${base_dir}/sounds/ivr/invalid.wav  nr \d+"/>
<action application="system" data="$${base_dir}/scripts/datechk.sh ${nr}
" />

       <action application="transfer" data="${destination_number}"/>
     </condition>
<condition field="${dateerror}" expression="^(.*)\d" break="on-true" >
break="never">
   <action application="playback" data="/ivr/date.ok.wav"/>
   <anti-action application="playback" data="/ivr/invalid-date.wav"/>

     </condition>
   </extension>

I can.t to export the variable from the shellscript to fs.

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
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From admin at blindi.net  Sat May 14 07:00:39 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 14 May 2011 05:00:39 +0200 (CEST)
Subject: [Freeswitch-users] iax connections breaks
In-Reply-To: <BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
	<Pine.LNX.4.64.1105132228380.6270@debian>
	<BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105140451280.10767@debian>

Hi all, i don.t become a iax-connection fs break the connection.
This is my consoleoutput:
2011-05-14 04:49:54.362667 [DEBUG] switch_ivr_originate.c:1873 Parsing 
global va
riables
2011-05-14 04:49:54.362667 [ERR] switch_core_session.c:413
Could not locate channel type iax
2011-05-14 04:49:54.362667 [ERR] switch_ivr_originate.c:2447 Cannot create 
outgo
ing channel of type [iax] cause: [CHAN_NOT_IMPLEMENTED]

I don.t find a mod_iax entry in the modules.conf
my dialplansyntax is:
    <action application="bridge" data="iax/user:password at server.net/3311"/>

Do your  have a tip? 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 william.suffill at gmail.com  Sat May 14 08:00:22 2011
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 14 May 2011 00:00:22 -0400
Subject: [Freeswitch-users] iax connections breaks
In-Reply-To: <Pine.LNX.4.64.1105140451280.10767@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com>
	<Pine.LNX.4.64.1105132228380.6270@debian>
	<BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com>
	<Pine.LNX.4.64.1105140451280.10767@debian>
Message-ID: <BANLkTi=RBRx4YUgueD4DCE7_xJK0LZOfRA@mail.gmail.com>

mod_iax was removed [2010-01-22]
http://thread.gmane.org/gmane.comp.telephony.freeswitch.devel/2432/focus=22573

-- W
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From infos at madovsky.org  Sat May 14 09:37:57 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 14 May 2011 01:37:57 -0400
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
	not working
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com><Pine.LNX.4.64.1105120530140.9184@debian><BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com><Pine.LNX.4.64.1105132046460.5405@debian><51E0B1392ADF44069FCB8537C2AE75B1@e1705><BANLkTi=XXNoxkZDe1gZzCRrPsYjZHnD=GA@mail.gmail.com><Pine.LNX.4.64.1105132228380.6270@debian><BANLkTi=O8oUxiW6dib2YtngCad7GbAKVog@mail.gmail.com><Pine.LNX.4.64.1105140030150.8087@debian><BANLkTikBvmqMSrqYSWsrU5pD5vKga82+YA@mail.gmail.com>
	<Pine.LNX.4.64.1105140315200.9833@debian>
Message-ID: <0F189258C16A4A178DAA1D1C7669A03D@e1705>

I think you have to escape the $ 3 times
\\\${nir}
\
----- Original Message ----- 
From: "Thomas Hoellriegel" <admin at blindi.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 13, 2011 9:49 PM
Subject: Re: [Freeswitch-users] problem exporting variables from shellsctipt 
not working


Am 13.05.11 um 17:15 schrieb Gabriel Gunderson:

This is a bashscript. I call this script from dialplan i compate with
play_and_get_digits::

   <extension name="datechecking">
     <condition field="destination_number" expression="^4411$"/>
     <condition field="${dateerror}" expression="^$" break="never">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="play_and_get_digits"
       data="0 8 1 1000 #
       $${base_dir}/sounds/date/enter_correct_date.wav
        $${base_dir}/sounds/ivr/invalid.wav  nr \d+"/>
<action application="system" data="$${base_dir}/scripts/datechk.sh ${nr}
" />

       <action application="transfer" data="${destination_number}"/>
     </condition>
<condition field="${dateerror}" expression="^(.*)\d" break="on-true" >
break="never">
   <action application="playback" data="/ivr/date.ok.wav"/>
   <anti-action application="playback" data="/ivr/invalid-date.wav"/>

     </condition>
   </extension>

I can.t to export the variable from the shellscript to fs.

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



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


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sat May 14 11:54:42 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 14 May 2011 15:54:42 +0800
Subject: [Freeswitch-users] call somebody into conference after
	conferenceestablished
In-Reply-To: <201105121532017963789@asiainfo-linkage.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<201105121532017963789@asiainfo-linkage.com>
Message-ID: <BANLkTi=wbd8ivSA4Gom3P2QsJ9AY7GRJUA@mail.gmail.com>

Hi Michael,

Thanks for your detailed suggestion.

i test it as below steps, it works fine, however, there is a limiation that
i have to fill fixed callee number in conference.conf.xml
(data="execute_extension 5001"), while actually i what to dial arbitrary
number after "*" dynamically(e.g *1234, *2345), i can think out it could
realize by a IVR script like "data="execute_extension IVR_scripts" and after
IVR give some voice prompt (pls input the callee number) then ... however,
considering simplicity, can i convey the DTMF into the dialplan (just
directly press *1234, and then 1234 ring...)? could you please provide any
hints or any suggestion... Thanks a lot!

1. in conference.conf.xml, set as below,
<control action="execute_application" digits="*" data="execute_extension
5001"/>

2. in dial plan,

<!-- for confernce named conf100 -->
<extension name="conf100" >
   <condition field="destination_number" expression="^666$">
       <action application="answer"/>
       <action application="set" data="conference_user_list=|"/>
       <action application="conference" data="conf100 at default"/>
   </condition>
</extension>

and

<!-- for dial out 5001 from the conference conf100 -->

<extension name="5_to_conf_out" >
   <condition field="destination_number" expression="^(5\d+)$">
       <!-- <action application="conference" data="conf100 dial
sofia/internal/5001 at 192.168.200.201 100 Conference_Caller_Name"/> -->
     <!-- <action application="bridge" data="sofia/internal/$
1 at 192.168.200.201"/> -->
     <action application="set" data="api_result=${conference(conf100 dial
sofia/internal/$1 at 192.168.200.201 100 Conference_Caller_Name)}"/>
   </condition>
</extension>

3. register a extension 3001, call 666 and join a conference, and then press
"*", the FS will call 5001 on a IAD.
then 5001 and 3001 join the same conference.


2011/5/12 liuyp2 <liuyp2 at asiainfo-linkage.com>

>  conference Your-Conf-Name dial user/1002
>
>  ------------------------------
>  liuyp2
> 2011-05-12
>  ------------------------------
>  *????* fieldpeak
> *?????* 2011-05-12 11:11:41
> *????* FreeSWITCH-users
> *???*
> *???* [Freeswitch-users] call somebody into conference after
> conferenceestablished
>
> Hi Gurus,
>
> i dial 666 and enter a conference, then i need call some body's phone
> number to join him into this conference...
> is there anyone can advise how can realize this scenario?
>
> thanks.
>
> 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 boris at tagnet.ru  Sat May 14 11:59:16 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 14 May 2011 13:59:16 +0600
Subject: [Freeswitch-users] CDR event
Message-ID: <4DCE3654.7080002@tagnet.ru>

Hello!

     May I fire custom cdr event for a LUA script? I've read 
http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events. 
Would You please point me to proper documentation or give an example?

-- 
Regards,
Boris




From admin at blindi.net  Sat May 14 16:11:24 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 14 May 2011 14:11:24 +0200 (CEST)
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <0F189258C16A4A178DAA1D1C7669A03D@e1705>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com><Pine.LNX.4.64.1105120530140.9184@debian><BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com><Pine.LNX.4.64.1105132046460.5405@debian><51E0B1392ADF44069FCB8537C2AE75B1@e1705><BANLkTi=XXNoxkZDe1gZzCR
	<Pine.LNX.4.64.1105140315200.9833@debian>
	<0F189258C16A4A178DAA1D1C7669A03D@e1705>
Message-ID: <Pine.LNX.4.64.1105141407100.2351@debian>

Am 14.05.11 um 01:37 schrieb Madovsky:

> I think you have to escape the $ 3 times
I don.t no. $3? I like to export the variable: dateerror  from the 
shellscript in my 
fs dialplan, to make a selection from condidion field.
> \\\${nir}
what is the syntax in the dialplan?
thank you.
---------------
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 eagle.antonio at gmail.com  Sat May 14 17:25:55 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Sat, 14 May 2011 14:25:55 +0100
Subject: [Freeswitch-users] ESL Command Not Found
In-Reply-To: <BANLkTi=-Ftkf42wv-aMogFhWM-=-kOW4sg@mail.gmail.com>
References: <BANLkTi=q0-sUZyxb0AsJsyhonjE8Z-tuoQ@mail.gmail.com>
	<BANLkTi=O8SGiY=4Y_eUX3pH6KaMo0L3dOA@mail.gmail.com>
	<BANLkTi=-Ftkf42wv-aMogFhWM-=-kOW4sg@mail.gmail.com>
Message-ID: <BANLkTinrATweNzBugzGP5=zvandyQofdMQ@mail.gmail.com>

Well oopsy , typo on the e-mail but the problem persists with ESL using
python even when using FULL Control.

A/T

2011/5/13 curriegrad2004 <curriegrad2004 at gmail.com>

> Ah yes, the dreaded cup of coffee is needed...
>
> On Fri, May 13, 2011 at 10:04 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > haha
> > uuid_getvar <uuid> <var_name>
> > -MC
> >
> > On Fri, May 13, 2011 at 10:01 AM, Antonio Teixeira <
> eagle.antonio at gmail.com>
> > wrote:
> >>
> >> Hello
> >>
> >> I'm using ESL to Fork a call inside another , this is the commom example
> >> of User (A) pressing 1 and calling another party (B).
> >>
> >> I'm forking the call by opening an inbound socket and issuing the
> command
> >>
> >>
> >>
> api('originate',"{step=1,origination_caller_id_number=XXXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s}sofia/external/%s at 2XXXXXXX
> >> &socket(192.168.0.12:8040 sync full)
> >>
> >>
> >> All works great ( i get the call) and as you can see the call to B is
> >> redirected to the IVR Server
> >>
> >> The problem is if i do on the channel B ( By esl)
> >> uuid_getvar step XXXXXXXXXXXXXXXXXXXXXXXXXX
> >>
> >> I get
> >> Event-Name: SOCKET_DATA
> >> Content-Type: command/reply
> >> Reply-Text: -ERR%20command%20not%20found
> >>
> >> Has anyone faced this ??
> >> BTW using the python ESL:
> >>
> >> Best Regards
> >> Antonio Teixeira
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.afzali2003 at gmail.com  Sat May 14 17:55:33 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 14 May 2011 18:25:33 +0430
Subject: [Freeswitch-users] Sending Message from background LUA script to an
 unregistered user prevents sofia profile to be restarted
Message-ID: <BANLkTikMBSv41i=fmBRVct=RwAjvmeLR1g@mail.gmail.com>

Hi Guys,

I have a background Lua script which send messages to specified users.I have
noticed that if the script sends a message to unregistered user then I'll
not be able to restart sofia profile properly. To clarify the issue, I sent
a message to unregistered user by a softphone instead of background
script.In this case, the sofia profile can be restarted properly! In the
following logs you see first I've tried to send message by a softphone
(sofia_presence.c:149)
after that I've restarted sofia correctly. After the background Lua tried to
send message (mod_sofia.c:4629), restarting sofia failed.

FreeSWITCH build from last GIT.
Should I open a ticket on JIRA?

BEST,
-- afshin


freeswitch at opxi2Server>
freeswitch at opxi2Server> 2011-05-14 09:34:25.471083 [ERR]
sofia_presence.c:149 Can't find registered user noname2 at fslab

freeswitch at opxi2Server>
freeswitch at opxi2Server> sofia status

                     Name
Type                                       Data      State
=================================================================================================
           sipinterface_1       profile
sip:mod_sofia at 192.168.128.2:5060      RUNNING (0)
              voicemail_1         alias
sipinterface_1      ALIASED
           sipinterface_3       profile
sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
                    fslab         alias
sipinterface_1      ALIASED
           sipinterface_2       profile
sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
=================================================================================================
3 profiles 2 aliases

freeswitch at opxi2Server> sofia profile sipinterface_1 restart

Reload XML [Success]
restarting: sipinterface_1
2011-05-14 09:34:59.771082 [INFO] mod_enum.c:765 ENUM Reloaded
2011-05-14 09:34:59.771082 [INFO] switch_time.c:1020 Timezone reloaded 530
definitions
freeswitch at opxi2Server> 2011-05-14 09:35:00.791085 [NOTICE] sofia.c:1677
Waiting for worker thread
2011-05-14 09:35:00.791085 [NOTICE] sofia.c:2274 Adding Alias [voicemail_1]
for profile [sipinterface_1]
2011-05-14 09:35:00.791085 [NOTICE] sofia.c:2274 Adding Alias [fslab] for
profile [sipinterface_1]
2011-05-14 09:35:00.791085 [NOTICE] sofia.c:4018 Started Profile
sipinterface_1 [sofia_reg_sipinterface_1]

freeswitch at opxi2Server> sofia status

                     Name
Type                                       Data      State
=================================================================================================
           sipinterface_1       profile
sip:mod_sofia at 192.168.128.2:5060      RUNNING (0)
              voicemail_1         alias
sipinterface_1      ALIASED
           sipinterface_3       profile
sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
                    fslab         alias
sipinterface_1      ALIASED
           sipinterface_2       profile
sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
=================================================================================================
3 profiles 2 aliases

freeswitch at opxi2Server> 2011-05-14 09:35:39.171084 [ERR] mod_sofia.c:4629
Can't find registered user noname at fslab

freeswitch at opxi2Server> sofia status

                     Name
Type                                       Data      State
=================================================================================================
           sipinterface_1       profile
sip:mod_sofia at 192.168.128.2:5060      RUNNING (0)
              voicemail_1         alias
sipinterface_1      ALIASED
           sipinterface_3       profile
sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
                    fslab         alias
sipinterface_1      ALIASED
           sipinterface_2       profile
sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
=================================================================================================
3 profiles 2 aliases

freeswitch at opxi2Server> sofia profile sipinterface_1 restart

Reload XML [Success]
restarting: sipinterface_1
2011-05-14 09:35:55.871085 [INFO] mod_enum.c:765 ENUM Reloaded
2011-05-14 09:35:55.871085 [INFO] switch_time.c:1020 Timezone reloaded 530
definitions
freeswitch at opxi2Server>
freeswitch at opxi2Server> sofia status

                     Name
Type                                       Data      State
=================================================================================================
           sipinterface_3       profile
sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
           sipinterface_2       profile
sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
=================================================================================================
2 profiles 0 aliases

freeswitch at opxi2Server> 2011-05-14 09:36:40.171083 [ERR] mod_sofia.c:4624
Can't find profile sipinterface_1
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From jcasale at activenetwerx.com  Sat May 14 19:09:41 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 14 May 2011 15:09:41 +0000
Subject: [Freeswitch-users] Domain tag in sofia profile
In-Reply-To: <BANLkTikW71RCcXC_eMYCuLrG_AwBFdLczA@mail.gmail.com>
References: <AcwNmD7jVFWFCWTPSrGfoQKT9fi7EA==>
	<assp.01095cb6c2.92097A6A775D5147B1078E3F15430B924B993D@prato.activenetwerx.local>
	<BANLkTikW71RCcXC_eMYCuLrG_AwBFdLczA@mail.gmail.com>
Message-ID: <assp.0115344b4e.92097A6A775D5147B1078E3F15430B925106EF@prato.activenetwerx.local>

>The domain in the directory tells FS what domain the user belongs to.
>This becomes more important as you run multiple domains on the same
>server.

Yup, that much I get. I was specifically looking to understand exactly what
the name/alias/parse parameters in the domain tags inside a sofia profile
xml definition accomplished and how you would utilize them.

I had some issues creating and utilizing aliases as all the users register by
ip addresses, but I think it's all resolved now.

Thanks Gabe,
jlc

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


From steveayre at gmail.com  Sat May 14 19:19:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 14 May 2011 16:19:31 +0100
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <Pine.LNX.4.64.1105141407100.2351@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<Pine.LNX.4.64.1105140315200.9833@debian>
	<0F189258C16A4A178DAA1D1C7669A03D@e1705>
	<Pine.LNX.4.64.1105141407100.2351@debian>
Message-ID: <BANLkTi=H5=jMnYrKZmEAQnXo3vZAtHbU=g@mail.gmail.com>

You can't export variables from a system command such as a shellscript.

-Steve


On 14 May 2011 13:11, Thomas Hoellriegel <admin at blindi.net> wrote:
> Am 14.05.11 um 01:37 schrieb Madovsky:
>
>> I think you have to escape the $ 3 times
>
> I don.t no. $3? I like to export the variable: dateerror ?from the
> shellscript in my fs dialplan, to make a selection from condidion field.
>>
>> \\\${nir}
>
> what is the syntax in the dialplan?
> thank you.
> ---------------
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Sat May 14 19:38:21 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 14 May 2011 11:38:21 -0400
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
	not working
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com><Pine.LNX.4.64.1105120530140.9184@debian><BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com><Pine.LNX.4.64.1105132046460.5405@debian><51E0B1392ADF44069FCB8537C2AE75B1@e1705><BANLkTi=XXNoxkZDe1gZzCR<Pine.LNX.4.64.1105140315200.9833@debian><0F189258C16A4A178DAA1D1C7669A03D@e1705>
	<Pine.LNX.4.64.1105141407100.2351@debian>
Message-ID: <C43FCE089095473DBA508CE55879B5B6@e1705>

I mean you need 3 backslashes in front of every variable dollar

----- Original Message ----- 
From: "Thomas Hoellriegel" <admin at blindi.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 14, 2011 8:11 AM
Subject: Re: [Freeswitch-users] problem exporting variables from shellsctipt 
not working


Am 14.05.11 um 01:37 schrieb Madovsky:

> I think you have to escape the $ 3 times
I don.t no. $3? I like to export the variable: dateerror  from the
shellscript in my
fs dialplan, to make a selection from condidion field.
> \\\${nir}
what is the syntax in the dialplan?
thank you.
---------------
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



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


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



From anton.vazir at gmail.com  Sat May 14 20:09:40 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 14 May 2011 21:09:40 +0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
Message-ID: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>

I'm trying to catch an error, in case I would dial wrong (non existent) gateway

I'm running ESL outbound listener, subscribing to all events,
if I do proper bgapi 'originate' - there are normal events flow, and I
can track what is happening.
But if issue originate to a gateway, which is not configured - there
is simply no any events fired. I have an error on FS console

2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway
following by
2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot
create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

But HOW to catch the given in ESL?

Trying trick like checking for UUID exist or bgapi job ID - all unsuccessfull.

sofia.c seems just does not have event code for that cases

if (profile_name && !profile_found) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such
Profile '%s'\n", profile_name);
>status = SWITCH_STATUS_FALSE;
}

Any clue?


From anton.vazir at gmail.com  Sat May 14 20:23:53 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 14 May 2011 21:23:53 +0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
Message-ID: <BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>

The same goes for gateway, which is just down. No events, signalling
that call will not succeed. And no events fired.

2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]

Am I missing the way to get info in the ESL about gateways, which are
out of order, or there is simple no way, without hacking the code?


From infos at madovsky.org  Sat May 14 20:50:55 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 14 May 2011 12:50:55 -0400
Subject: [Freeswitch-users] ESL: No events fired when there is error on
	submitted API command, like originate sofia to non-existent gateway
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
Message-ID: <23EBA3985DA74FA99ED187BAD3DA0014@e1705>

maybe your gateway is blocking some numbers

----- Original Message ----- 
From: "Anton VG" <anton.vazir at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 14, 2011 12:23 PM
Subject: Re: [Freeswitch-users] ESL: No events fired when there is error on 
submitted API command, like originate sofia to non-existent gateway


> The same goes for gateway, which is just down. No events, signalling
> that call will not succeed. And no events fired.
>
> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
> create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
>
> Am I missing the way to get info in the ESL about gateways, which are
> out of order, or there is simple no way, without hacking the code?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anton.vazir at gmail.com  Sat May 14 21:10:57 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 14 May 2011 22:10:57 +0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <23EBA3985DA74FA99ED187BAD3DA0014@e1705>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
Message-ID: <BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>

You did not understand. I INTENTIONALLY dialing the bad gateway, and
I'm looking for a proper way to determine that gateway is bad in my
ESL dialplan, by catching the proper event/reply/whatever,
And much preferably without tricks, like esl.api('sofia status gateway
GatewayWhichIsDown')

When in production, and there is more than a single route, there will
be plenty of cases, when you dial a bad gateway, so there should be a
way for ESL dialplan to determine that a gateway is not callable for a
moment, the reason WHY and  to retry with another one.

The trick above is bad, since:
1. blocking api query, before evey single gateway call attempt.
2. Gateway maybe known in UP state, but the state is stale, in dial in
fact will go to DOWN gateway. So, dialplan will screw

Possibly I should ask in DEV list...

2011/5/14 Madovsky <infos at madovsky.org>:
> maybe your gateway is blocking some numbers
>
> ----- Original Message -----
> From: "Anton VG" <anton.vazir at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, May 14, 2011 12:23 PM
> Subject: Re: [Freeswitch-users] ESL: No events fired when there is error on
> submitted API command, like originate sofia to non-existent gateway
>
>
>> The same goes for gateway, which is just down. No events, signalling
>> that call will not succeed. And no events fired.
>>
>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>> create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
>>
>> Am I missing the way to get info in the ESL about gateways, which are
>> out of order, or there is simple no way, without hacking the code?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anton.vazir at gmail.com  Sat May 14 21:17:12 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 14 May 2011 22:17:12 +0500
Subject: [Freeswitch-users] ESL: not bein able to determine when SIP gateway
 is down when originating a call
Message-ID: <BANLkTik9R9xqEEQic9miZbvN7T8MWCF-JA@mail.gmail.com>

I'm trying to catch an error, in case I would dial wrong (non
existent) gateway (intentionally!)

I'm running ESL outbound listener, subscribing to all events,
if I do bgapi 'originate' to a live gateway - there are normal events
flow, and I
can track what is happening.
But if I issue originate to a gateway, which is not configured or
simply down - there
are no any events fired.

I only have an error on FS console

2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway
following by
2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot
create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

But HOW to catch the given in ESL?

sofia.c seems just does not have event code for that cases

if (profile_name && !profile_found) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such
Profile '%s'\n", profile_name);
>status = SWITCH_STATUS_FALSE;
}

logically there should be a proper way to determine that gateway is bad in my
ESL dialplan, by catching the proper event/reply/whatever,
For the moment i did trick: esl.api('sofia status gateway
GatewayWhichIsDown')

When in production, and there is more than a single route, there will
be plenty of cases, when you dial a bad gateway, so there should be a
way for ESL dialplan to determine that a gateway is not callable for a
moment, the reason WHY and  to retry with another one.

The trick above is bad, since:
1. blocking api query, before evey single gateway call attempt.
2. Gateway maybe known in UP state, but the state is stale, in dial in
fact will go to DOWN gateway. So, ESL dialplan will screw in that case

Any clue?


From anthony.minessale at gmail.com  Sat May 14 21:48:31 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 14 May 2011 12:48:31 -0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
	<BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
Message-ID: <BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>

Read up on mod distributor on the wiki.
On May 14, 2011 12:12 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
> You did not understand. I INTENTIONALLY dialing the bad gateway, and
> I'm looking for a proper way to determine that gateway is bad in my
> ESL dialplan, by catching the proper event/reply/whatever,
> And much preferably without tricks, like esl.api('sofia status gateway
> GatewayWhichIsDown')
>
> When in production, and there is more than a single route, there will
> be plenty of cases, when you dial a bad gateway, so there should be a
> way for ESL dialplan to determine that a gateway is not callable for a
> moment, the reason WHY and to retry with another one.
>
> The trick above is bad, since:
> 1. blocking api query, before evey single gateway call attempt.
> 2. Gateway maybe known in UP state, but the state is stale, in dial in
> fact will go to DOWN gateway. So, dialplan will screw
>
> Possibly I should ask in DEV list...
>
> 2011/5/14 Madovsky <infos at madovsky.org>:
>> maybe your gateway is blocking some numbers
>>
>> ----- Original Message -----
>> From: "Anton VG" <anton.vazir at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, May 14, 2011 12:23 PM
>> Subject: Re: [Freeswitch-users] ESL: No events fired when there is error
on
>> submitted API command, like originate sofia to non-existent gateway
>>
>>
>>> The same goes for gateway, which is just down. No events, signalling
>>> that call will not succeed. And no events fired.
>>>
>>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>>> create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
>>>
>>> Am I missing the way to get info in the ESL about gateways, which are
>>> out of order, or there is simple no way, without hacking the code?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anton.vazir at gmail.com  Sat May 14 22:00:20 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 14 May 2011 23:00:20 +0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
	<BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
	<BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>
Message-ID: <BANLkTim3HG5YHG6sKRnU+PZmgg3pKL2Q9Q@mail.gmail.com>

That good one, but not for the my case.

i use originate &park, and than bridge_uuid, when there is an early_media.
I have a number of gateways, which support specific destinations each,
so it's up to my billing to decide what gateway should be dialed and
in which order. But I still need to determine if gateway could be
reached or not, or if while calling, it gives an error, and which one.

I see gwlist down could be used for bridge, but bridge does not give
flexibility I try to achieve. Will see what it gives if used for
originate...

So, considering your brief reply, there just no support for the case I
need, so will try to get inside sofia.c ...

Regards,
Anton.

2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
> Read up on mod distributor on the wiki.
>
> On May 14, 2011 12:12 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>> You did not understand. I INTENTIONALLY dialing the bad gateway, and
>> I'm looking for a proper way to determine that gateway is bad in my
>> ESL dialplan, by catching the proper event/reply/whatever,
>> And much preferably without tricks, like esl.api('sofia status gateway
>> GatewayWhichIsDown')
>>
>> When in production, and there is more than a single route, there will
>> be plenty of cases, when you dial a bad gateway, so there should be a
>> way for ESL dialplan to determine that a gateway is not callable for a
>> moment, the reason WHY and to retry with another one.
>>
>> The trick above is bad, since:
>> 1. blocking api query, before evey single gateway call attempt.
>> 2. Gateway maybe known in UP state, but the state is stale, in dial in
>> fact will go to DOWN gateway. So, dialplan will screw
>>
>> Possibly I should ask in DEV list...
>>
>> 2011/5/14 Madovsky <infos at madovsky.org>:
>>> maybe your gateway is blocking some numbers
>>>
>>> ----- Original Message -----
>>> From: "Anton VG" <anton.vazir at gmail.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Saturday, May 14, 2011 12:23 PM
>>> Subject: Re: [Freeswitch-users] ESL: No events fired when there is error
>>> on
>>> submitted API command, like originate sofia to non-existent gateway
>>>
>>>
>>>> The same goes for gateway, which is just down. No events, signalling
>>>> that call will not succeed. And no events fired.
>>>>
>>>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>>>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>>>> create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
>>>>
>>>> Am I missing the way to get info in the ESL about gateways, which are
>>>> out of order, or there is simple no way, without hacking the code?
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anton.vazir at gmail.com  Sat May 14 22:18:55 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 14 May 2011 23:18:55 +0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTim3HG5YHG6sKRnU+PZmgg3pKL2Q9Q@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
	<BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
	<BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>
	<BANLkTim3HG5YHG6sKRnU+PZmgg3pKL2Q9Q@mail.gmail.com>
Message-ID: <BANLkTin91Senm8SAoRNxuS+TDJWyxcLN1w@mail.gmail.com>

If I'm not wrong mod_distribute just provides a list for dialing the
set of gateways, but not generating events itself. So seems not for
the case anyway...

2011/5/14 Anton VG <anton.vazir at gmail.com>:
> That good one, but not for the my case.
>
> i use originate &park, and than bridge_uuid, when there is an early_media.
> I have a number of gateways, which support specific destinations each,
> so it's up to my billing to decide what gateway should be dialed and
> in which order. But I still need to determine if gateway could be
> reached or not, or if while calling, it gives an error, and which one.
>
> I see gwlist down could be used for bridge, but bridge does not give
> flexibility I try to achieve. Will see what it gives if used for
> originate...
>
> So, considering your brief reply, there just no support for the case I
> need, so will try to get inside sofia.c ...
>
> Regards,
> Anton.
>
> 2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
>> Read up on mod distributor on the wiki.
>>
>> On May 14, 2011 12:12 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>>> You did not understand. I INTENTIONALLY dialing the bad gateway, and
>>> I'm looking for a proper way to determine that gateway is bad in my
>>> ESL dialplan, by catching the proper event/reply/whatever,
>>> And much preferably without tricks, like esl.api('sofia status gateway
>>> GatewayWhichIsDown')
>>>
>>> When in production, and there is more than a single route, there will
>>> be plenty of cases, when you dial a bad gateway, so there should be a
>>> way for ESL dialplan to determine that a gateway is not callable for a
>>> moment, the reason WHY and to retry with another one.
>>>
>>> The trick above is bad, since:
>>> 1. blocking api query, before evey single gateway call attempt.
>>> 2. Gateway maybe known in UP state, but the state is stale, in dial in
>>> fact will go to DOWN gateway. So, dialplan will screw
>>>
>>> Possibly I should ask in DEV list...
>>>
>>> 2011/5/14 Madovsky <infos at madovsky.org>:
>>>> maybe your gateway is blocking some numbers
>>>>
>>>> ----- Original Message -----
>>>> From: "Anton VG" <anton.vazir at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Saturday, May 14, 2011 12:23 PM
>>>> Subject: Re: [Freeswitch-users] ESL: No events fired when there is error
>>>> on
>>>> submitted API command, like originate sofia to non-existent gateway
>>>>
>>>>
>>>>> The same goes for gateway, which is just down. No events, signalling
>>>>> that call will not succeed. And no events fired.
>>>>>
>>>>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>>>>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>>>>> create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
>>>>>
>>>>> Am I missing the way to get info in the ESL about gateways, which are
>>>>> out of order, or there is simple no way, without hacking the code?
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From anthony.minessale at gmail.com  Sat May 14 22:43:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 14 May 2011 13:43:23 -0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTin91Senm8SAoRNxuS+TDJWyxcLN1w@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
	<BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
	<BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>
	<BANLkTim3HG5YHG6sKRnU+PZmgg3pKL2Q9Q@mail.gmail.com>
	<BANLkTin91Senm8SAoRNxuS+TDJWyxcLN1w@mail.gmail.com>
Message-ID: <BANLkTikSqNJKLSguKbx7xpxCx-5_mP+HCA@mail.gmail.com>

It works with mod sofia dialplan app to test if a gw is down.
That's your hint ;)
On May 14, 2011 1:19 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
> If I'm not wrong mod_distribute just provides a list for dialing the
> set of gateways, but not generating events itself. So seems not for
> the case anyway...
>
> 2011/5/14 Anton VG <anton.vazir at gmail.com>:
>> That good one, but not for the my case.
>>
>> i use originate &park, and than bridge_uuid, when there is an
early_media.
>> I have a number of gateways, which support specific destinations each,
>> so it's up to my billing to decide what gateway should be dialed and
>> in which order. But I still need to determine if gateway could be
>> reached or not, or if while calling, it gives an error, and which one.
>>
>> I see gwlist down could be used for bridge, but bridge does not give
>> flexibility I try to achieve. Will see what it gives if used for
>> originate...
>>
>> So, considering your brief reply, there just no support for the case I
>> need, so will try to get inside sofia.c ...
>>
>> Regards,
>> Anton.
>>
>> 2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
>>> Read up on mod distributor on the wiki.
>>>
>>> On May 14, 2011 12:12 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>>>> You did not understand. I INTENTIONALLY dialing the bad gateway, and
>>>> I'm looking for a proper way to determine that gateway is bad in my
>>>> ESL dialplan, by catching the proper event/reply/whatever,
>>>> And much preferably without tricks, like esl.api('sofia status gateway
>>>> GatewayWhichIsDown')
>>>>
>>>> When in production, and there is more than a single route, there will
>>>> be plenty of cases, when you dial a bad gateway, so there should be a
>>>> way for ESL dialplan to determine that a gateway is not callable for a
>>>> moment, the reason WHY and to retry with another one.
>>>>
>>>> The trick above is bad, since:
>>>> 1. blocking api query, before evey single gateway call attempt.
>>>> 2. Gateway maybe known in UP state, but the state is stale, in dial in
>>>> fact will go to DOWN gateway. So, dialplan will screw
>>>>
>>>> Possibly I should ask in DEV list...
>>>>
>>>> 2011/5/14 Madovsky <infos at madovsky.org>:
>>>>> maybe your gateway is blocking some numbers
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Anton VG" <anton.vazir at gmail.com>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Saturday, May 14, 2011 12:23 PM
>>>>> Subject: Re: [Freeswitch-users] ESL: No events fired when there is
error
>>>>> on
>>>>> submitted API command, like originate sofia to non-existent gateway
>>>>>
>>>>>
>>>>>> The same goes for gateway, which is just down. No events, signalling
>>>>>> that call will not succeed. And no events fired.
>>>>>>
>>>>>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>>>>>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>>>>>> create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
>>>>>>
>>>>>> Am I missing the way to get info in the ESL about gateways, which are
>>>>>> out of order, or there is simple no way, without hacking the code?
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anton.vazir at gmail.com  Sat May 14 22:51:58 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 14 May 2011 23:51:58 +0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTikSqNJKLSguKbx7xpxCx-5_mP+HCA@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
	<BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
	<BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>
	<BANLkTim3HG5YHG6sKRnU+PZmgg3pKL2Q9Q@mail.gmail.com>
	<BANLkTin91Senm8SAoRNxuS+TDJWyxcLN1w@mail.gmail.com>
	<BANLkTikSqNJKLSguKbx7xpxCx-5_mP+HCA@mail.gmail.com>
Message-ID: <BANLkTi=N-+vxKRoA-u4ajZeRJ-Mb3P-3JQ@mail.gmail.com>

Yes, it does :) And I can use it to test is gw is up or down.
But question digs slightly deeper, it looks a kind odd, that when I
issue the non-working 'originate' or 'bridge'  - there is no ERROR
event at least. So seems there is no way to determine in ESL, that
'originate' or other command have fauiled have failed. On successful
execution - there is "BACKGROUD_JOB" event. But no event on error.
Don't you think that errors should fire events either, to inform ESL
dialplan that issued command have failed?


2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
> It works with mod sofia dialplan app to test if a gw is down.
> That's your hint ;)
>
> On May 14, 2011 1:19 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>> If I'm not wrong mod_distribute just provides a list for dialing the
>> set of gateways, but not generating events itself. So seems not for
>> the case anyway...
>>
>> 2011/5/14 Anton VG <anton.vazir at gmail.com>:
>>> That good one, but not for the my case.
>>>
>>> i use originate &park, and than bridge_uuid, when there is an
>>> early_media.
>>> I have a number of gateways, which support specific destinations each,
>>> so it's up to my billing to decide what gateway should be dialed and
>>> in which order. But I still need to determine if gateway could be
>>> reached or not, or if while calling, it gives an error, and which one.
>>>
>>> I see gwlist down could be used for bridge, but bridge does not give
>>> flexibility I try to achieve. Will see what it gives if used for
>>> originate...
>>>
>>> So, considering your brief reply, there just no support for the case I
>>> need, so will try to get inside sofia.c ...
>>>
>>> Regards,
>>> Anton.
>>>
>>> 2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
>>>> Read up on mod distributor on the wiki.
>>>>
>>>> On May 14, 2011 12:12 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>>>>> You did not understand. I INTENTIONALLY dialing the bad gateway, and
>>>>> I'm looking for a proper way to determine that gateway is bad in my
>>>>> ESL dialplan, by catching the proper event/reply/whatever,
>>>>> And much preferably without tricks, like esl.api('sofia status gateway
>>>>> GatewayWhichIsDown')
>>>>>
>>>>> When in production, and there is more than a single route, there will
>>>>> be plenty of cases, when you dial a bad gateway, so there should be a
>>>>> way for ESL dialplan to determine that a gateway is not callable for a
>>>>> moment, the reason WHY and to retry with another one.
>>>>>
>>>>> The trick above is bad, since:
>>>>> 1. blocking api query, before evey single gateway call attempt.
>>>>> 2. Gateway maybe known in UP state, but the state is stale, in dial in
>>>>> fact will go to DOWN gateway. So, dialplan will screw
>>>>>
>>>>> Possibly I should ask in DEV list...
>>>>>
>>>>> 2011/5/14 Madovsky <infos at madovsky.org>:
>>>>>> maybe your gateway is blocking some numbers
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Anton VG" <anton.vazir at gmail.com>
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Saturday, May 14, 2011 12:23 PM
>>>>>> Subject: Re: [Freeswitch-users] ESL: No events fired when there is
>>>>>> error
>>>>>> on
>>>>>> submitted API command, like originate sofia to non-existent gateway
>>>>>>
>>>>>>
>>>>>>> The same goes for gateway, which is just down. No events, signalling
>>>>>>> that call will not succeed. And no events fired.
>>>>>>>
>>>>>>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>>>>>>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>>>>>>> create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
>>>>>>>
>>>>>>> Am I missing the way to get info in the ESL about gateways, which are
>>>>>>> out of order, or there is simple no way, without hacking the code?
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From Nabble at slickdeals.endjunk.com  Sat May 14 13:12:30 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 14 May 2011 02:12:30 -0700 (PDT)
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
Message-ID: <1305364350375-6362642.post@n2.nabble.com>


Avi Marcus-2 wrote:
> 
> I've tried this cool formula for streaming TTS via google:
> 
> http://wiki.freeswitch.org/wiki/TTS
> 
> and while the link produces a pretty darn nice sounding MP3, I get:
> 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at
> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123:
> Invalid
> mpg123 handle. (code 10)
> 
> Is anyone else using this reliably?
I ran into some problem with streaming TTS via Google as shown 
http://freeswitch-users.2379917.n2.nabble.com/Playing-Google-translation-tts-tp5916138p5920422.html
here  and 
http://freeswitch-users.2379917.n2.nabble.com/Problem-with-mod-shout-on-FS-git-tp5977722p5977722.html
here . However, I thought to give it a try and it works now on my FreeSWITCH
Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500) hosted on a Seagate
DockStar running on an OpenWRT firmware.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Is-Anyone-Using-Google-TTS-tp6325535p6362642.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Sat May 14 23:29:07 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 14 May 2011 12:29:07 -0700 (PDT)
Subject: [Freeswitch-users] How to eliminate acoustic echo from server
	side(FreeSwitch)
In-Reply-To: <4DC75B8D.9090008@163.com>
References: <4DC75B8D.9090008@163.com>
Message-ID: <1305401347581-6363765.post@n2.nabble.com>


vivid333 wrote:
> 
> In the multi-party conference, because some termianl phone does not 
> process echo, which causes the call have echo, so SERVER side needs to 
> do echo processing. How to eliminate echo in this scenario?
This sounded like an acoustic echo. If so, the best way to eliminate the
echo is to reduce the sensitivity of miccrophone while reducing the volume
on speaker to a certain level so that the microphone won't be able to pick
up the signals from speaker(s).

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-eliminate-acoustic-echo-from-server-side-FreeSwitch-tp6343051p6363765.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From banhi.dutra at gmail.com  Sat May 14 12:16:53 2011
From: banhi.dutra at gmail.com (Banhi Dutra)
Date: Sat, 14 May 2011 10:16:53 +0200
Subject: [Freeswitch-users] Different SAS with ZRTP
In-Reply-To: <BANLkTi=bv703hO6Xjq2Z=+1F9vnEJJ3q_w@mail.gmail.com>
References: <BANLkTinsdQvkvgtuV8SEYOY7pP46TYYeow@mail.gmail.com>
	<BANLkTi=bv703hO6Xjq2Z=+1F9vnEJJ3q_w@mail.gmail.com>
Message-ID: <BANLkTinsd3AxT_y6KukEX7jhFpd4xMqHnA@mail.gmail.com>

Thank you for your reply.

How can I check this ? I have default configuration and I searched for the
string "<action application="set" data="zrtp_enrollment=true"/>" but I
didn't found anything. I'm sorry for stupid question, but I'm beginning with
 FS.

Banhi

2011/5/14 Michael Collins <msc at freeswitch.org>

> Is FS a trusted man in the middle in this scenario?
> -MC
>
> On Fri, May 13, 2011 at 2:50 PM, Banhi Dutra <banhi.dutra at gmail.com>wrote:
>
>> Hi all,
>>
>>
>> I installed FS in test environment with ZRTP enabled. I have two softphone
>> (I tested with Acrobits and Jitsi) registered and all works fine, except for
>> SAS exchange, which are different. I'm using libzrtp 0.81.514 and
>>  FreeSWITCH Version 1.0.head (git-23d8658 2011-05-07 00-27-20 -0400). I read
>> older post about a similar problem. Could be a version problem ?
>>
>>
>> Any help will be appreciated
>>
>>
>> Thank you
>>
>> Banhi
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Sun May 15 00:33:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 14 May 2011 15:33:35 -0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTi=N-+vxKRoA-u4ajZeRJ-Mb3P-3JQ@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
	<BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
	<BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>
	<BANLkTim3HG5YHG6sKRnU+PZmgg3pKL2Q9Q@mail.gmail.com>
	<BANLkTin91Senm8SAoRNxuS+TDJWyxcLN1w@mail.gmail.com>
	<BANLkTikSqNJKLSguKbx7xpxCx-5_mP+HCA@mail.gmail.com>
	<BANLkTi=N-+vxKRoA-u4ajZeRJ-Mb3P-3JQ@mail.gmail.com>
Message-ID: <BANLkTikdFenZKZOFahh8sqQn=CW4kJ115A@mail.gmail.com>

There is always a result to originate and its always in a background job
event success or fail. In the body of the event you will find the return
value.
On May 14, 2011 1:52 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
> Yes, it does :) And I can use it to test is gw is up or down.
> But question digs slightly deeper, it looks a kind odd, that when I
> issue the non-working 'originate' or 'bridge' - there is no ERROR
> event at least. So seems there is no way to determine in ESL, that
> 'originate' or other command have fauiled have failed. On successful
> execution - there is "BACKGROUD_JOB" event. But no event on error.
> Don't you think that errors should fire events either, to inform ESL
> dialplan that issued command have failed?
>
>
> 2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
>> It works with mod sofia dialplan app to test if a gw is down.
>> That's your hint ;)
>>
>> On May 14, 2011 1:19 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>>> If I'm not wrong mod_distribute just provides a list for dialing the
>>> set of gateways, but not generating events itself. So seems not for
>>> the case anyway...
>>>
>>> 2011/5/14 Anton VG <anton.vazir at gmail.com>:
>>>> That good one, but not for the my case.
>>>>
>>>> i use originate &park, and than bridge_uuid, when there is an
>>>> early_media.
>>>> I have a number of gateways, which support specific destinations each,
>>>> so it's up to my billing to decide what gateway should be dialed and
>>>> in which order. But I still need to determine if gateway could be
>>>> reached or not, or if while calling, it gives an error, and which one.
>>>>
>>>> I see gwlist down could be used for bridge, but bridge does not give
>>>> flexibility I try to achieve. Will see what it gives if used for
>>>> originate...
>>>>
>>>> So, considering your brief reply, there just no support for the case I
>>>> need, so will try to get inside sofia.c ...
>>>>
>>>> Regards,
>>>> Anton.
>>>>
>>>> 2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>> Read up on mod distributor on the wiki.
>>>>>
>>>>> On May 14, 2011 12:12 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>>>>>> You did not understand. I INTENTIONALLY dialing the bad gateway, and
>>>>>> I'm looking for a proper way to determine that gateway is bad in my
>>>>>> ESL dialplan, by catching the proper event/reply/whatever,
>>>>>> And much preferably without tricks, like esl.api('sofia status
gateway
>>>>>> GatewayWhichIsDown')
>>>>>>
>>>>>> When in production, and there is more than a single route, there will
>>>>>> be plenty of cases, when you dial a bad gateway, so there should be a
>>>>>> way for ESL dialplan to determine that a gateway is not callable for
a
>>>>>> moment, the reason WHY and to retry with another one.
>>>>>>
>>>>>> The trick above is bad, since:
>>>>>> 1. blocking api query, before evey single gateway call attempt.
>>>>>> 2. Gateway maybe known in UP state, but the state is stale, in dial
in
>>>>>> fact will go to DOWN gateway. So, dialplan will screw
>>>>>>
>>>>>> Possibly I should ask in DEV list...
>>>>>>
>>>>>> 2011/5/14 Madovsky <infos at madovsky.org>:
>>>>>>> maybe your gateway is blocking some numbers
>>>>>>>
>>>>>>> ----- Original Message -----
>>>>>>> From: "Anton VG" <anton.vazir at gmail.com>
>>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>> Sent: Saturday, May 14, 2011 12:23 PM
>>>>>>> Subject: Re: [Freeswitch-users] ESL: No events fired when there is
>>>>>>> error
>>>>>>> on
>>>>>>> submitted API command, like originate sofia to non-existent gateway
>>>>>>>
>>>>>>>
>>>>>>>> The same goes for gateway, which is just down. No events,
signalling
>>>>>>>> that call will not succeed. And no events fired.
>>>>>>>>
>>>>>>>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>>>>>>>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>>>>>>>> create outgoing channel of type [sofia] cause:
[NETWORK_OUT_OF_ORDER]
>>>>>>>>
>>>>>>>> Am I missing the way to get info in the ESL about gateways, which
are
>>>>>>>> out of order, or there is simple no way, without hacking the code?
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anton.vazir at gmail.com  Sun May 15 01:04:54 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 15 May 2011 02:04:54 +0500
Subject: [Freeswitch-users] ESL: No events fired when there is error on
 submitted API command, like originate sofia to non-existent gateway
In-Reply-To: <BANLkTikdFenZKZOFahh8sqQn=CW4kJ115A@mail.gmail.com>
References: <BANLkTimJF+5d8aFDi=k++uNmpf=Az34Bnw@mail.gmail.com>
	<BANLkTi=y2e+1TA-XBv_AU3z1hXvwJB5qZg@mail.gmail.com>
	<23EBA3985DA74FA99ED187BAD3DA0014@e1705>
	<BANLkTimqc-RHL5vy=6j9JW5vwkS3ZQ+A=Q@mail.gmail.com>
	<BANLkTinMf+A-2xJxAkg3FKLMd1kT0oC_OQ@mail.gmail.com>
	<BANLkTim3HG5YHG6sKRnU+PZmgg3pKL2Q9Q@mail.gmail.com>
	<BANLkTin91Senm8SAoRNxuS+TDJWyxcLN1w@mail.gmail.com>
	<BANLkTikSqNJKLSguKbx7xpxCx-5_mP+HCA@mail.gmail.com>
	<BANLkTi=N-+vxKRoA-u4ajZeRJ-Mb3P-3JQ@mail.gmail.com>
	<BANLkTikdFenZKZOFahh8sqQn=CW4kJ115A@mail.gmail.com>
Message-ID: <BANLkTikUpSCn1JdpLjwGmNDZFt_VV3pbCQ@mail.gmail.com>

What the shame... Reimplementation in plain event receiver catches the
BACKGROUND_JOB event on bad gateway. Seems I'm loosing some events in
multi-threaded app.
I was so sure I'm getting all events. Thanks so much Anthony, and
sorry for bothering, usually I double check in advance. Seems I have
to get some sleep.

2011/5/15 Anthony Minessale <anthony.minessale at gmail.com>:
> There is always a result to originate and its always in a background job
> event success or fail. In the body of the event you will find the return
> value.
>
> On May 14, 2011 1:52 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>> Yes, it does :) And I can use it to test is gw is up or down.
>> But question digs slightly deeper, it looks a kind odd, that when I
>> issue the non-working 'originate' or 'bridge' - there is no ERROR
>> event at least. So seems there is no way to determine in ESL, that
>> 'originate' or other command have fauiled have failed. On successful
>> execution - there is "BACKGROUD_JOB" event. But no event on error.
>> Don't you think that errors should fire events either, to inform ESL
>> dialplan that issued command have failed?
>>
>>
>> 2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
>>> It works with mod sofia dialplan app to test if a gw is down.
>>> That's your hint ;)
>>>
>>> On May 14, 2011 1:19 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>>>> If I'm not wrong mod_distribute just provides a list for dialing the
>>>> set of gateways, but not generating events itself. So seems not for
>>>> the case anyway...
>>>>
>>>> 2011/5/14 Anton VG <anton.vazir at gmail.com>:
>>>>> That good one, but not for the my case.
>>>>>
>>>>> i use originate &park, and than bridge_uuid, when there is an
>>>>> early_media.
>>>>> I have a number of gateways, which support specific destinations each,
>>>>> so it's up to my billing to decide what gateway should be dialed and
>>>>> in which order. But I still need to determine if gateway could be
>>>>> reached or not, or if while calling, it gives an error, and which one.
>>>>>
>>>>> I see gwlist down could be used for bridge, but bridge does not give
>>>>> flexibility I try to achieve. Will see what it gives if used for
>>>>> originate...
>>>>>
>>>>> So, considering your brief reply, there just no support for the case I
>>>>> need, so will try to get inside sofia.c ...
>>>>>
>>>>> Regards,
>>>>> Anton.
>>>>>
>>>>> 2011/5/14 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>>> Read up on mod distributor on the wiki.
>>>>>>
>>>>>> On May 14, 2011 12:12 PM, "Anton VG" <anton.vazir at gmail.com> wrote:
>>>>>>> You did not understand. I INTENTIONALLY dialing the bad gateway, and
>>>>>>> I'm looking for a proper way to determine that gateway is bad in my
>>>>>>> ESL dialplan, by catching the proper event/reply/whatever,
>>>>>>> And much preferably without tricks, like esl.api('sofia status
>>>>>>> gateway
>>>>>>> GatewayWhichIsDown')
>>>>>>>
>>>>>>> When in production, and there is more than a single route, there will
>>>>>>> be plenty of cases, when you dial a bad gateway, so there should be a
>>>>>>> way for ESL dialplan to determine that a gateway is not callable for
>>>>>>> a
>>>>>>> moment, the reason WHY and to retry with another one.
>>>>>>>
>>>>>>> The trick above is bad, since:
>>>>>>> 1. blocking api query, before evey single gateway call attempt.
>>>>>>> 2. Gateway maybe known in UP state, but the state is stale, in dial
>>>>>>> in
>>>>>>> fact will go to DOWN gateway. So, dialplan will screw
>>>>>>>
>>>>>>> Possibly I should ask in DEV list...
>>>>>>>
>>>>>>> 2011/5/14 Madovsky <infos at madovsky.org>:
>>>>>>>> maybe your gateway is blocking some numbers
>>>>>>>>
>>>>>>>> ----- Original Message -----
>>>>>>>> From: "Anton VG" <anton.vazir at gmail.com>
>>>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>>>> Sent: Saturday, May 14, 2011 12:23 PM
>>>>>>>> Subject: Re: [Freeswitch-users] ESL: No events fired when there is
>>>>>>>> error
>>>>>>>> on
>>>>>>>> submitted API command, like originate sofia to non-existent gateway
>>>>>>>>
>>>>>>>>
>>>>>>>>> The same goes for gateway, which is just down. No events,
>>>>>>>>> signalling
>>>>>>>>> that call will not succeed. And no events fired.
>>>>>>>>>
>>>>>>>>> 2011-05-14 21:19:32.002929 [ERR] mod_sofia.c:4050 Gateway is down!
>>>>>>>>> 2011-05-14 21:19:32.002929 [ERR] switch_ivr_originate.c:2447 Cannot
>>>>>>>>> create outgoing channel of type [sofia] cause:
>>>>>>>>> [NETWORK_OUT_OF_ORDER]
>>>>>>>>>
>>>>>>>>> Am I missing the way to get info in the ESL about gateways, which
>>>>>>>>> are
>>>>>>>>> out of order, or there is simple no way, without hacking the code?
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sun May 15 01:01:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 14 May 2011 14:01:09 -0700
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <Pine.LNX.4.64.1105141407100.2351@debian>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<Pine.LNX.4.64.1105140315200.9833@debian>
	<0F189258C16A4A178DAA1D1C7669A03D@e1705>
	<Pine.LNX.4.64.1105141407100.2351@debian>
Message-ID: <BANLkTikRVJjprzQOdqxWLw651m5qFu_JEA@mail.gmail.com>

ZOMG I wish I was blind like Thomas, Covici, and DelphiWorld so I wouldn't
have to read this thread any more!! :P </rant>

What you want to do is not possible directly from the dialplan, or even from
the console. There are probably several ways to address this problem. Since
I like Lua I thought a simple demonstration using a small amount of Lua to
get what you want would be good. Here is a generic Lua script that you can
call from the command line and feed it an argument which it will in turn
execute as a system call, then return the system's output:

-- sys_call.lua
--
-- simple read from pipe
-- Be sure to escape spaces in the argument cuz I was to lazy to parse args
properly...



cmd = argv[1];
freeswitch.consoleLog("DEBUG","sys_call.lua call with arg: '" .. cmd ..
"'\n");

f = assert (io.popen(cmd));
res_data = '';
for line in f:lines() do
  res_data = res_data .. line .. "\n";
end



stream:write(res_data);


freeswitch.consoleLog("DEBUG","sys_call.lua result: '" .. res_data ..
"'\n");

f:close();

Now you can go to fs_cli and do things like:
lua sys_call.lua ls\ -l

Now, Thomas, in your specific case you are trying to capture input from the
caller and compare that to the output from the command: date -d <date
argument> +%s and you have a script that does this comparison for you. To
call your shell script from the command line with my Lua script would be
this:

lua sys_call.lua datechk.sh\ <date value argument>

(Don't forget to put a backslash in front of the space character.) To do
this in the dialplan you need to use the special notation that lets you
execute an API call using the "set" dialplan application. Let's say the
input you capture from your caller is in variable ${nr}. You can set a new
channel variable that captures the output of your script:

<action application="set" data="my_result=${lua sys_call.lua datechk.sh\
${nr}}"/>

Now you have a new channel variable ${my_result} that contains the result of
the system call to your date script. From here you will need use the
dialplan "transfer" or "execute_extension" in order to do something with the
result. I recommend that you create a separate dialplan context and transfer
the call into that context for further processing.

<action application="transfer" data="${my_result} XML handle_date_test"/>

Then your new context could be something like this:
<context name="handle_date_test">
  <extension name="date match is good">
    <condition field="destination_number" expression="^good\s*$">
      <action application="log" data="INFO date matches, do stuff in action
tags..."/>
      <anti-action application="log" data="INFO date does not match, handle
this in anti-action tags..."/>
    </condtion>
  </extension>
</context>

For all of this to work you will have to change your script so that if the
dates match then the script will echo "good" to the console. If the dates do
not match then your script will need to echo anything other than "good".

Thomas, I respect you for trying so hard to make this work with only a
braille reader and in a language that is not your first. Keep up the good
work. I hope this helps.

-MC

On Sat, May 14, 2011 at 5:11 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Am 14.05.11 um 01:37 schrieb Madovsky:
>
>
>  I think you have to escape the $ 3 times
>>
> I don.t no. $3? I like to export the variable: dateerror  from the
> shellscript in my fs dialplan, to make a selection from condidion field.
>
>> \\\${nir}
>>
> what is the syntax in the dialplan?
> thank you.
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From grsingh750 at gmail.com  Sun May 15 02:31:49 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sun, 15 May 2011 04:01:49 +0530
Subject: [Freeswitch-users] mod_freetdm not compiling on archlinux
	2.6.32-lts kernel x86_64
In-Reply-To: <BANLkTikyTkQsiCMRs8FVoO-yBCtO-5JS-g@mail.gmail.com>
References: <BANLkTikyTkQsiCMRs8FVoO-yBCtO-5JS-g@mail.gmail.com>
Message-ID: <BANLkTims-9izKUrkTbn8pPhGT5fsdf+TgA@mail.gmail.com>

Just FYI
Fixed on latest git. Thanks stkn
commit 9cceb8e62ca2473df5cb564d97b0b404c96212e4
Author: Stefan Knoblich <s.knoblich at axsentis.de>
Date:   Sat May 14 23:59:14 2011 +0200

    FreeTDM: gcc-4.6 fix (-Wunused-but-set)

    ftmod_wanpipe.c:
        Remove myerrno variable in wanpipe_read(),
        snprintf does not set errno (according to the manpage),
        so no need to save it (without even using it later).

regards
gsin


On Sat, May 14, 2011 at 2:07 AM, guru singh <grsingh750 at gmail.com> wrote:
> Hi,
>
> Trying to install FS with mod_freetdm.
> Fails to compile. I have wanpipe installed
> Error : http://pastebin.freeswitch.org/16296
>
> Can anybody with a working setup on this platform comment on this please?
>
> Thanks
> guru
>


From garbytrash at gmail.com  Sun May 15 03:26:47 2011
From: garbytrash at gmail.com (Zenny)
Date: Sun, 15 May 2011 01:26:47 +0200
Subject: [Freeswitch-users] Compiling in Debian Squeeze fails
Message-ID: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>

Tried to compile with the Makefile downloaded from wget
http://www.freeswitch.org/eg/Makefile to Debian Squeeze, it popped up an
error message while compiling mod_spandsp (missing -ljpeg):

libtool: link: ranlib .libs/libspandsp.a
> libtool: link: ( cd ".libs" && rm -f "libspandsp.la" && ln -s "../
> libspandsp.la" "libspandsp.la" )
> make[8]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp/src'
> make[7]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp/src'
> make[6]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp'
> Creating mod_spandsp.la
> /usr/bin/ld: cannot find -ljpeg
> collect2: ld returned 1 exit status
> quiet_libtool: link: gcc -shared  .libs/mod_spandsp_la-mod_spandsp.o
> .libs/mod_spandsp_la-udptl.o .libs/mod_spandsp_la-mod_spandsp_fax.o
> .libs/mod_spandsp_la-mod_spandsp_dsp.o
> .libs/mod_spandsp_la-mod_spandsp_codecs.o   -Wl,-rpath
> -Wl,/usr/src/freeswitch.git/.libs -Wl,-rpath -Wl,/usr/local/freeswitch/lib
> -ljpeg /usr/src/freeswitch.git/.libs/libfreeswitch.so
> /usr/src/freeswitch.git/libs/spandsp/src/.libs/libspandsp.a
> /usr/src/freeswitch.git/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -lncurses
> -pthread -Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so
> make[5]: *** [mod_spandsp.la] Error 1
> make[5]: Leaving directory
> `/usr/src/freeswitch.git/src/mod/applications/mod_spandsp'
> make[4]: *** [mod_spandsp-all] Error 1
> make[4]: Leaving directory `/usr/src/freeswitch.git/src/mod'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory `/usr/src/freeswitch.git/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch.git'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/src/freeswitch.git'
> make: *** [freeswitch] Error 2
>
>
What package exactly satisfies this dependency problem?

Thanks!

/zenny
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From philippe at ppmt.org  Sun May 15 03:45:14 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sat, 14 May 2011 19:45:14 -0400
Subject: [Freeswitch-users] Compiling in Debian Squeeze fails
In-Reply-To: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
References: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
Message-ID: <4DCF140A.6030005@ppmt.org>

it complains that it can't find ljpeg which is prereq and contained in
libjpeg

so install it. On mine I actually installed libjpeg8-dev and libjpeg8

for more info see:

http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix

Towards the end there is a section on the pre requisites



On 11-05-14 07:26 PM, Zenny wrote:
> Tried to compile with the Makefile downloaded from wget
> http://www.freeswitch.org/eg/Makefile to Debian Squeeze, it popped up
> an error message while compiling mod_spandsp (missing -ljpeg):
>
>     libtool: link: ranlib .libs/libspandsp.a
>     libtool: link: ( cd ".libs" && rm -f "libspandsp.la
>     <http://libspandsp.la>" && ln -s "../libspandsp.la
>     <http://libspandsp.la>" "libspandsp.la <http://libspandsp.la>" )
>     make[8]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp/src'
>     make[7]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp/src'
>     make[6]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp'
>     Creating mod_spandsp.la <http://mod_spandsp.la>
>     /usr/bin/ld: cannot find -ljpeg
>     collect2: ld returned 1 exit status
>     quiet_libtool: link: gcc -shared 
>     .libs/mod_spandsp_la-mod_spandsp.o .libs/mod_spandsp_la-udptl.o
>     .libs/mod_spandsp_la-mod_spandsp_fax.o
>     .libs/mod_spandsp_la-mod_spandsp_dsp.o
>     .libs/mod_spandsp_la-mod_spandsp_codecs.o   -Wl,-rpath
>     -Wl,/usr/src/freeswitch.git/.libs -Wl,-rpath
>     -Wl,/usr/local/freeswitch/lib -ljpeg
>     /usr/src/freeswitch.git/.libs/libfreeswitch.so
>     /usr/src/freeswitch.git/libs/spandsp/src/.libs/libspandsp.a
>     /usr/src/freeswitch.git/libs/tiff-3.8.2/libtiff/.libs/libtiff.a
>     -lncurses    -pthread -Wl,-soname -Wl,mod_spandsp.so -o
>     .libs/mod_spandsp.so
>     make[5]: *** [mod_spandsp.la <http://mod_spandsp.la>] Error 1
>     make[5]: Leaving directory
>     `/usr/src/freeswitch.git/src/mod/applications/mod_spandsp'
>     make[4]: *** [mod_spandsp-all] Error 1
>     make[4]: Leaving directory `/usr/src/freeswitch.git/src/mod'
>     make[3]: *** [all-recursive] Error 1
>     make[3]: Leaving directory `/usr/src/freeswitch.git/src'
>     make[2]: *** [all-recursive] Error 1
>     make[2]: Leaving directory `/usr/src/freeswitch.git'
>     make[1]: *** [all] Error 2
>     make[1]: Leaving directory `/usr/src/freeswitch.git'
>     make: *** [freeswitch] Error 2
>
>
> What package exactly satisfies this dependency problem?
>
> Thanks!
>
> /zenny
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From admin at blindi.net  Sun May 15 04:20:16 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 15 May 2011 02:20:16 +0200 (CEST)
Subject: [Freeswitch-users] Error to play multiple file with
	play_and_get_digits
In-Reply-To: <4DCF140A.6030005@ppmt.org>
References: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
	<4DCF140A.6030005@ppmt.org>
Message-ID: <Pine.LNX.4.64.1105150211500.7979@debian>

Hi all, fs, don.t play multiple files.
my dialplan:
  <action application="set" data="playback_delimiter=!"/>
  <action application="set" data="playback_sleep_val=500"/>
   <action application="play_and_get_digits" data="0 42 1 10000 #
   file_string:///tmp/1.alaw,/tmp/2.alaw 
$${base_dir}/sounds/callback/invalid.wav   NR1 \d+"/>
I become the following error on the console:

EXECUTE sofia/internal/1000 at sip2.blindi.net set(playback_delimiter=!)
2011-05-15 02:09:03.316798 [DEBUG] mod_dptools.c:1060 
sofia/internal/1000 at sip2.b
lindi.net SET [playback_delimiter]=[!]
EXECUTE sofia/internal/1000 at sip2.blindi.net set(playback_sleep_val=500)
2011-05-15 02:09:03.316798 [DEBUG] mod_dptools.c:1060 
sofia/internal/1000 at sip2.b
lindi.net SET [playback_sleep_val]=[500]
EXECUTE sofia/internal/1000 at sip2.blindi.net play_and_get_digits(0 42 1 
10000 #
  file_string:///tmp/1.alaw,/tmp/2.alaw 
/usr/local/freeswitch/sounds/callback/i
nvalid.alaw NR1 \d+)
2011-05-15 02:09:03.316798 [ERR] mod_sndfile.c:194 Error Opening File 
[/tmp/1.alaw,/tmp/2.alaw
aw,/tmp/2.alaw] [System error : No such file or directory.]

The files are exists.
What ist the problem 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 admin at blindi.net  Sun May 15 04:48:30 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 15 May 2011 02:48:30 +0200 (CEST)
Subject: [Freeswitch-users] problem exporting variables from shellsctipt
 not working
In-Reply-To: <BANLkTikRVJjprzQOdqxWLw651m5qFu_JEA@mail.gmail.com>
References: <750685.42888.qm@web59408.mail.ac4.yahoo.com>
	<Pine.LNX.4.64.1105120530140.9184@debian>
	<BANLkTikrwWhBpzYgyBJWCsTN3j5Vdy6KZw@mail.gmail.com>
	<Pine.LNX.4.64.1105132046460.5405@debian>
	<51E0B1392ADF44069FCB8537C2AE75B1@e1705>
	<Pine.LNX.4.64.1105140315200.9833@debian>
	<0F189258C16A4A178DAA1D1C7669A03D@e1705>
	<Pine.LNX.4.64.1105141407100.2351@debian>
	<BANLkTikRVJjprzQOdqxWLw651m5qFu_JEA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105150240580.7979@debian>


Hi Michael, thank you for your help.
From day to day I get more and more forward with fs.
I have already call back, call trough, call handling, email for missed 
calls,
and hammer out sms missed call in to.
All beginnings are difficult.




---------------
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 math.parent at gmail.com  Sun May 15 14:30:45 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Sun, 15 May 2011 12:30:45 +0200
Subject: [Freeswitch-users] Compiling in Debian Squeeze fails
In-Reply-To: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
References: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
Message-ID: <BANLkTi=TG_psccGjAjai0yrdhVqRCTxS=Q@mail.gmail.com>

2011/5/15 Zenny <garbytrash at gmail.com>:
> Tried to compile with the Makefile downloaded from wget
> http://www.freeswitch.org/eg/Makefile to Debian Squeeze, it popped up an
> error message while compiling mod_spandsp (missing -ljpeg):

I suggest using the "deb" method, as described by Gabriel at
<http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-November/004478.html>

Regards

-- 
Mathieu


From steveayre at gmail.com  Sun May 15 15:03:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 15 May 2011 12:03:05 +0100
Subject: [Freeswitch-users] Compiling in Debian Squeeze fails
In-Reply-To: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
References: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
Message-ID: <BANLkTim4pgnXhpUNP5LkmLnJOBA9a5nBOw@mail.gmail.com>

/usr/bin/ld: cannot find -ljpeg

You're missing libjpeg-dev.

-Steve


On 15 May 2011 00:26, Zenny <garbytrash at gmail.com> wrote:
> Tried to compile with the Makefile downloaded from wget
> http://www.freeswitch.org/eg/Makefile to Debian Squeeze, it popped up an
> error message while compiling mod_spandsp (missing -ljpeg):
>
>> libtool: link: ranlib .libs/libspandsp.a
>> libtool: link: ( cd ".libs" && rm -f "libspandsp.la" && ln -s
>> "../libspandsp.la" "libspandsp.la" )
>> make[8]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp/src'
>> make[7]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp/src'
>> make[6]: Leaving directory `/usr/src/freeswitch.git/libs/spandsp'
>> Creating mod_spandsp.la
>> /usr/bin/ld: cannot find -ljpeg
>> collect2: ld returned 1 exit status
>> quiet_libtool: link: gcc -shared? .libs/mod_spandsp_la-mod_spandsp.o
>> .libs/mod_spandsp_la-udptl.o .libs/mod_spandsp_la-mod_spandsp_fax.o
>> .libs/mod_spandsp_la-mod_spandsp_dsp.o
>> .libs/mod_spandsp_la-mod_spandsp_codecs.o?? -Wl,-rpath
>> -Wl,/usr/src/freeswitch.git/.libs -Wl,-rpath -Wl,/usr/local/freeswitch/lib
>> -ljpeg /usr/src/freeswitch.git/.libs/libfreeswitch.so
>> /usr/src/freeswitch.git/libs/spandsp/src/.libs/libspandsp.a
>> /usr/src/freeswitch.git/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -lncurses
>> -pthread -Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so
>> make[5]: *** [mod_spandsp.la] Error 1
>> make[5]: Leaving directory
>> `/usr/src/freeswitch.git/src/mod/applications/mod_spandsp'
>> make[4]: *** [mod_spandsp-all] Error 1
>> make[4]: Leaving directory `/usr/src/freeswitch.git/src/mod'
>> make[3]: *** [all-recursive] Error 1
>> make[3]: Leaving directory `/usr/src/freeswitch.git/src'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/usr/src/freeswitch.git'
>> make[1]: *** [all] Error 2
>> make[1]: Leaving directory `/usr/src/freeswitch.git'
>> make: *** [freeswitch] Error 2
>>
>
> What package exactly satisfies this dependency problem?
>
> Thanks!
>
> /zenny
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From covici at ccs.covici.com  Sun May 15 15:27:03 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 15 May 2011 07:27:03 -0400
Subject: [Freeswitch-users] call waiting problem
In-Reply-To: <BANLkTi=gS8AiqmqYtRWHAem=Pd9ZE590HA@mail.gmail.com>
References: <6364.1305238946@ccs.covici.com>
	<BANLkTi=gS8AiqmqYtRWHAem=Pd9ZE590HA@mail.gmail.com>
Message-ID: <10432.1305458823@ccs.covici.com>

The logs are at http://pastebin.freeswitch.org/16298

Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, May 12, 2011 at 4:22 PM,  <covici at ccs.covici.com> wrote:
> > Hi. ?I am using a Digium card for a phone extension. ?Now when someone
> > calls that extension through my IVR they hear a nice ring after
> > selecting the menu option. ?However, if I am on another call and they do
> > this, they get silence. ?I do hear the call waiting beep, but why don't
> > they hear the ringback or transfer_ringback? ?I tried it myself and
> > indeed get silence.
> 
> Did you make any progress on this?  If not, it would be helpful to see
> some logs.  What do you have set as the "transfer_ringback"?
> 
> Best,
> 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

-- 
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 moises.silva at gmail.com  Mon May 16 06:58:13 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 15 May 2011 22:58:13 -0400
Subject: [Freeswitch-users] wanpipe fails to compile on arch64
In-Reply-To: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>
References: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>
Message-ID: <BANLkTik9MY+6S5341+0rug1jD_Ouv3eZtw@mail.gmail.com>

On Wed, May 11, 2011 at 7:18 PM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
>
> I am trying to setup an arch box for FS. I cant install wanpipe.
> make freetdm fails with this error http://pastebin.freeswitch.org/16273
> uname -r 2.6.38-ARCH
> gcc version 4.6.0 20110429 (prerelease) (GCC)
>
> Could with a working arch box suggest a solution to this
>
>
This is the kernel API that changed yet again. You'll have to wait until a
Wanpipe driver update is released.

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
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From OSchenk at wnr.com.au  Mon May 16 09:08:17 2011
From: OSchenk at wnr.com.au (Schenk, Oliver)
Date: Mon, 16 May 2011 13:08:17 +0800
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
Message-ID: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>

Hey,

 

I was wondering if anyone out there in the FreeSwitch community has
successfully used FXO cards (e.g. VIC2-4FXO) in a CISCO router (e.g.
2811) with FreeSwitch.

 

Was this achieved via SIP communication between the CISCO and
FreeSwitch?

 

Any documentation out there? I can't seem to find much for FreeSwitch
regarding what would need to be done on the FreeSwitch end.

 

 

 

Regards,

 

Oliver Schenk

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
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From grsingh750 at gmail.com  Mon May 16 11:22:53 2011
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 16 May 2011 12:52:53 +0530
Subject: [Freeswitch-users] wanpipe fails to compile on arch64
In-Reply-To: <BANLkTik9MY+6S5341+0rug1jD_Ouv3eZtw@mail.gmail.com>
References: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>
	<BANLkTik9MY+6S5341+0rug1jD_Ouv3eZtw@mail.gmail.com>
Message-ID: <BANLkTi=KPCSwQKYe=OMs0X9b7ATz5YmEJA@mail.gmail.com>

Hi Moises

Thanks for your reply
I tried it on lts-kernel and it works.

regards
guru

On Mon, May 16, 2011 at 8:28 AM, Moises Silva <moises.silva at gmail.com> wrote:
> On Wed, May 11, 2011 at 7:18 PM, guru singh <grsingh750 at gmail.com> wrote:
>>
>> Hi,
>>
>> I am trying to setup an arch box for FS. I cant install wanpipe.
>> make freetdm fails with this error http://pastebin.freeswitch.org/16273
>> uname -r 2.6.38-ARCH
>> gcc version 4.6.0 20110429 (prerelease) (GCC)
>>
>> Could with a working arch box suggest a solution to this
>>
>
> This is the kernel API that changed yet again. You'll have to wait until a
> Wanpipe driver update is released.
> 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
>
>


From eagle.antonio at gmail.com  Mon May 16 12:08:55 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 16 May 2011 08:08:55 +0000
Subject: [Freeswitch-users] ESL Command Not Found
In-Reply-To: <BANLkTi=-Ftkf42wv-aMogFhWM-=-kOW4sg@mail.gmail.com>
References: <BANLkTi=q0-sUZyxb0AsJsyhonjE8Z-tuoQ@mail.gmail.com>
	<BANLkTi=O8SGiY=4Y_eUX3pH6KaMo0L3dOA@mail.gmail.com>
	<BANLkTi=-Ftkf42wv-aMogFhWM-=-kOW4sg@mail.gmail.com>
Message-ID: <BANLkTimE_dJo3-8HmkS9EMw9xObnDGX0GQ@mail.gmail.com>

Here It Goes a Simple Example
http://pastebin.freeswitch.org/16300

Play & Get Digits Is unable to get var "bacalhau" on the B Leg.
The exact same code works on the A Leg of the Call , it appears that on the
B leg maybe socket full is ignore by parser.

I Will test if i transfer the call into a FS xml context , and then into a
socket conn .
Let's see if this will make the channel commands work properly.

2011/5/13 curriegrad2004 <curriegrad2004 at gmail.com>

> Ah yes, the dreaded cup of coffee is needed...
>
> On Fri, May 13, 2011 at 10:04 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > haha
> > uuid_getvar <uuid> <var_name>
> > -MC
> >
> > On Fri, May 13, 2011 at 10:01 AM, Antonio Teixeira <
> eagle.antonio at gmail.com>
> > wrote:
> >>
> >> Hello
> >>
> >> I'm using ESL to Fork a call inside another , this is the commom example
> >> of User (A) pressing 1 and calling another party (B).
> >>
> >> I'm forking the call by opening an inbound socket and issuing the
> command
> >>
> >>
> >>
> api('originate',"{step=1,origination_caller_id_number=XXXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s}sofia/external/%s at 2XXXXXXX
> >> &socket(192.168.0.12:8040 sync full)
> >>
> >>
> >> All works great ( i get the call) and as you can see the call to B is
> >> redirected to the IVR Server
> >>
> >> The problem is if i do on the channel B ( By esl)
> >> uuid_getvar step XXXXXXXXXXXXXXXXXXXXXXXXXX
> >>
> >> I get
> >> Event-Name: SOCKET_DATA
> >> Content-Type: command/reply
> >> Reply-Text: -ERR%20command%20not%20found
> >>
> >> Has anyone faced this ??
> >> BTW using the python ESL:
> >>
> >> Best Regards
> >> Antonio Teixeira
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 16 12:14:56 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 16 May 2011 08:14:56 +0000
Subject: [Freeswitch-users] ESL Command Not Found
In-Reply-To: <BANLkTimE_dJo3-8HmkS9EMw9xObnDGX0GQ@mail.gmail.com>
References: <BANLkTi=q0-sUZyxb0AsJsyhonjE8Z-tuoQ@mail.gmail.com>
	<BANLkTi=O8SGiY=4Y_eUX3pH6KaMo0L3dOA@mail.gmail.com>
	<BANLkTi=-Ftkf42wv-aMogFhWM-=-kOW4sg@mail.gmail.com>
	<BANLkTimE_dJo3-8HmkS9EMw9xObnDGX0GQ@mail.gmail.com>
Message-ID: <BANLkTi=RRXjTAN4YCDP9jmnP92Nar9qkEQ@mail.gmail.com>

Ok I Have tried , if i transfer the call into a dialplan context and from
that into a socket it works as expect if I do originate &socket ....)
The channel will be unable to run some commands.

Tell me if you need something else :D

Regards
A/T


2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>

> Here It Goes a Simple Example
> http://pastebin.freeswitch.org/16300
>
> Play & Get Digits Is unable to get var "bacalhau" on the B Leg.
> The exact same code works on the A Leg of the Call , it appears that on the
> B leg maybe socket full is ignore by parser.
>
> I Will test if i transfer the call into a FS xml context , and then into a
> socket conn .
> Let's see if this will make the channel commands work properly.
>
> 2011/5/13 curriegrad2004 <curriegrad2004 at gmail.com>
>
>> Ah yes, the dreaded cup of coffee is needed...
>>
>> On Fri, May 13, 2011 at 10:04 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > haha
>> > uuid_getvar <uuid> <var_name>
>> > -MC
>> >
>> > On Fri, May 13, 2011 at 10:01 AM, Antonio Teixeira <
>> eagle.antonio at gmail.com>
>> > wrote:
>> >>
>> >> Hello
>> >>
>> >> I'm using ESL to Fork a call inside another , this is the commom
>> example
>> >> of User (A) pressing 1 and calling another party (B).
>> >>
>> >> I'm forking the call by opening an inbound socket and issuing the
>> command
>> >>
>> >>
>> >>
>> api('originate',"{step=1,origination_caller_id_number=XXXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s}sofia/external/%s at 2XXXXXXX
>> >> &socket(192.168.0.12:8040 sync full)
>> >>
>> >>
>> >> All works great ( i get the call) and as you can see the call to B is
>> >> redirected to the IVR Server
>> >>
>> >> The problem is if i do on the channel B ( By esl)
>> >> uuid_getvar step XXXXXXXXXXXXXXXXXXXXXXXXXX
>> >>
>> >> I get
>> >> Event-Name: SOCKET_DATA
>> >> Content-Type: command/reply
>> >> Reply-Text: -ERR%20command%20not%20found
>> >>
>> >> Has anyone faced this ??
>> >> BTW using the python ESL:
>> >>
>> >> Best Regards
>> >> Antonio Teixeira
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From math.parent at gmail.com  Mon May 16 12:21:26 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Mon, 16 May 2011 10:21:26 +0200
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
Message-ID: <BANLkTimGPxuvrCNhTvNyMPf-i4PhRh=7uQ@mail.gmail.com>

2011/5/16 Schenk, Oliver <OSchenk at wnr.com.au>:
> Hey,

Hello,

>
> I was wondering if anyone out there in the FreeSwitch community has
> successfully used FXO cards (e.g. VIC2-4FXO) in a CISCO router (e.g. 2811)
> with FreeSwitch.
>
>
>
> Was this achieved via SIP communication between the CISCO and FreeSwitch?

Are you using those with CUCM currently? If yes, what is the used
protocol? I f this is SCCP, I'm interrested by a network capture.


>
> Any documentation out there? I can?t seem to find much for FreeSwitch
> regarding what would need to be done on the FreeSwitch end.

I don't know.

Regards
-- 
Mathieu

>
> NOTICE - This e-mail and any files transmitted with it are confidential and
> are only for the use of the person to whom they are addressed.
> If you are not the intended recipient then you have received this e-mail in
> error; please advise us immediately if this is the case.
> Any views expressed in this message are those of the individual sender,
> except where the sender specifically states them to be the views of WestNet
> Rail.

Really?


From oseslija at gmail.com  Mon May 16 12:46:03 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 16 May 2011 10:46:03 +0200
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
Message-ID: <BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>

I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.


On Mon, May 16, 2011 at 7:08 AM, Schenk, Oliver <OSchenk at wnr.com.au> wrote:

>  Hey,
>
>
>
> I was wondering if anyone out there in the FreeSwitch community has
> successfully used FXO cards (e.g. VIC2-4FXO) in a CISCO router (e.g. 2811)
> with FreeSwitch.
>
>
>
> Was this achieved via SIP communication between the CISCO and FreeSwitch?
>
>
>
> Any documentation out there? I can?t seem to find much for FreeSwitch
> regarding what would need to be done on the FreeSwitch end.
>
>
>
>
>
>
>
> Regards,
>
>
>
> *Oliver Schenk*
>
>
>
> NOTICE - This e-mail and any files transmitted with it are confidential and
> are only for the use of the person to whom they are addressed.
> If you are not the intended recipient then you have received this e-mail in
> error; please advise us immediately if this is the case.
> Any views expressed in this message are those of the individual sender,
> except where the sender specifically states them to be the views of WestNet
> Rail.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From math.parent at gmail.com  Mon May 16 12:57:55 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Mon, 16 May 2011 10:57:55 +0200
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
Message-ID: <BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>

2011/5/16 Ognjen Seslija <oseslija at gmail.com>:
> I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.
>

No ;-) I want to know the protocol used between CCME and the Cisco
2811: MGCP, H323, SCCP, SIP?


-- 
Mathieu


From frankie.k.yiu at gmail.com  Mon May 16 13:54:58 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 16 May 2011 02:54:58 -0700
Subject: [Freeswitch-users] how to fix "STUN Failed!
	stun.freeswitch.org:3478 [Bind Error!]" ?
Message-ID: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>

Hi there,

I am doing a stress test of sending 30 calls to another freeSWITCH box that
is in different network/location.  And I keep getting 1 call with the
following error:

2011-05-16 02:29:14.894260 [ERR] sofia_glue.c:912 STUN Failed!
stun.freeswitch.o
rg:3478 [Bind Error!]
2011-05-16 02:29:14.894260 [NOTICE] mod_sofia.c:670 Hangup sofia/external/<a
phone number>@<some <2@<some> IP address> [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]


Could someone please let me know what exactly is wrong and how to fix this?


Thanks,

Frankie
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From eagle.antonio at gmail.com  Mon May 16 14:16:06 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 16 May 2011 10:16:06 +0000
Subject: [Freeswitch-users] ESL Bridge
Message-ID: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>

Good Morning.

Here i continue my fight against ESL :)
On Leg A I'm forking a call that will become Leg B after a short IVR.

Anyway
Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers , Presses
1 and is merged by uuid_bridge
Then...
Both calls are disconnected with Executed Last Dialplan Command .

I have tried everything from freezing the IVR with sleep(in python and using
the freeswitch sleep).

But i'm unable to merge both calls.

As anyone tried something similar.

Regards
A/T
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From tayeb.meftah at gmail.com  Mon May 16 16:29:08 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 16 May 2011 14:29:08 +0200
Subject: [Freeswitch-users] how to fix "STUN Failed!
 stun.freeswitch.org:3478 [Bind Error!]" ?
In-Reply-To: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
References: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
Message-ID: <4DD11894.4040903@gmail.com>

stun.freeswitch is not a 100% uptime stun server
use another stun server like stun.voipuser.org, stun.counterpath.com or 
look for another in voipinfo.org
thank you
On 16/05/2011 11:54, Frankie Yiu wrote:
> Hi there,
> I am doing a stress test of sending 30 calls to another freeSWITCH box 
> that is in different network/location.  And I keep getting 1 call with 
> the following error:
> 2011-05-16 02:29:14.894260 [ERR] sofia_glue.c:912 STUN Failed! 
> stun.freeswitch.o
> rg:3478 [Bind Error!]
> 2011-05-16 02:29:14.894260 [NOTICE] mod_sofia.c:670 Hangup 
> sofia/external/<a phone number>@<some <mailto:2@%3Csome> IP 
> address> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
> Could someone please let me know what exactly is wrong and how to fix 
> this?
> Thanks,
>
> Frankie
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883

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From tayeb.meftah at gmail.com  Mon May 16 16:32:06 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 16 May 2011 14:32:06 +0200
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
	<BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>
Message-ID: <4DD11946.8080100@gmail.com>

Mathieu,
CCME is runing allready inside C28XX
CCME will talk to freeswitch through Sip or H.323.
SCCP i'm not sure, but sip is sure.
thank you
On 16/05/2011 10:57, Mathieu Parent wrote:
> 2011/5/16 Ognjen Seslija<oseslija at gmail.com>:
>    
>> I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.
>>
>>      
> No ;-) I want to know the protocol used between CCME and the Cisco
> 2811: MGCP, H323, SCCP, SIP?
>
>
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883



From admin at blindi.net  Mon May 16 16:57:34 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 16 May 2011 14:57:34 +0200 (CEST)
Subject: [Freeswitch-users] how to fix "STUN Failed!
 stun.freeswitch.org:3478 [Bind Error!]" ?
In-Reply-To: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
References: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105161451300.2152@debian>

Hi,
  It does not matter what stunserver you use.
I use any server in germany.
For example: stun.sipgate.de or stun.1und1.de
I hope this helps you.

---------------
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 anton.vazir at gmail.com  Mon May 16 17:48:43 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 16 May 2011 18:48:43 +0500
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
Message-ID: <BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>

I do it with originate &park - all works, i bridge them by uuid_bridge
successfully

2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
> Good Morning.
>
> Here i continue my fight against ESL :)
> On Leg A I'm forking a call that will become Leg B after a short IVR.
>
> Anyway
> Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers , Presses
> 1 and is merged by uuid_bridge
> Then...
> Both calls are disconnected with Executed Last Dialplan Command .
>
> I have tried everything from freezing the IVR with sleep(in python and using
> the freeswitch sleep).
>
> But i'm unable to merge both calls.
>
> As anyone tried something similar.
>
> Regards
> A/T
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anton.vazir at gmail.com  Mon May 16 17:51:38 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 16 May 2011 18:51:38 +0500
Subject: [Freeswitch-users] wanpipe fails to compile on arch64
In-Reply-To: <BANLkTi=KPCSwQKYe=OMs0X9b7ATz5YmEJA@mail.gmail.com>
References: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>
	<BANLkTik9MY+6S5341+0rug1jD_Ouv3eZtw@mail.gmail.com>
	<BANLkTi=KPCSwQKYe=OMs0X9b7ATz5YmEJA@mail.gmail.com>
Message-ID: <BANLkTimBeyP6cYBQN7rL-ppkDfJ5=MBEDA@mail.gmail.com>

Sangoma drivers are little late with newer kernel support. THere are
some -dev drivers on their FTP, which usually gets newer kernel
support first

2011/5/16 guru singh <grsingh750 at gmail.com>:
> Hi Moises
>
> Thanks for your reply
> I tried it on lts-kernel and it works.
>
> regards
> guru
>
> On Mon, May 16, 2011 at 8:28 AM, Moises Silva <moises.silva at gmail.com> wrote:
>> On Wed, May 11, 2011 at 7:18 PM, guru singh <grsingh750 at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>> I am trying to setup an arch box for FS. I cant install wanpipe.
>>> make freetdm fails with this error http://pastebin.freeswitch.org/16273
>>> uname -r 2.6.38-ARCH
>>> gcc version 4.6.0 20110429 (prerelease) (GCC)
>>>
>>> Could with a working arch box suggest a solution to this
>>>
>>
>> This is the kernel API that changed yet again. You'll have to wait until a
>> Wanpipe driver update is released.
>> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon May 16 18:16:37 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 16 May 2011 14:16:37 +0000
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
Message-ID: <BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>

Hello Anton.

The problem is the following  :

I receive an Inbound Call Leg A.
I say something using TTS and place it on Park , grab the channel uuid.

Fork another call pass the uuid as argument.

As soon as Leg B picks up , TTS fires UP asks the user a digit and after we
have got the digit .

I send uuid_bridge mycurrentchanel + argument uuid from channel A

And i got +OK great :D

The problem , Park is now broken and the call is terminated with last
dialplan extension executed or simply the call is terminated HANGUP.

If i do some time.sleep(40) the calls don't end but they aren't merge i
can't see no RTP flowing trough.

Hope you guys can help me out.

Regards
A/T



2011/5/16 Anton VG <anton.vazir at gmail.com>

> I do it with originate &park - all works, i bridge them by uuid_bridge
> successfully
>
> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
> > Good Morning.
> >
> > Here i continue my fight against ESL :)
> > On Leg A I'm forking a call that will become Leg B after a short IVR.
> >
> > Anyway
> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
> Presses
> > 1 and is merged by uuid_bridge
> > Then...
> > Both calls are disconnected with Executed Last Dialplan Command .
> >
> > I have tried everything from freezing the IVR with sleep(in python and
> using
> > the freeswitch sleep).
> >
> > But i'm unable to merge both calls.
> >
> > As anyone tried something similar.
> >
> > Regards
> > A/T
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kheimerl at cs.berkeley.edu  Sun May 15 02:47:32 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sat, 14 May 2011 15:47:32 -0700
Subject: [Freeswitch-users] Project Announcement and Request for Direction
Message-ID: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>

Hello Freeswitch Users!

My name is Kurtis Heimerl, and I'm a graduate researcher at the
University of California, Berkeley in the Technology and
Infrastructure for Emerging Regions (TIER) group under Eric Brewer.
We're currently investigating the use of OpenBTS
(http://openbts.sourceforge.net/) for providing cellular coverage via
low-cost base stations in low-density parts of the world. This project
is called The Village Base Station. In support of that project, we're
also investigating the use of Freeswitch to support OpenBTS and
provide a flexible, extensible, and simple platform for deploying
voice/sms/data applications on the basestation itself. Towards this
end, I'm asking you, the freeswitch community, for advice and
direction on some of our research goals.

The basic story is simple. OpenBTS uses a software radio and generic
PC to provide cellular service. As a byproduct of this architecture,
we also gain the ability to run freeswitch applications "locally"
(concurrently with OpenBTS), and take advantage of the benefits this
tight coupling provides us. These benefits are numerous; calls/SMS
between BTS users can be much cheaper, as they do not use any backhaul
bandwidth. Applications can query the system for more information
about users, such as location or status. As an example, unlike
traditional GSM telephony applications, we are able to query if users
are currently available on the network. This could be used to create
voice "chat lists", which tell participants which of their friends are
currently within cellular range.

We foresee 6 features required to support these "local" freeswitch
applications on our OpenBTS system. I'm very curious how the
freeswitch community feels about these possible additions, as well how
they might be implemented.

There are a few that are already available in freeswitch, but may be
rougher than we would like. These include:

1) Identity: The ability to query for user's status, numbers, etc.
This seems simple enough in the existing system. However, we'd like to
provide hooks for applications to act on these sign-ons or offs. For
instance, an app may hold messages until a phone logs onto the system
and push them then. My understanding is that this should be simple,
probably hooking onto "SIP presence" events?
2) Storage: Freeswitch currently seems to support only per-application
storage, with limited support for cross-application storage (mostly
user directories). This is occasionally problematic: one issue we've
heard is that it is difficult to place messages into voice mailboxes
from other apps. We'd like a more unified storage framework. Or...
3) "Pipes": This is the ability to pass messages between freeswitch
apps. This seems pretty well supported though simple dialplan
interactions, though the modules themselves may not provide enough
functionality. Is there a way to do this inside of apps? I consider
this an alternative to the storage framework discussed in #2

Lastly, there are three functions we don't believe are well-supported
in freeswitch. These are...
1) Privacy: We expect our BTS to be used in politically sensitive
areas. Given this, freeswitch could provide an anonymity layer,
providing short term phone/SMS numbers, or directing communications
through more secure layers (e.g., Tor).
2) Asynchrony: While freeswitch seems to support basic asynchrony
though its event system, I couldn't find any way to delay events for
indeterminate times. For instance, we may want to schedule a traffic
warning for 1PM every Wednesday to every phone currently on the
system. Is there a way to do that currently?
3) SMS: Freeswitch seems to currently support SIP chat messages (using
SIMPLE?). We need to either extend OpenBTS to speak SIMPLE, or extend
freeswitch to speak OpenBTS's SMS protocol. Neither seems particularly
difficult. This will allow our apps to send and receive both voice and
SMS messages from users.

We believe these core functions will enable a wide variety of BTS
applications. I have a laundry list of those, but I'll omit them for
sake of space.

If any members of the community (that means you!) have any directions,
ideas, projects, or thoughts, please pass them on! We're just
beginning this part of the project, and getting the lay of the land.
Feedback is critical at this point.

Thanks!


From sireeps at gmail.com  Sun May 15 23:43:19 2011
From: sireeps at gmail.com (Kamen)
Date: Sun, 15 May 2011 12:43:19 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <1305407454535-6363912.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
Message-ID: <1305488599172-6366400.post@n2.nabble.com>

I spent a few more hours and I figured a few more things.  It looks like
there was a lot of excessive configuration for the purpose.  I think what
should work (although it still does not) is as follows. No need for any java
scripts in this new version.

I set the public.xml extension:

 <extension name="extname">
  	<condition field="caller_id_number" expression="^(9051212121)$">
 		<action application="set" data="domain_name=$${domain}"/>
 		<action application="transfer" data="3472 XML default"/>
 	</condition>
 </extension>

And the default.xml goes like this:

    <extension name="extname">
        <condition field="destination_number" expression="3472">
	<action application="set" data="ringback=$${us-ring}"/>		
	<action application="set" data="transfer_ringback=$${us-ring}"/>		
	<action application="set" data="hangup_after_bridge=true"/>		
	<action application="bridge"
data="sofia/gateway/mysipprovider/15191212121"/>
        </condition>
    </extension>

So this is way simpler, although the result is the same - the channel is
opened (2011-05-15 5:33:45.687500 [NOTICE] switch_channel.c:812 New Channel
sofia/external/15191212121 [430cf1e6-0876-4cc7-ba87-9c962619b341]), but I
still get paused for 10 seconds.  After which I get short busy signals.  

What I noticed, and that seems to be a clue, if I do not hang up on calling
side and continue with the busy signal, the connection gets through to the
destination number and it shows the caller ID and rings shortly.  The way I
see it  it is a break through!  I never got any rings before on the
destination side.

Anyway, so the question I have now, why the heck it seems not calling on
destination right away as soon as the channel is opened but pauses until
disconnected? And then actually still calls, but only a short ring, enough
to display a caller ID.

If anyone had similar problem solved, please, let me know. I would really
appreciate it.

Regards,

Kamen


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From sireeps at gmail.com  Mon May 16 03:16:31 2011
From: sireeps at gmail.com (Kamen)
Date: Sun, 15 May 2011 16:16:31 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <1305407454535-6363912.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
Message-ID: <1305501391976-6366874.post@n2.nabble.com>

Hi there!

I still have that problem with pausing for 10 seconds when transferring an
outside incoming call back to another external number.  I would really
really appreciate anyone with FS knowledge having a peek at it.  I think I
am giving enough information here.  But if I am not, I am sure willing to
provide more. However, my gutt feeling tells me the solution is easy and
this is something silly I missed while configuring my FS.

SO, here it goes, more info on the problem.  I enabled debug info and here
is what I see:

2011-05-15 18:14:12.187500 [DEBUG] switch_core_state_machine.c:157
sofia/external/9051212121 at 218.165.240.142 Standard EXECUTE EXECUTE
sofia/external/9051212121 at 218.165.240.142 log(INFO dialing destination
number)
2011-05-15 18:14:12.203125 [INFO] mod_dptools.c:1183 dialing destination
number EXECUTE sofia/external/9051212121 at 218.165.240.142
bridge(sofia/gateway/mysipprovider/15191212121)
2011-05-15 18:14:12.203125 [NOTICE] switch_channel.c:812 New Channel
sofia/external/15191212121 [55ef3edb-82b7-44dd-b961-3f6c33a5bd5e]

So, new channel is created.  And it looks like it is dialing. Then it goes

2011-05-15 18:14:12.203125 [DEBUG] mod_sofia.c:4286
(sofia/external/15191212121) State Change CS_NEW -> CS_INIT
2011-05-15 18:14:12.203125 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-15 18:14:12.203125 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_INIT
2011-05-15 18:14:12.203125 [DEBUG] switch_core_state_machine.c:356
(sofia/external/15191212121) State INIT
2011-05-15 18:14:12.203125 [DEBUG] mod_sofia.c:84 sofia/external/15191212121
SOFIA INIT

An that was the last statement at the time of creating a channel.  It pauses
for 5 seconds and gives me:


2011-05-15 18:14:17.203125 [DEBUG] switch_nat.c:502 mapped public port 31306
protocol UDP to localport 31306


Well, ok, not sure why it needed 5 seconds for that.  And then it pauses for
another 5 seconds and eventually goes to the busy signal with the following
debug log.

2011-05-15 18:14:22.203125 [DEBUG] switch_nat.c:502 mapped public port 31307
protocol UDP to localport 31307
2011-05-15 18:14:22.203125 [DEBUG] mod_sofia.c:124
(sofia/external/15191212121)State Change CS_INIT -> CS_ROUTING
2011-05-15 18:14:22.203125 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:356
(sofia/external/15191212121) State INIT going to sleep
2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_ROUTING
2011-05-15 18:14:22.203125 [DEBUG] switch_channel.c:1668
(sofia/external/15191212121) Callstate Change DOWN -> RINGING

hm... interesting.  Why does it go to RINGING after 10 seconds pause?  Thats
the main question.

2011-05-15 18:14:22.203125 [DEBUG] sofia.c:4744 Channel
sofia/external/15191212121 entering state [calling][0]
2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:359
(sofia/external/15191212121) State ROUTING
2011-05-15 18:14:22.203125 [DEBUG] mod_sofia.c:147
sofia/external/15191212121 SOFIA ROUTING
2011-05-15 18:14:22.203125 [DEBUG] switch_ivr_originate.c:66
(sofia/external/15191212121) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-05-15 18:14:22.203125 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:359
(sofia/external/15191212121) State ROUTING going to sleep
2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_CONSUME_MEDIA
2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:378
(sofia/external/15191212121) State CONSUME_MEDIA
2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:378
(sofia/external/15191212121) State CONSUME_MEDIA going to sleep

So, this is briefly the debug trace.  I really hope anybody might tell me
what is going on in here.  Why sofia goes into that pause mode and tries to
ring after 10 seconds, after actually it is disconnected.  Most likely this
has to do with my config.  But I am being a total novice to FS have no idea
what possibly can go wrong.  

Thanks in advance.

Kamen

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View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6366874.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Mon May 16 17:09:29 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 16 May 2011 06:09:29 -0700 (PDT)
Subject: [Freeswitch-users] how to fix "STUN Failed!
 stun.freeswitch.org:3478 [Bind Error!]" ?
In-Reply-To: <4DD11894.4040903@gmail.com>
References: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
	<4DD11894.4040903@gmail.com>
Message-ID: <1305551369509-6368571.post@n2.nabble.com>


Meftah Tayeb wrote:
> stun.freeswitch is not a 100% uptime stun server
Perhaps, it is time to add a support for multiple STUN servers to FS. This
way, if the default STUn server fails, FS will try the other STUN servers.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-fix-STUN-Failed-stun-freeswitch-org-3478-Bind-Error-tp6368105p6368571.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anton.vazir at gmail.com  Mon May 16 19:01:30 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 16 May 2011 20:01:30 +0500
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
Message-ID: <BANLkTiniKZtLkfoVtou3S2g6hTBbEJbxBw@mail.gmail.com>

I do exactly the same, except a single step - I do not ask user to
press "1" for uuid_bridge, and bridge the channels on EARLY_MEDIA -
works for me.
But I also do monitor DTMF before and while in bridge for "*" to disconnect legs

2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
> Hello Anton.
>
> The problem is the following? :
>
> I receive an Inbound Call Leg A.
> I say something using TTS and place it on Park , grab the channel uuid.
>
> Fork another call pass the uuid as argument.
>
> As soon as Leg B picks up , TTS fires UP asks the user a digit and after we
> have got the digit .
>
> I send uuid_bridge mycurrentchanel + argument uuid from channel A
>
> And i got +OK great :D
>
> The problem , Park is now broken and the call is terminated with last
> dialplan extension executed or simply the call is terminated HANGUP.
>
> If i do some time.sleep(40) the calls don't end but they aren't merge i
> can't see no RTP flowing trough.
>
> Hope you guys can help me out.
>
> Regards
> A/T
>
>
>
> 2011/5/16 Anton VG <anton.vazir at gmail.com>
>>
>> I do it with originate &park - all works, i bridge them by uuid_bridge
>> successfully
>>
>> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
>> > Good Morning.
>> >
>> > Here i continue my fight against ESL :)
>> > On Leg A I'm forking a call that will become Leg B after a short IVR.
>> >
>> > Anyway
>> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
>> > Presses
>> > 1 and is merged by uuid_bridge
>> > Then...
>> > Both calls are disconnected with Executed Last Dialplan Command .
>> >
>> > I have tried everything from freezing the IVR with sleep(in python and
>> > using
>> > the freeswitch sleep).
>> >
>> > But i'm unable to merge both calls.
>> >
>> > As anyone tried something similar.
>> >
>> > Regards
>> > A/T
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anton.vazir at gmail.com  Mon May 16 19:03:47 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 16 May 2011 20:03:47 +0500
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTiniKZtLkfoVtou3S2g6hTBbEJbxBw@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<BANLkTiniKZtLkfoVtou3S2g6hTBbEJbxBw@mail.gmail.com>
Message-ID: <BANLkTik8D7rjTnstvfDC3tFjgGxFwrL81w@mail.gmail.com>

BTW - i do not invoke sleep(x) either,
tts i use - cepstral (I found it buggy though, not always works as expected.)

2011/5/16 Anton VG <anton.vazir at gmail.com>:
> I do exactly the same, except a single step - I do not ask user to
> press "1" for uuid_bridge, and bridge the channels on EARLY_MEDIA -
> works for me.
> But I also do monitor DTMF before and while in bridge for "*" to disconnect legs
>
> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
>> Hello Anton.
>>
>> The problem is the following? :
>>
>> I receive an Inbound Call Leg A.
>> I say something using TTS and place it on Park , grab the channel uuid.
>>
>> Fork another call pass the uuid as argument.
>>
>> As soon as Leg B picks up , TTS fires UP asks the user a digit and after we
>> have got the digit .
>>
>> I send uuid_bridge mycurrentchanel + argument uuid from channel A
>>
>> And i got +OK great :D
>>
>> The problem , Park is now broken and the call is terminated with last
>> dialplan extension executed or simply the call is terminated HANGUP.
>>
>> If i do some time.sleep(40) the calls don't end but they aren't merge i
>> can't see no RTP flowing trough.
>>
>> Hope you guys can help me out.
>>
>> Regards
>> A/T
>>
>>
>>
>> 2011/5/16 Anton VG <anton.vazir at gmail.com>
>>>
>>> I do it with originate &park - all works, i bridge them by uuid_bridge
>>> successfully
>>>
>>> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
>>> > Good Morning.
>>> >
>>> > Here i continue my fight against ESL :)
>>> > On Leg A I'm forking a call that will become Leg B after a short IVR.
>>> >
>>> > Anyway
>>> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
>>> > Presses
>>> > 1 and is merged by uuid_bridge
>>> > Then...
>>> > Both calls are disconnected with Executed Last Dialplan Command .
>>> >
>>> > I have tried everything from freezing the IVR with sleep(in python and
>>> > using
>>> > the freeswitch sleep).
>>> >
>>> > But i'm unable to merge both calls.
>>> >
>>> > As anyone tried something similar.
>>> >
>>> > Regards
>>> > A/T
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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  Mon May 16 19:08:20 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 16 May 2011 15:08:20 +0000
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTik8D7rjTnstvfDC3tFjgGxFwrL81w@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<BANLkTiniKZtLkfoVtou3S2g6hTBbEJbxBw@mail.gmail.com>
	<BANLkTik8D7rjTnstvfDC3tFjgGxFwrL81w@mail.gmail.com>
Message-ID: <BANLkTimmc+tOekRtidjHe8BHYFvR5RtqRQ@mail.gmail.com>

The main problem is that i need user confirmation before making the bridge .
:\
For TTS we use Loquendo , a great MRCP Server.

Is like FS is unable to see the calls bridged ?!?



2011/5/16 Anton VG <anton.vazir at gmail.com>

> BTW - i do not invoke sleep(x) either,
> tts i use - cepstral (I found it buggy though, not always works as
> expected.)
>
> 2011/5/16 Anton VG <anton.vazir at gmail.com>:
> > I do exactly the same, except a single step - I do not ask user to
> > press "1" for uuid_bridge, and bridge the channels on EARLY_MEDIA -
> > works for me.
> > But I also do monitor DTMF before and while in bridge for "*" to
> disconnect legs
> >
> > 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
> >> Hello Anton.
> >>
> >> The problem is the following  :
> >>
> >> I receive an Inbound Call Leg A.
> >> I say something using TTS and place it on Park , grab the channel uuid.
> >>
> >> Fork another call pass the uuid as argument.
> >>
> >> As soon as Leg B picks up , TTS fires UP asks the user a digit and after
> we
> >> have got the digit .
> >>
> >> I send uuid_bridge mycurrentchanel + argument uuid from channel A
> >>
> >> And i got +OK great :D
> >>
> >> The problem , Park is now broken and the call is terminated with last
> >> dialplan extension executed or simply the call is terminated HANGUP.
> >>
> >> If i do some time.sleep(40) the calls don't end but they aren't merge i
> >> can't see no RTP flowing trough.
> >>
> >> Hope you guys can help me out.
> >>
> >> Regards
> >> A/T
> >>
> >>
> >>
> >> 2011/5/16 Anton VG <anton.vazir at gmail.com>
> >>>
> >>> I do it with originate &park - all works, i bridge them by uuid_bridge
> >>> successfully
> >>>
> >>> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
> >>> > Good Morning.
> >>> >
> >>> > Here i continue my fight against ESL :)
> >>> > On Leg A I'm forking a call that will become Leg B after a short IVR.
> >>> >
> >>> > Anyway
> >>> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
> >>> > Presses
> >>> > 1 and is merged by uuid_bridge
> >>> > Then...
> >>> > Both calls are disconnected with Executed Last Dialplan Command .
> >>> >
> >>> > I have tried everything from freezing the IVR with sleep(in python
> and
> >>> > using
> >>> > the freeswitch sleep).
> >>> >
> >>> > But i'm unable to merge both calls.
> >>> >
> >>> > As anyone tried something similar.
> >>> >
> >>> > Regards
> >>> > A/T
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Mon May 16 19:13:58 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 16 May 2011 17:13:58 +0200
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F794371B@cooper>

What does the originate string look like?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira
Skickat: den 16 maj 2011 16:17
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] ESL Bridge

Hello Anton.

The problem is the following  :

I receive an Inbound Call Leg A.
I say something using TTS and place it on Park , grab the channel uuid.

Fork another call pass the uuid as argument.

As soon as Leg B picks up , TTS fires UP asks the user a digit and after we have got the digit .

I send uuid_bridge mycurrentchanel + argument uuid from channel A

And i got +OK great :D

The problem , Park is now broken and the call is terminated with last dialplan extension executed or simply the call is terminated HANGUP.

If i do some time.sleep(40) the calls don't end but they aren't merge i can't see no RTP flowing trough.

Hope you guys can help me out.

Regards
A/T


2011/5/16 Anton VG <anton.vazir at gmail.com<mailto:anton.vazir at gmail.com>>
I do it with originate &park - all works, i bridge them by uuid_bridge
successfully

2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>>:
> Good Morning.
>
> Here i continue my fight against ESL :)
> On Leg A I'm forking a call that will become Leg B after a short IVR.
>
> Anyway
> Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers , Presses
> 1 and is merged by uuid_bridge
> Then...
> Both calls are disconnected with Executed Last Dialplan Command .
>
> I have tried everything from freezing the IVR with sleep(in python and using
> the freeswitch sleep).
>
> But i'm unable to merge both calls.
>
> As anyone tried something similar.
>
> Regards
> A/T
>
>
>
> _______________________________________________
> 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<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 eagle.antonio at gmail.com  Mon May 16 19:19:24 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 16 May 2011 15:19:24 +0000
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58F794371B@cooper>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F794371B@cooper>
Message-ID: <BANLkTimhApdsRbrAk9DT8533jOWE0AObtw@mail.gmail.com>

bgapi('originate',"{strategy=108,step=1,origination_caller_id_number=ETERNANUMBER,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s,position=%s}sofia/external/%
s at 213.141.11.107 4998 XML default" % (self.channel_uuid,position,'PHONE
NUMBER'))

4998 translates into this

    <condition field="destination_number" expression="^4998$"
break="on-true">
      <action application="set" data="strategy=108"/>
      <action application="socket" data="192.168.0.12:8040 sync full"/>
      <action application="hangup" />
    </condition>


Using BGAPI ou just API will end up in the same problem .

2011/5/16 Peter Olsson <peter.olsson at visionutveckling.se>

> What does the originate string look like?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Antonio Teixeira
> *Skickat:* den 16 maj 2011 16:17
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] ESL Bridge
>
>
>
> Hello Anton.
>
> The problem is the following  :
>
> I receive an Inbound Call Leg A.
> I say something using TTS and place it on Park , grab the channel uuid.
>
> Fork another call pass the uuid as argument.
>
> As soon as Leg B picks up , TTS fires UP asks the user a digit and after we
> have got the digit .
>
> I send uuid_bridge mycurrentchanel + argument uuid from channel A
>
> And i got +OK great :D
>
> The problem , Park is now broken and the call is terminated with last
> dialplan extension executed or simply the call is terminated HANGUP.
>
> If i do some time.sleep(40) the calls don't end but they aren't merge i
> can't see no RTP flowing trough.
>
> Hope you guys can help me out.
>
> Regards
> A/T
>
>
> 2011/5/16 Anton VG <anton.vazir at gmail.com>
>
> I do it with originate &park - all works, i bridge them by uuid_bridge
> successfully
>
> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
>
> > Good Morning.
> >
> > Here i continue my fight against ESL :)
> > On Leg A I'm forking a call that will become Leg B after a short IVR.
> >
> > Anyway
> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
> Presses
> > 1 and is merged by uuid_bridge
> > Then...
> > Both calls are disconnected with Executed Last Dialplan Command .
> >
> > I have tried everything from freezing the IVR with sleep(in python and
> using
> > the freeswitch sleep).
> >
> > But i'm unable to merge both calls.
> >
> > As anyone tried something similar.
> >
> > Regards
> > A/T
> >
> >
> >
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4dd1331a32761499047613!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Mon May 16 19:38:23 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 16 May 2011 17:38:23 +0200
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTimhApdsRbrAk9DT8533jOWE0AObtw@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F794371B@cooper>
	<BANLkTimhApdsRbrAk9DT8533jOWE0AObtw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F794372D@cooper>

I've done the sime kind of setup, but I'm using only inbound socket, so that make things work a bit different. What happens if you try to use async instead of sync? I see two possible problems.


1.       The call will hangup when the socket is disconnected

2.       If you loop (sleep) inside the socket, the thread will lock up (that's why you get no audio when trying this?).

Personally I like the inbound mode better, it gives me better control, and since I'm never used outbound mode, I'm not 100% sure my conclusions are correct :)

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira
Skickat: den 16 maj 2011 17:19
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] ESL Bridge

bgapi('originate',"{strategy=108,step=1,origination_caller_id_number=ETERNANUMBER,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s,position=%s}sofia/external/%s at 213.141.11.107<mailto:s at 213.141.11.107> 4998 XML default" % (self.channel_uuid,position,'PHONE NUMBER'))

4998 translates into this

    <condition field="destination_number" expression="^4998$" break="on-true">
      <action application="set" data="strategy=108"/>
      <action application="socket" data="192.168.0.12:8040<http://192.168.0.12:8040> sync full"/>
      <action application="hangup" />
    </condition>


Using BGAPI ou just API will end up in the same problem .
2011/5/16 Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
What does the originate string look like?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Antonio Teixeira
Skickat: den 16 maj 2011 16:17
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] ESL Bridge

Hello Anton.

The problem is the following  :

I receive an Inbound Call Leg A.
I say something using TTS and place it on Park , grab the channel uuid.

Fork another call pass the uuid as argument.

As soon as Leg B picks up , TTS fires UP asks the user a digit and after we have got the digit .

I send uuid_bridge mycurrentchanel + argument uuid from channel A

And i got +OK great :D

The problem , Park is now broken and the call is terminated with last dialplan extension executed or simply the call is terminated HANGUP.

If i do some time.sleep(40) the calls don't end but they aren't merge i can't see no RTP flowing trough.

Hope you guys can help me out.

Regards
A/T

2011/5/16 Anton VG <anton.vazir at gmail.com<mailto:anton.vazir at gmail.com>>
I do it with originate &park - all works, i bridge them by uuid_bridge
successfully

2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>>:
> Good Morning.
>
> Here i continue my fight against ESL :)
> On Leg A I'm forking a call that will become Leg B after a short IVR.
>
> Anyway
> Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers , Presses
> 1 and is merged by uuid_bridge
> Then...
> Both calls are disconnected with Executed Last Dialplan Command .
>
> I have tried everything from freezing the IVR with sleep(in python and using
> the freeswitch sleep).
>
> But i'm unable to merge both calls.
>
> As anyone tried something similar.
>
> Regards
> A/T
>
>
>
> _______________________________________________
> 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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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From freeswitch at priv.de  Mon May 16 20:45:08 2011
From: freeswitch at priv.de (=?ISO-8859-1?Q?Markus_M=FCller?=)
Date: Mon, 16 May 2011 18:45:08 +0200
Subject: [Freeswitch-users] BRI-ISDN-Card Alternatives to Sangoma A500 -
 Do you have a working EuroISDN-Setup with a different card? Please comment!
In-Reply-To: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
References: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
Message-ID: <4DD15494.1090701@priv.de>

Hi Christian,

I am using a now working version with trhee HFC Singleport cards, costs 
about 10 Euro per ISDN Channel.

After investigating and resolving some problem (with help of Stefan 
Knoblich, thank you very much!) in freeftdm and libpri, there is now 
just an kernel memory leak in slib memory. Currently I am investigating 
this and hope to be able to fix this within the next weeks.  The leak is 
480 byte per minute, not regarding if there are calls or not. I am just 
rebooting the server every week on sunday night, so currently it works 
as I need.

Regards,
Markus M?ller
> I'm having a hard time finding a working ISDN-Card for FreeSWITCH.
> I've succesfully tested the Sangoma A500 and it would be a great card
> - but the card is just a litte bit too long for the mini-ITX-Hardware
> we want to use. Therefore i'm desperately looking for alternatives:
>
> - I've tested an OpenVox B100P, without success so far(See
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071872.html)
> - Today i've installed an Eicon Diva BRI-2 card but so far i was not
> able to get a working setup, right now i have no clue how to continue
> marrying FreeTDM with the official CAPI-drivers
>
> So if anyone has a working setup with EuroISDN in PTP- and TE-mode and
> a card other than Sangoma A500, please let me know! Or if you know
> there's nothing working BUT Sangoma A500, i'm happy too(Then i have a
> final answer at least).
>
> In case anyone knows how to get a Eicon/Dialogic Diva BRI-2 card to
> work with FreeTDM i'd also be very thankful for a comment. I've
> installed the drivers provided by Dialogic(Diva4Linux 9.50111-98) and
> in an blind attempt tried to configure FreeSWITCH with libpri, so far
> the card has not been recognized.
> Unfortunately i have a limited knowledge about ISDN beyond vanilla
> installation of drivers and getting the drivers/devices/modules to
> show up in Linux(i.e. i don't really know what's going on between ftdm
> and the diverse ISDN-drivers).
>
> Best regards
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From chrisg.lists at gmail.com  Mon May 16 19:14:14 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Mon, 16 May 2011 17:14:14 +0200
Subject: [Freeswitch-users] mod_lcr
Message-ID: <BANLkTikBf7xTR=UEhf-NTwyhOBY0QyXMiQ@mail.gmail.com>

Hi All,
I am having an issue with mod_lcr, when I query from the CLI it
returns the expected values as per:

freeswitch at internal> lcr 0112341234 peak
?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Limit |
Dialstring
? ? ? ?|
?| 011 ? ? ? ? | Telkom ?| 0.32000 ?| ? ? ? | ? ? ? ? ? ?| ? ? ? |
[lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
|
freeswitch at internal> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:1687
data passed to lcr is [0112341234 peak]
2011-05-16 17:06:19.847934 [WARNING] mod_lcr.c:1731 Using default CID
[18005551212]
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1]
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:771 0112341234 doesn't
appear to be a NANP number
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:852 intra routing
[state:0 lata:0] so rate field is [rate]
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:872 we have an event
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:896 SQL: ?SELECT l.digits
AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS
lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(0112341234, 011234123, 01123412, 0112341, 011234, 01123, 0112, 011,
01, 0) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY
digits DESC, ?rate, ?quality DESC, ?reliability DESC, rand();
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
[lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:664 Adding Telkom to head of list
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
[lcr_carrier=Neotel,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:682 Ignoring Duplicate
route for termination point (sofia/gateway/pp-ast-trunk-01/:)
2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
[lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234


When calling from the dialplan, it fails with:

2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1563 intrastate channel
var is [undef]
2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1566 Select routes based
on interstate rates
2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1585 LCR Lookup on $1
using profile peak
2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1]
2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:771 $1 doesn't appear to
be a NANP number
2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:852 intra routing
[state:0 lata:0] so rate field is [rate]
2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:857 we have a session
2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:896 SQL: ?SELECT l.digits
AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS
lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(1) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY
digits DESC, ?rate, ?quality DESC, ?reliability DESC, rand();
EXECUTE sofia/internal/1000 at 192.168.16.136 bridge()
2011-05-16 17:08:33.100261 [NOTICE] switch_core_state_machine.c:189
sofia/internal/1000 at 192.168.16.136 has executed the last dialplan
instruction, hanging up.


The xml entry called:

?? ? ? ?<extension name="Test">
?? ? ? ? ? ? ? ?<condition field="destination_number" expression="^0112341234$">
?? ? ? ? ? ? ? ? ? ? ? ?<action application="lcr" data="$1 peak"/>
?? ? ? ? ? ? ? ? ? ? ? ?<action application="bridge" data="${lcr_auto_route}"/>
?? ? ? ? ? ? ? ?</condition>
?? ? ? ?</extension>


My custom query:

?? ? ? ?<profile name="peak">
?? ? ? ? ? ? ? ?<param name="custom_sql" value="
SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name,
l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
date_start AND date_end ORDER BY digits DESC, ?rate, ?quality DESC,
reliability DESC, rand();
"/>
?? ? ? ?</profile>

Thanks in advance,
Chris


From Nabble at slickdeals.endjunk.com  Mon May 16 20:37:50 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 16 May 2011 09:37:50 -0700 (PDT)
Subject: [Freeswitch-users] wanpipe fails to compile on arch64
In-Reply-To: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>
References: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>
Message-ID: <1305563870080-6369372.post@n2.nabble.com>

You might want to try a git pull to get the latest codes and see if that will
fix the issue.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/wanpipe-fails-to-compile-on-arch64-tp6353964p6369372.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From roger.castaldo at gmail.com  Mon May 16 21:27:45 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 16 May 2011 13:27:45 -0400
Subject: [Freeswitch-users] Project Announcement and Request for
	Direction
In-Reply-To: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>
References: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>
Message-ID: <BANLkTinpJyBahFEBddVJabOGoOeCF9qO3w@mail.gmail.com>

In my humble opinion and experience with writing an application to interface
into freeswitch, it sounds as if a centralized deployment server sending out
xml responses to the module that allows for freeswitch to be configured
remotely ( I believe its called mod_curl or xml_rpc something like that I
cannot remember) would solve a couple of items on the list.  As for the
other items, one suggestion I would like to make is to run something
independent of freeswitch that links into the event socket, this would allow
you to implement something to handle most if not all of the features without
a need to add some things to freeswitch.  The event socket from my
experience has been quite the effective API, as well as it is quite
effective for gathering messages from the system itself.  Something along
those lines would allow you to "pipe" items between multiple servers.

On Sat, May 14, 2011 at 6:47 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>wrote:

> Hello Freeswitch Users!
>
> My name is Kurtis Heimerl, and I'm a graduate researcher at the
> University of California, Berkeley in the Technology and
> Infrastructure for Emerging Regions (TIER) group under Eric Brewer.
> We're currently investigating the use of OpenBTS
> (http://openbts.sourceforge.net/) for providing cellular coverage via
> low-cost base stations in low-density parts of the world. This project
> is called The Village Base Station. In support of that project, we're
> also investigating the use of Freeswitch to support OpenBTS and
> provide a flexible, extensible, and simple platform for deploying
> voice/sms/data applications on the basestation itself. Towards this
> end, I'm asking you, the freeswitch community, for advice and
> direction on some of our research goals.
>
> The basic story is simple. OpenBTS uses a software radio and generic
> PC to provide cellular service. As a byproduct of this architecture,
> we also gain the ability to run freeswitch applications "locally"
> (concurrently with OpenBTS), and take advantage of the benefits this
> tight coupling provides us. These benefits are numerous; calls/SMS
> between BTS users can be much cheaper, as they do not use any backhaul
> bandwidth. Applications can query the system for more information
> about users, such as location or status. As an example, unlike
> traditional GSM telephony applications, we are able to query if users
> are currently available on the network. This could be used to create
> voice "chat lists", which tell participants which of their friends are
> currently within cellular range.
>
> We foresee 6 features required to support these "local" freeswitch
> applications on our OpenBTS system. I'm very curious how the
> freeswitch community feels about these possible additions, as well how
> they might be implemented.
>
> There are a few that are already available in freeswitch, but may be
> rougher than we would like. These include:
>
> 1) Identity: The ability to query for user's status, numbers, etc.
> This seems simple enough in the existing system. However, we'd like to
> provide hooks for applications to act on these sign-ons or offs. For
> instance, an app may hold messages until a phone logs onto the system
> and push them then. My understanding is that this should be simple,
> probably hooking onto "SIP presence" events?
> 2) Storage: Freeswitch currently seems to support only per-application
> storage, with limited support for cross-application storage (mostly
> user directories). This is occasionally problematic: one issue we've
> heard is that it is difficult to place messages into voice mailboxes
> from other apps. We'd like a more unified storage framework. Or...
> 3) "Pipes": This is the ability to pass messages between freeswitch
> apps. This seems pretty well supported though simple dialplan
> interactions, though the modules themselves may not provide enough
> functionality. Is there a way to do this inside of apps? I consider
> this an alternative to the storage framework discussed in #2
>
> Lastly, there are three functions we don't believe are well-supported
> in freeswitch. These are...
> 1) Privacy: We expect our BTS to be used in politically sensitive
> areas. Given this, freeswitch could provide an anonymity layer,
> providing short term phone/SMS numbers, or directing communications
> through more secure layers (e.g., Tor).
> 2) Asynchrony: While freeswitch seems to support basic asynchrony
> though its event system, I couldn't find any way to delay events for
> indeterminate times. For instance, we may want to schedule a traffic
> warning for 1PM every Wednesday to every phone currently on the
> system. Is there a way to do that currently?
> 3) SMS: Freeswitch seems to currently support SIP chat messages (using
> SIMPLE?). We need to either extend OpenBTS to speak SIMPLE, or extend
> freeswitch to speak OpenBTS's SMS protocol. Neither seems particularly
> difficult. This will allow our apps to send and receive both voice and
> SMS messages from users.
>
> We believe these core functions will enable a wide variety of BTS
> applications. I have a laundry list of those, but I'll omit them for
> sake of space.
>
> If any members of the community (that means you!) have any directions,
> ideas, projects, or thoughts, please pass them on! We're just
> beginning this part of the project, and getting the lay of the land.
> Feedback is critical at this point.
>
> 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 frankie.k.yiu at gmail.com  Mon May 16 21:37:20 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 16 May 2011 10:37:20 -0700
Subject: [Freeswitch-users] how to fix "STUN Failed!
 stun.freeswitch.org:3478 [Bind Error!]" ?
In-Reply-To: <4DD11894.4040903@gmail.com>
References: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
	<4DD11894.4040903@gmail.com>
Message-ID: <BANLkTim55CVLQGqUqgJohMtErAwX7HgKcg@mail.gmail.com>

I am not sure how to add multiple stun server.
Currently in my vars.xml, I have these 2 lines:

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>

Do I just add the following 4 lines in the vars.xml?

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.voipuser.org"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.voipuser.org"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.counterpath.com"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.counterpath.com"/>

Thanks,
Frankie


On Mon, May 16, 2011 at 5:29 AM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  stun.freeswitch is not a 100% uptime stun server
> use another stun server like stun.voipuser.org, stun.counterpath.com or
> look for another in voipinfo.org
> thank you
>
> On 16/05/2011 11:54, Frankie Yiu wrote:
>
> Hi there,
>
> I am doing a stress test of sending 30 calls to another freeSWITCH box that
> is in different network/location.  And I keep getting 1 call with the
> following error:
>
> 2011-05-16 02:29:14.894260 [ERR] sofia_glue.c:912 STUN Failed!
> stun.freeswitch.o
> rg:3478 [Bind Error!]
> 2011-05-16 02:29:14.894260 [NOTICE] mod_sofia.c:670 Hangup
> sofia/external/<a phone number>@<some <2@%3Csome> IP address> [CS_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
>
>
> Could someone please let me know what exactly is wrong and how to fix
> this?
>
> Thanks,
>
> Frankie
>
>
> _______________________________________________
> 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
>
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> phone: +13477595883
>
>
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From admin at blindi.net  Mon May 16 21:46:36 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 16 May 2011 19:46:36 +0200 (CEST)
Subject: [Freeswitch-users] Problems multiple condition fields in
	databaseselection
In-Reply-To: <4DD15494.1090701@priv.de>
References: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
	<4DD15494.1090701@priv.de>
Message-ID: <Pine.LNX.4.64.1105161923040.4518@debian>

Hi all,
Is this normal: i can make only a single selection from a database in 
single 
context?
Example:
  <extension name="cforward_only_db" >
<condition field="destination_number" expression="^cforward_only_db$"
break="never">
</condition>
  <condition field="${dummy}" expression="^$" break="never">
</condition>

<condition field="${db(select/thomasfwdstatus/thomas)}"
expression="^1$" break="on-true" >
   <action application="playback"
   data="$${play_prefix}/cforward/alle-callfw-ist-an.wav"/>
<anti-action application="playback"
data="$${play_prefix}/cforward/all-fw-is-off.wav"/>

</condition>
</extension>

When i make a new extension:
<condition field="${db(select/thomasfwdstatus1/thomas)}"
expression="^1$" break="on-true" >
fs fails with an error message: regexp fails.
When i create a new context, and the syntax is the same, fs accept this.
  there is a opportunity in an existing contexty with multipe database 
selections? thank you




---------------
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From kris at kriskinc.com  Mon May 16 22:37:19 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 16 May 2011 14:37:19 -0400
Subject: [Freeswitch-users] how to fix "STUN Failed!
 stun.freeswitch.org:3478 [Bind Error!]" ?
In-Reply-To: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
References: <BANLkTikGg7EPzV1YTxgVfUqvvNNy2yWjSQ@mail.gmail.com>
Message-ID: <BANLkTi=VtKsWbtQqzqqrZ6k+_az+BmMxLg@mail.gmail.com>

The default configuration is NOT designed for stress testing.  It's
designed for maximum usability for standard PBX installs out of the
box.

There's a lot you're going to need to do for any relavant stress
tests.  One of those steps is disabling STUN completely.

On Mon, May 16, 2011 at 5:54 AM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:
> Hi there,
>
> I am doing a stress test of sending 30 calls to another freeSWITCH box that
> is in different network/location.? And I keep getting 1 call with the
> following error:
>
> 2011-05-16 02:29:14.894260 [ERR] sofia_glue.c:912 STUN Failed!
> stun.freeswitch.o
> rg:3478 [Bind Error!]
> 2011-05-16 02:29:14.894260 [NOTICE] mod_sofia.c:670 Hangup sofia/external/<a
> phone number>@<some IP address>?[CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
>
> Could someone please let me know what exactly is wrong and how to fix this?
>
> Thanks,
> Frankie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon May 16 22:41:58 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 16 May 2011 13:41:58 -0500
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
Message-ID: <BANLkTinQ3jSZTSVqpsC8E8i=6WPmFM6s4w@mail.gmail.com>

set the channel variable park_after_bridge=true on all the channels so
they go back to park again when the call ends.


On Mon, May 16, 2011 at 9:16 AM, Antonio Teixeira
<eagle.antonio at gmail.com> wrote:
> Hello Anton.
>
> The problem is the following? :
>
> I receive an Inbound Call Leg A.
> I say something using TTS and place it on Park , grab the channel uuid.
>
> Fork another call pass the uuid as argument.
>
> As soon as Leg B picks up , TTS fires UP asks the user a digit and after we
> have got the digit .
>
> I send uuid_bridge mycurrentchanel + argument uuid from channel A
>
> And i got +OK great :D
>
> The problem , Park is now broken and the call is terminated with last
> dialplan extension executed or simply the call is terminated HANGUP.
>
> If i do some time.sleep(40) the calls don't end but they aren't merge i
> can't see no RTP flowing trough.
>
> Hope you guys can help me out.
>
> Regards
> A/T
>
>
>
> 2011/5/16 Anton VG <anton.vazir at gmail.com>
>>
>> I do it with originate &park - all works, i bridge them by uuid_bridge
>> successfully
>>
>> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
>> > Good Morning.
>> >
>> > Here i continue my fight against ESL :)
>> > On Leg A I'm forking a call that will become Leg B after a short IVR.
>> >
>> > Anyway
>> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
>> > Presses
>> > 1 and is merged by uuid_bridge
>> > Then...
>> > Both calls are disconnected with Executed Last Dialplan Command .
>> >
>> > I have tried everything from freezing the IVR with sleep(in python and
>> > using
>> > the freeswitch sleep).
>> >
>> > But i'm unable to merge both calls.
>> >
>> > As anyone tried something similar.
>> >
>> > Regards
>> > A/T
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 banhi.dutra at gmail.com  Mon May 16 22:52:14 2011
From: banhi.dutra at gmail.com (Banhi Dutra)
Date: Mon, 16 May 2011 20:52:14 +0200
Subject: [Freeswitch-users] Different SAS with ZRTP
Message-ID: <64710029-617A-4648-8BF7-68249B3BD8F0@gmail.com>

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

From leonardo.bidinoto at voicetechnology.com.br  Mon May 16 23:12:24 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Mon, 16 May 2011 16:12:24 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
Message-ID: <BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>

hehe, ok michael.

here is the pastebin link:
http://pastebin.freeswitch.org/16303

2011/5/13 Michael Collins <msc at freeswitch.org>

> Pastebin this info and select "FreeSWITCH Log" as the syntax highlighting.
> I need the colorized output to read logs. (I'm getting older and it's hard
> for me to ready black and white in an email.)
>
> -MC
>
>
> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <
> leonardo.bidinoto at voicetechnology.com.br> wrote:
>
>> Hi Michael,
>>
>> Just succeeded to reproduce the problem.
>>
>> The condition is: when a channel inside a conference is using a ESL
>> connection(lets call it "A") through socket application and another ESL
>> connection(lets call it "B") executes a command with this channel, the "B"
>> ESL connection will wait the "A" ESL connection close to execute its
>> command.
>> If the channel hangs up before the "A" ESL connection is closed, then "B"
>> ESL command will never be executed and the stucked channel will still be
>> there, into sofia and the conference too.
>> To verify that, just do "show channels" and "conference list". with
>> "uuid_exists" command, return "false".
>>
>> Here are the actions done by the channel before get stucked:
>>
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321 [NOTICE]
>> switch_channel.c:816 New Channel sofia/external/1000123402 at 192.168.0.154[16e09413-9cb0-4011-a635-f91933a35c0f]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>> state [received][100]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia.c:4772 Remote SDP:
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia_glue.c:4656 Audio Codec Compare
>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia_glue.c:4656 Audio Codec Compare
>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia_glue.c:4656 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia_glue.c:2788 Set Codec sofia/external/1000123402 at 192.168.0.154PCMU/8000 20 ms 160 samples 64000 bits
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change
>> CS_NEW -> CS_INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_session.c:1114 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>> Running State Change CS_INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
>> State INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change
>> CS_INIT -> CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_session.c:1114 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
>> State INIT going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>> Running State Change CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154) Callstate
>> Change DOWN -> RINGING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>> State ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO]
>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>> public
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true) =~
>> /^true$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~
>> /^true$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 parsing [public->public_extensions]
>> continue=false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
>> destination_number(1234567890) =~ /^(\d*)$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>> switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
>> State Change CS_ROUTING -> CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>> switch_core_session.c:1114 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>> State ROUTING going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>> Running State Change CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
>> State EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 transfer(1234567890 XML default)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State Change
>> CS_EXECUTE -> CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_core_session.c:1114 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_core_session.c:707 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [NOTICE]
>> switch_ivr.c:1603 Transfer sofia/external/1000123402 at 192.168.0.154 to
>> XML[1234567890 at default]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
>> State EXECUTE going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>> Running State Change CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>> State ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO]
>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>> default
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 parsing [default->flex] continue=false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Regex (PASS) [flex]
>> destination_number(1234567890) =~ /^(\d+)$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Action log(INFO VOICE received dest=1234567890)
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Action set(playback_terminators=#)
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall we?)
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Action
>> set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>> 1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>> switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
>> State Change CS_ROUTING -> CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>> switch_core_session.c:1114 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>> State ROUTING going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>> Running State Change CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
>> State EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO]
>> mod_dptools.c:1184 VOICE received dest=1234567890
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 set(playback_terminators=#)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>> [playback_terminators]=[#]
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO]
>> mod_dptools.c:1184 Let's do some ivrd, shall we?
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>> [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 socket(localhost:8084 full)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> answer()
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 answer()
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>> sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154]
>> 192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>> switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>> sofia_glue.c:3284 Set 2833 dtmf send payload to 101
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>> sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>> mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>> switch_core_session.c:707 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>> switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154) Callstate
>> Change RINGING -> ACTIVE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [NOTICE]
>> mod_dptools.c:930 Channel [sofia/external/1000123402 at 192.168.0.154] has
>> been answered
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG]
>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>> state [completed][200]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG]
>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>> state [ready][200]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav
>> flex_digits 5000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 read(1 1
>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>> )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>> [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> read(1 1
>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>> flex_digits 5000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 read(1 1
>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>> flex_digits 5000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF 8:640
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>> [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>> flex_digits 5000 #,*)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 read(1 11
>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>> #,*)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF #:960
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>> [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF #:800
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>> mod_dptools.c:1664 Digit #
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> conference(15646 at teste+flags{waste})
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>> mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1 channel
>> 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>> mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1 channel
>> 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>> codec L16:70
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>> switch_core_session.c:707 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>> mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF *:960
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
>> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
>> previous codec PCMU:0.
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>> flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 read(1 1
>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>> [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink
>> session from object
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>> codec L16:70
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF *:800
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
>> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
>> previous codec PCMU:0.
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>
>> ====================================================================================================================================================
>> While Inside this connection, a "conference 15646 kick [member_id of this
>> channels]" command is executed by a fs_cli console and get stuck while
>> waiting response.
>>
>> ====================================================================================================================================================
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>> flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 read(1 1
>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG]
>> switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG]
>> switch_rtp.c:3280 RTP RECV DTMF 1:960
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>> set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>> 1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>> [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>> switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154) Callstate
>> Change ACTIVE -> HANGUP
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [NOTICE]
>> sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154 [CS_EXECUTE]
>> [NORMAL_CLEARING]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>> switch_channel.c:2576 Send signal sofia/external/1000123402 at 192.168.0.154[KILL]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>> switch_core_session.c:1114 Send signal sofia/external/
>> 1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
>> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink
>> session from object
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
>> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>> codec L16:70
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>> mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG]
>> mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip
>> receive message [UNBRIDGE] (channel is hungup already)
>>
>> I hope this info helps.
>>
>> 2011/5/12 Michael Collins <msc at freeswitch.org>
>>
>>>
>>>
>>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>
>>>> Hi Michael,
>>>>
>>>> Im not using to any cdr module.
>>>
>>>
>>> I would recommend that you do several things:
>>>
>>> #1 - update to latest git
>>> #2 - rotate logs
>>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>>> #4 - reproduce the symptom with a single call (if possible)
>>> #5 - pastebin the log for the uuid in question and link to it in this
>>> thread
>>>
>>> From there hopefully we'll get a clue as to what is happening.
>>> -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
>>>
>>>
>>
>>
>> --
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From msc at freeswitch.org  Mon May 16 23:18:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 12:18:17 -0700
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
	<BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
Message-ID: <BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>

Can you tcpdump or otherwise capture the traffic on port 8085? I am curious
what is happening with that.
-MC

On Mon, May 16, 2011 at 12:12 PM, Leonardo P. Bidinoto <
leonardo.bidinoto at voicetechnology.com.br> wrote:

> hehe, ok michael.
>
> here is the pastebin link:
> http://pastebin.freeswitch.org/16303
>
>
> 2011/5/13 Michael Collins <msc at freeswitch.org>
>
>> Pastebin this info and select "FreeSWITCH Log" as the syntax highlighting.
>> I need the colorized output to read logs. (I'm getting older and it's hard
>> for me to ready black and white in an email.)
>>
>> -MC
>>
>>
>> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <
>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>
>>> Hi Michael,
>>>
>>> Just succeeded to reproduce the problem.
>>>
>>> The condition is: when a channel inside a conference is using a ESL
>>> connection(lets call it "A") through socket application and another ESL
>>> connection(lets call it "B") executes a command with this channel, the "B"
>>> ESL connection will wait the "A" ESL connection close to execute its
>>> command.
>>> If the channel hangs up before the "A" ESL connection is closed, then "B"
>>> ESL command will never be executed and the stucked channel will still be
>>> there, into sofia and the conference too.
>>> To verify that, just do "show channels" and "conference list". with
>>> "uuid_exists" command, return "false".
>>>
>>> Here are the actions done by the channel before get stucked:
>>>
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321 [NOTICE]
>>> switch_channel.c:816 New Channel sofia/external/1000123402 at 192.168.0.154[16e09413-9cb0-4011-a635-f91933a35c0f]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>> state [received][100]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia.c:4772 Remote SDP:
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia_glue.c:4656 Audio Codec Compare
>>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia_glue.c:4656 Audio Codec Compare
>>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia_glue.c:4656 Audio Codec Compare
>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia_glue.c:2788 Set Codec sofia/external/1000123402 at 192.168.0.154PCMU/8000 20 ms 160 samples 64000 bits
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change
>>> CS_NEW -> CS_INIT
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_session.c:1114 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>>> Running State Change CS_INIT
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
>>> State INIT
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change
>>> CS_INIT -> CS_ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_session.c:1114 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154)
>>> State INIT going to sleep
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>>> Running State Change CS_ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154)
>>> Callstate Change DOWN -> RINGING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>>> State ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO]
>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>> public
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true) =~
>>> /^true$/ break=on-false
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~
>>> /^true$/ break=on-false
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 parsing [public->public_extensions]
>>> continue=false
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
>>> destination_number(1234567890) =~ /^(\d*)$/ break=on-false
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>> switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
>>> State Change CS_ROUTING -> CS_EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>> switch_core_session.c:1114 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>>> State ROUTING going to sleep
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>>> Running State Change CS_EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
>>> State EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 transfer(1234567890 XML default)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State Change
>>> CS_EXECUTE -> CS_ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_core_session.c:1114 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_core_session.c:707 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [NOTICE]
>>> switch_ivr.c:1603 Transfer sofia/external/1000123402 at 192.168.0.154 to
>>> XML[1234567890 at default]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
>>> State EXECUTE going to sleep
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>>> Running State Change CS_ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>>> State ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO]
>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>> default
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 parsing [default->flex] continue=false
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Regex (PASS) [flex]
>>> destination_number(1234567890) =~ /^(\d+)$/ break=on-false
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Action log(INFO VOICE received dest=1234567890)
>>>
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Action set(playback_terminators=#)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall we?)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Action
>>> set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>> 1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>> switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154)
>>> State Change CS_ROUTING -> CS_EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>> switch_core_session.c:1114 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>> switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154)
>>> State ROUTING going to sleep
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>> switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154)
>>> Running State Change CS_EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>> switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154)
>>> State EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO]
>>> mod_dptools.c:1184 VOICE received dest=1234567890
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 set(playback_terminators=#)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>> [playback_terminators]=[#]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO]
>>> mod_dptools.c:1184 Let's do some ivrd, shall we?
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>> [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 socket(localhost:8084 full)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> answer()
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 answer()
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>> sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154]
>>> 192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>> switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>> sofia_glue.c:3284 Set 2833 dtmf send payload to 101
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>> sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>> mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>> switch_core_session.c:707 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>> switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154)
>>> Callstate Change RINGING -> ACTIVE
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [NOTICE]
>>> mod_dptools.c:930 Channel [sofia/external/1000123402 at 192.168.0.154] has
>>> been answered
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG]
>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>> state [completed][200]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG]
>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>> state [ready][200]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav
>>> flex_digits 5000 )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 read(1 1
>>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>>> )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>> [flex_digits]=[UNDEF]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> read(1 1
>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>> flex_digits 5000 )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 read(1 1
>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>> flex_digits 5000 )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF 8:640
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>> [flex_digits]=[UNDEF]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>> flex_digits 5000 #,*)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 read(1 11
>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>>> #,*)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF #:960
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>> [flex_digits]=[UNDEF]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF #:800
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>> mod_dptools.c:1664 Digit #
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> conference(15646 at teste+flags{waste})
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>> mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1 channel
>>> 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>> mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1 channel
>>> 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>> codec L16:70
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>>> switch_core_session.c:707 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>>> mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF *:960
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
>>> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
>>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
>>> previous codec PCMU:0.
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>> flex_digits 2000 )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 read(1 1
>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>> [flex_digits]=[UNDEF]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>> codec L16:70
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF *:800
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
>>> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
>>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
>>> previous codec PCMU:0.
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>
>>> ====================================================================================================================================================
>>> While Inside this connection, a "conference 15646 kick [member_id of this
>>> channels]" command is executed by a fs_cli console and get stuck while
>>> waiting response.
>>>
>>> ====================================================================================================================================================
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>> flex_digits 2000 )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 read(1 1
>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG]
>>> switch_ivr_play_say.c:1649 done playing file
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG]
>>> switch_rtp.c:3280 RTP RECV DTMF 1:960
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute
>>> set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>> [flex_digits]=[UNDEF]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>> switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154)
>>> Callstate Change ACTIVE -> HANGUP
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [NOTICE]
>>> sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154 [CS_EXECUTE]
>>> [NORMAL_CLEARING]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>> switch_channel.c:2576 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [KILL]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>> switch_core_session.c:1114 Send signal sofia/external/
>>> 1000123402 at 192.168.0.154 [BREAK]
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
>>> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
>>> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>> codec L16:70
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>> mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG]
>>> mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip
>>> receive message [UNBRIDGE] (channel is hungup already)
>>>
>>> I hope this info helps.
>>>
>>> 2011/5/12 Michael Collins <msc at freeswitch.org>
>>>
>>>>
>>>>
>>>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>
>>>>> Hi Michael,
>>>>>
>>>>> Im not using to any cdr module.
>>>>
>>>>
>>>> I would recommend that you do several things:
>>>>
>>>> #1 - update to latest git
>>>> #2 - rotate logs
>>>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>>>> #4 - reproduce the symptom with a single call (if possible)
>>>> #5 - pastebin the log for the uuid in question and link to it in this
>>>> thread
>>>>
>>>> From there hopefully we'll get a clue as to what is happening.
>>>> -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
>>>>
>>>>
>>>
>>>
>>> --
>>> Leonardo Pires Bidinoto
>>> Voice Technology
>>> www.voicetechnology.com.br
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May 16 23:37:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 12:37:45 -0700
Subject: [Freeswitch-users] Problems multiple condition fields in
	databaseselection
In-Reply-To: <Pine.LNX.4.64.1105161923040.4518@debian>
References: <BANLkTin+GLZdZSNiwUwVyXUHVL3JL0Z_cQ@mail.gmail.com>
	<4DD15494.1090701@priv.de>
	<Pine.LNX.4.64.1105161923040.4518@debian>
Message-ID: <BANLkTimovSMeVk1OXYk7Jo7MOw19oZbP_g@mail.gmail.com>

Thomas,

Could you pastebin the extension that does not work? Also, please pastebin
two call examples: one that works and one that fails. We should be able to
assist from there.

Thanks,
MC

On Mon, May 16, 2011 at 10:46 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi all,
> Is this normal: i can make only a single selection from a database in
> single context?
> Example:
>  <extension name="cforward_only_db" >
> <condition field="destination_number" expression="^cforward_only_db$"
> break="never">
> </condition>
>  <condition field="${dummy}" expression="^$" break="never">
> </condition>
>
> <condition field="${db(select/thomasfwdstatus/thomas)}"
> expression="^1$" break="on-true" >
>  <action application="playback"
>  data="$${play_prefix}/cforward/alle-callfw-ist-an.wav"/>
> <anti-action application="playback"
> data="$${play_prefix}/cforward/all-fw-is-off.wav"/>
>
> </condition>
> </extension>
>
> When i make a new extension:
> <condition field="${db(select/thomasfwdstatus1/thomas)}"
> expression="^1$" break="on-true" >
> fs fails with an error message: regexp fails.
> When i create a new context, and the syntax is the same, fs accept this.
>  there is a opportunity in an existing contexty with multipe database
> selections? thank you
>
>
>
>
> ---------------
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May 16 23:39:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 12:39:46 -0700
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <1305488599172-6366400.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
Message-ID: <BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>

Time to get a full debug log with sip trace and put on pastebin. Be sure to
use "FreeSWITCH Log" as the syntax highlighting.

-MC

On Sun, May 15, 2011 at 12:43 PM, Kamen <sireeps at gmail.com> wrote:

> I spent a few more hours and I figured a few more things.  It looks like
> there was a lot of excessive configuration for the purpose.  I think what
> should work (although it still does not) is as follows. No need for any
> java
> scripts in this new version.
>
> I set the public.xml extension:
>
>  <extension name="extname">
>        <condition field="caller_id_number" expression="^(9051212121)$">
>                <action application="set" data="domain_name=$${domain}"/>
>                <action application="transfer" data="3472 XML default"/>
>        </condition>
>  </extension>
>
> And the default.xml goes like this:
>
>    <extension name="extname">
>        <condition field="destination_number" expression="3472">
>        <action application="set" data="ringback=$${us-ring}"/>
>        <action application="set" data="transfer_ringback=$${us-ring}"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="bridge"
> data="sofia/gateway/mysipprovider/15191212121"/>
>        </condition>
>    </extension>
>
> So this is way simpler, although the result is the same - the channel is
> opened (2011-05-15 5:33:45.687500 [NOTICE] switch_channel.c:812 New Channel
> sofia/external/15191212121 [430cf1e6-0876-4cc7-ba87-9c962619b341]), but I
> still get paused for 10 seconds.  After which I get short busy signals.
>
> What I noticed, and that seems to be a clue, if I do not hang up on calling
> side and continue with the busy signal, the connection gets through to the
> destination number and it shows the caller ID and rings shortly.  The way I
> see it  it is a break through!  I never got any rings before on the
> destination side.
>
> Anyway, so the question I have now, why the heck it seems not calling on
> destination right away as soon as the channel is opened but pauses until
> disconnected? And then actually still calls, but only a short ring, enough
> to display a caller ID.
>
> If anyone had similar problem solved, please, let me know. I would really
> appreciate it.
>
> Regards,
>
> Kamen
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6366400.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 msc at freeswitch.org  Mon May 16 23:43:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 12:43:24 -0700
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <4DD11946.8080100@gmail.com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
	<BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>
	<4DD11946.8080100@gmail.com>
Message-ID: <BANLkTine+R41wM72nNe=-PrN3sP3CzKNcg@mail.gmail.com>

I think what Mathieu is saying is that since he is writing mod_skinny that
it would be extremely valuable for him to see some raw network captures
between the CUCM and the 2811 - assuming, of course, that they are not both
running inside the 2811 as Meftah noted. In other words, in addition to the
OP's issue Mathieu was asking for extra information about how the OP is
currently using the FXO cards.

-MC

On Mon, May 16, 2011 at 5:32 AM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

> Mathieu,
> CCME is runing allready inside C28XX
> CCME will talk to freeswitch through Sip or H.323.
> SCCP i'm not sure, but sip is sure.
> thank you
> On 16/05/2011 10:57, Mathieu Parent wrote:
> > 2011/5/16 Ognjen Seslija<oseslija at gmail.com>:
> >
> >> I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.
> >>
> >>
> > No ;-) I want to know the protocol used between CCME and the Cisco
> > 2811: MGCP, H323, SCCP, SIP?
> >
> >
> >
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> phone: +13477595883
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peder at networkoblivion.com  Mon May 16 23:57:39 2011
From: peder at networkoblivion.com (Peder)
Date: Mon, 16 May 2011 14:57:39 -0500
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <BANLkTine+R41wM72nNe=-PrN3sP3CzKNcg@mail.gmail.com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>	<BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>	<4DD11946.8080100@gmail.com>
	<BANLkTine+R41wM72nNe=-PrN3sP3CzKNcg@mail.gmail.com>
Message-ID: <026801cc1403$82c7c790$885756b0$@com>

FYI, I am about 80% sure that sccp is only for phones or ATAs.  Routers use
MGCP or H.323 to talk to CCM  or CUCM or CCME or whatever they are calling
it today..  He appears to be running CCME on a router, so you won't get any
captures as it will all be internal.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, May 16, 2011 2:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] CISCO FXO cards with FreeSwitch

 

I think what Mathieu is saying is that since he is writing mod_skinny that
it would be extremely valuable for him to see some raw network captures
between the CUCM and the 2811 - assuming, of course, that they are not both
running inside the 2811 as Meftah noted. In other words, in addition to the
OP's issue Mathieu was asking for extra information about how the OP is
currently using the FXO cards.

 

-MC

On Mon, May 16, 2011 at 5:32 AM, Meftah Tayeb <tayeb.meftah at gmail.com>
wrote:

Mathieu,
CCME is runing allready inside C28XX
CCME will talk to freeswitch through Sip or H.323.
SCCP i'm not sure, but sip is sure.
thank you

On 16/05/2011 10:57, Mathieu Parent wrote:
> 2011/5/16 Ognjen Seslija<oseslija at gmail.com>:
>
>> I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.
>>
>>
> No ;-) I want to know the protocol used between CCME and the Cisco
> 2811: MGCP, H323, SCCP, SIP?
>
>
>


--

Meftah Tayeb
inum: +883510001288000
phone: +13477595883



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

 

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From grsingh750 at gmail.com  Mon May 16 23:59:13 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 17 May 2011 01:29:13 +0530
Subject: [Freeswitch-users] wanpipe fails to compile on arch64
In-Reply-To: <1305563870080-6369372.post@n2.nabble.com>
References: <BANLkTikUnvK8gXpx9sAkevd2Bb0H-hhiPA@mail.gmail.com>
	<1305563870080-6369372.post@n2.nabble.com>
Message-ID: <BANLkTiko15Kwa7M4215mMKs533g-4Twcfg@mail.gmail.com>

Hi

Anton, I didn't know of that, I'll have a look. Thanks

mazilo, There was nothing wrong with FS, wanpipe itself didn't compile
on arch. on arch-lts-kernel it worked, and so does FS with mod_freetdm
on latest git.

Regards
guru

On Mon, May 16, 2011 at 10:07 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> You might want to try a git pull to get the latest codes and see if that will
> fix the issue.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/wanpipe-fails-to-compile-on-arch64-tp6353964p6369372.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 mgende at gendesign.com  Tue May 17 00:01:22 2011
From: mgende at gendesign.com (Michael Gende)
Date: Mon, 16 May 2011 15:01:22 -0500
Subject: [Freeswitch-users] fs_cli question
Message-ID: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>

Hello,

Say, I'm setting up a new FS box (no big issues there, using standard CentOS
and latest FS from the site).

Works fine, registering with provider, handset's, sends/receives calls, etc.
One weird thing though:

when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.

Locally, fs_cli works, but only without flags of any kind. Just invoking the
executable without arguments works every time.

I have a few FS's running and want to use the fs_cli on my local computer to
connect when need be. This works fine for all but my latest creation (two
prior ones are a year or more older).

Something foolish I've overlooked? No firewall on the new FS box, routing
and LAN networking look/act fine.

Any commentary welcome, thanks in advance.

Mike G.
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From msc at freeswitch.org  Tue May 17 00:03:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 13:03:28 -0700
Subject: [Freeswitch-users] Error to play multiple file with
	play_and_get_digits
In-Reply-To: <Pine.LNX.4.64.1105150211500.7979@debian>
References: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
	<4DCF140A.6030005@ppmt.org>
	<Pine.LNX.4.64.1105150211500.7979@debian>
Message-ID: <BANLkTimAy-Mk9j662D8DC-37TMCNp4h60A@mail.gmail.com>

Use ! instead of , to separate your file names:

 <action application="play_and_get_digits" data="0 42 1 10000
# file_string:///tmp/1.alaw!/tmp/2.alaw
$${base_dir}/sounds/callback/invalid.wav   NR1 \d+"/>

-MC

On Sat, May 14, 2011 at 5:20 PM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi all, fs, don.t play multiple files.
> my dialplan:
>  <action application="set" data="playback_delimiter=!"/>
>  <action application="set" data="playback_sleep_val=500"/>
>  <action application="play_and_get_digits" data="0 42 1 10000 #
>  file_string:///tmp/1.alaw,/tmp/2.alaw
> $${base_dir}/sounds/callback/invalid.wav   NR1 \d+"/>
> I become the following error on the console:
>
> EXECUTE sofia/internal/1000 at sip2.blindi.net set(playback_delimiter=!)
> 2011-05-15 02:09:03.316798 [DEBUG] mod_dptools.c:1060
> sofia/internal/1000 at sip2.b
> lindi.net SET [playback_delimiter]=[!]
> EXECUTE sofia/internal/1000 at sip2.blindi.net set(playback_sleep_val=500)
> 2011-05-15 02:09:03.316798 [DEBUG] mod_dptools.c:1060
> sofia/internal/1000 at sip2.b
> lindi.net SET [playback_sleep_val]=[500]
> EXECUTE sofia/internal/1000 at sip2.blindi.net play_and_get_digits(0 42 1
> 10000 #
>  file_string:///tmp/1.alaw,/tmp/2.alaw
> /usr/local/freeswitch/sounds/callback/i
> nvalid.alaw NR1 \d+)
> 2011-05-15 02:09:03.316798 [ERR] mod_sndfile.c:194 Error Opening File
> [/tmp/1.alaw,/tmp/2.alaw
> aw,/tmp/2.alaw] [System error : No such file or directory.]
>
> The files are exists.
> What ist the problem 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 17 00:04:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 13:04:54 -0700
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
Message-ID: <BANLkTikjBQ=jC0CooxQzmMfZrW2nknRn+A@mail.gmail.com>

Do you have a way to monitor the remote FS boxes for test purposes? I'd be
interested in knowing if the fs_cli on the local machine is making it all
the way to the remote FS or not.

-MC

On Mon, May 16, 2011 at 1:01 PM, Michael Gende <mgende at gendesign.com> wrote:

> Hello,
>
> Say, I'm setting up a new FS box (no big issues there, using standard
> CentOS and latest FS from the site).
>
> Works fine, registering with provider, handset's, sends/receives calls,
> etc. One weird thing though:
>
> when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
> xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
>
> Locally, fs_cli works, but only without flags of any kind. Just invoking
> the executable without arguments works every time.
>
> I have a few FS's running and want to use the fs_cli on my local computer
> to connect when need be. This works fine for all but my latest creation (two
> prior ones are a year or more older).
>
> Something foolish I've overlooked? No firewall on the new FS box, routing
> and LAN networking look/act fine.
>
> Any commentary welcome, thanks in advance.
>
> Mike G.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 17 00:06:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 13:06:53 -0700
Subject: [Freeswitch-users] Sending Message from background LUA script
 to an unregistered user prevents sofia profile to be restarted
In-Reply-To: <BANLkTikMBSv41i=fmBRVct=RwAjvmeLR1g@mail.gmail.com>
References: <BANLkTikMBSv41i=fmBRVct=RwAjvmeLR1g@mail.gmail.com>
Message-ID: <BANLkTim4pxj9v2=8yoM3uYpPwX6_1by47w@mail.gmail.com>

Definitely open a Jira and include all this information. Also, include a
copy of your Lua script or at least a simple Lua script that can be used for
testing this behavior.

-MC

On Sat, May 14, 2011 at 6:55 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> Hi Guys,
>
> I have a background Lua script which send messages to specified users.I
> have noticed that if the script sends a message to unregistered user then
> I'll not be able to restart sofia profile properly. To clarify the issue, I
> sent a message to unregistered user by a softphone instead of background
> script.In this case, the sofia profile can be restarted properly! In the
> following logs you see first I've tried to send message by a softphone (sofia_presence.c:149)
> after that I've restarted sofia correctly. After the background Lua tried
> to send message (mod_sofia.c:4629), restarting sofia failed.
>
> FreeSWITCH build from last GIT.
> Should I open a ticket on JIRA?
>
> BEST,
> -- afshin
>
>
> freeswitch at opxi2Server>
> freeswitch at opxi2Server> 2011-05-14 09:34:25.471083 [ERR]
> sofia_presence.c:149 Can't find registered user noname2 at fslab
>
> freeswitch at opxi2Server>
> freeswitch at opxi2Server> sofia status
>
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>            sipinterface_1       profile
> sip:mod_sofia at 192.168.128.2:5060      RUNNING (0)
>               voicemail_1         alias
> sipinterface_1      ALIASED
>            sipinterface_3       profile
> sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
>                     fslab         alias
> sipinterface_1      ALIASED
>            sipinterface_2       profile
> sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
>
> =================================================================================================
> 3 profiles 2 aliases
>
> freeswitch at opxi2Server> sofia profile sipinterface_1 restart
>
> Reload XML [Success]
> restarting: sipinterface_1
> 2011-05-14 09:34:59.771082 [INFO] mod_enum.c:765 ENUM Reloaded
> 2011-05-14 09:34:59.771082 [INFO] switch_time.c:1020 Timezone reloaded 530
> definitions
> freeswitch at opxi2Server> 2011-05-14 09:35:00.791085 [NOTICE] sofia.c:1677
> Waiting for worker thread
> 2011-05-14 09:35:00.791085 [NOTICE] sofia.c:2274 Adding Alias [voicemail_1]
> for profile [sipinterface_1]
> 2011-05-14 09:35:00.791085 [NOTICE] sofia.c:2274 Adding Alias [fslab] for
> profile [sipinterface_1]
> 2011-05-14 09:35:00.791085 [NOTICE] sofia.c:4018 Started Profile
> sipinterface_1 [sofia_reg_sipinterface_1]
>
> freeswitch at opxi2Server> sofia status
>
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>            sipinterface_1       profile
> sip:mod_sofia at 192.168.128.2:5060      RUNNING (0)
>               voicemail_1         alias
> sipinterface_1      ALIASED
>            sipinterface_3       profile
> sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
>                     fslab         alias
> sipinterface_1      ALIASED
>            sipinterface_2       profile
> sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
>
> =================================================================================================
> 3 profiles 2 aliases
>
> freeswitch at opxi2Server> 2011-05-14 09:35:39.171084 [ERR] mod_sofia.c:4629
> Can't find registered user noname at fslab
>
> freeswitch at opxi2Server> sofia status
>
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>            sipinterface_1       profile
> sip:mod_sofia at 192.168.128.2:5060      RUNNING (0)
>               voicemail_1         alias
> sipinterface_1      ALIASED
>            sipinterface_3       profile
> sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
>                     fslab         alias
> sipinterface_1      ALIASED
>            sipinterface_2       profile
> sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
>
> =================================================================================================
> 3 profiles 2 aliases
>
> freeswitch at opxi2Server> sofia profile sipinterface_1 restart
>
> Reload XML [Success]
> restarting: sipinterface_1
> 2011-05-14 09:35:55.871085 [INFO] mod_enum.c:765 ENUM Reloaded
> 2011-05-14 09:35:55.871085 [INFO] switch_time.c:1020 Timezone reloaded 530
> definitions
> freeswitch at opxi2Server>
> freeswitch at opxi2Server> sofia status
>
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>            sipinterface_3       profile
> sip:mod_sofia at 192.168.128.2:5080      RUNNING (0)
>            sipinterface_2       profile
> sip:mod_sofia at 192.168.128.2:5070      RUNNING (0)
>
> =================================================================================================
> 2 profiles 0 aliases
>
> freeswitch at opxi2Server> 2011-05-14 09:36:40.171083 [ERR] mod_sofia.c:4624
> Can't find profile sipinterface_1
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Tue May 17 00:08:26 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 16 May 2011 16:08:26 -0400
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
Message-ID: <BANLkTimReEA_UX6HK_tX72o5Jo_o3kW7jQ@mail.gmail.com>

The event socket in the default configuration binds to 127.0.0.1 for
security purposes.  Have you changed that to a real network IP?

On Mon, May 16, 2011 at 4:01 PM, Michael Gende <mgende at gendesign.com> wrote:
> Hello,
>
> Say, I'm setting up a new FS box (no big issues there, using standard CentOS
> and latest FS from the site).
>
> Works fine, registering with provider, handset's, sends/receives calls, etc.
> One weird thing though:
>
> when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
> xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
>
> Locally, fs_cli works, but only without flags of any kind. Just invoking the
> executable without arguments works every time.
>
> I have a few FS's running and want to use the fs_cli on my local computer to
> connect when need be. This works fine for all but my latest creation (two
> prior ones are a year or more older).
>
> Something foolish I've overlooked? No firewall on the new FS box, routing
> and LAN networking look/act fine.
>
> Any commentary welcome, thanks in advance.
>
> Mike G.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mgende at gendesign.com  Tue May 17 00:28:19 2011
From: mgende at gendesign.com (Michael Gende)
Date: Mon, 16 May 2011 15:28:19 -0500
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTikZXFzBK5AiHmFiyUqYU2evKGqwCQ@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
	<BANLkTikjBQ=jC0CooxQzmMfZrW2nknRn+A@mail.gmail.com>
	<BANLkTikZXFzBK5AiHmFiyUqYU2evKGqwCQ@mail.gmail.com>
Message-ID: <BANLkTimkBuV9u+8kSyufx3PEQWD-SzLS+w@mail.gmail.com>

Hey Michael,

What we've tried is this: put fs_cli on a non-FS linux station on the LAN,
then tried to connect to this FS or that, both on the LAN.

Connects fine to the older FS. For the new one, I can only use fs_cli being
ssh'd into the new FS or from its console, and only without arguments.
Weird.
On May 16, 2011 3:05 PM, "Michael Collins" <msc at freeswitch.org> wrote:
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From mgende at gendesign.com  Tue May 17 01:09:58 2011
From: mgende at gendesign.com (Michael Gende)
Date: Mon, 16 May 2011 16:09:58 -0500
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTimReEA_UX6HK_tX72o5Jo_o3kW7jQ@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
	<BANLkTimReEA_UX6HK_tX72o5Jo_o3kW7jQ@mail.gmail.com>
Message-ID: <BANLkTi=Rc_91k3g6MJj5OFWN3dqtHWDdcw@mail.gmail.com>

Hello Kristian,

That was it. Many thanks. We found the right XML file, changed the default
binding from 127.0.0.1 to the LAN IP of the FS box itself and are "in like
the burglar".

Hopefully, we've not compromised security by doing so, as you intimate in
your initial post!

Many Thanks Again,

Mike G.

On Mon, May 16, 2011 at 3:08 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> The event socket in the default configuration binds to 127.0.0.1 for
> security purposes.  Have you changed that to a real network IP?
>
> On Mon, May 16, 2011 at 4:01 PM, Michael Gende <mgende at gendesign.com>
> wrote:
> > Hello,
> >
> > Say, I'm setting up a new FS box (no big issues there, using standard
> CentOS
> > and latest FS from the site).
> >
> > Works fine, registering with provider, handset's, sends/receives calls,
> etc.
> > One weird thing though:
> >
> > when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
> > xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
> >
> > Locally, fs_cli works, but only without flags of any kind. Just invoking
> the
> > executable without arguments works every time.
> >
> > I have a few FS's running and want to use the fs_cli on my local computer
> to
> > connect when need be. This works fine for all but my latest creation (two
> > prior ones are a year or more older).
> >
> > Something foolish I've overlooked? No firewall on the new FS box, routing
> > and LAN networking look/act fine.
> >
> > Any commentary welcome, thanks in advance.
> >
> > Mike G.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 admin at blindi.net  Tue May 17 02:01:53 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 17 May 2011 00:01:53 +0200 (CEST)
Subject: [Freeswitch-users] Error to play multiple file with
	play_and_get_digits
In-Reply-To: <BANLkTimAy-Mk9j662D8DC-37TMCNp4h60A@mail.gmail.com>
References: <BANLkTikNLthc8vz8k4+C97NpRcYRSOrOww@mail.gmail.com>
	<4DCF140A.6030005@ppmt.org> <Pine.LNX.4.64.1105150211500.7979@debian>
	<BANLkTimAy-Mk9j662D8DC-37TMCNp4h60A@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105162352570.6838@debian>

Hi michael, thanks for you nice help!!
I have fix the error.
  my editor has ben wraps all lines.

---------------
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 steveayre at gmail.com  Tue May 17 02:33:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 16 May 2011 23:33:38 +0100
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
Message-ID: <BANLkTi=2yvzhLA=4eirOTQ=Nzm4wq6tZCQ@mail.gmail.com>

By default mod_event_socket only listens on 127.0.0.1 (localhost) so
you won't be able to connect remotely unless you change the listen-ip
setting in conf/autoload_configs/event_socket.conf.xml

-Steve


On 16 May 2011 21:01, Michael Gende <mgende at gendesign.com> wrote:
> Hello,
>
> Say, I'm setting up a new FS box (no big issues there, using standard CentOS
> and latest FS from the site).
>
> Works fine, registering with provider, handset's, sends/receives calls, etc.
> One weird thing though:
>
> when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
> xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
>
> Locally, fs_cli works, but only without flags of any kind. Just invoking the
> executable without arguments works every time.
>
> I have a few FS's running and want to use the fs_cli on my local computer to
> connect when need be. This works fine for all but my latest creation (two
> prior ones are a year or more older).
>
> Something foolish I've overlooked? No firewall on the new FS box, routing
> and LAN networking look/act fine.
>
> Any commentary welcome, thanks in advance.
>
> Mike G.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue May 17 02:36:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 16 May 2011 23:36:56 +0100
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTi=Rc_91k3g6MJj5OFWN3dqtHWDdcw@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
	<BANLkTimReEA_UX6HK_tX72o5Jo_o3kW7jQ@mail.gmail.com>
	<BANLkTi=Rc_91k3g6MJj5OFWN3dqtHWDdcw@mail.gmail.com>
Message-ID: <BANLkTi=zz7VTKK5OwEnzMzHJE8Y_3a_TRA@mail.gmail.com>

You can firewall the port to control who can connect to it. It still
requires a password of course (you should change it from the default).
The protocol is plaintext though, which you should bear in mind.

If you're worried about security you could tunnel a connection to
127.0.0.1 through a SSH tunnel, or look at mod_ssh (a new module, I
haven't had a proper look at it yet).

Something to bear in mind is that anyone on ESL can use any of the FS
commands, which includes the system command which'll allow them to run
any program on your server as the user FS runs as. That's probably the
biggest security risk of opening the ESL port up.

-Steve


On 16 May 2011 22:09, Michael Gende <mgende at gendesign.com> wrote:
> Hello Kristian,
>
> That was it. Many thanks. We found the right XML file, changed the default
> binding from 127.0.0.1 to the LAN IP of the FS box itself and are "in like
> the burglar".
>
> Hopefully, we've not compromised security by doing so, as you intimate in
> your initial post!
>
> Many Thanks Again,
>
> Mike G.
>
> On Mon, May 16, 2011 at 3:08 PM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
>>
>> The event socket in the default configuration binds to 127.0.0.1 for
>> security purposes. ?Have you changed that to a real network IP?
>>
>> On Mon, May 16, 2011 at 4:01 PM, Michael Gende <mgende at gendesign.com>
>> wrote:
>> > Hello,
>> >
>> > Say, I'm setting up a new FS box (no big issues there, using standard
>> > CentOS
>> > and latest FS from the site).
>> >
>> > Works fine, registering with provider, handset's, sends/receives calls,
>> > etc.
>> > One weird thing though:
>> >
>> > when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
>> > xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
>> >
>> > Locally, fs_cli works, but only without flags of any kind. Just invoking
>> > the
>> > executable without arguments works every time.
>> >
>> > I have a few FS's running and want to use the fs_cli on my local
>> > computer to
>> > connect when need be. This works fine for all but my latest creation
>> > (two
>> > prior ones are a year or more older).
>> >
>> > Something foolish I've overlooked? No firewall on the new FS box,
>> > routing
>> > and LAN networking look/act fine.
>> >
>> > Any commentary welcome, thanks in advance.
>> >
>> > Mike G.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>
>


From gcd at i.ph  Tue May 17 04:58:33 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 17 May 2011 08:58:33 +0800
Subject: [Freeswitch-users] mod_lcr
In-Reply-To: <BANLkTikBf7xTR=UEhf-NTwyhOBY0QyXMiQ@mail.gmail.com>
References: <BANLkTikBf7xTR=UEhf-NTwyhOBY0QyXMiQ@mail.gmail.com>
Message-ID: <BANLkTikAU2TesJwPWWjjLoN_rtZpJqfXGg@mail.gmail.com>

u hv to enclose the destination number with () so its value will be assigned
to $1
  <condition field="destination_number" expression="^(0112341234)$">


On Mon, May 16, 2011 at 11:14 PM, Chris Graham <chrisg.lists at gmail.com>wrote:

> Hi All,
> I am having an issue with mod_lcr, when I query from the CLI it
> returns the expected values as per:
>
> freeswitch at internal> lcr 0112341234 peak
>  | Digit Match | Carrier | Rate     | Codec | CID Regexp | Limit |
> Dialstring
>        |
>  | 011         | Telkom  | 0.32000  |       |            |       |
>
> [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
> |
> freeswitch at internal> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:1687
> data passed to lcr is [0112341234 peak]
> 2011-05-16 17:06:19.847934 [WARNING] mod_lcr.c:1731 Using default CID
> [18005551212]
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1]
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:771 0112341234 doesn't
> appear to be a NANP number
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:852 intra routing
> [state:0 lata:0] so rate field is [rate]
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:872 we have an event
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:896 SQL:  SELECT l.digits
> AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS
> lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
> lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
> lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
> cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (0112341234, 011234123, 01123412, 0112341, 011234, 01123, 0112, 011,
> 01, 0) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY
> digits DESC,  rate,  quality DESC,  reliability DESC, rand();
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
>
> [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:664 Adding Telkom to head of
> list
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
>
> [lcr_carrier=Neotel,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:682 Ignoring Duplicate
> route for termination point (sofia/gateway/pp-ast-trunk-01/:)
> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
>
> [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
>
>
> When calling from the dialplan, it fails with:
>
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1563 intrastate channel
> var is [undef]
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1566 Select routes based
> on interstate rates
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1585 LCR Lookup on $1
> using profile peak
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1]
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:771 $1 doesn't appear to
> be a NANP number
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:852 intra routing
> [state:0 lata:0] so rate field is [rate]
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:857 we have a session
> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:896 SQL:  SELECT l.digits
> AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS
> lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
> lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
> lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
> cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (1) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY
> digits DESC,  rate,  quality DESC,  reliability DESC, rand();
> EXECUTE sofia/internal/1000 at 192.168.16.136 bridge()
> 2011-05-16 17:08:33.100261 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1000 at 192.168.16.136 has executed the last dialplan
> instruction, hanging up.
>
>
> The xml entry called:
>
>         <extension name="Test">
>                 <condition field="destination_number"
> expression="^0112341234$">
>                         <action application="lcr" data="$1 peak"/>
>                         <action application="bridge"
> data="${lcr_auto_route}"/>
>                 </condition>
>         </extension>
>
>
> My custom query:
>
>         <profile name="peak">
>                 <param name="custom_sql" value="
> SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name,
> l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
> lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
> lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
> cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
> date_start AND date_end ORDER BY digits DESC,  rate,  quality DESC,
> reliability DESC, rand();
> "/>
>         </profile>
>
> Thanks in advance,
> Chris
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From OSchenk at wnr.com.au  Tue May 17 05:02:07 2011
From: OSchenk at wnr.com.au (Schenk, Oliver)
Date: Tue, 17 May 2011 09:02:07 +0800
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
Message-ID: <A0AF161BE92F1B4CB28FDCBF2524B0420610C8@prdexch01.wnr.internal>

I'm very limited in CISCO speak so I actually had to look up what CCME
means. From your comments I gather that CME is something that runs
inside the CISCO and not something that needs to be installed on a
server? My IOS is version 12.4. Is CME something that is an add-on or
something that is usually built into 2811 routers?

 

 

I'm yet to purchase the FXO cards, that's why I posted this question to
the mailing list in case someone would know FS will work with the
hardware I mentioned.

 

 

Thanks for everyone's reply.

 

 

Cheers!

 

 

From: Ognjen Seslija [mailto:oseslija at gmail.com] 
Sent: 16/05/11 16:46
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] CISCO FXO cards with FreeSwitch

 

I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.



On Mon, May 16, 2011 at 7:08 AM, Schenk, Oliver <OSchenk at wnr.com.au>
wrote:

Hey,

 

I was wondering if anyone out there in the FreeSwitch community has
successfully used FXO cards (e.g. VIC2-4FXO) in a CISCO router (e.g.
2811) with FreeSwitch.

 

Was this achieved via SIP communication between the CISCO and
FreeSwitch?

 

Any documentation out there? I can't seem to find much for FreeSwitch
regarding what would need to be done on the FreeSwitch end.

 

 

 

Regards,

 

Oliver Schenk

 


NOTICE - This e-mail and any files transmitted with it are confidential
and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail
in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender,
except where the sender specifically states them to be the views of
WestNet Rail.


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

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.
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From OSchenk at wnr.com.au  Tue May 17 05:11:35 2011
From: OSchenk at wnr.com.au (Schenk, Oliver)
Date: Tue, 17 May 2011 09:11:35 +0800
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
Message-ID: <A0AF161BE92F1B4CB28FDCBF2524B0420610D2@prdexch01.wnr.internal>

Any chance of some sample CISCO and/or FreeSwitch configs?

 

You can send to my email if you can't post public.

 

 

Thanks all,

 

Oliver





 

 

From: Ognjen Seslija [mailto:oseslija at gmail.com] 
Sent: 16/05/11 16:46
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] CISCO FXO cards with FreeSwitch

 

I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.



On Mon, May 16, 2011 at 7:08 AM, Schenk, Oliver <OSchenk at wnr.com.au>
wrote:

Hey,

 

I was wondering if anyone out there in the FreeSwitch community has
successfully used FXO cards (e.g. VIC2-4FXO) in a CISCO router (e.g.
2811) with FreeSwitch.

 

Was this achieved via SIP communication between the CISCO and
FreeSwitch?

 

Any documentation out there? I can't seem to find much for FreeSwitch
regarding what would need to be done on the FreeSwitch end.

 

 

 

Regards,

 

Oliver Schenk

 


NOTICE - This e-mail and any files transmitted with it are confidential
and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail
in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender,
except where the sender specifically states them to be the views of
WestNet Rail.


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

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.
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From gabe at gundy.org  Tue May 17 06:22:11 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 16 May 2011 20:22:11 -0600
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
Message-ID: <BANLkTi=NjOeHoJMnPvYnRN6bHa1ssVwn+A@mail.gmail.com>

On Mon, May 16, 2011 at 2:01 PM, Michael Gende <mgende at gendesign.com> wrote:
> when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
> xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
>
> Locally, fs_cli works, but only without flags of any kind. Just invoking the
> executable without arguments works every time.

What do you see with netstat -tlp ?

It could be as simple as the event socket just listening on localhost
(that is the default).


Best,
Gabe


From fieldpeak at gmail.com  Tue May 17 07:19:41 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 17 May 2011 11:19:41 +0800
Subject: [Freeswitch-users] call somebody into conference after
	conferenceestablished
In-Reply-To: <BANLkTi=wbd8ivSA4Gom3P2QsJ9AY7GRJUA@mail.gmail.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<201105121532017963789@asiainfo-linkage.com>
	<BANLkTi=wbd8ivSA4Gom3P2QsJ9AY7GRJUA@mail.gmail.com>
Message-ID: <BANLkTikPq5GteE1Y5w56oL_=oAQ3ieVC1w@mail.gmail.com>

Hi Gurus,

Is there anybody can help this issue? thanks!

Regards,
Charles

? 2011?5?14? ??3:54?fieldpeak <fieldpeak at gmail.com>???

> Hi Michael,
>
> Thanks for your detailed suggestion.
>
> i test it as below steps, it works fine, however, there is a limiation that
> i have to fill fixed callee number in conference.conf.xml
> (data="execute_extension 5001"), while actually i what to dial arbitrary
> number after "*" dynamically(e.g *1234, *2345), i can think out it could
> realize by a IVR script like "data="execute_extension IVR_scripts" and after
> IVR give some voice prompt (pls input the callee number) then ... however,
> considering simplicity, can i convey the DTMF into the dialplan (just
> directly press *1234, and then 1234 ring...)? could you please provide any
> hints or any suggestion... Thanks a lot!
>
> 1. in conference.conf.xml, set as below,
> <control action="execute_application" digits="*" data="execute_extension
> 5001"/>
>
> 2. in dial plan,
>
> <!-- for confernce named conf100 -->
> <extension name="conf100" >
>    <condition field="destination_number" expression="^666$">
>        <action application="answer"/>
>        <action application="set" data="conference_user_list=|"/>
>        <action application="conference" data="conf100 at default"/>
>    </condition>
> </extension>
>
> and
>
> <!-- for dial out 5001 from the conference conf100 -->
>
> <extension name="5_to_conf_out" >
>    <condition field="destination_number" expression="^(5\d+)$">
>        <!-- <action application="conference" data="conf100 dial
> sofia/internal/5001 at 192.168.200.201 100 Conference_Caller_Name"/> -->
>      <!-- <action application="bridge" data="sofia/internal/$
> 1 at 192.168.200.201"/> -->
>      <action application="set" data="api_result=${conference(conf100 dial
> sofia/internal/$1 at 192.168.200.201 100 Conference_Caller_Name)}"/>
>    </condition>
> </extension>
>
> 3. register a extension 3001, call 666 and join a conference, and then
> press "*", the FS will call 5001 on a IAD.
> then 5001 and 3001 join the same conference.
>
>
> 2011/5/12 liuyp2 <liuyp2 at asiainfo-linkage.com>
>
>>  conference Your-Conf-Name dial user/1002
>>
>>  ------------------------------
>>  liuyp2
>> 2011-05-12
>>  ------------------------------
>>  *????* fieldpeak
>> *?????* 2011-05-12 11:11:41
>> *????* FreeSWITCH-users
>> *???*
>> *???* [Freeswitch-users] call somebody into conference after
>> conferenceestablished
>>
>> Hi Gurus,
>>
>> i dial 666 and enter a conference, then i need call some body's phone
>> number to join him into this conference...
>> is there anyone can advise how can realize this scenario?
>>
>> thanks.
>>
>> 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 gabe at gundy.org  Tue May 17 08:04:50 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 16 May 2011 22:04:50 -0600
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTi=NjOeHoJMnPvYnRN6bHa1ssVwn+A@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
	<BANLkTi=NjOeHoJMnPvYnRN6bHa1ssVwn+A@mail.gmail.com>
Message-ID: <BANLkTikx-whY1q-OO9df4M8UHM=FWRgKVw@mail.gmail.com>

Sorry, I don't know how I missed Kristian and Steven's emails.  Sounds
like they helped you get it figured out.  Great minds think alike ;)

Best,
Gabe


On Mon, May 16, 2011 at 8:22 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Mon, May 16, 2011 at 2:01 PM, Michael Gende <mgende at gendesign.com> wrote:
>> when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
>> xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
>>
>> Locally, fs_cli works, but only without flags of any kind. Just invoking the
>> executable without arguments works every time.
>
> What do you see with netstat -tlp ?
>
> It could be as simple as the event socket just listening on localhost
> (that is the default).
>
>
> Best,
> Gabe
>


From all.eforums at gmail.com  Tue May 17 10:19:07 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Tue, 17 May 2011 02:19:07 -0400
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
	<BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>
Message-ID: <BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>

On Fri, May 13, 2011 at 8:44 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, May 2, 2011 at 4:27 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > I've tried this cool formula for streaming TTS via google:
> > http://wiki.freeswitch.org/wiki/TTS
> > and while the link produces a pretty darn nice sounding MP3, I get:
> > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at
> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123:
> Invalid
> > mpg123 handle. (code 10)
> > Is anyone else using this reliably?
>
> Yes, I too can confirm that others are using this reliably.
>
> What more can you tell us about your problem?
>
> Best,
> Gabe
>
> Hello,

Just wanted to find out when you say "reliably" is this being used in
production on a relatively small/medium load, like maybe 10-20
look-ups/translations per second? The last I checked (about 4-5 mths ago)
after about 10 or so (can't remember the exact number) Google throws out a
Captcha as a challenge. Now, that could also be since I was doing this over
a web-browser. Does it change or doesn't have that limitation when using
mod_shout in freeswitch?

Thanks so much
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From math.parent at gmail.com  Tue May 17 11:51:20 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 17 May 2011 09:51:20 +0200
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <026801cc1403$82c7c790$885756b0$@com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
	<BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>
	<4DD11946.8080100@gmail.com>
	<BANLkTine+R41wM72nNe=-PrN3sP3CzKNcg@mail.gmail.com>
	<026801cc1403$82c7c790$885756b0$@com>
Message-ID: <BANLkTim7vOdqFW62WV_R_fTKj+-9pcO6wQ@mail.gmail.com>

2011/5/16 Michael Collins <msc at freeswitch.org>:
> I think what Mathieu is saying is that since he is writing mod_skinny that
> it would be extremely valuable for him to see some raw network captures
> between the CUCM and the 2811 - assuming, of course, that they are not both
> running inside the 2811 as Meftah noted. In other words, in addition to the
> OP's issue Mathieu was asking for extra information about how the OP is
> currently using the FXO cards.

Yes, this is what I meant.

2011/5/16 Peder <peder at networkoblivion.com>:
> FYI, I am about 80% sure that sccp is only for phones or ATAs.? Routers use
> MGCP or H.323 to talk to CCM? or CUCM or CCME or whatever they are calling
> it today?.? He appears to be running CCME on a router, so you won?t get any
> captures as it will all be internal.

What I need is an ATA capture, especially with T.38. This is not
urgent, as I have a lot to do before.

-- 
Mathieu


From oseslija at gmail.com  Tue May 17 12:06:06 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 17 May 2011 10:06:06 +0200
Subject: [Freeswitch-users] CISCO FXO cards with FreeSwitch
In-Reply-To: <BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>
References: <A0AF161BE92F1B4CB28FDCBF2524B04206102F@prdexch01.wnr.internal>
	<BANLkTinS8meY7GjRANd0jZzZ5sJn8fvfHA@mail.gmail.com>
	<BANLkTinc4hVfkLnkhPVzXwj6gni252-jQQ@mail.gmail.com>
Message-ID: <BANLkTikfqQJXcU0SxtRmk8jy_ec1rYv34Q@mail.gmail.com>

Cisco 2811 is the platform CCME or CUCME (whatever the name is these days)
is ran on (like Call Manager appliance lite hosted on the router itself). It
speaks SCCP or SIP to phones and SIP/H323/PSTN protos for trunks.
I have working CCMEs with SCCP phones and SIP trunks to FS for DID routing,
voicemail etc.



On Mon, May 16, 2011 at 10:57 AM, Mathieu Parent <math.parent at gmail.com>wrote:

> 2011/5/16 Ognjen Seslija <oseslija at gmail.com>:
> > I have working SIP trunks from CCME 4.3 to FS, if that's what you mean.
> >
>
> No ;-) I want to know the protocol used between CCME and the Cisco
> 2811: MGCP, H323, SCCP, SIP?
>
>
> --
> Mathieu
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eagle.antonio at gmail.com  Tue May 17 13:43:34 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 17 May 2011 09:43:34 +0000
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTinQ3jSZTSVqpsC8E8i=6WPmFM6s4w@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<BANLkTinQ3jSZTSVqpsC8E8i=6WPmFM6s4w@mail.gmail.com>
Message-ID: <BANLkTi=HL2e3Qb26bJ9zR11hfWiZz5Qi6g@mail.gmail.com>

Hello Anthony.

Thanks for your input i have added the channel variable and still the same :

I bridge the call using uuid_bridge and after this LEG B is disconnected.
LEG A is still alive with due to a time.sleep python (not freeswitch sleep)
If i do bridge with a time.sleep() on LEG B the two calls are alive but no
transit of audio.


I also tried intercept but still no luck :\

Regards
A/T




2011/5/16 Anthony Minessale <anthony.minessale at gmail.com>

> set the channel variable park_after_bridge=true on all the channels so
> they go back to park again when the call ends.
>
>
> On Mon, May 16, 2011 at 9:16 AM, Antonio Teixeira
> <eagle.antonio at gmail.com> wrote:
> > Hello Anton.
> >
> > The problem is the following  :
> >
> > I receive an Inbound Call Leg A.
> > I say something using TTS and place it on Park , grab the channel uuid.
> >
> > Fork another call pass the uuid as argument.
> >
> > As soon as Leg B picks up , TTS fires UP asks the user a digit and after
> we
> > have got the digit .
> >
> > I send uuid_bridge mycurrentchanel + argument uuid from channel A
> >
> > And i got +OK great :D
> >
> > The problem , Park is now broken and the call is terminated with last
> > dialplan extension executed or simply the call is terminated HANGUP.
> >
> > If i do some time.sleep(40) the calls don't end but they aren't merge i
> > can't see no RTP flowing trough.
> >
> > Hope you guys can help me out.
> >
> > Regards
> > A/T
> >
> >
> >
> > 2011/5/16 Anton VG <anton.vazir at gmail.com>
> >>
> >> I do it with originate &park - all works, i bridge them by uuid_bridge
> >> successfully
> >>
> >> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
> >> > Good Morning.
> >> >
> >> > Here i continue my fight against ESL :)
> >> > On Leg A I'm forking a call that will become Leg B after a short IVR.
> >> >
> >> > Anyway
> >> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
> >> > Presses
> >> > 1 and is merged by uuid_bridge
> >> > Then...
> >> > Both calls are disconnected with Executed Last Dialplan Command .
> >> >
> >> > I have tried everything from freezing the IVR with sleep(in python and
> >> > using
> >> > the freeswitch sleep).
> >> >
> >> > But i'm unable to merge both calls.
> >> >
> >> > As anyone tried something similar.
> >> >
> >> > Regards
> >> > A/T
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jcasale at activenetwerx.com  Tue May 17 16:02:00 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 17 May 2011 12:02:00 +0000
Subject: [Freeswitch-users] Custom ivr prompts
Message-ID: <assp.0118c95d84.92097A6A775D5147B1078E3F15430B92522DC2@prato.activenetwerx.local>

Anyone know where the fs ivr/vm prompts came from, I need to add a few and
it would be helpful to utilize all of what exists as most are fine for my need.

Being able to pay for only the couple custom ones I need would certainly be
nice if they matched.

Thanks,
jlc


From rhuddleston at gmail.com  Tue May 17 16:45:12 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 17 May 2011 08:45:12 -0400
Subject: [Freeswitch-users] Custom ivr prompts
In-Reply-To: <assp.0118c95d84.92097A6A775D5147B1078E3F15430B92522DC2@prato.activenetwerx.local>
References: <assp.0118c95d84.92097A6A775D5147B1078E3F15430B92522DC2@prato.activenetwerx.local>
Message-ID: <152901cc1490$441450d0$cc3cf270$@com>

GM Voices?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joseph
L. Casale
Sent: Tuesday, May 17, 2011 8:02 AM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] Custom ivr prompts

Anyone know where the fs ivr/vm prompts came from, I need to add a few and
it would be helpful to utilize all of what exists as most are fine for my
need.

Being able to pay for only the couple custom ones I need would certainly be
nice if they matched.

Thanks,
jlc

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



From jcasale at activenetwerx.com  Tue May 17 17:19:28 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 17 May 2011 13:19:28 +0000
Subject: [Freeswitch-users] Custom ivr prompts
In-Reply-To: <152901cc1490$441450d0$cc3cf270$@com>
References: <92097A6A775D5147B1078E3F15430B92522DC2@prato.activenetwerx.local>
	<152901cc1490$441450d0$cc3cf270$@com>
Message-ID: <assp.0118f51921.92097A6A775D5147B1078E3F15430B925234A8@prato.activenetwerx.local>

>GM Voices?

Thanks, they are going to quote it as they say they can replicate the voice perfectly.
Appreciate that,
jlc


From leonardo.bidinoto at voicetechnology.com.br  Tue May 17 18:25:17 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Tue, 17 May 2011 11:25:17 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
	<BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
	<BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>
Message-ID: <BANLkTim=ueSmFA_ug2MbbLFMBh0362_2Sg@mail.gmail.com>

Sure. Im sending a pcap file made by tcpdump and one that i made by ngrep.
In both files, it was registering whats happening when i stuck the channel
by hanging up while using a ESL connection inside a conference(app socket
8085 sync full). I did a "conference kick" command in this channel while its
was waiting to close the ESL connection.


2011/5/16 Michael Collins <msc at freeswitch.org>

> Can you tcpdump or otherwise capture the traffic on port 8085? I am curious
> what is happening with that.
> -MC
>
>
> On Mon, May 16, 2011 at 12:12 PM, Leonardo P. Bidinoto <
> leonardo.bidinoto at voicetechnology.com.br> wrote:
>
>> hehe, ok michael.
>>
>> here is the pastebin link:
>> http://pastebin.freeswitch.org/16303
>>
>>
>> 2011/5/13 Michael Collins <msc at freeswitch.org>
>>
>>> Pastebin this info and select "FreeSWITCH Log" as the syntax
>>> highlighting. I need the colorized output to read logs. (I'm getting older
>>> and it's hard for me to ready black and white in an email.)
>>>
>>> -MC
>>>
>>>
>>> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <
>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>
>>>> Hi Michael,
>>>>
>>>> Just succeeded to reproduce the problem.
>>>>
>>>> The condition is: when a channel inside a conference is using a ESL
>>>> connection(lets call it "A") through socket application and another ESL
>>>> connection(lets call it "B") executes a command with this channel, the "B"
>>>> ESL connection will wait the "A" ESL connection close to execute its
>>>> command.
>>>> If the channel hangs up before the "A" ESL connection is closed, then
>>>> "B" ESL command will never be executed and the stucked channel will still be
>>>> there, into sofia and the conference too.
>>>> To verify that, just do "show channels" and "conference list". with
>>>> "uuid_exists" command, return "false".
>>>>
>>>> Here are the actions done by the channel before get stucked:
>>>>
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321 [NOTICE]
>>>> switch_channel.c:816 New Channel sofia/external/
>>>> 1000123402 at 192.168.0.154 [16e09413-9cb0-4011-a635-f91933a35c0f]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>>> state [received][100]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia.c:4772 Remote SDP:
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia_glue.c:4656 Audio Codec Compare
>>>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia_glue.c:4656 Audio Codec Compare
>>>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia_glue.c:4656 Audio Codec Compare
>>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia_glue.c:2788 Set Codec sofia/external/1000123402 at 192.168.0.154PCMU/8000 20 ms 160 samples 64000 bits
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change
>>>> CS_NEW -> CS_INIT
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_state_machine.c:325 (sofia/external/
>>>> 1000123402 at 192.168.0.154) Running State Change CS_INIT
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_state_machine.c:361 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State INIT
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change
>>>> CS_INIT -> CS_ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_state_machine.c:361 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State INIT going to sleep
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_state_machine.c:325 (sofia/external/
>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154)
>>>> Callstate Change DOWN -> RINGING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_state_machine.c:364 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO]
>>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>>> public
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true) =~
>>>> /^true$/ break=on-false
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~
>>>> /^true$/ break=on-false
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 parsing [public->public_extensions]
>>>> continue=false
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
>>>> destination_number(1234567890) =~ /^(\d*)$/ break=on-false
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>> switch_core_state_machine.c:119 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>> switch_core_state_machine.c:364 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>> switch_core_state_machine.c:325 (sofia/external/
>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>> switch_core_state_machine.c:371 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 transfer(1234567890 XML default)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State
>>>> Change CS_EXECUTE -> CS_ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_core_session.c:707 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [NOTICE]
>>>> switch_ivr.c:1603 Transfer sofia/external/1000123402 at 192.168.0.154 to
>>>> XML[1234567890 at default]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_core_state_machine.c:371 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State EXECUTE going to sleep
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_core_state_machine.c:325 (sofia/external/
>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_core_state_machine.c:364 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO]
>>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>>> default
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 parsing [default->flex] continue=false
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Regex (PASS) [flex]
>>>> destination_number(1234567890) =~ /^(\d+)$/ break=on-false
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Action log(INFO VOICE received
>>>> dest=1234567890)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Action set(playback_terminators=#)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall we?)
>>>>
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Action
>>>> set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>> 1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>> switch_core_state_machine.c:119 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>> switch_core_state_machine.c:364 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>> switch_core_state_machine.c:325 (sofia/external/
>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>> switch_core_state_machine.c:371 (sofia/external/
>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>> switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154Standard EXECUTE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO]
>>>> mod_dptools.c:1184 VOICE received dest=1234567890
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 set(playback_terminators=#)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>> [playback_terminators]=[#]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO]
>>>> mod_dptools.c:1184 Let's do some ivrd, shall we?
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>> [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 socket(localhost:8084 full)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute answer()
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 answer()
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>>> sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154]
>>>> 192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>>> switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>> sofia_glue.c:3284 Set 2833 dtmf send payload to 101
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>> sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>> mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>> switch_core_session.c:707 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>> switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154)
>>>> Callstate Change RINGING -> ACTIVE
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [NOTICE]
>>>> mod_dptools.c:930 Channel [sofia/external/1000123402 at 192.168.0.154] has
>>>> been answered
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG]
>>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>>> state [completed][200]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG]
>>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>>> state [ready][200]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute
>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav
>>>> flex_digits 5000 )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 read(1 1
>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>>>> )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>> [flex_digits]=[UNDEF]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute read(1 1
>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>> flex_digits 5000 )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 read(1 1
>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>> flex_digits 5000 )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF 8:640
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>> [flex_digits]=[UNDEF]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>>> flex_digits 5000 #,*)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 read(1 11
>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>>>> #,*)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF #:960
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>> [flex_digits]=[UNDEF]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute
>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF #:800
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>>> mod_dptools.c:1664 Digit #
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute
>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute conference(15646 at teste+flags{waste})
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>>> mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1 channel
>>>> 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>>> mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1 channel
>>>> 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>>> codec L16:70
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>>>> switch_core_session.c:707 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>>>> mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF *:960
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
>>>> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
>>>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
>>>> previous codec PCMU:0.
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>>> flex_digits 2000 )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 read(1 1
>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>> [flex_digits]=[UNDEF]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute
>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>>>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>>> codec L16:70
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF *:800
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
>>>> mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
>>>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore
>>>> previous codec PCMU:0.
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>>
>>>> ====================================================================================================================================================
>>>> While Inside this connection, a "conference 15646 kick [member_id of
>>>> this channels]" command is executed by a fs_cli console and get stuck while
>>>> waiting response.
>>>>
>>>> ====================================================================================================================================================
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>>> flex_digits 2000 )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 read(1 1
>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG]
>>>> switch_ivr_play_say.c:1649 done playing file
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG]
>>>> switch_rtp.c:3280 RTP RECV DTMF 1:960
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>> Execute set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>> [flex_digits]=[UNDEF]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>>> switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154)
>>>> Callstate Change ACTIVE -> HANGUP
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [NOTICE]
>>>> sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154 [CS_EXECUTE]
>>>> [NORMAL_CLEARING]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>>> switch_channel.c:2576 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [KILL]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
>>>> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip
>>>> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>>> codec L16:70
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>> mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG]
>>>> mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip
>>>> receive message [UNBRIDGE] (channel is hungup already)
>>>>
>>>> I hope this info helps.
>>>>
>>>> 2011/5/12 Michael Collins <msc at freeswitch.org>
>>>>
>>>>>
>>>>>
>>>>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
>>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>>
>>>>>> Hi Michael,
>>>>>>
>>>>>> Im not using to any cdr module.
>>>>>
>>>>>
>>>>> I would recommend that you do several things:
>>>>>
>>>>> #1 - update to latest git
>>>>> #2 - rotate logs
>>>>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>>>>> #4 - reproduce the symptom with a single call (if possible)
>>>>> #5 - pastebin the log for the uuid in question and link to it in this
>>>>> thread
>>>>>
>>>>> From there hopefully we'll get a clue as to what is happening.
>>>>> -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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Leonardo Pires Bidinoto
>>>> Voice Technology
>>>> www.voicetechnology.com.br
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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interface: lo (127.0.0.0/255.0.0.0)
filter: (ip) and ( port 8085 )
####
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
connect


##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Event-Name: CHANNEL_DATA
Core-UUID: 3da23fcb-09e6-470f-9c1c-c8c55367088b
FreeSWITCH-Hostname: FSBR
FreeSWITCH-IPv4: 192.168.0.154
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-05-17%2011%3A12%3A04
Event-Date-GMT: Tue,%2017%20May%202011%2014%3A12%3A04%20GMT
Event-Date-Timestamp: 1305641524162945
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: parse_command
Event-Calling-Line-Number: 1843
Channel-Direction: inbound
Channel-Username: 1000123402
Channel-Dialplan: XML
Channel-Caller-ID-Name: VT_CallAll
Channel-Caller-ID-Number: 1000123402
Channel-Network-Addr: 192.168.0.111
Channel-ANI: 1000123402
Channel-Destination-Number: 1234567890
Channel-Unique-ID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Channel-Source: mod_sofia
Channel-Context: default
Channel-RDNIS: 1234567890
Channel-Channel-Name: sofia/external/1000123402%40192.168.0.154
Channel-Profile-Index: 2
Channel-Profile-Created-Time: 1305641505382431
Channel-Channel-Created-Time: 1305641505378432
Channel-Channel-Answered-Time: 1305641505433429
Channel-Channel-Progress-Time: 0
Channel-Channel-Progress-Media-Time: 0
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: true
Channel-Privacy-Hide-Name: false
Channel-Privacy-Hide-Number: false
Channel-State: CS_EXECUTE
Channel-Call-State: ACTIVE
Channel-State-Number: 4
Channel-Name: sofia/external/1000123402%40192.168.0.154
Unique-ID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Read-Codec-Bit-Rate: 64000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Channel-Write-Codec-Bit-Rate: 64000
Caller-Direction: inbound
Caller-Username: 1000123402
Caller-Dialplan: XML
Caller-Caller-ID-Name: VT_CallAll
Caller-Caller-ID-Number: 1000123402
Caller-Network-Addr: 192.168.0.111
Caller-ANI: 1000123402
Caller-Destination-Number: 1234567890
Caller-Unique-ID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Caller-Source: mod_sofia
Caller-Context: default
Caller-RDNIS: 1234567890
Caller-Channel-Name: sofia/external/1000123402%40192.168.0.154
Caller-Profile-Index: 2
Caller-Profile-Created-Time: 1305641505382431
Caller-Channel-Created-Time: 1305641505378432
Caller-Channel-Answered-Time: 1305641505433429
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
variable_direction: inbound
variable_uuid: 3d61dc59-c442-4a94-a213-2dd55cf24829
variable_sip_local_network_addr: 192.168.0.154
variable_sip_network_ip: 192.168.0.111
variable_sip_network_port: 28799
variable_sip_received_ip: 192.168.0.111
variable_sip_received_port: 28799
variable_sip_via_protocol: udp
variable_sip_from_user: 1000123402
variable_sip_from_uri: 1000123402%40192.168.0.154
variable_sip_from_host: 192.168.0.154
variable_sip_from_user_stripped: 1000123402
variable_sofia_profile_name: external
variable_sip_full_via: SIP/2.0/UDP%20192.168.0.111%3A28799%3Bbranch%3Dz9hG4bK-d8754z-c76a5027e57b8e55-1---d8754z-%3Brport%3D28799
variable_sip_req_user: 1234567890
variable_sip_req_uri: 1234567890%40192.168.0.154
variable_sip_req_host: 192.168.0.154
variable_sip_to_user: 1234567890
variable_sip_to_uri: 1234567890%40192.168.0.154
variable_sip_to_host: 192.168.0.154
variable_sip_contact_user: 1000123402
variable_sip_contact_port: 28799
variable_sip_contact_uri: 1000123402%40192.168.0.111%3A28799
variable_sip_contact_host: 192.168.0.111
variable_channel_name: sofia/external/1000123402%40192.168.0.154
variable_sip_user_agent: X-Lite%20release%201104o%20stamp%2056125
variable_sip_via_host: 192.168.0.111
variable_sip_via_port: 28799
variable_sip_via_rport: 28799
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%206%202%20IN%20IP4%20192.168.0.111%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20192.168.0.111%0D%0At%3D0%200%0D%0Am%3Daudio%2060172%20RTP/AVP%20107%200%208%20101%0D%0Aa%3Drtpmap%3A107%20BV32/16000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%208WLcGidE%20iJrX4Grh%20192.168.0.111%2060172%0D%0Aa%3Dalt%3A2%202%20%3A%20pvziSqb6%20Plo2W8CM%20169.254.200.97%2060172%0D%0Aa%3Dalt%3A3%201%20%3A%20HZe7dhMq%20R8wzjNTX%20192.168.129.1%2060172%0D%0A
variable_sip_audio_recv_pt: 0
variable_sip_use_codec_name: PCMU
variable_sip_use_codec_rate: 8000
variable_sip_use_codec_ptime: 20
variable_write_codec: PCMU
variable_write_rate: 8000
variable_max_forwards: 69
variable_playback_terminators: %23
variable_playback_delimiter: %26
variable_sip_local_sdp_str: v%3D0%0Ao%3DFreeSWITCH%201305624205%201305624206%20IN%20IP4%20192.168.0.154%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20192.168.0.154%0At%3D0%200%0Am%3Daudio%2017300%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A
variable_local_media_ip: 192.168.0.154
variable_local_media_port: 17300
variable_sip_use_pt: 0
variable_rtp_use_ssrc: 3923290733
variable_remote_media_ip: 192.168.0.111
variable_remote_media_port: 60172
variable_endpoint_disposition: ANSWER
variable_sip_to_tag: BFXeH2QpeZF0c
variable_sip_from_tag: 8b593c05
variable_sip_cseq: 1
variable_sip_call_id: MDg3YzY4YzYzYTRiNzUxMDgwODM3ZjM4NTliYjNkOWQ.
variable_sip_from_display: VT_CallAll
variable_sip_full_from: %22VT_CallAll%22%20%3Csip%3A1000123402%40192.168.0.154%3E%3Btag%3D8b593c05
variable_sip_to_display: 1234567890
variable_sip_full_to: %221234567890%22%20%3Csip%3A1234567890%40192.168.0.154%3E%3Btag%3DBFXeH2QpeZF0c
variable_read_terminator_used: %23
variable_read_result: failure
variable_session_conf_number: 15646
variable_RECORD_COPYRIGHT: Studio%2088
variable_RECORD_SOFTWARE: WaveConvertPro
variable_RECORD_DATE: 04/10/06%20%2013%3A57%3A51
variable_mute: false
variable_playback_seconds: 1
variable_playback_ms: 1910
variable_playback_samples: 15280
variable_conference_name: 15646
variable_conference_member_id: 1
variable_conference_uuid: 1b27ae62-f046-454a-8b10-fa43d82ae4cc
variable_conference_last_matching_digits: *
variable_read_codec: PCMU
variable_read_rate: 8000
variable_admin_menu: true
variable_current_application_data: localhost%3A8085%20sync%20full
variable_current_application: socket
variable_socket_host: localhost
Content-Type: command/reply
Reply-Text: %2BOK%0A
Socket-Mode: static
Control: full


##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_setvar 3d61dc59-c442-4a94-a213-2dd55cf24829 inside_conf_menu true
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 4


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
+OK

##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 session_conf_number
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 5


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
15646
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 conference_member_id
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 1


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
1
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 admin_menu
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 4


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
true
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_setvar 3d61dc59-c442-4a94-a213-2dd55cf24829 admin_menu 
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 4


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
+OK

##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 exit_conf
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 superuser
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
sendmsg
call-command: execute
execute-app-name: read
execute-app-arg: 1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 


##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: command/reply
Reply-Text: +OK


#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 flex_digits
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 1


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
*
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
sendmsg
call-command: execute
execute-app-name: set
execute-app-arg: flex_digits


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: command/reply
Reply-Text: +OK


#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 private
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 session_moderator_flag
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
sendmsg
call-command: execute
execute-app-name: playback
execute-app-arg: /usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav


##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: command/reply
Reply-Text: +OK


#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
api uuid_setvar 3d61dc59-c442-4a94-a213-2dd55cf24829 inside_conf_menu 
##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 4


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
+OK

##
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]
exit
#
T 127.0.0.1:8085 -> 127.0.0.1:35624 [AP]



##
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: command/reply
Reply-Text: +OK bye


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Content-Type: text/disconnect-notice
Controlled-Session-UUID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Content-Disposition: disconnect
Content-Length: 67


#
T 127.0.0.1:35624 -> 127.0.0.1:8085 [AP]
Disconnected, goodbye.
See you at ClueCon! http://www.cluecon.com/

######
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
connect


##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Event-Name: CHANNEL_DATA
Core-UUID: 3da23fcb-09e6-470f-9c1c-c8c55367088b
FreeSWITCH-Hostname: FSBR
FreeSWITCH-IPv4: 192.168.0.154
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-05-17%2011%3A12%3A20
Event-Date-GMT: Tue,%2017%20May%202011%2014%3A12%3A20%20GMT
Event-Date-Timestamp: 1305641540499173
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: parse_command
Event-Calling-Line-Number: 1843
Channel-Direction: inbound
Channel-Username: 1000123402
Channel-Dialplan: XML
Channel-Caller-ID-Name: VT_CallAll
Channel-Caller-ID-Number: 1000123402
Channel-Network-Addr: 192.168.0.111
Channel-ANI: 1000123402
Channel-Destination-Number: 1234567890
Channel-Unique-ID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Channel-Source: mod_sofia
Channel-Context: default
Channel-RDNIS: 1234567890
Channel-Channel-Name: sofia/external/1000123402%40192.168.0.154
Channel-Profile-Index: 2
Channel-Profile-Created-Time: 1305641505382431
Channel-Channel-Created-Time: 1305641505378432
Channel-Channel-Answered-Time: 1305641505433429
Channel-Channel-Progress-Time: 0
Channel-Channel-Progress-Media-Time: 0
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: true
Channel-Privacy-Hide-Name: false
Channel-Privacy-Hide-Number: false
Channel-State: CS_EXECUTE
Channel-Call-State: ACTIVE
Channel-State-Number: 4
Channel-Name: sofia/external/1000123402%40192.168.0.154
Unique-ID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Read-Codec-Bit-Rate: 64000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Channel-Write-Codec-Bit-Rate: 64000
Caller-Direction: inbound
Caller-Username: 1000123402
Caller-Dialplan: XML
Caller-Caller-ID-Name: VT_CallAll
Caller-Caller-ID-Number: 1000123402
Caller-Network-Addr: 192.168.0.111
Caller-ANI: 1000123402
Caller-Destination-Number: 1234567890
Caller-Unique-ID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Caller-Source: mod_sofia
Caller-Context: default
Caller-RDNIS: 1234567890
Caller-Channel-Name: sofia/external/1000123402%40192.168.0.154
Caller-Profile-Index: 2
Caller-Profile-Created-Time: 1305641505382431
Caller-Channel-Created-Time: 1305641505378432
Caller-Channel-Answered-Time: 1305641505433429
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
variable_direction: inbound
variable_uuid: 3d61dc59-c442-4a94-a213-2dd55cf24829
variable_sip_local_network_addr: 192.168.0.154
variable_sip_network_ip: 192.168.0.111
variable_sip_network_port: 28799
variable_sip_received_ip: 192.168.0.111
variable_sip_received_port: 28799
variable_sip_via_protocol: udp
variable_sip_from_user: 1000123402
variable_sip_from_uri: 1000123402%40192.168.0.154
variable_sip_from_host: 192.168.0.154
variable_sip_from_user_stripped: 1000123402
variable_sofia_profile_name: external
variable_sip_full_via: SIP/2.0/UDP%20192.168.0.111%3A28799%3Bbranch%3Dz9hG4bK-d8754z-c76a5027e57b8e55-1---d8754z-%3Brport%3D28799
variable_sip_req_user: 1234567890
variable_sip_req_uri: 1234567890%40192.168.0.154
variable_sip_req_host: 192.168.0.154
variable_sip_to_user: 1234567890
variable_sip_to_uri: 1234567890%40192.168.0.154
variable_sip_to_host: 192.168.0.154
variable_sip_contact_user: 1000123402
variable_sip_contact_port: 28799
variable_sip_contact_uri: 1000123402%40192.168.0.111%3A28799
variable_sip_contact_host: 192.168.0.111
variable_channel_name: sofia/external/1000123402%40192.168.0.154
variable_sip_user_agent: X-Lite%20release%201104o%20stamp%2056125
variable_sip_via_host: 192.168.0.111
variable_sip_via_port: 28799
variable_sip_via_rport: 28799
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%206%202%20IN%20IP4%20192.168.0.111%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20192.168.0.111%0D%0At%3D0%200%0D%0Am%3Daudio%2060172%20RTP/AVP%20107%200%208%20101%0D%0Aa%3Drtpmap%3A107%20BV32/16000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dalt%3A1%203%20%3A%208WLcGidE%20iJrX4Grh%20192.168.0.111%2060172%0D%0Aa%3Dalt%3A2%202%20%3A%20pvziSqb6%20Plo2W8CM%20169.254.200.97%2060172%0D%0Aa%3Dalt%3A3%201%20%3A%20HZe7dhMq%20R8wzjNTX%20192.168.129.1%2060172%0D%0A
variable_sip_audio_recv_pt: 0
variable_sip_use_codec_name: PCMU
variable_sip_use_codec_rate: 8000
variable_sip_use_codec_ptime: 20
variable_write_codec: PCMU
variable_write_rate: 8000
variable_max_forwards: 69
variable_playback_terminators: %23
variable_playback_delimiter: %26
variable_sip_local_sdp_str: v%3D0%0Ao%3DFreeSWITCH%201305624205%201305624206%20IN%20IP4%20192.168.0.154%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20192.168.0.154%0At%3D0%200%0Am%3Daudio%2017300%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A
variable_local_media_ip: 192.168.0.154
variable_local_media_port: 17300
variable_sip_use_pt: 0
variable_rtp_use_ssrc: 3923290733
variable_remote_media_ip: 192.168.0.111
variable_remote_media_port: 60172
variable_endpoint_disposition: ANSWER
variable_sip_to_tag: BFXeH2QpeZF0c
variable_sip_from_tag: 8b593c05
variable_sip_cseq: 1
variable_sip_call_id: MDg3YzY4YzYzYTRiNzUxMDgwODM3ZjM4NTliYjNkOWQ.
variable_sip_from_display: VT_CallAll
variable_sip_full_from: %22VT_CallAll%22%20%3Csip%3A1000123402%40192.168.0.154%3E%3Btag%3D8b593c05
variable_sip_to_display: 1234567890
variable_sip_full_to: %221234567890%22%20%3Csip%3A1234567890%40192.168.0.154%3E%3Btag%3DBFXeH2QpeZF0c
variable_read_terminator_used: %23
variable_session_conf_number: 15646
variable_mute: false
variable_conference_name: 15646
variable_conference_member_id: 1
variable_conference_uuid: 1b27ae62-f046-454a-8b10-fa43d82ae4cc
variable_read_result: success
variable_RECORD_COPYRIGHT: Studio%2088
variable_RECORD_SOFTWARE: WaveConvertPro
variable_RECORD_DATE: 04/10/06%20%2013%3A57%3A51
variable_playback_terminator_used: %23
variable_playback_seconds: 4
variable_playback_ms: 4096
variable_playback_samples: 32768
variable_conference_last_matching_digits: *
variable_read_codec: PCMU
variable_read_rate: 8000
variable_admin_menu: true
variable_current_application_data: localhost%3A8085%20sync%20full
variable_current_application: socket
variable_socket_host: localhost
Content-Type: command/reply
Reply-Text: %2BOK%0A
Socket-Mode: static
Control: full


##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_setvar 3d61dc59-c442-4a94-a213-2dd55cf24829 inside_conf_menu true
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 4


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
+OK

##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 session_conf_number
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 5


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
15646
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 conference_member_id
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 1


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
1
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 admin_menu
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 4


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
true
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_setvar 3d61dc59-c442-4a94-a213-2dd55cf24829 admin_menu 
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 4


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
+OK

##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 exit_conf
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 superuser
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
sendmsg
call-command: execute
execute-app-name: read
execute-app-arg: 1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 


##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: command/reply
Reply-Text: +OK


#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 flex_digits
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 1


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
1
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
sendmsg
call-command: execute
execute-app-name: set
execute-app-arg: flex_digits


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: command/reply
Reply-Text: +OK


#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 private
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 session_moderator_flag
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 7


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
_undef_
##
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]
api uuid_getvar 3d61dc59-c442-4a94-a213-2dd55cf24829 mute
#
T 127.0.0.1:8085 -> 127.0.0.1:35626 [AP]



##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: api/response
Content-Length: 5


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
false
##
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Content-Type: text/disconnect-notice
Controlled-Session-UUID: 3d61dc59-c442-4a94-a213-2dd55cf24829
Content-Disposition: disconnect
Content-Length: 67


#
T 127.0.0.1:35626 -> 127.0.0.1:8085 [AP]
Disconnected, goodbye.
See you at ClueCon! http://www.cluecon.com/

##exit
348 received, 0 dropped
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From anthony.minessale at gmail.com  Tue May 17 18:47:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 17 May 2011 09:47:19 -0500
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTi=HL2e3Qb26bJ9zR11hfWiZz5Qi6g@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<BANLkTinQ3jSZTSVqpsC8E8i=6WPmFM6s4w@mail.gmail.com>
	<BANLkTi=HL2e3Qb26bJ9zR11hfWiZz5Qi6g@mail.gmail.com>
Message-ID: <BANLkTinf894JH=JPV8tJgwK32pUs8J=-Gg@mail.gmail.com>

You should explain what "forking a call" means. This does not exist in
our terminology.
My advice is to start thinking about what you need to do to make it
work vs how you want it to work.

You MUST use async mode for what you are trying to do.  The reason the
call won't bridge in sync mode is because you are asking the channel
to do 2 things at once and it can only do 1 because sync mode means to
block on every command.

If you are still stuck:

Please pastebin a console log on debug level at http://pastebin.freeswitch.org



On Tue, May 17, 2011 at 4:43 AM, Antonio Teixeira
<eagle.antonio at gmail.com> wrote:
> Hello Anthony.
>
> Thanks for your input i have added the channel variable and still the same :
>
> I bridge the call using uuid_bridge and after this LEG B is disconnected.
> LEG A is still alive with due to a time.sleep python (not freeswitch sleep)
> If i do bridge with a time.sleep() on LEG B the two calls are alive but no
> transit of audio.
>
>
> I also tried intercept but still no luck :\
>
> Regards
> A/T
>
>
>
>
> 2011/5/16 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> set the channel variable park_after_bridge=true on all the channels so
>> they go back to park again when the call ends.
>>
>>
>> On Mon, May 16, 2011 at 9:16 AM, Antonio Teixeira
>> <eagle.antonio at gmail.com> wrote:
>> > Hello Anton.
>> >
>> > The problem is the following? :
>> >
>> > I receive an Inbound Call Leg A.
>> > I say something using TTS and place it on Park , grab the channel uuid.
>> >
>> > Fork another call pass the uuid as argument.
>> >
>> > As soon as Leg B picks up , TTS fires UP asks the user a digit and after
>> > we
>> > have got the digit .
>> >
>> > I send uuid_bridge mycurrentchanel + argument uuid from channel A
>> >
>> > And i got +OK great :D
>> >
>> > The problem , Park is now broken and the call is terminated with last
>> > dialplan extension executed or simply the call is terminated HANGUP.
>> >
>> > If i do some time.sleep(40) the calls don't end but they aren't merge i
>> > can't see no RTP flowing trough.
>> >
>> > Hope you guys can help me out.
>> >
>> > Regards
>> > A/T
>> >
>> >
>> >
>> > 2011/5/16 Anton VG <anton.vazir at gmail.com>
>> >>
>> >> I do it with originate &park - all works, i bridge them by uuid_bridge
>> >> successfully
>> >>
>> >> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
>> >> > Good Morning.
>> >> >
>> >> > Here i continue my fight against ESL :)
>> >> > On Leg A I'm forking a call that will become Leg B after a short IVR.
>> >> >
>> >> > Anyway
>> >> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
>> >> > Presses
>> >> > 1 and is merged by uuid_bridge
>> >> > Then...
>> >> > Both calls are disconnected with Executed Last Dialplan Command .
>> >> >
>> >> > I have tried everything from freezing the IVR with sleep(in python
>> >> > and
>> >> > using
>> >> > the freeswitch sleep).
>> >> >
>> >> > But i'm unable to merge both calls.
>> >> >
>> >> > As anyone tried something similar.
>> >> >
>> >> > Regards
>> >> > A/T
>> >> >
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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 mgende at gendesign.com  Tue May 17 19:21:34 2011
From: mgende at gendesign.com (Michael Gende)
Date: Tue, 17 May 2011 10:21:34 -0500
Subject: [Freeswitch-users] fs_cli question
In-Reply-To: <BANLkTikx-whY1q-OO9df4M8UHM=FWRgKVw@mail.gmail.com>
References: <BANLkTikM_uX3jtq2o=A7Oan_sfi2UULGJQ@mail.gmail.com>
	<BANLkTi=NjOeHoJMnPvYnRN6bHa1ssVwn+A@mail.gmail.com>
	<BANLkTikx-whY1q-OO9df4M8UHM=FWRgKVw@mail.gmail.com>
Message-ID: <BANLkTin6C-8h_krMCgF15J9j80ARRzNJJA@mail.gmail.com>

No worries Gabe, I appreciate you taking the time to look at it.

I do wonder about why my older FS "didn't care" about connections but my new
one does. I'll have to look at the .XML file for the binding and see if the
old and new are the same.

I do worry a little about security issues I may have created for myself. I
guess I'll have to mitigate that via a firewall setting.

Thanks Again,

Mike G.

On Mon, May 16, 2011 at 11:04 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> Sorry, I don't know how I missed Kristian and Steven's emails.  Sounds
> like they helped you get it figured out.  Great minds think alike ;)
>
> Best,
> Gabe
>
>
> On Mon, May 16, 2011 at 8:22 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> > On Mon, May 16, 2011 at 2:01 PM, Michael Gende <mgende at gendesign.com>
> wrote:
> >> when I use -/bin/fs_cli, I've found that using the flags I'm used to "-H
> >> xxx.xxx.xxx.xxx -p password" doesn't seem to work remotely.
> >>
> >> Locally, fs_cli works, but only without flags of any kind. Just invoking
> the
> >> executable without arguments works every time.
> >
> > What do you see with netstat -tlp ?
> >
> > It could be as simple as the event socket just listening on localhost
> > (that is the default).
> >
> >
> > Best,
> > 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
>
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From jcasale at activenetwerx.com  Tue May 17 20:01:15 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 17 May 2011 16:01:15 +0000
Subject: [Freeswitch-users] Voicemail macro error in log
Message-ID: <assp.01182d9298.92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>

While looking at something, I noticed a
[WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not match any patterns
pass by but this match exists in the voicemail_ack macro. I looked in freeswitch.xml.fsxml and
it is being loaded.

What could cause this?

Thanks,
jlc


From eagle.antonio at gmail.com  Tue May 17 20:37:03 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 17 May 2011 16:37:03 +0000
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTinf894JH=JPV8tJgwK32pUs8J=-Gg@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<BANLkTinQ3jSZTSVqpsC8E8i=6WPmFM6s4w@mail.gmail.com>
	<BANLkTi=HL2e3Qb26bJ9zR11hfWiZz5Qi6g@mail.gmail.com>
	<BANLkTinf894JH=JPV8tJgwK32pUs8J=-Gg@mail.gmail.com>
Message-ID: <BANLkTinrvRJ5ciSbqDrW0wpG0Sn4F=PvDw@mail.gmail.com>

Hello Anthony.

Forking A Call :

bgapi('originate',"{strategy=108,step=1,origination_caller_id_number=XXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s,position=%s}sofia/external/%s at XXXXXX4998
XML default" % (self.channel_uuid,position,'XXXXXX'))

Means running this command Against The Outbound Socket and that goes well.

The Problem with async is that i have a multi-level IVR system not a
generally simple ( press 1 for sales , press 2 for support ) type of IVR.
This means lots of monitoring on the events to get it right , and some APPs
fail to send events, lack of documentation ,etc.

Anyway i was able to solve this problem :

1- Process ESL and transfer the call to a extension&ValetPark and call
disconnect() on the ESL connection , now open a new connection and issue
uuid_bridge , all goes well connection with ESL is no longer in the middle
:D

I have Tried to do Park from inside the ESL even declaring
setAsyncExecute('1') does not execute ValetPark in async it will just fail.
If i don't use setAsyncExecute  the call parks and ESL processing stops , so
i just bypassed that using the XML dialplan.

I know this is not the recommended way , but will have to do for now ,
anyway i will try to review this situation and will see what i can do with
the async mode.

If you want me to do any special testing feel free to drop me an e-mail off
list.

Regards
A/T

2011/5/17 Anthony Minessale <anthony.minessale at gmail.com>

> You should explain what "forking a call" means. This does not exist in
> our terminology.
> My advice is to start thinking about what you need to do to make it
> work vs how you want it to work.
>
> You MUST use async mode for what you are trying to do.  The reason the
> call won't bridge in sync mode is because you are asking the channel
> to do 2 things at once and it can only do 1 because sync mode means to
> block on every command.
>
> If you are still stuck:
>
> Please pastebin a console log on debug level at
> http://pastebin.freeswitch.org
>
>
>
> On Tue, May 17, 2011 at 4:43 AM, Antonio Teixeira
> <eagle.antonio at gmail.com> wrote:
> > Hello Anthony.
> >
> > Thanks for your input i have added the channel variable and still the
> same :
> >
> > I bridge the call using uuid_bridge and after this LEG B is disconnected.
> > LEG A is still alive with due to a time.sleep python (not freeswitch
> sleep)
> > If i do bridge with a time.sleep() on LEG B the two calls are alive but
> no
> > transit of audio.
> >
> >
> > I also tried intercept but still no luck :\
> >
> > Regards
> > A/T
> >
> >
> >
> >
> > 2011/5/16 Anthony Minessale <anthony.minessale at gmail.com>
> >>
> >> set the channel variable park_after_bridge=true on all the channels so
> >> they go back to park again when the call ends.
> >>
> >>
> >> On Mon, May 16, 2011 at 9:16 AM, Antonio Teixeira
> >> <eagle.antonio at gmail.com> wrote:
> >> > Hello Anton.
> >> >
> >> > The problem is the following  :
> >> >
> >> > I receive an Inbound Call Leg A.
> >> > I say something using TTS and place it on Park , grab the channel
> uuid.
> >> >
> >> > Fork another call pass the uuid as argument.
> >> >
> >> > As soon as Leg B picks up , TTS fires UP asks the user a digit and
> after
> >> > we
> >> > have got the digit .
> >> >
> >> > I send uuid_bridge mycurrentchanel + argument uuid from channel A
> >> >
> >> > And i got +OK great :D
> >> >
> >> > The problem , Park is now broken and the call is terminated with last
> >> > dialplan extension executed or simply the call is terminated HANGUP.
> >> >
> >> > If i do some time.sleep(40) the calls don't end but they aren't merge
> i
> >> > can't see no RTP flowing trough.
> >> >
> >> > Hope you guys can help me out.
> >> >
> >> > Regards
> >> > A/T
> >> >
> >> >
> >> >
> >> > 2011/5/16 Anton VG <anton.vazir at gmail.com>
> >> >>
> >> >> I do it with originate &park - all works, i bridge them by
> uuid_bridge
> >> >> successfully
> >> >>
> >> >> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
> >> >> > Good Morning.
> >> >> >
> >> >> > Here i continue my fight against ESL :)
> >> >> > On Leg A I'm forking a call that will become Leg B after a short
> IVR.
> >> >> >
> >> >> > Anyway
> >> >> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers ,
> >> >> > Presses
> >> >> > 1 and is merged by uuid_bridge
> >> >> > Then...
> >> >> > Both calls are disconnected with Executed Last Dialplan Command .
> >> >> >
> >> >> > I have tried everything from freezing the IVR with sleep(in python
> >> >> > and
> >> >> > using
> >> >> > the freeswitch sleep).
> >> >> >
> >> >> > But i'm unable to merge both calls.
> >> >> >
> >> >> > As anyone tried something similar.
> >> >> >
> >> >> > Regards
> >> >> > A/T
> >> >> >
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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  Tue May 17 20:48:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 May 2011 09:48:04 -0700
Subject: [Freeswitch-users] Voicemail macro error in log
In-Reply-To: <assp.01182d9298.92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
References: <assp.01182d9298.92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
Message-ID: <BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>

What is in your "voicemail_ack" macro?
-MC

On Tue, May 17, 2011 at 9:01 AM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> While looking at something, I noticed a
> [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not
> match any patterns
> pass by but this match exists in the voicemail_ack macro. I looked in
> freeswitch.xml.fsxml and
> it is being loaded.
>
> What could cause this?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue May 17 20:50:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 May 2011 09:50:37 -0700
Subject: [Freeswitch-users] Custom ivr prompts
In-Reply-To: <assp.0118f51921.92097A6A775D5147B1078E3F15430B925234A8@prato.activenetwerx.local>
References: <92097A6A775D5147B1078E3F15430B92522DC2@prato.activenetwerx.local>
	<152901cc1490$441450d0$cc3cf270$@com>
	<assp.0118f51921.92097A6A775D5147B1078E3F15430B925234A8@prato.activenetwerx.local>
Message-ID: <BANLkTinkngwC-=O7EBQ_-3UGNB_7eHnCLA@mail.gmail.com>

If you are going to place an order with GM Voices then please contact me off
list. We might be able to help you get a discounted rate. Also, it might be
good to ask around and see if anyone else wants to get some custom prompts
recorded and to split the session setup fee.

-MC

On Tue, May 17, 2011 at 6:19 AM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >GM Voices?
>
> Thanks, they are going to quote it as they say they can replicate the voice
> perfectly.
> Appreciate that,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue May 17 20:55:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 May 2011 09:55:20 -0700
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
	<BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>
	<BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>
Message-ID: <BANLkTimR+9=wmZd5REfT+1mpbCjD53uwTg@mail.gmail.com>

I have to wonder if Google thinks 10-20 lookups per second from a single
client is a "light/medium" load. That's 36K - 72K lookups per hour. If I
were you I would tcpdump the traffic and see what Google is "really" sending
back. I doubt that Google cares about the distinction between a browser and
other types of clients.

-MC

On Mon, May 16, 2011 at 11:19 PM, A E [Gmail] <all.eforums at gmail.com> wrote:

> On Fri, May 13, 2011 at 8:44 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Mon, May 2, 2011 at 4:27 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> > I've tried this cool formula for streaming TTS via google:
>> > http://wiki.freeswitch.org/wiki/TTS
>> > and while the link produces a pretty darn nice sounding MP3, I get:
>> > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at
>> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
>> > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123:
>> Invalid
>> > mpg123 handle. (code 10)
>> > Is anyone else using this reliably?
>>
>> Yes, I too can confirm that others are using this reliably.
>>
>> What more can you tell us about your problem?
>>
>> Best,
>> Gabe
>>
>> Hello,
>
> Just wanted to find out when you say "reliably" is this being used in
> production on a relatively small/medium load, like maybe 10-20
> look-ups/translations per second? The last I checked (about 4-5 mths ago)
> after about 10 or so (can't remember the exact number) Google throws out a
> Captcha as a challenge. Now, that could also be since I was doing this over
> a web-browser. Does it change or doesn't have that limitation when using
> mod_shout in freeswitch?
>
> Thanks so much
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 17 20:58:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 May 2011 09:58:52 -0700
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DCE3654.7080002@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>
Message-ID: <BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>

You want your Lua script to *send* an event? It most certainly can fire
events:

http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event

I suppose the question is: what do you want to have happen when you fire an
event? What will receive this event?

-MC

On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     May I fire custom cdr event for a LUA script? I've read
> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
> Would You please point me to proper documentation or give an example?
>
> --
> 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 infos at madovsky.org  Tue May 17 21:03:58 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 17 May 2011 13:03:58 -0400
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com><BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com><BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>
	<BANLkTimR+9=wmZd5REfT+1mpbCjD53uwTg@mail.gmail.com>
Message-ID: <DE77F089BB49408C9C928FCEEECB5C81@e1705>

they will do if overload occurs
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, May 17, 2011 12:55 PM
  Subject: Re: [Freeswitch-users] Is Anyone Using Google TTS?


  I have to wonder if Google thinks 10-20 lookups per second from a single client is a "light/medium" load. That's 36K - 72K lookups per hour. If I were you I would tcpdump the traffic and see what Google is "really" sending back. I doubt that Google cares about the distinction between a browser and other types of clients.


  -MC


  On Mon, May 16, 2011 at 11:19 PM, A E [Gmail] <all.eforums at gmail.com> wrote:

    On Fri, May 13, 2011 at 8:44 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

      On Mon, May 2, 2011 at 4:27 PM, Avi Marcus <avi at avimarcus.net> wrote:
      > I've tried this cool formula for streaming TTS via google:
      > http://wiki.freeswitch.org/wiki/TTS
      > and while the link produces a pretty darn nice sounding MP3, I get:
      > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at
      > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
      > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123: Invalid
      > mpg123 handle. (code 10)
      > Is anyone else using this reliably?


      Yes, I too can confirm that others are using this reliably.

      What more can you tell us about your problem?

      Best,
      Gabe



    Hello,


    Just wanted to find out when you say "reliably" is this being used in production on a relatively small/medium load, like maybe 10-20 look-ups/translations per second? The last I checked (about 4-5 mths ago) after about 10 or so (can't remember the exact number) Google throws out a Captcha as a challenge. Now, that could also be since I was doing this over a web-browser. Does it change or doesn't have that limitation when using mod_shout in freeswitch?


    Thanks so much 

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






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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jcasale at activenetwerx.com  Tue May 17 21:14:46 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 17 May 2011 17:14:46 +0000
Subject: [Freeswitch-users] Voicemail macro error in log
In-Reply-To: <BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>
References: <92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
	<BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>
Message-ID: <assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>

>What is in your "voicemail_ack" macro?

Hi Michael,

It looks like:
  <macro name="voicemail_ack"> 
    <input pattern="^(too-small)$">
      <match>
	<action function="play-file" data="voicemail/vm-too-small.wav"/>
      </match>
    </input>
    <input pattern="^(deleted)$">
      <match>
	<action function="play-file" data="voicemail/vm-message.wav"/>
	<action function="play-file" data="voicemail/vm-$1.wav"/>
      </match>
    </input>
    <input pattern="^(saved)$">
      <match>
	<action function="play-file" data="voicemail/vm-message.wav"/>
	<action function="play-file" data="voicemail/vm-$1.wav"/>
      </match>
    </input>
    <input pattern="^(emailed)$">
      <match>
	<action function="play-file" data="voicemail/vm-message.wav"/>
	<action function="play-file" data="voicemail/vm-$1.wav"/>
      </match>
    </input>
    <input pattern="^(marked-urgent)$">
      <match>
	<action function="play-file" data="voicemail/vm-message.wav"/>
	<action function="play-file" data="voicemail/vm-$1.wav"/>
      </match>
    </input>
  </macro>


From nico at clickfono.com  Tue May 17 21:21:51 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 17 May 2011 13:21:51 -0400
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <DE77F089BB49408C9C928FCEEECB5C81@e1705>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
	<BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>
	<BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>
	<BANLkTimR+9=wmZd5REfT+1mpbCjD53uwTg@mail.gmail.com>
	<DE77F089BB49408C9C928FCEEECB5C81@e1705>
Message-ID: <BANLkTinKn_97R9Y_=nCRrUFzKjJ_nY4Amg@mail.gmail.com>

Be careful about the legal implications of using Google's TTS in your
system. There are a few discussions on Google Groups (e.g.
http://goo.gl/TvRVR) about the possibility of using Google's TTS for
applications other than just Google Translate on the web, and there's a
Google employee (DeWitt) saying that there are no Terms of Service (ToS),
nor an official API for their TTS system, so that you should abide by their
general ToS, and that "[TTS wasn't] designed for internal usage only, will
likely change or break over time, and thus shouldn't be accessed by external
developers"



On Tue, May 17, 2011 at 1:03 PM, Madovsky <infos at madovsky.org> wrote:

>  they will do if overload occurs
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, May 17, 2011 12:55 PM
> *Subject:* Re: [Freeswitch-users] Is Anyone Using Google TTS?
>
> I have to wonder if Google thinks 10-20 lookups per second from a single
> client is a "light/medium" load. That's 36K - 72K lookups per hour. If I
> were you I would tcpdump the traffic and see what Google is "really" sending
> back. I doubt that Google cares about the distinction between a browser and
> other types of clients.
>
> -MC
>
> On Mon, May 16, 2011 at 11:19 PM, A E [Gmail] <all.eforums at gmail.com>wrote:
>
>>  On Fri, May 13, 2011 at 8:44 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>>  On Mon, May 2, 2011 at 4:27 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>> > I've tried this cool formula for streaming TTS via google:
>>> > http://wiki.freeswitch.org/wiki/TTS
>>> > and while the link produces a pretty darn nice sounding MP3, I get:
>>> > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:804 Error: MPG123 Error at
>>> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
>>> > 2011-05-03 01:16:47.828769 [ERR] mod_shout.c:807 Error from mpg123:
>>> Invalid
>>> > mpg123 handle. (code 10)
>>> > Is anyone else using this reliably?
>>>
>>> Yes, I too can confirm that others are using this reliably.
>>>
>>> What more can you tell us about your problem?
>>>
>>> Best,
>>> Gabe
>>>
>>> Hello,
>>
>> Just wanted to find out when you say "reliably" is this being used in
>> production on a relatively small/medium load, like maybe 10-20
>> look-ups/translations per second? The last I checked (about 4-5 mths ago)
>> after about 10 or so (can't remember the exact number) Google throws out a
>> Captcha as a challenge. Now, that could also be since I was doing this over
>> a web-browser. Does it change or doesn't have that limitation when using
>> mod_shout in freeswitch?
>>
>> Thanks so much
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 17 23:54:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 May 2011 12:54:21 -0700
Subject: [Freeswitch-users] Voicemail macro error in log
In-Reply-To: <assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>
References: <92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
	<BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>
	<assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>
Message-ID: <BANLkTi=c0mso4-zxp__TM1sqNS7UE+QWxg@mail.gmail.com>

Can you pb the debug log when this gets called as well as the dialplan
snippet where it gets called?
-MC

On Tue, May 17, 2011 at 10:14 AM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> >What is in your "voicemail_ack" macro?
>
> Hi Michael,
>
> It looks like:
>  <macro name="voicemail_ack">
>    <input pattern="^(too-small)$">
>      <match>
>        <action function="play-file" data="voicemail/vm-too-small.wav"/>
>      </match>
>    </input>
>    <input pattern="^(deleted)$">
>      <match>
>        <action function="play-file" data="voicemail/vm-message.wav"/>
>        <action function="play-file" data="voicemail/vm-$1.wav"/>
>      </match>
>    </input>
>    <input pattern="^(saved)$">
>      <match>
>        <action function="play-file" data="voicemail/vm-message.wav"/>
>        <action function="play-file" data="voicemail/vm-$1.wav"/>
>      </match>
>    </input>
>    <input pattern="^(emailed)$">
>      <match>
>        <action function="play-file" data="voicemail/vm-message.wav"/>
>        <action function="play-file" data="voicemail/vm-$1.wav"/>
>      </match>
>    </input>
>    <input pattern="^(marked-urgent)$">
>      <match>
>        <action function="play-file" data="voicemail/vm-message.wav"/>
>        <action function="play-file" data="voicemail/vm-$1.wav"/>
>      </match>
>    </input>
>  </macro>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 18 00:43:58 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 17 May 2011 16:43:58 -0400
Subject: [Freeswitch-users] skype child accounts and FS
Message-ID: <8C410090745C40EB81929054AC7A70A1@e1705>

Is it possible to register multiple skype child accounts
from http://www.skype.com/intl/en/business/
with FS and skype_open ?

Thanks
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From all.eforums at gmail.com  Wed May 18 00:53:41 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Tue, 17 May 2011 16:53:41 -0400
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTimR+9=wmZd5REfT+1mpbCjD53uwTg@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
	<BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>
	<BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>
	<BANLkTimR+9=wmZd5REfT+1mpbCjD53uwTg@mail.gmail.com>
Message-ID: <BANLkTik60iy4M+eUgGaz4FwuWptjJ1rTKw@mail.gmail.com>

On Tue, May 17, 2011 at 12:55 PM, Michael Collins <msc at freeswitch.org>wrote:

> I have to wonder if Google thinks 10-20 lookups per second from a single
> client is a "light/medium" load. That's 36K - 72K lookups per hour. If I
> were you I would tcpdump the traffic and see what Google is "really" sending
> back. I doubt that Google cares about the distinction between a browser and
> other types of clients.
>
> -MC
>
>
> Haha sorry for that air-headed comment. Didn't really do the math in my
head. Really should have said  "heavy" load :) As for it caring whether it's
a client or a browser, I'm not sure. I read somewhere that seeing some of
the header attributes (that typically browsers send) it will either send you
an MP3 file or return a "404" but that doesn't seem to be the problem anyone
else has mentioned when using it with mod_shout. I do know about the captcha
as I saw it on 2-3 occasions but it might have its own rules i.e. 10
translations in one minute from the same IP might throw it out to prevent
abuse. No one else seems to have run into it, it seems though so that might
mean that kind of a rule doesn't exist. Needs more investigation
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From frankie.k.yiu at gmail.com  Wed May 18 01:03:35 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 17 May 2011 14:03:35 -0700
Subject: [Freeswitch-users] How to resolve AUDIO RTP REPORTS ERROR: [Bind
	Error!]
Message-ID: <BANLkTimbgPXB1N+fXFHu3=CyOQH_S+UDow@mail.gmail.com>

Hi there,

When I make 30 calls to the other freeSwitch box, I got 1 or 2 errors on the
receiving side.  Could someone please let me know how to resolve this?

2011-05-17 12:40:36.061473 [ERR] sofia_glue.c:3449 AUDIO RTP REPORTS ERROR:
[Bin
d Error!]


Thanks,
Frankie
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From all.eforums at gmail.com  Wed May 18 00:58:51 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Tue, 17 May 2011 16:58:51 -0400
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTinKn_97R9Y_=nCRrUFzKjJ_nY4Amg@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
	<BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>
	<BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>
	<BANLkTimR+9=wmZd5REfT+1mpbCjD53uwTg@mail.gmail.com>
	<DE77F089BB49408C9C928FCEEECB5C81@e1705>
	<BANLkTinKn_97R9Y_=nCRrUFzKjJ_nY4Amg@mail.gmail.com>
Message-ID: <BANLkTin9OVDcjw5JkEz_yktT7v06U-mXiA@mail.gmail.com>

On Tue, May 17, 2011 at 1:21 PM, Nicolas Brenner <nico at clickfono.com> wrote:

> Be careful about the legal implications of using Google's TTS in your
> system. There are a few discussions on Google Groups (e.g.
> http://goo.gl/TvRVR) about the possibility of using Google's TTS for
> applications other than just Google Translate on the web, and there's a
> Google employee (DeWitt) saying that there are no Terms of Service (ToS),
> nor an official API for their TTS system, so that you should abide by their
> general ToS, and that "[TTS wasn't] designed for internal usage only, will
> likely change or break over time, and thus shouldn't be accessed by
> external developers"
>
>
> Yeah that seems to be getting into a territory that only the big boys can
play in. Last thing I want is to be in trouble with the Big 'G', triple OG!
Maybe we'll keep the usage of Google TTS just for playing with it and
occasional translations to use as a benchmark coz the quality *is* damn
good.
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From anton.vazir at gmail.com  Wed May 18 01:19:00 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 18 May 2011 02:19:00 +0500
Subject: [Freeswitch-users] Clustering/ storing registration server IP in DB
Message-ID: <BANLkTi=mqLgdOFVQLgWkXO3EZObzFEpt7A@mail.gmail.com>

Trying to figure how to get info where (on which FS server) the SIP
endpoint have been registered.
There is an XML_CURL - but it get's only registration requests, and
than not notified by the FS server, whether endpoint have been
accepted for registration by the server or not. Should I listen for an
proper event, and than update the DB ? Or there is a simplier/proper
way to update DB upon endpoint registration?

2. Also - if there is a multiple regs allowed, and the same user have
been registered on several FS server?


From steveayre at gmail.com  Wed May 18 02:06:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 17 May 2011 23:06:59 +0100
Subject: [Freeswitch-users] Clustering/ storing registration server IP
	in DB
In-Reply-To: <BANLkTi=mqLgdOFVQLgWkXO3EZObzFEpt7A@mail.gmail.com>
References: <BANLkTi=mqLgdOFVQLgWkXO3EZObzFEpt7A@mail.gmail.com>
Message-ID: <BANLkTikrARQNCvQJnwFdoSAMkL_1+AXhiw@mail.gmail.com>

You can move the core db and set a dsn for mod_sofia pointing to a
ODBC database. Registrations will then be stored in that database,
providing you are on latest git.

-Steve


On 17 May 2011 22:19, Anton VG <anton.vazir at gmail.com> wrote:
> Trying to figure how to get info where (on which FS server) the SIP
> endpoint have been registered.
> There is an XML_CURL - but it get's only registration requests, and
> than not notified by the FS server, whether endpoint have been
> accepted for registration by the server or not. Should I listen for an
> proper event, and than update the DB ? Or there is a simplier/proper
> way to update DB upon endpoint registration?
>
> 2. Also - if there is a multiple regs allowed, and the same user have
> been registered on several FS 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 jcasale at activenetwerx.com  Wed May 18 02:43:50 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 17 May 2011 22:43:50 +0000
Subject: [Freeswitch-users] Voicemail macro error in log
In-Reply-To: <BANLkTi=c0mso4-zxp__TM1sqNS7UE+QWxg@mail.gmail.com>
References: <92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
	<BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>
	<assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>
	<BANLkTi=c0mso4-zxp__TM1sqNS7UE+QWxg@mail.gmail.com>
Message-ID: <assp.0118eb1d2f.92097A6A775D5147B1078E3F15430B92526907@prato.activenetwerx.local>

>Can you pb the debug log when this gets called as well as the dialplan snippet where it gets called?

Hi Michael,
http://pastebin.freeswitch.org/16326
http://pastebin.freeswitch.org/16327

This instance is running the OpenWRT 1.0.6 packages, appreciate the help!
jlc


From bwibowo at gmail.com  Wed May 18 04:02:49 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 18 May 2011 07:02:49 +0700
Subject: [Freeswitch-users] mod_manage install
Message-ID: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>

hi
i have fs running on centos 5.5, any body knows the combination version of
software need to be installed like mono etc.
have tried few combination but always failed compiling


thx
budi
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From david.villasmil.work at gmail.com  Wed May 18 05:29:49 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 18 May 2011 03:29:49 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
Message-ID: <BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>

Hello and sorry for taking so long to answer.

What do you mean what's involved?

- You need to have a MySQL Server and a Web Server (same host is fine, but
preferably different than FS)
- You would need to compile your own FS, then copy all configs and scripts
to the FS and to the Webserver as well as importing a MySQL schema.

Everything else is done via the web interface.

I don't have a contrib, would be nice to have one, or a github so i can work
from there. (I've never used those for develoment, only for checkout ;) so
be patient, please.

David

On Thu, May 12, 2011 at 10:09 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, May 12, 2011 at 12:49 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Guys,
>>
>> Well, except for a few things, I think this is about-ready to go out to
>> you guys.
>>
>> Problem is, I have NO IDEA about Gira or anything like that. I developed
>> this on my own free time and making no money out of it at all.
>>
>> So, I would appreciate it if someone can guide me to publish this somehow
>> ;)
>>
>
> Well, everyone is pretty anxious to see this. First question: what's
> involved in the installation? What are the dependencies? Secondly, do you
> have a sub-folder on the freeswitch-contrib repo? If not then Raymond can
> create one for you. You could also publish it to github and then link from
> github to your freeswitch-contrib folder at a later date.
>
> -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 admin at blindi.net  Wed May 18 05:58:42 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 18 May 2011 03:58:42 +0200 (CEST)
Subject: [Freeswitch-users] Problem ringback does not working by
	orignatecalls
In-Reply-To: <BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105180345430.9120@debian>

Hi all, i generage a callback via originate.
the orinatecommand:
/usr/local/freeswitch/bin/fs_cli -x "bgapi originate 
{ignore_early_media=true,bypass_media=true,originate_retries=10,origination_caller_id_name=Callback,originate_retry_sleep_ms=60000,originate_timeout=900}loopback/XXXXXXX/callback_only_out 
&bridge(loopback/callback_only_sendex1/callback_only_send)"
My phonenumber is ringing. here the prompt. I don.t here a transferrington 
or music.
My extension:
<extension name="callback_only_sendex1">
<condition field="destination_number" 
expression="^callback_only_sendex1$">
  <action application="answer" />
  <action application="playback" data="silence_stream://10"/>
<action application="set" data="ringback=${de-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="ring_ready"/>
<action application="set" data="call_timeout=30"/>
  <action application="set" data="hangup_after_bridge=true"/>

  <action application="set" data="ignore_early_media=true" />

<action application="bridge"
data="loopback/0174XXXXX/disa"/>
</condition>
</extension>


Can your help please thank you


---------------
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 adminjew at gmail.com  Wed May 18 06:06:09 2011
From: adminjew at gmail.com (Yitzchok)
Date: Tue, 17 May 2011 22:06:09 -0400
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
Message-ID: <BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>

Make sure mono is installed and check which version using "mono -V" and
follow the instructions on http://wiki.freeswitch.org/wiki/Mod_managed let
us know if you get into any errors.

This helped me for installing mono
http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum

<http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
Yitzchok


On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com> wrote:

> hi
> i have fs running on centos 5.5, any body knows the combination version of
> software need to be installed like mono etc.
> have tried few combination but always failed compiling
>
>
> thx
> budi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From admin at blindi.net  Wed May 18 08:40:44 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 18 May 2011 06:40:44 +0200 (CEST)
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and
	exec_after_bridge_app
In-Reply-To: <BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105180620370.10537@debian>

Hi all, i gernerate a callback with the originate command.
I do forward all calls to my cellphone with a bridge
(loppback in and out).
If I.m not the option add: bypass_media=true in the originate string,
  fs aborts the connection to my cellphone.
if i add this option, the connection is esstablisht, bind_meta_app is not 
working.
I need this feature to transfer callers from my cellphone to others.
exec_after_bridge_app
  also does not work unfortunately.
  Is there a way to solve the problem?

thank you


---------------
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 gabe at gundy.org  Wed May 18 09:56:08 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 17 May 2011 23:56:08 -0600
Subject: [Freeswitch-users] How to resolve AUDIO RTP REPORTS ERROR:
	[Bind Error!]
In-Reply-To: <BANLkTimbgPXB1N+fXFHu3=CyOQH_S+UDow@mail.gmail.com>
References: <BANLkTimbgPXB1N+fXFHu3=CyOQH_S+UDow@mail.gmail.com>
Message-ID: <BANLkTi=-WUJVL6OUGwYZWO5pDqe2OrDVcA@mail.gmail.com>

On Tue, May 17, 2011 at 3:03 PM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:
> 2011-05-17 12:40:36.061473 [ERR] sofia_glue.c:3449 AUDIO RTP REPORTS ERROR:
> [Bind Error!]

Sounds like the IP/port is in use already.  Can you pastebin your sofia conf?

Best,
Gabe


From gabe at gundy.org  Wed May 18 10:02:13 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 18 May 2011 00:02:13 -0600
Subject: [Freeswitch-users] Clustering/ storing registration server IP
	in DB
In-Reply-To: <BANLkTikrARQNCvQJnwFdoSAMkL_1+AXhiw@mail.gmail.com>
References: <BANLkTi=mqLgdOFVQLgWkXO3EZObzFEpt7A@mail.gmail.com>
	<BANLkTikrARQNCvQJnwFdoSAMkL_1+AXhiw@mail.gmail.com>
Message-ID: <BANLkTim03Aw8qFNxvYmrY_vFcuPXpe8GAQ@mail.gmail.com>

On Tue, May 17, 2011 at 4:06 PM, Steven Ayre <steveayre at gmail.com> wrote:
> You can move the core db and set a dsn for mod_sofia pointing to a
> ODBC database. Registrations will then be stored in that database,
> providing you are on latest git.


Ditto.

Also, I have no idea what you're trying to do, but you can watch for
custom events when an SIP UA registers too.  That might be of interest
depending on your application.

Best,
Gabe


From anton.vazir at gmail.com  Wed May 18 10:07:41 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 18 May 2011 11:07:41 +0500
Subject: [Freeswitch-users] ESL Bridge
In-Reply-To: <BANLkTinrvRJ5ciSbqDrW0wpG0Sn4F=PvDw@mail.gmail.com>
References: <BANLkTi=bE_hRyrb13yCNW6eGZykLSM+Ypg@mail.gmail.com>
	<BANLkTimqzkLw5ygRVazyWN159XDfc=6usw@mail.gmail.com>
	<BANLkTikqvb5+DMa=Ygc-_e4HA6kCjaL0cQ@mail.gmail.com>
	<BANLkTinQ3jSZTSVqpsC8E8i=6WPmFM6s4w@mail.gmail.com>
	<BANLkTi=HL2e3Qb26bJ9zR11hfWiZz5Qi6g@mail.gmail.com>
	<BANLkTinf894JH=JPV8tJgwK32pUs8J=-Gg@mail.gmail.com>
	<BANLkTinrvRJ5ciSbqDrW0wpG0Sn4F=PvDw@mail.gmail.com>
Message-ID: <BANLkTikAUghgUJs-7eNCM+q1_zK+o=qi8w@mail.gmail.com>

I've been unsuccessful in trying to get use of SetAsyncExecute and
SetEventLock to temporary switch to sync, while in outbound async full
mode. System behaves as is. and finally I've decided to stick with
async paradigm, and store events in spool, and than analyse regardless
of the order. Maybe I was doing something wrong, but more
test/analysis definitely required there...

2011/5/17 Antonio Teixeira <eagle.antonio at gmail.com>:
> Hello Anthony.
>
> Forking A Call :
>
> bgapi('originate',"{strategy=108,step=1,origination_caller_id_number=XXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s,position=%s}sofia/external/%s at XXXXXX
> 4998 XML default" % (self.channel_uuid,position,'XXXXXX'))
>
> Means running this command Against The Outbound Socket and that goes well.
>
> The Problem with async is that i have a multi-level IVR system not a
> generally simple ( press 1 for sales , press 2 for support ) type of IVR.
> This means lots of monitoring on the events to get it right , and some APPs
> fail to send events, lack of documentation ,etc.
>
> Anyway i was able to solve this problem :
>
> 1- Process ESL and transfer the call to a extension&ValetPark and call
> disconnect() on the ESL connection , now open a new connection and issue
> uuid_bridge , all goes well connection with ESL is no longer in the middle
> :D
>
> I have Tried to do Park from inside the ESL even declaring
> setAsyncExecute('1') does not execute ValetPark in async it will just fail.
> If i don't use setAsyncExecute? the call parks and ESL processing stops , so
> i just bypassed that using the XML dialplan.
>
> I know this is not the recommended way , but will have to do for now ,
> anyway i will try to review this situation and will see what i can do with
> the async mode.
>
> If you want me to do any special testing feel free to drop me an e-mail off
> list.
>
> Regards
> A/T
>
> 2011/5/17 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> You should explain what "forking a call" means. This does not exist in
>> our terminology.
>> My advice is to start thinking about what you need to do to make it
>> work vs how you want it to work.
>>
>> You MUST use async mode for what you are trying to do. ?The reason the
>> call won't bridge in sync mode is because you are asking the channel
>> to do 2 things at once and it can only do 1 because sync mode means to
>> block on every command.
>>
>> If you are still stuck:
>>
>> Please pastebin a console log on debug level at
>> http://pastebin.freeswitch.org
>>
>>
>>
>> On Tue, May 17, 2011 at 4:43 AM, Antonio Teixeira
>> <eagle.antonio at gmail.com> wrote:
>> > Hello Anthony.
>> >
>> > Thanks for your input i have added the channel variable and still the
>> > same :
>> >
>> > I bridge the call using uuid_bridge and after this LEG B is
>> > disconnected.
>> > LEG A is still alive with due to a time.sleep python (not freeswitch
>> > sleep)
>> > If i do bridge with a time.sleep() on LEG B the two calls are alive but
>> > no
>> > transit of audio.
>> >
>> >
>> > I also tried intercept but still no luck :\
>> >
>> > Regards
>> > A/T
>> >
>> >
>> >
>> >
>> > 2011/5/16 Anthony Minessale <anthony.minessale at gmail.com>
>> >>
>> >> set the channel variable park_after_bridge=true on all the channels so
>> >> they go back to park again when the call ends.
>> >>
>> >>
>> >> On Mon, May 16, 2011 at 9:16 AM, Antonio Teixeira
>> >> <eagle.antonio at gmail.com> wrote:
>> >> > Hello Anton.
>> >> >
>> >> > The problem is the following? :
>> >> >
>> >> > I receive an Inbound Call Leg A.
>> >> > I say something using TTS and place it on Park , grab the channel
>> >> > uuid.
>> >> >
>> >> > Fork another call pass the uuid as argument.
>> >> >
>> >> > As soon as Leg B picks up , TTS fires UP asks the user a digit and
>> >> > after
>> >> > we
>> >> > have got the digit .
>> >> >
>> >> > I send uuid_bridge mycurrentchanel + argument uuid from channel A
>> >> >
>> >> > And i got +OK great :D
>> >> >
>> >> > The problem , Park is now broken and the call is terminated with last
>> >> > dialplan extension executed or simply the call is terminated HANGUP.
>> >> >
>> >> > If i do some time.sleep(40) the calls don't end but they aren't merge
>> >> > i
>> >> > can't see no RTP flowing trough.
>> >> >
>> >> > Hope you guys can help me out.
>> >> >
>> >> > Regards
>> >> > A/T
>> >> >
>> >> >
>> >> >
>> >> > 2011/5/16 Anton VG <anton.vazir at gmail.com>
>> >> >>
>> >> >> I do it with originate &park - all works, i bridge them by
>> >> >> uuid_bridge
>> >> >> successfully
>> >> >>
>> >> >> 2011/5/16 Antonio Teixeira <eagle.antonio at gmail.com>:
>> >> >> > Good Morning.
>> >> >> >
>> >> >> > Here i continue my fight against ESL :)
>> >> >> > On Leg A I'm forking a call that will become Leg B after a short
>> >> >> > IVR.
>> >> >> >
>> >> >> > Anyway
>> >> >> > Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers
>> >> >> > ,
>> >> >> > Presses
>> >> >> > 1 and is merged by uuid_bridge
>> >> >> > Then...
>> >> >> > Both calls are disconnected with Executed Last Dialplan Command .
>> >> >> >
>> >> >> > I have tried everything from freezing the IVR with sleep(in python
>> >> >> > and
>> >> >> > using
>> >> >> > the freeswitch sleep).
>> >> >> >
>> >> >> > But i'm unable to merge both calls.
>> >> >> >
>> >> >> > As anyone tried something similar.
>> >> >> >
>> >> >> > Regards
>> >> >> > A/T
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> 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
>
>


From gabe at gundy.org  Wed May 18 10:08:20 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 18 May 2011 00:08:20 -0600
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and
	exec_after_bridge_app
In-Reply-To: <Pine.LNX.4.64.1105180620370.10537@debian>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
Message-ID: <BANLkTikuPa-DrKgkGx6jb3h+HTbYxE8-9g@mail.gmail.com>

On Tue, May 17, 2011 at 10:40 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Hi all, i gernerate a callback with the originate command.
> I do forward all calls to my cellphone with a bridge
> (loppback in and out).

<SNIP>

> ?Is there a way to solve the problem?

I don't know, but are you sure you really need to use loopbacks for
this?  While they are helpful in some situations, it seems like they
get used to solve problems that should be solved in other ways.

BTW, am I the only one who thinks this?


Best,
Gabe


From vivid333 at 163.com  Wed May 18 11:06:12 2011
From: vivid333 at 163.com (vivid)
Date: Wed, 18 May 2011 15:06:12 +0800
Subject: [Freeswitch-users] FreeSwitch Compile Error
Message-ID: <4DD36FE4.2000205@163.com>

Can anyone tell me  how to solve this problem?


./configure


make
///////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
Compiling src/switch_apr.c ...
Compiling src/switch_buffer.c ...
Compiling src/switch_caller.c ...
Compiling src/switch_channel.c ...
Compiling src/switch_console.c ...
Compiling src/switch_mprintf.c ...
Compiling src/switch_core_media_bug.c ...
Compiling src/switch_core_timer.c ...
Compiling src/switch_core_asr.c ...
Compiling src/switch_core_event_hook.c ...
Compiling src/switch_core_speech.c ...
Compiling src/switch_core_memory.c ...
Compiling src/switch_core_codec.c ...
Compiling src/switch_core_file.c ...
Compiling src/switch_core_hash.c ...
Compiling src/switch_core_sqldb.c ...
cc1: warnings being treated as errors
src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle':
src/switch_core_sqldb.c:314:47: error: comparison between 
'switch_odbc_status_t' and 'enum <anonymous>'
src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real':
src/switch_core_sqldb.c:389:90: error: comparison between 
'switch_status_t' and 'enum <anonymous>'
make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Error 1
make: *** [all] Error 2




From admin at blindi.net  Wed May 18 11:08:15 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 18 May 2011 09:08:15 +0200 (CEST)
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and
 exec_after_bridge_app
In-Reply-To: <BANLkTikuPa-DrKgkGx6jb3h+HTbYxE8-9g@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
	<BANLkTikuPa-DrKgkGx6jb3h+HTbYxE8-9g@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105180848300.2494@debian>

  Hi Gabriel, I hope to be able to explain the situation.
I have a webcallback. People can enter the phonenummer on my website and 
fs call back.
I send a originatcommand
example:
orginate loopback/0891111disa 0891122
Fs call leg a also calls to the specified person.
    After the contract extension is made I will be called back on my 
cellphone legb.
This is my callforwarding.
  The problem is that both lines are connected, it is not possible to 
give an Dtmf command to fs, to transfer the call to a extenion or so on.
    I hope I have explained well.

---------------
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 admin at vci.net  Wed May 18 05:18:30 2011
From: admin at vci.net (Bill Dunn - VCI Internet Services)
Date: Tue, 17 May 2011 20:18:30 -0500
Subject: [Freeswitch-users] AC Problem?
Message-ID: <D2E1A1B8635B45C1A5039C20EBF389F1@bildun>

I have Freeswitch 1.0 running. I had been using it successfully until I took 
a VOIP phone home and couldn't get it to register. I tried different VOIP 
accounts on the box but they all do the same thing - 401 Unauthorized. I 
suspected it may be ACL related but not sure. I have another VOIP phone at 
another employee's house and it works properly without any modification to 
the ACLs. How can I determine if it is an ACL problem or something else? I 
don't get enough debugging output at the cli with "/log 7" and I've tried 
watching the parkcets with wireshark. Neither give me enough info to 
determine if the problem is an ACL or something else.

 Bill Dunn 



From anton.vazir at gmail.com  Wed May 18 11:59:53 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 18 May 2011 12:59:53 +0500
Subject: [Freeswitch-users] Clustering/ storing registration server IP
	in DB
In-Reply-To: <BANLkTim03Aw8qFNxvYmrY_vFcuPXpe8GAQ@mail.gmail.com>
References: <BANLkTi=mqLgdOFVQLgWkXO3EZObzFEpt7A@mail.gmail.com>
	<BANLkTikrARQNCvQJnwFdoSAMkL_1+AXhiw@mail.gmail.com>
	<BANLkTim03Aw8qFNxvYmrY_vFcuPXpe8GAQ@mail.gmail.com>
Message-ID: <BANLkTinEVjQ0x1E+c2y7245+AUoTCNZAVQ@mail.gmail.com>

Yea, the custom events I mean.

What I'm trying to do:
Suppose there are several FS servers (cluster) - managed via ESL and
xml_curl for users,
user (SIP) register to one of the servers in round-robin fashion. So
to call from one user to another, if given users are registered on
different FS servers, before cunstructing the 'originate', I have to
determine, which server is responsible for the given user, and place
call there as to gateway, or if the user found on the same server,
call him as internal user. Any better way?

My billing done in Postgres, with quite a few schemes, so overall
structure is quite complex, so I'd like to manage the registration
process, rather to give access to a public schema for FS instances and
than adopt FS-created it data to my needs.

PS: I'm on git, which is a week old, seems it supports the given.

2011/5/18 Gabriel Gunderson <gabe at gundy.org>:
> On Tue, May 17, 2011 at 4:06 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> You can move the core db and set a dsn for mod_sofia pointing to a
>> ODBC database. Registrations will then be stored in that database,
>> providing you are on latest git.
>
>
> Ditto.
>
> Also, I have no idea what you're trying to do, but you can watch for
> custom events when an SIP UA registers too. ?That might be of interest
> depending on your application.
>
> Best,
> 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 anton.vazir at gmail.com  Wed May 18 12:01:43 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 18 May 2011 13:01:43 +0500
Subject: [Freeswitch-users] CDR event
In-Reply-To: <BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
References: <4DCE3654.7080002@tagnet.ru>
	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
Message-ID: <BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>

I suppose he's asking for asterisk analog of fork_cdr

2011/5/17 Michael Collins <msc at freeswitch.org>:
> You want your Lua script to *send* an event? It most certainly can fire
> events:
> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
> I suppose the question is: what do you want to have happen when you fire an
> event? What will receive this event?
> -MC
>
> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>> Hello!
>>
>> ? ? May I fire custom cdr event for a LUA script? I've read
>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>> Would You please point me to proper documentation or give an example?
>>
>> --
>> 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 list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From zetruger at gmail.com  Wed May 18 12:20:01 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Wed, 18 May 2011 12:20:01 +0400
Subject: [Freeswitch-users] how to handling REFER from remote subscriber on
	the FreeSWICH?
Message-ID: <BANLkTikD2Uu9D7qVhwvXXZjQLnLMe1_Vdw@mail.gmail.com>

how to handling REFER from remote subscriber on the FreeSWICH?


From boris at tagnet.ru  Wed May 18 12:34:39 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 18 May 2011 14:34:39 +0600
Subject: [Freeswitch-users] CDR event
In-Reply-To: <BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
Message-ID: <4DD3849F.40409@tagnet.ru>

Hello!

I don't know what is asterisk fork_cdr :) I want to bridge two calls and 
get 2 cdr records for A leg: one the standard cdr, and another with 
variables I need.

> I suppose he's asking for asterisk analog of fork_cdr
>
> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>> You want your Lua script to *send* an event? It most certainly can fire
>> events:
>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>> I suppose the question is: what do you want to have happen when you fire an
>> event? What will receive this event?
>> -MC
>>
>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> Hello!
>>>
>>>      May I fire custom cdr event for a LUA script? I've read
>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>>> Would You please point me to proper documentation or give an example?
>>>
>>> --
>>> 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 list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Wed May 18 12:34:33 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 18 May 2011 10:34:33 +0200
Subject: [Freeswitch-users] AC Problem?
In-Reply-To: <D2E1A1B8635B45C1A5039C20EBF389F1@bildun>
References: <D2E1A1B8635B45C1A5039C20EBF389F1@bildun>
Message-ID: <AFD5158E-4090-4340-8BFA-6E19A48AA0D9@ipeva.fr>

Bill,

The 401 Unauthorized is normal.
The client is supposed then to send back another REGISTER, with digest auth this time.
if you never added any specific ACL for auth, there is no reason the behavior to be different depending on the location of the phone.
But, I suppose a nasty router with a buggy SIP ALG could mess it up.
Also, check the packets coming from the not-working phone.
What does the phone do after receiving the 401 ?
Does it send back immediately another REGISTER which again is answered with a 401 ?
That could be because of a phone with a buggy digest auth implementation.

What brand/modem of phone is that ?

Are both phones the same ?

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

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

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




Le 18/05/2011 ? 03:18, Bill Dunn - VCI Internet Services a ?crit :

> I have Freeswitch 1.0 running. I had been using it successfully until I took 
> a VOIP phone home and couldn't get it to register. I tried different VOIP 
> accounts on the box but they all do the same thing - 401 Unauthorized. I 
> suspected it may be ACL related but not sure. I have another VOIP phone at 
> another employee's house and it works properly without any modification to 
> the ACLs. How can I determine if it is an ACL problem or something else? I 
> don't get enough debugging output at the cli with "/log 7" and I've tried 
> watching the parkcets with wireshark. Neither give me enough info to 
> determine if the problem is an ACL or something else.
> 
> Bill Dunn 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Wed May 18 12:37:00 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 18 May 2011 10:37:00 +0200
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD3849F.40409@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
	<4DD3849F.40409@tagnet.ru>
Message-ID: <B5266EE9-823F-4E74-BE80-95339B55F612@ipeva.fr>

Boris,

Can't you add the variables you need to the standard CDRs ?
Can't you use the XML CDRs, which already contain all available variables ?

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

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

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




Le 18/05/2011 ? 10:34, Boris Kovalenko a ?crit :

> Hello!
> 
> I don't know what is asterisk fork_cdr :) I want to bridge two calls and 
> get 2 cdr records for A leg: one the standard cdr, and another with 
> variables I need.
> 
>> I suppose he's asking for asterisk analog of fork_cdr
>> 
>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>> You want your Lua script to *send* an event? It most certainly can fire
>>> events:
>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>> I suppose the question is: what do you want to have happen when you fire an
>>> event? What will receive this event?
>>> -MC
>>> 
>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>>> Hello!
>>>> 
>>>>     May I fire custom cdr event for a LUA script? I've read
>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>>>> Would You please point me to proper documentation or give an example?
>>>> 
>>>> --
>>>> 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 list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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

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From steveayre at gmail.com  Wed May 18 12:51:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 18 May 2011 09:51:09 +0100
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD3849F.40409@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>
	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
	<4DD3849F.40409@tagnet.ru>
Message-ID: <BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>

There's no 'standard' CDR. The XML CDR will contain everything,
including all variables. If you mean the default mod_cdr_csv, you can
customise which variables are saved in its configuration.

-Steve



On 18 May 2011 09:34, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> I don't know what is asterisk fork_cdr :) I want to bridge two calls and
> get 2 cdr records for A leg: one the standard cdr, and another with
> variables I need.
>
>> I suppose he's asking for asterisk analog of fork_cdr
>>
>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>> You want your Lua script to *send* an event? It most certainly can fire
>>> events:
>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>> I suppose the question is: what do you want to have happen when you fire an
>>> event? What will receive this event?
>>> -MC
>>>
>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>>> Hello!
>>>>
>>>> ? ? ?May I fire custom cdr event for a LUA script? I've read
>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>>>> Would You please point me to proper documentation or give an example?
>>>>
>>>> --
>>>> 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 list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 gmaruzz at gmail.com  Wed May 18 13:50:15 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 18 May 2011 11:50:15 +0200
Subject: [Freeswitch-users] skype child accounts and FS
In-Reply-To: <8C410090745C40EB81929054AC7A70A1@e1705>
References: <8C410090745C40EB81929054AC7A70A1@e1705>
Message-ID: <BANLkTinrwntXg_Joq8SL+hU2eXZ9s+9dUQ@mail.gmail.com>

On Tue, May 17, 2011 at 10:43 PM, Madovsky <infos at madovsky.org> wrote:
> Is it possible to register multiple skype child accounts

Yes
Actually using the Skype Manager seems to be the preferred way in
corporate or multiline installation:
http://www.skype.com/intl/en/business/skype-manager

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From frankie.k.yiu at gmail.com  Wed May 18 15:26:19 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 18 May 2011 04:26:19 -0700
Subject: [Freeswitch-users] How to resolve AUDIO RTP REPORTS ERROR:
	[Bind Error!]
Message-ID: <BANLkTi=E_yp9gjfmhaS=HgRMUV=2UmUDnQ@mail.gmail.com>

Gabe,


Here are my conf files ( I have included sofia, external, and internal)





#####  *sofia.conf*
**
**
-<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/autoload_configs/sofia.conf.xml#>
<configuration name="*sofia.conf*" description="*sofia Endpoint*">
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/autoload_configs/sofia.conf.xml#>
<global_settings>
   <param name="*log-level*" value="*0*" />
- <!--

 <param name="auto-restart" value="false"/>

  -->
   <param name="*debug-presence*" value="*0*" />
  </global_settings>
- <!--

      The rabbit hole goes deep.  This includes all the
      profiles in the sip_profiles directory that is up
      one level from this directory.


  -->
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/autoload_configs/sofia.conf.xml#>
<profiles>
   <X-PRE-PROCESS cmd="*include*" data="*../sip_profiles/*.xml*" />
  </profiles>
  </configuration>


*#####  External*

-<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#>
<profile name="*external*">
 - <!--

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

  -->
- <!--

 This profile is only for outbound registrations to providers

  -->
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#>
<gateways>
   <X-PRE-PROCESS cmd="*include*" data="*external/*.xml*" />
  </gateways>
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#>
<aliases>
 - <!--


    <alias name="outbound"/>
    <alias name="nat"/>


  -->
  </aliases>
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#>
<domains>
   <domain name="*all*" alias="*false*" parse="*true*" />
  </domains>
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#>
<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="*rfc2833-pt*" value="*101*" />
   <param name="*sip-port*" value="*$${external_sip_port}*" />
   <param name="*dialplan*" value="*XML*" />
   <param name="*context*" value="*public*" />
   <param name="*dtmf-duration*" value="*2000*" />
   <param name="*inbound-codec-prefs*" value="*$${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"/>

  -->
- <!--

 This could be set to "passive"

  -->
   <param name="*local-network-acl*" value="*localnet.auto*" />
   <param name="*manage-presence*" value="*false*" />
- <!--

 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="*generous*" />
   <param name="*nonce-ttl*" value="*60*" />
   <param name="*auth-calls*" value="*false*" />
- <!--

	DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!


  -->
   <param name="*rtp-ip*" value="*$${local_ip_v4}*" />
   <param name="*sip-ip*" value="*$${local_ip_v4}*" />
   <param name="*ext-rtp-ip*" value="*$${external_rtp_ip}*" />
   <param name="*ext-sip-ip*" value="*$${external_sip_ip}*" />
   <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>


*######   Another file inside External folder*
-<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external/external_sip.xml#>
<include>
  <param name="*expire-seconds*" value="*60*" />
- <!--

/// do not register ///

  -->
   <param name="*register*" value="*true*" />
- <!--

 which transport to use for register

  -->
   <param name="*register-transport*" value="*tcp*" />
- <!--

How many seconds before a retry when a failure or timeout occurs

  -->
   <param name="*retry-seconds*" value="*30*" />
- <!--

Use the callerid of an inbound call in the from field on outbound
calls via this gateway

  -->
- <!--

<param name="caller-id-in-from" value="false"/>

  -->
- <!--

extra sip params to send in the contact

  -->
   <param name="*contact-params*" value="*tport=tcp*" />
- <!--

send an options ping every x seconds, failure will unregister and/or
mark it down

  -->
   <param name="*ping*" value="*25*" />
- <!--

</gateway>

  -->
  </include>


*#####  Internal*

-<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#>
<profile name="*internal*">
 - <!--

      This is a sofia sip profile/user agent.  This will service
exactly one ip and port.
      In FreeSWITCH you can run multiple sip user agents on their own
ip and port.

      When you hear someone say "sofia profile" this is what they are
talking about.


  -->
- <!--

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

  -->
- <!--

aliases are other names that will work as a valid profile name for this profile

  -->
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#>
<aliases>
 - <!--

    <alias name="default"/>


  -->
  </aliases>
- <!--

 Outbound Registrations

  -->
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#>
<gateways>
   <X-PRE-PROCESS cmd="*include*" data="*internal/*.xml*" />
  </gateways>
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#>
<domains>
 - <!--

 indicator to parse the directory for domains with parse="true" to get gateways

  -->
- <!--

<domain name="$${domain}" parse="true"/>

  -->
- <!--

 indicator to parse the directory for domains with parse="true" to get
gateways and alias every domain to this profile

  -->
- <!--

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

  -->
   <domain name="*all*" alias="*true*" parse="*false*" />
  </domains>
 -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#>
<settings>
 - <!--

	When calls are in no media this will bring them back to media
	when you press the hold button.


  -->
- <!--

<param name="media-option" value="resume-media-on-hold"/>

  -->
- <!--

	This will allow a call after an attended transfer go back to
	 bypass media after an attended transfer.


  -->
- <!--

<param name="media-option" value="bypass-media-after-att-xfer"/>

  -->
- <!--

 <param name="user-agent-string" value="FreeSWITCH Rocks!"/>

  -->
   <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*" />
- <!--


        Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
        responding. These options allow you to enable and control a watchdog
        on the Sofia SIP stack so that if it stops responding for the
        specified number of milliseconds, it will cause FreeSWITCH to shut
        down immediately. This is useful if you run in an HA environment and
        need to ensure automated recovery from such a condition. Note that if
        your server is idle a lot, the watchdog may fire due to not receiving
        any SIP messages. Thus, if you expect your system to be idle, you
        should leave the watchdog disabled. It can be toggled on and off
        through the FreeSWITCH CLI either on an individual profile basis or
        globally for all profiles. So, if you run in an HA environment with a
        master and slave, you should use the CLI to make sure the watchdog is
        only enabled on the master.


  -->
   <param name="*watchdog-enabled*" value="*no*" />
   <param name="*watchdog-step-timeout*" value="*30000*" />
   <param name="*watchdog-event-timeout*" value="*30000*" />
   <param name="*log-auth-failures*" value="*true*" />
   <param name="*forward-unsolicited-mwi-notify*" value="*false*" />
   <param name="*context*" value="*public*" />
   <param name="*rfc2833-pt*" value="*101*" />
- <!--

 port to bind to for sip traffic

  -->
   <param name="*sip-port*" value="*$${internal_sip_port}*" />
   <param name="*dialplan*" value="*XML*" />
   <param name="*dtmf-duration*" value="*2000*" />
   <param name="*inbound-codec-prefs*" value="*$${global_codec_prefs}*" />
   <param name="*outbound-codec-prefs*" value="*$${global_codec_prefs}*" />
   <param name="*rtp-timer-name*" value="*soft*" />
- <!--

 ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES

  -->
   <param name="*rtp-ip*" value="*$${local_ip_v4}*" />
- <!--

 ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES

  -->
   <param name="*sip-ip*" value="*$${local_ip_v4}*" />
   <param name="*hold-music*" value="*$${hold_music}*" />
   <param name="*apply-nat-acl*" value="*nat.auto*" />
- <!--

 (default true) set to false if you do not wish to have called party
info in 1XX responses

  -->
- <!--

 <param name="cid-in-1xx" value="false"/>

  -->
- <!--

 extended info parsing

  -->
- <!--

 <param name="extended-info-parsing" value="true"/>

  -->
- <!--

<param name="aggressive-nat-detection" value="true"/>

  -->
- <!--

	There are known issues (asserts and segfaults) when 100rel is enabled.
	It is not recommended to enable 100rel at this time.


  -->
- <!--

<param name="enable-100rel" value="true"/>

  -->
- <!--

 Enable Compact SIP headers.

  -->
- <!--

<param name="enable-compact-headers" value="true"/>

  -->
- <!--

	enable/disable session timers


  -->
- <!--

<param name="enable-timer" value="false"/>

  -->
- <!--

<param name="minimum-session-expires" value="120"/>

  -->
   <param name="*apply-inbound-acl*" value="*domains*" />
- <!--

	This defines your local network, by default we detect your local network
	and create this localnet.auto ACL for this.


  -->
   <param name="*local-network-acl*" value="*localnet.auto*" />
- <!--

<param name="apply-register-acl" value="domains"/>

  -->
- <!--

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

  -->
- <!--

 'true' means every time 'first-only' means on the first register

  -->
- <!--

<param name="send-message-query-on-register" value="true"/>

  -->
- <!--

 'true' means every time 'first-only' means on the first register

  -->
- <!--

<param name="send-presence-on-register" value="first-only"/>

  -->
- <!--

 Caller-ID type (choose one, can be overridden by inbound call type
and/or sip_cid_type channel variable

  -->
- <!--

 Remote-Party-ID header

  -->
- <!--

<param name="caller-id-type" value="rpid"/>

  -->
- <!--

 P-*-Identity family of headers

  -->
- <!--

<param name="caller-id-type" value="pid"/>

  -->
- <!--

 neither one

  -->
- <!--

<param name="caller-id-type" value="none"/>

  -->
   <param name="*record-path*" value="*$${recordings_dir}*" />
   <param name="*record-template*" value="*
${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav*" />
- <!--

enable to use presence

  -->
   <param name="*manage-presence*" value="*true*" />
- <!--

 send a presence probe on each register to query devices to send
presence instead of sending presence with less info

  -->
- <!--

<param name="presence-probe-on-register" value="true"/>

  -->
- <!--

<param name="manage-shared-appearance" value="true"/>

  -->
- <!--

 used to share presence info across sofia profiles

  -->
- <!--

 Name of the db to use for this profile

  -->
- <!--

<param name="dbname" value="share_presence"/>

  -->
   <param name="*presence-hosts*" value="*$${domain},$${local_ip_v4}*" />
- <!--

 *************************************************

  -->
- <!--

 This setting is for AAL2 bitpacking on G726

  -->
- <!--

 <param name="bitpacking" value="aal2"/>

  -->
- <!--

max number of open dialogs in proceeding

  -->
- <!--

<param name="max-proceeding" value="1000"/>

  -->
- <!--

session timers for all call to expire after the specified seconds

  -->
- <!--

<param name="session-timeout" value="1800"/>

  -->
- <!--

 Can be 'true' or 'contact'

  -->
- <!--

<param name="multiple-registrations" value="contact"/>

  -->
- <!--

set to 'greedy' if you want your codec list to take precedence

  -->
   <param name="*inbound-codec-negotiation*" value="*generous*" />
- <!--

 if you want to send any special bind params of your own

  -->
- <!--

<param name="bind-params" value="transport=udp"/>

  -->
- <!--

<param name="unregister-on-options-fail" value="true"/>

  -->
- <!--

 TLS: disabled by default, set to "true" to enable

  -->
   <param name="*tls*" value="*$${internal_ssl_enable}*" />
- <!--

 additional bind parameters for TLS

  -->
   <param name="*tls-bind-params*" value="*transport=tls*" />
- <!--

 Port to listen on for TLS requests. (5061 will be used if unspecified)

  -->
   <param name="*tls-sip-port*" value="*$${internal_tls_port}*" />
- <!--

 Location of the agent.pem and cafile.pem ssl certificates (needed for
TLS server)

  -->
   <param name="*tls-cert-dir*" value="*$${internal_ssl_dir}*" />
- <!--

 TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work
with TLSv1

  -->
   <param name="*tls-version*" value="*$${sip_tls_version}*" />
- <!--

 turn on auto-flush during bridge (skip timer sleep when the socket
already has data)
	 (reduces delay on latent connections default true, must be disabled
explicitly)

  -->
- <!--

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

  -->
- <!--

If you don't want to pass through timestamps from 1 RTP call to
another (on a per call basis with rtp_rewrite_timestamps chanvar)

  -->
- <!--

<param name="rtp-rewrite-timestamps" value="true"/>

  -->
- <!--

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

  -->
- <!--

If you have ODBC support and a working dsn you can use it instead of SQLite

  -->
- <!--

<param name="odbc-dsn" value="dsn:user:pass"/>

  -->
- <!--

Uncomment to set all inbound calls to no media mode

  -->
- <!--

<param name="inbound-bypass-media" value="true"/>

  -->
- <!--

Uncomment to set all inbound calls to proxy media mode

  -->
- <!--

<param name="inbound-proxy-media" value="true"/>

  -->
- <!--

Uncomment to let calls hit the dialplan *before* you decide if the codec is ok

  -->
- <!--

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

  -->
- <!--

 this lets anything register

  -->
- <!--

  comment the next line and uncomment one or both of the other 2 lines
for call authentication

  -->
- <!--

 <param name="accept-blind-reg" value="true"/>

  -->
- <!--

 accept any authentication without actually checking (not a good
feature for most people)

  -->
- <!--

 <param name="accept-blind-auth" value="true"/>

  -->
- <!--

 suppress CNG on this profile or per call with the 'suppress_cng' variable

  -->
- <!--

 <param name="suppress-cng" value="true"/>

  -->
- <!--

TTL for nonce in sip auth

  -->
   <param name="*nonce-ttl*" value="*60*" />
- <!--

Uncomment if you want to force the outbound leg of a bridge to only
offer the codec
	that the originator is using

  -->
- <!--

<param name="disable-transcoding" value="true"/>

  -->
- <!--

 Handle 302 Redirect in the dialplan

  -->
- <!--

<param name="manual-redirect" value="true"/>

  -->
- <!--

 Disable Transfer

  -->
- <!--

<param name="disable-transfer" value="true"/>

  -->
- <!--

 Disable Register

  -->
- <!--

<param name="disable-register" value="true"/>

  -->
- <!--

 Used for when phones respond to a challenged ACK with method INVITE
in the hash

  -->
- <!--

<param name="NDLB-broken-auth-hash" value="true"/>

  -->
- <!--

 add a ;received="<ip>:<port>" to the contact when replying to
register for nat handling

  -->
- <!--

<param name="NDLB-received-in-nat-reg-contact" value="true"/>

  -->
   <param name="*auth-calls*" value="*$${internal_auth_calls}*" />
- <!--

 Force the user and auth-user to match.

  -->
   <param name="*inbound-reg-force-matching-username*" value="*true*" />
- <!--

 on authed calls, authenticate *all* the packets not just invite

  -->
   <param name="*auth-all-packets*" value="*false*" />
- <!--

 external_sip_ip
      Used as the public IP address for SDP.
      Can be an one of:
           ip address            - "12.34.56.78"
           a stun server lookup  - "stun:stun.server.com"
           a DNS name            - "host:host.server.com"
           auto                  - Use guessed ip.
           auto-nat              - Use ip learned from NAT-PMP or UPNP


  -->
   <param name="*ext-rtp-ip*" value="*auto-nat*" />
   <param name="*ext-sip-ip*" value="*auto-nat*" />
- <!--

 rtp inactivity timeout

  -->
   <param name="*rtp-timeout-sec*" value="*300*" />
   <param name="*rtp-hold-timeout-sec*" value="*1800*" />
- <!--

 VAD choose one (out is a good choice);

  -->
- <!--

 <param name="vad" value="in"/>

  -->
- <!--

 <param name="vad" value="out"/>

  -->
- <!--

 <param name="vad" value="both"/>

  -->
- <!--

<param name="alias" value="sip:10.0.1.251:5555"/>

  -->
- <!--

	These are enabled to make the default config work better out of the box.
	If you need more than ONE domain you'll need to not use these options.



  -->
- <!--

all inbound reg will look in this domain for the users

  -->
   <param name="*force-register-domain*" value="*$${domain}*" />
- <!--

force the domain in subscriptions to this value

  -->
   <param name="*force-subscription-domain*" value="*$${domain}*" />
- <!--

all inbound reg will stored in the db using this domain

  -->
   <param name="*force-register-db-domain*" value="*$${domain}*" />
- <!--

<param name="delete-subs-on-register" value="false"/>

  -->
- <!--

 enable rtcp on every channel also can be done per leg basis with
rtcp_audio_interval_msec variable set to passthru to pass it across a
call

  -->
- <!--

<param name="rtcp-audio-interval-msec" value="5000"/>

  -->
- <!--

<param name="rtcp-video-interval-msec" value="5000"/>

  -->
- <!--

force suscription expires to a lower value than requested

  -->
- <!--

<param name="force-subscription-expires" value="60"/>

  -->
- <!--

 disable register and transfer which may be undesirable in a public switch

  -->
- <!--

<param name="disable-transfer" value="true"/>

  -->
- <!--

<param name="disable-register" value="true"/>

  -->
- <!--


	 enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
	 right away, proxy waits until the call has been answered then sends accepts


  -->
- <!--

<param name="enable-3pcc" value="true"/>

  -->
- <!--

 use at your own risk or if you know what this does.

  -->
- <!--

<param name="NDLB-force-rport" value="true"/>

  -->
- <!--

	Choose the realm challenge key. Default is auto_to if not set.
	
	auto_from  - uses the from field as the value for the sip realm.
	auto_to    - uses the to field as the value for the sip realm.
	<anyvalue> - you can input any value to use for the sip realm.

	If you want URL dialing to work you'll want to set this to auto_from.
	
	If you use any other value besides auto_to or auto_from you'll loose
	the ability to do multiple domains.
	
	Note: comment out to restore the behavior before 2008-09-29



  -->
   <param name="*challenge-realm*" value="*auto_from*" />
- <!--

<param name="disable-rtp-auto-adjust" value="true"/>

  -->
- <!--

 on inbound calls make the uuid of the session equal to the sip call
id of that call

  -->
- <!--

<param name="inbound-use-callid-as-uuid" value="true"/>

  -->
- <!--

 on outbound calls set the callid to match the uuid of the session

  -->
- <!--

<param name="outbound-use-uuid-as-callid" value="true"/>

  -->
- <!--

 set to false disable this feature

  -->
- <!--

<param name="rtp-autofix-timing" value="false"/>

  -->
- <!--

 set this param to false if your gateway for some reason hates X-
headers that it is supposed to ignore

  -->
- <!--

<param name="pass-callee-id" value="false"/>

  -->
- <!--

 clear clears them all or supply the name to add or the name prefixed
with ~ to remove
	 valid values:

	 clear
	 CISCO_SKIP_MARK_BIT_2833
	 SONUS_SEND_INVALID_TIMESTAMP_2833



  -->
- <!--

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

  -->
- <!--

 the following can be used as workaround with bogus SRV/NAPTR records

  -->
- <!--

<param name="disable-srv" value="false" />

  -->
- <!--

<param name="disable-naptr" value="false" />

  -->
- <!--

 The following can be used to fine-tune timers within sofia's transport layer
		 Those settings are for advanced users and can safely be left as-is

  -->
- <!--

 Initial retransmission interval (in milliseconds).
		Set the T1 retransmission interval used by the SIP transaction engine.
		The T1 is the initial duration used by request retransmission timers
A and E (UDP) as well as response retransmission timer G.

  -->
- <!--

 <param name="timer-T1" value="500" />

  -->
- <!--

  Transaction timeout (defaults to T1 * 64).
		Set the T1x64 timeout value used by the SIP transaction engine.
		The T1x64 is duration used for timers B, F, H, and J (UDP) by the
SIP transaction engine.
		The timeout value T1x64 can be adjusted separately from the initial
retransmission interval T1.

  -->
- <!--

 <param name="timer-T1X64" value="32000" />

  -->
- <!--

 Maximum retransmission interval (in milliseconds).
		Set the maximum retransmission interval used by the SIP transaction engine.
		The T2 is the maximum duration used for the timers E (UDP) and G by
the SIP transaction engine.
		Note that the timer A is not capped by T2. Retransmission interval
of INVITE requests grows exponentially
		until the timer B fires.

  -->
- <!--

 <param name="timer-T2" value="4000" />

  -->
- <!--

		Transaction lifetime (in milliseconds).
		Set the lifetime for completed transactions used by the SIP
transaction engine.
		A completed transaction is kept around for the duration of T4 in
order to catch late responses.
		The T4 is the maximum duration for the messages to stay in the
network and the duration of SIP timer K.

  -->
- <!--

 <param name="timer-T4" value="4000" />

  -->
- <!--

 Turn on a jitterbuffer for every call

  -->
- <!--

 <param name="auto-jitterbuffer-msec" value="60"/>

  -->
  </settings>
  </profile>



Thanks,
Frankie




>
> ---------- Forwarded message ----------
> From: Gabriel Gunderson <gabe at gundy.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 17 May 2011 23:56:08 -0600
> Subject: Re: [Freeswitch-users] How to resolve AUDIO RTP REPORTS ERROR:
> [Bind Error!]
> On Tue, May 17, 2011 at 3:03 PM, Frankie Yiu <frankie.k.yiu at gmail.com>
> wrote:
> > 2011-05-17 12:40:36.061473 [ERR] sofia_glue.c:3449 AUDIO RTP REPORTS
> ERROR:
> > [Bind Error!]
>
> Sounds like the IP/port is in use already.  Can you pastebin your sofia
> conf?
>
> Best,
> Gabe
>
>
>
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From frankie.k.yiu at gmail.com  Wed May 18 15:32:18 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 18 May 2011 04:32:18 -0700
Subject: [Freeswitch-users] Getting Error message " [ERR] switch_cpp.cpp:48
	Cannot queue any more events....."
Message-ID: <BANLkTinvEo_6fwC9yd9pjAiNi7y7_4AkEA@mail.gmail.com>

Hi there,

When I made 200 calls to the other freeswitch box in different network, I
got a lot of the following error messages:
 [ERR] switch_cpp.cpp:48 Cannot queue any more events.....

FYI--I have a 1 second delay for every 25 calls.

What did I do wrong and how can I solve this?  Because of these errors, I
did not get any Hangup completion event.

Thanks,
Frankie
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From avi at avimarcus.net  Wed May 18 15:45:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 18 May 2011 14:45:11 +0300
Subject: [Freeswitch-users] Clustering/ storing registration server IP
	in DB
In-Reply-To: <BANLkTinEVjQ0x1E+c2y7245+AUoTCNZAVQ@mail.gmail.com>
References: <BANLkTi=mqLgdOFVQLgWkXO3EZObzFEpt7A@mail.gmail.com>
	<BANLkTikrARQNCvQJnwFdoSAMkL_1+AXhiw@mail.gmail.com>
	<BANLkTim03Aw8qFNxvYmrY_vFcuPXpe8GAQ@mail.gmail.com>
	<BANLkTinEVjQ0x1E+c2y7245+AUoTCNZAVQ@mail.gmail.com>
Message-ID: <BANLkTi=Wv=w59wJJNXzJW8TnNMwax=hFdw@mail.gmail.com>

A master-master type of sql setup will share the registrations along with
the path to reach them. It might be easiest to do this in the sql database
rather than ESL.

-Avi

On Wed, May 18, 2011 at 10:59 AM, Anton VG <anton.vazir at gmail.com> wrote:

> Yea, the custom events I mean.
>
> What I'm trying to do:
> Suppose there are several FS servers (cluster) - managed via ESL and
> xml_curl for users,
> user (SIP) register to one of the servers in round-robin fashion. So
> to call from one user to another, if given users are registered on
> different FS servers, before cunstructing the 'originate', I have to
> determine, which server is responsible for the given user, and place
> call there as to gateway, or if the user found on the same server,
> call him as internal user. Any better way?
>
> My billing done in Postgres, with quite a few schemes, so overall
> structure is quite complex, so I'd like to manage the registration
> process, rather to give access to a public schema for FS instances and
> than adopt FS-created it data to my needs.
>
> PS: I'm on git, which is a week old, seems it supports the given.
>
> 2011/5/18 Gabriel Gunderson <gabe at gundy.org>:
> > On Tue, May 17, 2011 at 4:06 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> You can move the core db and set a dsn for mod_sofia pointing to a
> >> ODBC database. Registrations will then be stored in that database,
> >> providing you are on latest git.
> >
> >
> > Ditto.
> >
> > Also, I have no idea what you're trying to do, but you can watch for
> > custom events when an SIP UA registers too.  That might be of interest
> > depending on your application.
> >
> > Best,
> > 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
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed May 18 18:50:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 18 May 2011 09:50:55 -0500
Subject: [Freeswitch-users] Getting Error message " [ERR]
 switch_cpp.cpp:48 Cannot queue any more events....."
In-Reply-To: <BANLkTinvEo_6fwC9yd9pjAiNi7y7_4AkEA@mail.gmail.com>
References: <BANLkTinvEo_6fwC9yd9pjAiNi7y7_4AkEA@mail.gmail.com>
Message-ID: <BANLkTikzGX_-y9A0w12vpN_x4MrR4q+mNA@mail.gmail.com>

I am not sure why you think its a strange error.  A strange error to
me would be something like:

 [ERR] switch_cpp.cpp:48 There are 10 types of people, those who
understand binary and those who don't

or

 [ERR] switch_cpp.cpp:48 Are you in the house alone?


The error you posted is telling you that you have consumed too many
events in your event consumer object.
So you have created an event consumer then stopped regularly calling
the pop method until it filled up.
Now its trying to tell you that it has reached its max.





On Wed, May 18, 2011 at 6:32 AM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:
> Hi there,
>
> When I made 200 calls to the other freeswitch box in different network, I
> got a lot of the following error messages:
> ?[ERR] switch_cpp.cpp:48 Cannot queue any more events.....
>
> FYI--I have a 1 second delay for every 25?calls.
>
> What did I do wrong and how can I solve this?? Because of these errors, I
> did not get any Hangup completion event.
> Thanks,
> Frankie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From infos at madovsky.org  Wed May 18 18:53:49 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 18 May 2011 10:53:49 -0400
Subject: [Freeswitch-users] skype child accounts and FS
References: <8C410090745C40EB81929054AC7A70A1@e1705>
	<BANLkTinrwntXg_Joq8SL+hU2eXZ9s+9dUQ@mail.gmail.com>
Message-ID: <C6FAD86DFBDC4C63930279E6450D0E79@e1705>

Grazie mille giovanni ;)
I hope $MS won't acquire FS one day ;)...

----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, May 18, 2011 5:50 AM
Subject: Re: [Freeswitch-users] skype child accounts and FS


> On Tue, May 17, 2011 at 10:43 PM, Madovsky <infos at madovsky.org> wrote:
>> Is it possible to register multiple skype child accounts
> 
> Yes
> Actually using the Skype Manager seems to be the preferred way in
> corporate or multiline installation:
> http://www.skype.com/intl/en/business/skype-manager
> 
> -giovanni
> 
> -- 
> 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


From gmaruzz at gmail.com  Wed May 18 19:07:30 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 18 May 2011 17:07:30 +0200
Subject: [Freeswitch-users] Getting Error message " [ERR]
 switch_cpp.cpp:48 Cannot queue any more events....."
In-Reply-To: <BANLkTikzGX_-y9A0w12vpN_x4MrR4q+mNA@mail.gmail.com>
References: <BANLkTinvEo_6fwC9yd9pjAiNi7y7_4AkEA@mail.gmail.com>
	<BANLkTikzGX_-y9A0w12vpN_x4MrR4q+mNA@mail.gmail.com>
Message-ID: <BANLkTim-Qb8zYk6msg3vyHPr5gdhrHt7AQ@mail.gmail.com>

On Wed, May 18, 2011 at 4:50 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>
> ?[ERR] switch_cpp.cpp:48 Are you in the house alone?
>

:)))))))))))))))))))))))

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From curriegrad2004 at gmail.com  Wed May 18 19:34:12 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 18 May 2011 08:34:12 -0700
Subject: [Freeswitch-users] FreeSwitch Compile Error
In-Reply-To: <4DD36FE4.2000205@163.com>
References: <4DD36FE4.2000205@163.com>
Message-ID: <BANLkTimSJuEnJxbDbR3A9XfV2tB0v6GPZQ@mail.gmail.com>

Try removing the -Wall option on the sqlite makefile and see what happens.

On Wed, May 18, 2011 at 12:06 AM, vivid <vivid333 at 163.com> wrote:
> Can anyone tell me ?how to solve this problem?
>
>
> ./configure
>
>
> make
> ///////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
> Compiling src/switch_apr.c ...
> Compiling src/switch_buffer.c ...
> Compiling src/switch_caller.c ...
> Compiling src/switch_channel.c ...
> Compiling src/switch_console.c ...
> Compiling src/switch_mprintf.c ...
> Compiling src/switch_core_media_bug.c ...
> Compiling src/switch_core_timer.c ...
> Compiling src/switch_core_asr.c ...
> Compiling src/switch_core_event_hook.c ...
> Compiling src/switch_core_speech.c ...
> Compiling src/switch_core_memory.c ...
> Compiling src/switch_core_codec.c ...
> Compiling src/switch_core_file.c ...
> Compiling src/switch_core_hash.c ...
> Compiling src/switch_core_sqldb.c ...
> cc1: warnings being treated as errors
> src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle':
> src/switch_core_sqldb.c:314:47: error: comparison between
> 'switch_odbc_status_t' and 'enum <anonymous>'
> src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real':
> src/switch_core_sqldb.c:389:90: error: comparison between
> 'switch_status_t' and 'enum <anonymous>'
> make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Error 1
> make: *** [all] Error 2
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed May 18 19:46:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 18 May 2011 08:46:47 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTimYj3BueYFKnUuAVz7BwUE17F6r=Q@mail.gmail.com>

Hello all!

Today's agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_18

It's a bit light but there are a few things to talk about. Please bring your
questions for discussion.

Thanks,
MC
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From robert.hadley at teotech.com  Wed May 18 19:45:37 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 18 May 2011 08:45:37 -0700
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app
	and	exec_after_bridge_app
In-Reply-To: <Pine.LNX.4.64.1105180620370.10537@debian>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC09E9@VA3DIAXVS351.RED001.local>


If you bypass_media then FS is out of the call and bind_meta_app will not work.


-----Original Message-----
From: Thomas Hoellriegel [mailto:admin at blindi.net] 
Sent: Tuesday, May 17, 2011 9:41 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and exec_after_bridge_app

Hi all, i gernerate a callback with the originate command.
I do forward all calls to my cellphone with a bridge (loppback in and out).
If I.m not the option add: bypass_media=true in the originate string,
  fs aborts the connection to my cellphone.
if i add this option, the connection is esstablisht, bind_meta_app is not working.
I need this feature to transfer callers from my cellphone to others.
exec_after_bridge_app
  also does not work unfortunately.
  Is there a way to solve the problem?

thank you


---------------
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 sid.kshatriya at gmail.com  Wed May 18 20:02:43 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 18 May 2011 21:32:43 +0530
Subject: [Freeswitch-users] Something as powerful as voip.ms?
Message-ID: <BANLkTimhZoaZMPKdxvpX3hezrMnGKR-2=w@mail.gmail.com>

Ok guys, a few days ago I had asked people for a good round voip provider
and people pitched in with a lot of names. Among all of them, I checked out
voip.ms and I am totally impressed by the hundreds of options in their login
interface. Thats what really set it apart from others -- for me. Many other
providers seemed to have similar pricing but I haven't found any website
that gives that kind of login interface options + information transparency
that voip.ms does.

I really want to sign up for them. However there is a catch. They don't
accept Paypal / Credit cards from India!! This is unusual and strange as
I've never had any problems before. The only other route to pay voip.ms is
western union in which I need to pay in $100 increments or Bank transfers in
$500 increments. Needless to say, I'd like to avoid all that.

I am considering abandoning voip.ms but just can't seem to find any other
provider which are "equivalent" in terms of flexibility and power.

Any help? Suggestions?

I'm looking for inbound DID services only (to run an IVR)
* One US number non-toll free 5+ channels
* One US toll free number
* (Maybe) UK DID / UK toll free DID

Thanks,

Sidharth

-- 
Sidharth Kshatriya
www.sidk.info
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From sid.kshatriya at gmail.com  Wed May 18 20:05:01 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 18 May 2011 21:35:01 +0530
Subject: [Freeswitch-users] Something as powerful as voip.ms?
In-Reply-To: <BANLkTimhZoaZMPKdxvpX3hezrMnGKR-2=w@mail.gmail.com>
References: <BANLkTimhZoaZMPKdxvpX3hezrMnGKR-2=w@mail.gmail.com>
Message-ID: <BANLkTimZ7AM7YEZpNkSpuZrM6hhdUSS6+g@mail.gmail.com>

*Correction:* "good *all* *round* voip provider" and not "good round voip
provider"

On Wed, May 18, 2011 at 9:32 PM, Sidharth Kshatriya <sid.kshatriya at gmail.com
> wrote:

> Ok guys, a few days ago I had asked people for a good round voip provider
> and people pitched in with a lot of names. Among all of them, I checked out
> voip.ms and I am totally impressed by the hundreds of options in their
> login interface. Thats what really set it apart from others -- for me. Many
> other providers seemed to have similar pricing but I haven't found any
> website that gives that kind of login interface options + information
> transparency that voip.ms does.
>
> I really want to sign up for them. However there is a catch. They don't
> accept Paypal / Credit cards from India!! This is unusual and strange as
> I've never had any problems before. The only other route to pay voip.ms is
> western union in which I need to pay in $100 increments or Bank transfers in
> $500 increments. Needless to say, I'd like to avoid all that.
>
> I am considering abandoning voip.ms but just can't seem to find any other
> provider which are "equivalent" in terms of flexibility and power.
>
> Any help? Suggestions?
>
> I'm looking for inbound DID services only (to run an IVR)
> * One US number non-toll free 5+ channels
> * One US toll free number
> * (Maybe) UK DID / UK toll free DID
>
> Thanks,
>
> Sidharth
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From rhuddleston at gmail.com  Wed May 18 20:06:43 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Wed, 18 May 2011 12:06:43 -0400
Subject: [Freeswitch-users] Something as powerful as voip.ms?
In-Reply-To: <BANLkTimhZoaZMPKdxvpX3hezrMnGKR-2=w@mail.gmail.com>
References: <BANLkTimhZoaZMPKdxvpX3hezrMnGKR-2=w@mail.gmail.com>
Message-ID: <1b2a01cc1575$95941a90$c0bc4fb0$@com>

This is par for the course in my opinion. I personally like them too and
have nothing but good experiences with them.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sidharth
Kshatriya
Sent: Wednesday, May 18, 2011 12:03 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Something as powerful as voip.ms?

 

Ok guys, a few days ago I had asked people for a good round voip provider
and people pitched in with a lot of names. Among all of them, I checked out
voip.ms and I am totally impressed by the hundreds of options in their login
interface. Thats what really set it apart from others -- for me. Many other
providers seemed to have similar pricing but I haven't found any website
that gives that kind of login interface options + information transparency
that voip.ms does.

I really want to sign up for them. However there is a catch. They don't
accept Paypal / Credit cards from India!! This is unusual and strange as
I've never had any problems before. The only other route to pay voip.ms is
western union in which I need to pay in $100 increments or Bank transfers in
$500 increments. Needless to say, I'd like to avoid all that.

I am considering abandoning voip.ms but just can't seem to find any other
provider which are "equivalent" in terms of flexibility and power.

Any help? Suggestions?

I'm looking for inbound DID services only (to run an IVR)
* One US number non-toll free 5+ channels
* One US toll free number
* (Maybe) UK DID / UK toll free DID

Thanks,

Sidharth

-- 
Sidharth Kshatriya
www.sidk.info

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From admin at blindi.net  Wed May 18 21:05:09 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 18 May 2011 19:05:09 +0200 (CEST)
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and
 exec_after_bridge_app
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC09E9@VA3DIAXVS351.RED001.local>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC09E9@VA3DIAXVS351.RED001.local>
Message-ID: <Pine.LNX.4.64.1105181857550.7113@debian>

Hi robert,
  Is there another way to connect without bypass_media build?
Is this a known bug in fs?
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 chrisg.lists at gmail.com  Wed May 18 17:05:46 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Wed, 18 May 2011 15:05:46 +0200
Subject: [Freeswitch-users] mod_lcr
In-Reply-To: <BANLkTikAU2TesJwPWWjjLoN_rtZpJqfXGg@mail.gmail.com>
References: <BANLkTikBf7xTR=UEhf-NTwyhOBY0QyXMiQ@mail.gmail.com>
	<BANLkTikAU2TesJwPWWjjLoN_rtZpJqfXGg@mail.gmail.com>
Message-ID: <BANLkTik9N56Tpbn078iObvBwLK4gD0ShkQ@mail.gmail.com>

Thanks Nandy, have no idea how I overlooked that after three hours of debugging.


On Tue, May 17, 2011 at 2:58 AM, Nandy Dagondon <gcd at i.ph> wrote:
> u hv to enclose the destination number with () so its value will be assigned
> to $1
> ? <condition field="destination_number" expression="^(0112341234)$">
>
>
> On Mon, May 16, 2011 at 11:14 PM, Chris Graham <chrisg.lists at gmail.com>
> wrote:
>>
>> Hi All,
>> I am having an issue with mod_lcr, when I query from the CLI it
>> returns the expected values as per:
>>
>> freeswitch at internal> lcr 0112341234 peak
>> ?| Digit Match | Carrier | Rate ? ? | Codec | CID Regexp | Limit |
>> Dialstring
>> ? ? ? ?|
>> ?| 011 ? ? ? ? | Telkom ?| 0.32000 ?| ? ? ? | ? ? ? ? ? ?| ? ? ? |
>>
>> [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
>> |
>> freeswitch at internal> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:1687
>> data passed to lcr is [0112341234 peak]
>> 2011-05-16 17:06:19.847934 [WARNING] mod_lcr.c:1731 Using default CID
>> [18005551212]
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1]
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:771 0112341234 doesn't
>> appear to be a NANP number
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:852 intra routing
>> [state:0 lata:0] so rate field is [rate]
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:872 we have an event
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:896 SQL: ?SELECT l.digits
>> AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS
>> lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
>> lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
>> lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
>> cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
>> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>> (0112341234, 011234123, 01123412, 0112341, 011234, 01123, 0112, 011,
>> 01, 0) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY
>> digits DESC, ?rate, ?quality DESC, ?reliability DESC, rand();
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
>>
>> [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:664 Adding Telkom to head of
>> list
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
>>
>> [lcr_carrier=Neotel,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:682 Ignoring Duplicate
>> route for termination point (sofia/gateway/pp-ast-trunk-01/:)
>> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring
>>
>> [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234
>>
>>
>> When calling from the dialplan, it fails with:
>>
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1563 intrastate channel
>> var is [undef]
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1566 Select routes based
>> on interstate rates
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1585 LCR Lookup on $1
>> using profile peak
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1]
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:771 $1 doesn't appear to
>> be a NANP number
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:852 intra routing
>> [state:0 lata:0] so rate field is [rate]
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:857 we have a session
>> 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:896 SQL: ?SELECT l.digits
>> AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS
>> lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
>> lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
>> lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
>> cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
>> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>> (1) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY
>> digits DESC, ?rate, ?quality DESC, ?reliability DESC, rand();
>> EXECUTE sofia/internal/1000 at 192.168.16.136 bridge()
>> 2011-05-16 17:08:33.100261 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/1000 at 192.168.16.136 has executed the last dialplan
>> instruction, hanging up.
>>
>>
>> The xml entry called:
>>
>> ?? ? ? ?<extension name="Test">
>> ?? ? ? ? ? ? ? ?<condition field="destination_number"
>> expression="^0112341234$">
>> ?? ? ? ? ? ? ? ? ? ? ? ?<action application="lcr" data="$1 peak"/>
>> ?? ? ? ? ? ? ? ? ? ? ? ?<action application="bridge"
>> data="${lcr_auto_route}"/>
>> ?? ? ? ? ? ? ? ?</condition>
>> ?? ? ? ?</extension>
>>
>>
>> My custom query:
>>
>> ?? ? ? ?<profile name="peak">
>> ?? ? ? ? ? ? ? ?<param name="custom_sql" value="
>> SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name,
>> l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS
>> lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS
>> lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
>> cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON
>> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>> (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>> date_start AND date_end ORDER BY digits DESC, ?rate, ?quality DESC,
>> reliability DESC, rand();
>> "/>
>> ?? ? ? ?</profile>
>>
>> Thanks in advance,
>> Chris
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From admin at vci.net  Wed May 18 17:16:22 2011
From: admin at vci.net (Bill Dunn - VCI Internet Services)
Date: Wed, 18 May 2011 08:16:22 -0500
Subject: [Freeswitch-users] AC Problem?
References: <D2E1A1B8635B45C1A5039C20EBF389F1@bildun>
	<AFD5158E-4090-4340-8BFA-6E19A48AA0D9@ipeva.fr>
Message-ID: <D88FF8807C314C538F5BA7B655216C78@bildun>

The router is an old AT&T named router. I don't know who actually makes it. 
I'll have to research that.

The phone doesn't send back a 2nd REGISTER request in response to the 401.

I am now convinced the problem is somehow related to the router. I tried the 
same PAP2T behind a Cisco 3640 and it worked flawlessly. I also tried a 
different PAP2T purchased at the same time behind a Pardyne DSL modem/router 
and it worked perfectly. I also tried an X-Lite softphone and it did the 
same 401 errors. The router that seems to be the problem doesn't have any 
further firmware upgrades so I may have to replace it.

170.048156 98.93.145.xxx -> 207.162.18x.x SIP Request: REGISTER 
sip:207.162.18x.x
170.048512 207.162.18x.x -> 98.93.145.xxx SIP Status: 401 Unauthorized    (0 
bindings)
174.047965 98.93.145.x -> 207.162.18x.x SIP Request: REGISTER 
sip:207.162.18x.x
174.048332 207.162.18x.x -> 98.93.145.xxx SIP Status: 401 Unauthorized    (0 
bindings)


 Bill Dunn




----- Original Message ----- 
From: David Ponzone
To: FreeSWITCH Users Help
Sent: Wednesday, May 18, 2011 3:34 AM
Subject: Re: [Freeswitch-users] AC Problem?


Bill,


The 401 Unauthorized is normal.
The client is supposed then to send back another REGISTER, with digest auth 
this time.
if you never added any specific ACL for auth, there is no reason the 
behavior to be different depending on the location of the phone.
But, I suppose a nasty router with a buggy SIP ALG could mess it up.
Also, check the packets coming from the not-working phone.
What does the phone do after receiving the 401 ?
Does it send back immediately another REGISTER which again is answered with 
a 401 ?
That could be because of a phone with a buggy digest auth implementation.


What brand/modem of phone is that ?

Are both phones the same ?


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com
Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? 
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion 
non autoris?e est interdite. Tout message ?lectronique est susceptible 
d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il 
a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce 
message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 18/05/2011 ? 03:18, Bill Dunn - VCI Internet Services a ?crit :
I have Freeswitch 1.0 running. I had been using it successfully until I took
a VOIP phone home and couldn't get it to register. I tried different VOIP
accounts on the box but they all do the same thing - 401 Unauthorized. I
suspected it may be ACL related but not sure. I have another VOIP phone at
another employee's house and it works properly without any modification to
the ACLs. How can I determine if it is an ACL problem or something else? I
don't get enough debugging output at the cli with "/log 7" and I've tried
watching the parkcets with wireshark. Neither give me enough info to
determine if the problem is an ACL or something else.

Bill Dunn



From admin at vci.net  Wed May 18 18:17:16 2011
From: admin at vci.net (Bill Dunn - VCI Internet Services)
Date: Wed, 18 May 2011 09:17:16 -0500
Subject: [Freeswitch-users] vm_list sends Notify to phone
Message-ID: <762C266FAFA449F8A9571E80424DB4A5@bildun>

I am using the API. When I issue a "vm_list phone#@domain" command 
Freeswitch sends a Notify packet to the phone. My Polycom phones give me a 
audible alert each time they receive one of these. Is there a way to stop 
Freeswitch from sending a Notify packet to the phone each time a "vm_list" 
command is sent to the API?


 Bill Dunn 



From Nabble at slickdeals.endjunk.com  Wed May 18 04:56:53 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 17 May 2011 17:56:53 -0700 (PDT)
Subject: [Freeswitch-users] Is Anyone Using Google TTS?
In-Reply-To: <BANLkTin9OVDcjw5JkEz_yktT7v06U-mXiA@mail.gmail.com>
References: <BANLkTikZPPiJL+csehSwwwrNM85HcBsGHw@mail.gmail.com>
	<BANLkTinSR5JzXXJ0jj9NkE7aw-nSuoUBpg@mail.gmail.com>
	<BANLkTikz+S1PERxr4gQNTXAORQtxtFiRvQ@mail.gmail.com>
	<BANLkTimR+9=wmZd5REfT+1mpbCjD53uwTg@mail.gmail.com>
	<DE77F089BB49408C9C928FCEEECB5C81@e1705>
	<BANLkTinKn_97R9Y_=nCRrUFzKjJ_nY4Amg@mail.gmail.com>
	<BANLkTin9OVDcjw5JkEz_yktT7v06U-mXiA@mail.gmail.com>
Message-ID: <1305680212995-6375930.post@n2.nabble.com>


A E [Gmail] wrote:
> Maybe we'll keep the usage of Google TTS just for playing with it and
> occasional translations to use as a benchmark coz the quality *is* damn
> good.
Agree. That's what I configured my FS to use it for educational fun.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Is-Anyone-Using-Google-TTS-tp6325535p6375930.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Wed May 18 17:16:27 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 18 May 2011 06:16:27 -0700 (PDT)
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
In-Reply-To: <l2xa44688a1005051716wda417209sc62554678b4c8e6e@mail.gmail.com>
References: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
	<s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>
	<o2ya44688a1005051658q204f4507w2e9b118e4289e9ab@mail.gmail.com>
	<t2n191c3a031005051706za41ed9a6vfdb11c45ac4f9d57@mail.gmail.com>
	<l2xa44688a1005051716wda417209sc62554678b4c8e6e@mail.gmail.com>
Message-ID: <1305724587011-6377764.post@n2.nabble.com>


Woody Dickson wrote:
> I am using 1.0.6 for the ipk that I am building.
Try update to FS v1.0.7.

BTW, I am just curious what hardware is it with a 2GB RAM do you have to
host your FS.

-----
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-segfault-on-openwrt-tp5008807p6377764.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Wed May 18 17:50:39 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 18 May 2011 06:50:39 -0700 (PDT)
Subject: [Freeswitch-users] Voicemail macro error in log
In-Reply-To: <assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>
References: <assp.01182d9298.92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
	<BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>
	<assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>
Message-ID: <1305726639462-6377886.post@n2.nabble.com>

You are using the default voicemail_ack macro on lang/en/vm/sounds.xml,
right? If so, it will probably be some bugs that have been fixed on the
latest git version. I tried your dialplan on my FS  Version 1.0.head
(git-9795dd2 2011-03-26 11-07-34 -0500) and did not encounter the issue.
Perhaps, you will need to upgrade to FS v1.0.7

-----
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/Voicemail-macro-error-in-log-tp6373688p6377886.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sipmaillist at gmail.com  Wed May 18 09:44:32 2011
From: sipmaillist at gmail.com (Jakson Kalsson)
Date: Wed, 18 May 2011 13:44:32 +0800
Subject: [Freeswitch-users] ZRTP of freeswitch.
Message-ID: <BANLkTimx2OzttB_KVq=aWGaNsJWfZ9Oygg@mail.gmail.com>

Does freeswitch support ZRTP?

Thansk

-- 
havesoftware, Inc.
http://www.havesoftware.com


Jakson Kalsson
Senior Programmer
jakkalsoon at havesoftware.com
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From tang.du at hotmail.com  Wed May 18 16:50:53 2011
From: tang.du at hotmail.com (tangdu)
Date: Wed, 18 May 2011 05:50:53 -0700 (PDT)
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <4AD5AE3E.9070206@gmx.net>
References: <4AD5AE3E.9070206@gmx.net>
Message-ID: <1305723053176-6377655.post@n2.nabble.com>

Now, I had the same problem. Who can help me?
I use FS version 1.0.7.
Thank you. 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-tp3821994p6377655.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ttobin at ubq.thrupoint.net  Tue May 17 18:52:18 2011
From: ttobin at ubq.thrupoint.net (Tim Tobin)
Date: Tue, 17 May 2011 15:52:18 +0100
Subject: [Freeswitch-users] G.729A licencing validation error
Message-ID: <BANLkTi=42gOrfQqvKdsT4-4R-refv1D=Eg@mail.gmail.com>

Hi,

I've just purchased a license for the G.729 codec module. I've uploaded
fsg729-194-installer to my freeswitch machine and have executed it. When i
try to validate i get the following response:

Failed with error code 6


Does anybody know why this might be?

Cheers,

--------------------
Note: The information contained in this message may be privileged and confidential 
and protected from disclosure. If the reader of this message is not the intended 
recipient, or an employee or agent responsible for delivering this message to the 
intended recipient, you are hereby notified that any dissemination, distribution or 
copying of this communication is strictly prohibited. If you have received this 
communication in error, please notify us immediately by replying to the message and 
deleting it from your computer. Thank you. ThruPoint, Inc. - U5MJHZGXCFFDOYV
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From jcasale at activenetwerx.com  Wed May 18 21:17:17 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 18 May 2011 17:17:17 +0000
Subject: [Freeswitch-users] Voicemail macro error in log
In-Reply-To: <1305726639462-6377886.post@n2.nabble.com>
References: <92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
	<BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>
	<assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>
	<1305726639462-6377886.post@n2.nabble.com>
Message-ID: <assp.01195fbe59.92097A6A775D5147B1078E3F15430B9252AD67@prato.activenetwerx.local>

>You are using the default voicemail_ack macro on lang/en/vm/sounds.xml,
>right? If so, it will probably be some bugs that have been fixed on the
>latest git version. I tried your dialplan on my FS  Version 1.0.head
>(git-9795dd2 2011-03-26 11-07-34 -0500) and did not encounter the issue.
>Perhaps, you will need to upgrade to FS v1.0.7

Right,
Problem is they are OpenWRT packages for the ar71xx platform. I haven't built
packages for OpenWRT and given all that I followed with respect to FS packages
specifically, I can't say I am brave enough to try.

I recall you running OpenWRT, would you be willing to provide some guidance?


Thanks for following up!
jlc


From gabe at gundy.org  Wed May 18 21:41:34 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 18 May 2011 11:41:34 -0600
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and
	exec_after_bridge_app
In-Reply-To: <Pine.LNX.4.64.1105181857550.7113@debian>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC09E9@VA3DIAXVS351.RED001.local>
	<Pine.LNX.4.64.1105181857550.7113@debian>
Message-ID: <BANLkTin82WFw2Fir=2tVQ+DJEW7GTkD6JA@mail.gmail.com>

On May 18, 2011 11:06 AM, "Thomas Hoellriegel" <admin at blindi.net> wrote:
>
> Hi robert,
>  Is there another way to connect without bypass_media build?
> Is this a known bug in fs?
> thanks

It's wel> Hi robert,
> Is there another way to connect without bypass_media build?
> Is this a known bug in fs?
> 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
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From gabe at gundy.org  Wed May 18 21:47:01 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 18 May 2011 11:47:01 -0600
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and
	exec_after_bridge_app
In-Reply-To: <BANLkTin82WFw2Fir=2tVQ+DJEW7GTkD6JA@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC09E9@VA3DIAXVS351.RED001.local>
	<Pine.LNX.4.64.1105181857550.7113@debian>
	<BANLkTin82WFw2Fir=2tVQ+DJEW7GTkD6JA@mail.gmail.com>
Message-ID: <BANLkTimE4ehe5tOLwNvGgLG4cL2g-5YF-A@mail.gmail.com>

On Wed, May 18, 2011 at 11:41 AM, Gabriel Gunderson <gabe at gundy.org> wrote:
> On May 18, 2011 11:06 AM, "Thomas Hoellriegel" <admin at blindi.net> wrote:
>>
>> Hi robert,
>> ?Is there another way to connect without bypass_media build?
>> Is this a known bug in fs?
>> thanks
>
> It's wel> Hi robert,

Sorry about that... It was my Android phone getting me down :(


Let's try that again:

It's well known, but it's not a bug.  How can FreeSWITCH know if a
digit was pressed if it's not in the media path?

Best,
Gabe


From Nabble at slickdeals.endjunk.com  Wed May 18 21:50:28 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 18 May 2011 10:50:28 -0700 (PDT)
Subject: [Freeswitch-users] Voicemail macro error in log
In-Reply-To: <assp.01195fbe59.92097A6A775D5147B1078E3F15430B9252AD67@prato.activenetwerx.local>
References: <assp.01182d9298.92097A6A775D5147B1078E3F15430B9252424D@prato.activenetwerx.local>
	<BANLkTimMWeH+Pw2wOP6pEmHvN1dFDVv_ew@mail.gmail.com>
	<assp.011834193d.92097A6A775D5147B1078E3F15430B925249AB@prato.activenetwerx.local>
	<1305726639462-6377886.post@n2.nabble.com>
	<assp.01195fbe59.92097A6A775D5147B1078E3F15430B9252AD67@prato.activenetwerx.local>
Message-ID: <1305741028307-6378788.post@n2.nabble.com>


Joseph L. Casale wrote:
> Right,
> Problem is they are OpenWRT packages for the ar71xx platform. I haven't
> built
> packages for OpenWRT and given all that I followed with respect to FS
> packages
> specifically, I can't say I am brave enough to try.
If you are using any Linux distribution with development packages installed,
you can simply (not necessarily easy) compile your own firmware to include
FS v1.0.7. For further information on how to do this, you may want to visit
the  https://dev.openwrt.org/wiki/GetSource OpenWrt source repository
downloads . BTW, what hardware on AR71XX platform are you currently using to
run FS?


I recall you running OpenWRT, would you be willing to provide some guidance?
I reckon my signature (below) speaks very loud on this.

Providing help on OpenWRT in this mailing list isn't really a good idea if
you know what I mean. You will get better receptions and/or responses asking
your questions w.r.t OPenWRT on the  https://forum.openwrt.org/index.php
OpenWRT Forum . You can also try the #openwrt and/or #openwrt_devel IRC
channels on  http://freenode.net FreeNode . If your questions relate to FS,
I am sure experts here will be able to help you out in a time manner.

-----
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/Voicemail-macro-error-in-log-tp6373688p6378788.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed May 18 21:59:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 18 May 2011 18:59:24 +0100
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <1305723053176-6377655.post@n2.nabble.com>
References: <4AD5AE3E.9070206@gmx.net>
	<1305723053176-6377655.post@n2.nabble.com>
Message-ID: <BANLkTin_bOeRdb9q2BDpATWyeMXD3Tmdug@mail.gmail.com>

>
> I use FS version 1.0.7.


Which one? It's a nightly build of git head.

-Steve



On 18 May 2011 13:50, tangdu <tang.du at hotmail.com> wrote:
> Now, I had the same problem. Who can help me?
> I use FS version 1.0.7.
> Thank you.
>
>
>
> --
> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-tp3821994p6377655.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 steveayre at gmail.com  Wed May 18 22:00:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 18 May 2011 19:00:23 +0100
Subject: [Freeswitch-users] ZRTP of freeswitch.
In-Reply-To: <BANLkTimx2OzttB_KVq=aWGaNsJWfZ9Oygg@mail.gmail.com>
References: <BANLkTimx2OzttB_KVq=aWGaNsJWfZ9Oygg@mail.gmail.com>
Message-ID: <BANLkTikWZmrzSRHXduhDPiAFHqk3g868Fw@mail.gmail.com>

Yes.

http://wiki.freeswitch.org/wiki/ZRTP

-Steve


On 18 May 2011 06:44, Jakson Kalsson <sipmaillist at gmail.com> wrote:

> Does freeswitch support ZRTP?
>
> Thansk
>
> --
> havesoftware, Inc.
> http://www.havesoftware.com
>
>
> Jakson Kalsson
> Senior Programmer
> jakkalsoon at havesoftware.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From robert.hadley at teotech.com  Wed May 18 22:09:08 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 18 May 2011 11:09:08 -0700
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app
	and	exec_after_bridge_app
In-Reply-To: <BANLkTin82WFw2Fir=2tVQ+DJEW7GTkD6JA@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC09E9@VA3DIAXVS351.RED001.local>
	<Pine.LNX.4.64.1105181857550.7113@debian>
	<BANLkTin82WFw2Fir=2tVQ+DJEW7GTkD6JA@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC0B03@VA3DIAXVS351.RED001.local>

Hi,

You should be able to connect when bypass_media=false (the default).
Not aware of any FS bug in this situation.
There is also bypass_media PROXY mode, don't know if it would help in this case.

-Robert

From: Gabriel Gunderson [mailto:gabe at gundy.org]
Sent: Wednesday, May 18, 2011 10:42 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] problem bypass_media and bind_meta_app and exec_after_bridge_app


On May 18, 2011 11:06 AM, "Thomas Hoellriegel" <admin at blindi.net<mailto:admin at blindi.net>> wrote:
>
> Hi robert,
>  Is there another way to connect without bypass_media build?
> Is this a known bug in fs?
> 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
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From admin at vci.net  Wed May 18 23:58:20 2011
From: admin at vci.net (Bill Dunn - VCI Internet Services)
Date: Wed, 18 May 2011 14:58:20 -0500
Subject: [Freeswitch-users] vm_list sends Notify to phone
Message-ID: <71419FB5E78941B59A2BE50F375D48EE@bildun>

I found an answer to my problem at 
http://www.voip-info.org./wiki/view/Polycom+SoundPoint+IP+MWI+audio


 Bill Dunn



----- Original Message ----- 
From: Bill Dunn - VCI Internet Services
To: freeswitch-users at lists.freeswitch.org
Sent: Wednesday, May 18, 2011 9:17 AM
Subject: vm_list sends Notify to phone


I am using the API. When I issue a "vm_list phone#@domain" command
Freeswitch sends a Notify packet to the phone. My Polycom phones give me a
audible alert each time they receive one of these. Is there a way to stop
Freeswitch from sending a Notify packet to the phone each time a "vm_list"
command is sent to the API?


 Bill Dunn 



From admin at blindi.net  Thu May 19 02:45:47 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 19 May 2011 00:45:47 +0200 (CEST)
Subject: [Freeswitch-users] problem bypass_media and bind_meta_app and
 exec_after_bridge_app
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC0B03@VA3DIAXVS351.RED001.local>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180620370.10537@debian>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC09E9@VA3DIAXVS351.RED001.local>
	<Pine.LNX.4.64.1105181857550.7113@debian>
	<BANLkTin82WFw2Fir=2tVQ+DJEW7GTkD6JA@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BD8BC0B03@VA3DIAXVS351.RED001.local>
Message-ID: <Pine.LNX.4.64.1105190022560.9696@debian>


Hi Robert, now I begin to slowly clear.
  The problem will always exist as soon as I bridge 2 lines with 
originate.
Enable this option, but put through the call but no call screening is 
possible.
This can be a problem if one assumes only by forwarding the calls to 
outsidelines.
It is hoped that once a fix is.
  these problems may well have field sales staff.

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 Nabble at slickdeals.endjunk.com  Thu May 19 04:18:03 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 18 May 2011 17:18:03 -0700 (PDT)
Subject: [Freeswitch-users] Something as powerful as voip.ms?
In-Reply-To: <BANLkTimhZoaZMPKdxvpX3hezrMnGKR-2=w@mail.gmail.com>
References: <BANLkTimhZoaZMPKdxvpX3hezrMnGKR-2=w@mail.gmail.com>
Message-ID: <1305764283463-6380127.post@n2.nabble.com>


Sidharth Kshatriya wrote:
> Any help? Suggestions?
What about VoSPs on  http://www.dslreports.com/gbu The Good, The Bad, and
The Ugly  chart?

-----
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/Something-as-powerful-as-voip-ms-tp6378330p6380127.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Thu May 19 06:01:08 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 18 May 2011 19:01:08 -0700 (PDT)
Subject: [Freeswitch-users] Spoofed CID?
Message-ID: <1305770468329-6380373.post@n2.nabble.com>

Today, after my phone rang with a CID of (443)366-9222, I took a look at
/var/log/freeswitch/freeswitch.log file on my Seagate DockStar (excerpted
below). What caught my attention is the last line which shows two numbers,
namely 4433669222 and 4016488312. The question I have is if 4433669222 is a
spoofed number?

2011-05-18 12:54:03.258849 [NOTICE] switch_channel.c:812 New Channel
sofia/external/4433669222 at 64.154.
2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4744 Channel
sofia/external/4433669222 at 64.154.41.150 enteri
2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4755 Remote SDP:                                           
v=0^M                                                                                                 
o=root 241964608 241964608 IN IP4 64.154.41.150^M                                                     
s=Asterisk PBX 1.6.2.13^M                                                                             
c=IN IP4 64.154.41.150^M                                                                              
t=0 0^M                                                                                               
m=audio 10730 RTP/AVP 0 8 3 18 101^M                                                                  
a=rtpmap:0 PCMU/8000^M                                                                                
a=rtpmap:8 PCMA/8000^M                                                                                
a=rtpmap:3 GSM/8000^M                                                                                 
a=rtpmap:18 G729/8000^M                                                                               
a=fmtp:18 annexb=no^M                                                                                 
a=rtpmap:101 telephone-event/8000^M                                                                   
a=fmtp:101 0-16^M                                                                                     
a=ptime:20^M                                                                                          
                                                                                                      
2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221
2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221
2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:
2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:
2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:2760 Set Codec
sofia/external/4433669222 at 64.154.41.150
2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4751 Set 2833 dtmf send/recv
payload to 101           
2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
(sofia/external/4433669222 at 64.154.41.150) State Change

2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
(sofia/external/4433669222 at 64.154.41.150) State Change
2011-05-18 12:54:03.258849 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/4433669222 at 64
2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
(sofia/external/4433669222 at 64.154.4
2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
(sofia/external/4433669222 at 64.154.4
2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:84
sofia/external/4433669222 at 64.154.41.150 SOFIA INIT  
2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:124
(sofia/external/4433669222 at 64.154.41.150) State Cha
2011-05-18 12:54:03.265631 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/4433669222 at 64
2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
(sofia/external/4433669222 at 64.154.4
2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
(sofia/external/4433669222 at 64.154.4
2011-05-18 12:54:03.265631 [DEBUG] switch_channel.c:1668
(sofia/external/4433669222 at 64.154.41.150) Cal
2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:359
(sofia/external/4433669222 at 64.154.4
2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:147
sofia/external/4433669222 at 64.154.41.150 SOFIA ROUTI
2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:77
sofia/external/4433669222 at 64.154.41.
2011-05-18 12:54:03.265631 [INFO] mod_dialplan_xml.c:331 Processing
4433669222 <4433669222>->401648831


-----
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/Spoofed-CID-tp6380373p6380373.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jcasale at activenetwerx.com  Thu May 19 06:19:06 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 19 May 2011 02:19:06 +0000
Subject: [Freeswitch-users] Ptime and Aastra phone
Message-ID: <assp.012027e0a5.92097A6A775D5147B1078E3F15430B9252BF0D@prato.activenetwerx.local>

I have a few 6730i's that have codec choice set to All, as per the Aastra doc
ptime gets defaulted to 30 and won't accept 20 which seems to be what fs
is using. I am seeing the following lines in the log:
[WARNING] sofia_glue.c:213 Codec PCMU payload 0 added to sdp wanting ptime 20 but it's already 30 (PCMU:0:30), disabling ptime.

How does one correct this? This is on a recent trunk build.

Thanks,
jlc


From boris at tagnet.ru  Thu May 19 07:12:25 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 19 May 2011 09:12:25 +0600
Subject: [Freeswitch-users] CDR event
In-Reply-To: <BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>
	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>
Message-ID: <4DD48A99.4000700@tagnet.ru>

Hello!

     I know :) But with one call I get only one CDR record for A leg and 
one record for B leg. I need to get 2 records for A leg with two 
different sets of variables. Is this possible?

> There's no 'standard' CDR. The XML CDR will contain everything,
> including all variables. If you mean the default mod_cdr_csv, you can
> customise which variables are saved in its configuration.
>
> -Steve
>
>
>
> On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>> I don't know what is asterisk fork_cdr :) I want to bridge two calls and
>> get 2 cdr records for A leg: one the standard cdr, and another with
>> variables I need.
>>
>>> I suppose he's asking for asterisk analog of fork_cdr
>>>
>>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>>> You want your Lua script to *send* an event? It most certainly can fire
>>>> events:
>>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>>> I suppose the question is: what do you want to have happen when you fire an
>>>> event? What will receive this event?
>>>> -MC
>>>>
>>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>>>> Hello!
>>>>>
>>>>>       May I fire custom cdr event for a LUA script? I've read
>>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>>>>> Would You please point me to proper documentation or give an example?
>>>>>
>>>>> --
>>>>> 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 list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tang.du at hotmail.com  Thu May 19 08:27:57 2011
From: tang.du at hotmail.com (tangdu)
Date: Wed, 18 May 2011 21:27:57 -0700 (PDT)
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <BANLkTin_bOeRdb9q2BDpATWyeMXD3Tmdug@mail.gmail.com>
References: <4AD5AE3E.9070206@gmx.net>
	<1305723053176-6377655.post@n2.nabble.com>
	<BANLkTin_bOeRdb9q2BDpATWyeMXD3Tmdug@mail.gmail.com>
Message-ID: <1305779277804-6380608.post@n2.nabble.com>

I use FreeSWITCH Version 1.0.7 (hacked-20110418T002854Z)
The following scenario

FS-->TG-->PSTN FAX

|Time     |xxx.xxx.xxx.xxx                          |
|         |                   | xxx.xxx.xxx.xxx    |                   
|3.263    |         INVITE SDP ( g711A telephone-event CN)          |SIP
From: sip:TCAPI_User at xxx.xxx To:sip:12334 at xxx.xxx.xxx.xxx
|         |(5060)   ------------------>  (2080)   |
|3.323    |         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (2080)   |
|5.362    |         183 Session Progress SDP ( g711A)          |SIP Status
|         |(5060)   <------------------  (2080)   |
|5.372    |         183 Session Progress SDP ( g711A)          |SIP Status
|         |(5060)   <------------------  (2080)   |
|5.439    |         RTP (g711A)                   |RTP Num packets:577 
Duration:11.520s SSRC:0xDA48CD22
|         |(31962)  <------------------  (6276)   |
|5.697    |         RTP (g711A)                   |RTP Num packets:283 
Duration:5.639s SSRC:0x4896FE5
|         |(31962)  ------------------>  (6276)   |
|11.338   |         200 OK SDP ( g711A)           |SIP Status
|         |(5060)   <------------------  (2080)   |
|11.339   |         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (2080)   |
|11.357   |         RTP (g711A)                   |RTP Num packets:281 
Duration:5.620s SSRC:0x4896FE5
|         |(31962)  ------------------>  (6276)   |
|16.986   |         UPDATE SDP ( t38)             |SIP Request
|         |(5060)   <------------------  (2080)   |
|16.986   |         200 OK SDP ( g711A telephone-event CN)          |SIP
Status
|         |(5060)   ------------------>  (2080)   |
|53.000   |         BYE       |                   |SIP Request
|         |(5060)   ------------------>  (2080)   |
|53.064   |         200 OK    |                   |SIP Status
|         |(5060)   <------------------  (2080)   |

How should I set FS?

Thank you?


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-tp3821994p6380608.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Thu May 19 11:04:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 08:04:04 +0100
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD48A99.4000700@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>
	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
	<4DD3849F.40409@tagnet.ru>
	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>
	<4DD48A99.4000700@tagnet.ru>
Message-ID: <BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>

I don't understand why you would need this.

What would the difference between the variables be?

-Steve



On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     I know :) But with one call I get only one CDR record for A leg and
> one record for B leg. I need to get 2 records for A leg with two
> different sets of variables. Is this possible?
>
> > There's no 'standard' CDR. The XML CDR will contain everything,
> > including all variables. If you mean the default mod_cdr_csv, you can
> > customise which variables are saved in its configuration.
> >
> > -Steve
> >
> >
> >
> > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru>  wrote:
> >> Hello!
> >>
> >> I don't know what is asterisk fork_cdr :) I want to bridge two calls and
> >> get 2 cdr records for A leg: one the standard cdr, and another with
> >> variables I need.
> >>
> >>> I suppose he's asking for asterisk analog of fork_cdr
> >>>
> >>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
> >>>> You want your Lua script to *send* an event? It most certainly can
> fire
> >>>> events:
> >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
> >>>> I suppose the question is: what do you want to have happen when you
> fire an
> >>>> event? What will receive this event?
> >>>> -MC
> >>>>
> >>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>
>  wrote:
> >>>>> Hello!
> >>>>>
> >>>>>       May I fire custom cdr event for a LUA script? I've read
> >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr
> events.
> >>>>> Would You please point me to proper documentation or give an example?
> >>>>>
> >>>>> --
> >>>>> 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 list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
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From steveayre at gmail.com  Thu May 19 11:36:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 08:36:19 +0100
Subject: [Freeswitch-users] Spoofed CID?
In-Reply-To: <1305770468329-6380373.post@n2.nabble.com>
References: <1305770468329-6380373.post@n2.nabble.com>
Message-ID: <BANLkTikydmOJghbySDBe_v5TYanXUCNL=g@mail.gmail.com>

There's no way of knowing really.

You can send any CID you like in SIP. They might own 4433669222 in real
life, they might not. You only really get a guarantee it's genuine if it's
coming off the PSTN.

All I can tell you is 44336 is a valid UK phone number prefix (
http://en.wikipedia.org/wiki/03_numbers).

-Steve



On 19 May 2011 03:01, mazilo <Nabble at slickdeals.endjunk.com> wrote:

> Today, after my phone rang with a CID of (443)366-9222, I took a look at
> /var/log/freeswitch/freeswitch.log file on my Seagate DockStar (excerpted
> below). What caught my attention is the last line which shows two numbers,
> namely 4433669222 and 4016488312. The question I have is if 4433669222 is a
> spoofed number?
>
> 2011-05-18 12:54:03.258849 [NOTICE] switch_channel.c:812 New Channel
> sofia/external/4433669222 at 64.154.
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4744 Channel
> sofia/external/4433669222 at 64.154.41.150 enteri
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4755 Remote SDP:
> v=0^M
> o=root 241964608 241964608 IN IP4 64.154.41.150^M
> s=Asterisk PBX 1.6.2.13^M
> c=IN IP4 64.154.41.150^M
> t=0 0^M
> m=audio 10730 RTP/AVP 0 8 3 18 101^M
> a=rtpmap:0 PCMU/8000^M
> a=rtpmap:8 PCMA/8000^M
> a=rtpmap:3 GSM/8000^M
> a=rtpmap:18 G729/8000^M
> a=fmtp:18 annexb=no^M
> a=rtpmap:101 telephone-event/8000^M
> a=fmtp:101 0-16^M
> a=ptime:20^M
>
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G722:
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:2760 Set Codec
> sofia/external/4433669222 at 64.154.41.150
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4751 Set 2833 dtmf
> send/recv
> payload to 101
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
> (sofia/external/4433669222 at 64.154.41.150) State Change
>
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
> (sofia/external/4433669222 at 64.154.41.150) State Change
> 2011-05-18 12:54:03.258849 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/4433669222 at 64
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:84
> sofia/external/4433669222 at 64.154.41.150 SOFIA INIT
> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:124
> (sofia/external/4433669222 at 64.154.41.150) State Cha
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/4433669222 at 64
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] switch_channel.c:1668
> (sofia/external/4433669222 at 64.154.41.150) Cal
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:147
> sofia/external/4433669222 at 64.154.41.150 SOFIA ROUTI
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:77
> sofia/external/4433669222 at 64.154.41.
> 2011-05-18 12:54:03.265631 [INFO] mod_dialplan_xml.c:331 Processing
> 4433669222 <4433669222>->401648831
>
>
> -----
> 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/Spoofed-CID-tp6380373p6380373.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 boris at tagnet.ru  Thu May 19 11:54:51 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 19 May 2011 13:54:51 +0600
Subject: [Freeswitch-users] CDR event
In-Reply-To: <BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>
	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>
Message-ID: <4DD4CCCB.2080700@tagnet.ru>

Hello!

     It is the way I try to do manual redirect. May be I do it wrong, so 
may be somebody tell me another way. So... I use only A-leg accounting 
records. One of my users set the redirect from its phone to cellular 
phone (PAP2T device). So, when call is arrived from PBX
and redirected by user, in A-leg CDR record I see only _original_ 
information: pbx_caller_id_number and local_destination_number. And this 
is wrong for billing. With help of CDR I want to emulate two calls scenario:
First CDR record - PBX -> user number
Second CDR record - user number -> redirected number

> I don't understand why you would need this.
>
> What would the difference between the variables be?
>
> -Steve
>
>
>
> On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         I know :) But with one call I get only one CDR record for A
>     leg and
>     one record for B leg. I need to get 2 records for A leg with two
>     different sets of variables. Is this possible?
>
>     > There's no 'standard' CDR. The XML CDR will contain everything,
>     > including all variables. If you mean the default mod_cdr_csv,
>     you can
>     > customise which variables are saved in its configuration.
>     >
>     > -Steve
>     >
>     >
>     >
>     > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru
>     <mailto:boris at tagnet.ru>>  wrote:
>     >> Hello!
>     >>
>     >> I don't know what is asterisk fork_cdr :) I want to bridge two
>     calls and
>     >> get 2 cdr records for A leg: one the standard cdr, and another with
>     >> variables I need.
>     >>
>     >>> I suppose he's asking for asterisk analog of fork_cdr
>     >>>
>     >>> 2011/5/17 Michael Collins<msc at freeswitch.org
>     <mailto:msc at freeswitch.org>>:
>     >>>> You want your Lua script to *send* an event? It most
>     certainly can fire
>     >>>> events:
>     >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>     >>>> I suppose the question is: what do you want to have happen
>     when you fire an
>     >>>> event? What will receive this event?
>     >>>> -MC
>     >>>>
>     >>>> On Sat, May 14, 2011 at 12:59 AM, Boris
>     Kovalenko<boris at tagnet.ru <mailto:boris at tagnet.ru>>    wrote:
>     >>>>> Hello!
>     >>>>>
>     >>>>>       May I fire custom cdr event for a LUA script? I've read
>     >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found
>     cdr events.
>     >>>>> Would You please point me to proper documentation or give an
>     example?
>     >>>>>
>     >>>>> --
>     >>>>> 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
>     >>>>
>     >>>>
>     >>> _______________________________________________
>     >>> 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
>     >>
>     >> --
>     >> ? ?????????,
>     >>    ????? ?????????
>     >>    ??? "??????"
>     >>    ???. +7 (3435) 230001
>     >>    ???? +7 (3435) 230005
>     >>
>     >>
>     >> _______________________________________________
>     >> 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
>
>
>     --
>     ? ?????????,
>       ????? ?????????
>       ??? "??????"
>       ???. +7 (3435) 230001
>       ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     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


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

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From kbdfck at gmail.com  Thu May 19 12:26:03 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 19 May 2011 12:26:03 +0400
Subject: [Freeswitch-users] Spoofed CID?
In-Reply-To: <BANLkTikydmOJghbySDBe_v5TYanXUCNL=g@mail.gmail.com>
References: <1305770468329-6380373.post@n2.nabble.com>
	<BANLkTikydmOJghbySDBe_v5TYanXUCNL=g@mail.gmail.com>
Message-ID: <BANLkTik0YvEmM_b+sTz0WMPuEMvN=4wxvQ@mail.gmail.com>

Actually you can't get any guarantees in case of PSTN too :)

2011/5/19 Steven Ayre <steveayre at gmail.com>

> There's no way of knowing really.
>
> You can send any CID you like in SIP. They might own 4433669222 in real
> life, they might not. You only really get a guarantee it's genuine if it's
> coming off the PSTN.
>
> All I can tell you is 44336 is a valid UK phone number prefix (
> http://en.wikipedia.org/wiki/03_numbers).
>
> -Steve
>
>
>
>
> On 19 May 2011 03:01, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>> Today, after my phone rang with a CID of (443)366-9222, I took a look at
>> /var/log/freeswitch/freeswitch.log file on my Seagate DockStar (excerpted
>> below). What caught my attention is the last line which shows two numbers,
>> namely 4433669222 and 4016488312. The question I have is if 4433669222 is
>> a
>> spoofed number?
>>
>> 2011-05-18 12:54:03.258849 [NOTICE] switch_channel.c:812 New Channel
>> sofia/external/4433669222 at 64.154.
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4744 Channel
>> sofia/external/4433669222 at 64.154.41.150 enteri
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4755 Remote SDP:
>> v=0^M
>> o=root 241964608 241964608 IN IP4 64.154.41.150^M
>> s=Asterisk PBX 1.6.2.13^M
>> c=IN IP4 64.154.41.150^M
>> t=0 0^M
>> m=audio 10730 RTP/AVP 0 8 3 18 101^M
>> a=rtpmap:0 PCMU/8000^M
>> a=rtpmap:8 PCMA/8000^M
>> a=rtpmap:3 GSM/8000^M
>> a=rtpmap:18 G729/8000^M
>> a=fmtp:18 annexb=no^M
>> a=rtpmap:101 telephone-event/8000^M
>> a=fmtp:101 0-16^M
>> a=ptime:20^M
>>
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[G7221
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[G7221
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[G722:
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[PCMU:
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:2760 Set Codec
>> sofia/external/4433669222 at 64.154.41.150
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4751 Set 2833 dtmf
>> send/recv
>> payload to 101
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
>> (sofia/external/4433669222 at 64.154.41.150) State Change
>>
>> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
>> (sofia/external/4433669222 at 64.154.41.150) State Change
>> 2011-05-18 12:54:03.258849 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/4433669222 at 64
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/4433669222 at 64.154.4
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
>> (sofia/external/4433669222 at 64.154.4
>> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:84
>> sofia/external/4433669222 at 64.154.41.150 SOFIA INIT
>> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:124
>> (sofia/external/4433669222 at 64.154.41.150) State Cha
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/4433669222 at 64
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
>> (sofia/external/4433669222 at 64.154.4
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/4433669222 at 64.154.4
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_channel.c:1668
>> (sofia/external/4433669222 at 64.154.41.150) Cal
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:359
>> (sofia/external/4433669222 at 64.154.4
>> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:147
>> sofia/external/4433669222 at 64.154.41.150 SOFIA ROUTI
>> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:77
>> sofia/external/4433669222 at 64.154.41.
>> 2011-05-18 12:54:03.265631 [INFO] mod_dialplan_xml.c:331 Processing
>> 4433669222 <4433669222>->401648831
>>
>>
>> -----
>> 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/Spoofed-CID-tp6380373p6380373.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
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From eric at loopfx.com  Thu May 19 18:57:28 2011
From: eric at loopfx.com (Eric Beard)
Date: Thu, 19 May 2011 10:57:28 -0400
Subject: [Freeswitch-users] SIP Trace
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>

Hello,

In fs_cli, I can issue this command:

sofia global siptrace on

And I see all SIP traffic.  But the SIP does not get logged to the actual log file.  How do I get the same output to be sent to the log?  I've searched through the Wiki but I'm not seeing a clear way to do it.

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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From Nabble at slickdeals.endjunk.com  Thu May 19 19:15:08 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 19 May 2011 08:15:08 -0700 (PDT)
Subject: [Freeswitch-users] Spoofed CID?
In-Reply-To: <BANLkTikydmOJghbySDBe_v5TYanXUCNL=g@mail.gmail.com>
References: <1305770468329-6380373.post@n2.nabble.com>
	<BANLkTikydmOJghbySDBe_v5TYanXUCNL=g@mail.gmail.com>
Message-ID: <1305818108878-6382404.post@n2.nabble.com>


Steven Ayre wrote:
> 
> There's no way of knowing really.
> 
> You can send any CID you like in SIP. They might own 4433669222 in real
> life, they might not. You only really get a guarantee it's genuine if it's
> coming off the PSTN.
> 
> All I can tell you is 44336 is a valid UK phone number prefix (
> http://en.wikipedia.org/wiki/03_numbers).
Yes and 44336 is a valid UK phone number prefix. However, this is an
11-digit number coming from a Google Voice by a 
http://tnid.us/lookup/4433669222/ wireless caller .

-----
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/Spoofed-CID-tp6380373p6382404.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Thu May 19 19:18:09 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 19 May 2011 08:18:09 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch with Sipgate
In-Reply-To: <1305812333066-6381973.post@n2.nabble.com>
References: <1305812333066-6381973.post@n2.nabble.com>
Message-ID: <1305818289417-6382418.post@n2.nabble.com>

Have a look 
http://wiki.freeswitch.org/wiki/Provider_Configuration:_SipGate.de here  and
the proxy is sipgate.de.

-----
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-with-Sipgate-tp6381973p6382418.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rhuddleston at gmail.com  Thu May 19 19:18:19 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Thu, 19 May 2011 11:18:19 -0400
Subject: [Freeswitch-users] Spoofed CID?
In-Reply-To: <1305818108878-6382404.post@n2.nabble.com>
References: <1305770468329-6380373.post@n2.nabble.com>
	<BANLkTikydmOJghbySDBe_v5TYanXUCNL=g@mail.gmail.com>
	<1305818108878-6382404.post@n2.nabble.com>
Message-ID: <200701cc1637$fcf5eed0$f6e1cc70$@com>

Did you look at SIP headers to see if a P-Asserted-Indetity could have been
attached.. I don't think it's widely used - but I know some carriers require
it.

http://www.ietf.org/rfc/rfc3325.txt
 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
Sent: Thursday, May 19, 2011 11:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Spoofed CID?


Steven Ayre wrote:
> 
> There's no way of knowing really.
> 
> You can send any CID you like in SIP. They might own 4433669222 in real
> life, they might not. You only really get a guarantee it's genuine if it's
> coming off the PSTN.
> 
> All I can tell you is 44336 is a valid UK phone number prefix (
> http://en.wikipedia.org/wiki/03_numbers).
Yes and 44336 is a valid UK phone number prefix. However, this is an
11-digit number coming from a Google Voice by a 
http://tnid.us/lookup/4433669222/ wireless caller .

-----
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/Spoofed-CID-tp6380373p6382404.
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 Nabble at slickdeals.endjunk.com  Thu May 19 19:20:46 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 19 May 2011 08:20:46 -0700 (PDT)
Subject: [Freeswitch-users] G729 on windows???
In-Reply-To: <1305817765013-6382381.post@n2.nabble.com>
References: <81C2CEF80046FB4F863A60D4347DD33A105B39@server1.st.local>
	<9642110D-AA7F-460D-AEBA-E86F0A06EBF2@freeswitch.org>
	<1305817765013-6382381.post@n2.nabble.com>
Message-ID: <1305818446150-6382435.post@n2.nabble.com>


Mogsy.uk wrote:
> Is there a G729 codec available yet for windows?
I am just wondering some free windows softphones have a support for G729
CoDec. Can anyone make use this CoDec on Windows FS?

-----
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/G729-on-windows-tp5213277p6382435.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Thu May 19 19:42:09 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 19 May 2011 08:42:09 -0700 (PDT)
Subject: [Freeswitch-users] Spoofed CID?
In-Reply-To: <200701cc1637$fcf5eed0$f6e1cc70$@com>
References: <1305770468329-6380373.post@n2.nabble.com>
	<BANLkTikydmOJghbySDBe_v5TYanXUCNL=g@mail.gmail.com>
	<1305818108878-6382404.post@n2.nabble.com>
	<200701cc1637$fcf5eed0$f6e1cc70$@com>
Message-ID: <1305819729927-6382521.post@n2.nabble.com>


Robert Huddleston wrote:
> 
> Did you look at SIP headers to see if a P-Asserted-Indetity could have
> been
> attached.. I don't think it's widely used - but I know some carriers
> require
> it.
> 
> http://www.ietf.org/rfc/rfc3325.txt
I took a look at the RFC3325 you provided above, particularly on section 9.1
The P-Asserted-Identity Header. Unfortunately, I can't identify any
P-Asserted-Identity Header from a regular /var/log/freeswitch/freeswitch.log
file. :(

-----
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/Spoofed-CID-tp6380373p6382521.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at ipeva.fr  Thu May 19 20:10:03 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 19 May 2011 18:10:03 +0200
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD4CCCB.2080700@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>
	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>
	<4DD4CCCB.2080700@tagnet.ru>
Message-ID: <F3CEB067-D2A0-41D8-BC61-4265277AF106@ipeva.fr>

Boris,

so basically, you just need the second CDR to bill the customer, don't you ?
I suppose you don't bill for incoming calls, so the redirected call is the only one you care about, or am I missing something ?

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

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

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




Le 19/05/2011 ? 09:54, Boris Kovalenko a ?crit :

> Hello!
> 
>     It is the way I try to do manual redirect. May be I do it wrong, so may be somebody tell me another way. So... I use only A-leg accounting records. One of my users set the redirect from its phone to cellular phone (PAP2T device). So, when call is arrived from PBX
> and redirected by user, in A-leg CDR record I see only _original_ information: pbx_caller_id_number and local_destination_number. And this is wrong for billing. With help of CDR I want to emulate two calls scenario:
> First CDR record - PBX -> user number
> Second CDR record - user number -> redirected number
>  
>> I don't understand why you would need this.
>> 
>> What would the difference between the variables be?
>> 
>> -Steve
>> 
>> 
>> 
>> On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru> wrote:
>> Hello!
>> 
>>     I know :) But with one call I get only one CDR record for A leg and
>> one record for B leg. I need to get 2 records for A leg with two
>> different sets of variables. Is this possible?
>> 
>> > There's no 'standard' CDR. The XML CDR will contain everything,
>> > including all variables. If you mean the default mod_cdr_csv, you can
>> > customise which variables are saved in its configuration.
>> >
>> > -Steve
>> >
>> >
>> >
>> > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> >> Hello!
>> >>
>> >> I don't know what is asterisk fork_cdr :) I want to bridge two calls and
>> >> get 2 cdr records for A leg: one the standard cdr, and another with
>> >> variables I need.
>> >>
>> >>> I suppose he's asking for asterisk analog of fork_cdr
>> >>>
>> >>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>> >>>> You want your Lua script to *send* an event? It most certainly can fire
>> >>>> events:
>> >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>> >>>> I suppose the question is: what do you want to have happen when you fire an
>> >>>> event? What will receive this event?
>> >>>> -MC
>> >>>>
>> >>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>> >>>>> Hello!
>> >>>>>
>> >>>>>       May I fire custom cdr event for a LUA script? I've read
>> >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>> >>>>> Would You please point me to proper documentation or give an example?
>> >>>>>
>> >>>>> --
>> >>>>> 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 list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at 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
>> >>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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

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From david.ponzone at ipeva.fr  Thu May 19 20:12:05 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 19 May 2011 18:12:05 +0200
Subject: [Freeswitch-users] G729 on windows???
In-Reply-To: <1305818446150-6382435.post@n2.nabble.com>
References: <81C2CEF80046FB4F863A60D4347DD33A105B39@server1.st.local>
	<9642110D-AA7F-460D-AEBA-E86F0A06EBF2@freeswitch.org>
	<1305817765013-6382381.post@n2.nabble.com>
	<1305818446150-6382435.post@n2.nabble.com>
Message-ID: <76661DA5-5B56-49B9-A6FB-5B7613EAE5EF@ipeva.fr>

Free softphones  for windows with G729 support and OpenSource ?

Names!

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

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

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




Le 19/05/2011 ? 17:20, mazilo a ?crit :

> 
> Mogsy.uk wrote:
>> Is there a G729 codec available yet for windows?
> I am just wondering some free windows softphones have a support for G729
> CoDec. Can anyone make use this CoDec on Windows FS?
> 
> -----
> 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/G729-on-windows-tp5213277p6382435.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 steveayre at gmail.com  Thu May 19 20:14:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 17:14:08 +0100
Subject: [Freeswitch-users] G729 on windows???
In-Reply-To: <1305818446150-6382435.post@n2.nabble.com>
References: <81C2CEF80046FB4F863A60D4347DD33A105B39@server1.st.local>
	<9642110D-AA7F-460D-AEBA-E86F0A06EBF2@freeswitch.org>
	<1305817765013-6382381.post@n2.nabble.com>
	<1305818446150-6382435.post@n2.nabble.com>
Message-ID: <BANLkTi=c0_xjaifuqaXZ5cMzKPF5R596aQ@mail.gmail.com>

"I am just wondering some free windows softphones have a support for G729
CoDec"

No legal ones...





On 19 May 2011 16:20, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Mogsy.uk wrote:
> > Is there a G729 codec available yet for windows?
> I am just wondering some free windows softphones have a support for G729
> CoDec. Can anyone make use this CoDec on Windows FS?
>
> -----
> 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/G729-on-windows-tp5213277p6382435.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 steveayre at gmail.com  Thu May 19 20:15:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 17:15:45 +0100
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
Message-ID: <BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>

By default it only goes to console. 'sofia tracelevel debug' will let you
set a log level which'll let you log it to the logfile (you can choose any
of the log files).

-Steve


On 19 May 2011 15:57, Eric Beard <eric at loopfx.com> wrote:

> Hello,
>
>
>
> In fs_cli, I can issue this command:
>
>
>
> sofia global siptrace on
>
>
>
> And I see all SIP traffic.  But the SIP does not get logged to the actual
> log file.  How do I get the same output to be sent to the log?  I?ve
> searched through the Wiki but I?m not seeing a clear way to do it.
>
>
>
> Thanks!
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From steveayre at gmail.com  Thu May 19 20:18:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 17:18:17 +0100
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD4CCCB.2080700@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>
	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
	<4DD3849F.40409@tagnet.ru>
	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>
	<4DD48A99.4000700@tagnet.ru>
	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>
	<4DD4CCCB.2080700@tagnet.ru>
Message-ID: <BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>

The XML CDR contains a history of the call, which might be more suited to
you. One CDR but it'll contain everything you need. You can then choose to
create 2 DB records if that's what you wish to do.

Otherwise write all relevant variables to a CDR (if you're overwriting one
in the dialplan set it to another variable first otherwise it won't be
available any more when mod_cdr_csv writes it).

-Steve


On 19 May 2011 08:54, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     It is the way I try to do manual redirect. May be I do it wrong, so may
> be somebody tell me another way. So... I use only A-leg accounting records.
> One of my users set the redirect from its phone to cellular phone (PAP2T
> device). So, when call is arrived from PBX
> and redirected by user, in A-leg CDR record I see only _original_
> information: pbx_caller_id_number and local_destination_number. And this is
> wrong for billing. With help of CDR I want to emulate two calls scenario:
> First CDR record - PBX -> user number
> Second CDR record - user number -> redirected number
>
>
>
> I don't understand why you would need this.
>
> What would the difference between the variables be?
>
> -Steve
>
>
>
> On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     I know :) But with one call I get only one CDR record for A leg and
>> one record for B leg. I need to get 2 records for A leg with two
>> different sets of variables. Is this possible?
>>
>> > There's no 'standard' CDR. The XML CDR will contain everything,
>> > including all variables. If you mean the default mod_cdr_csv, you can
>> > customise which variables are saved in its configuration.
>> >
>> > -Steve
>> >
>> >
>> >
>> > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> >> Hello!
>> >>
>> >> I don't know what is asterisk fork_cdr :) I want to bridge two calls
>> and
>> >> get 2 cdr records for A leg: one the standard cdr, and another with
>> >> variables I need.
>> >>
>> >>> I suppose he's asking for asterisk analog of fork_cdr
>> >>>
>> >>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>> >>>> You want your Lua script to *send* an event? It most certainly can
>> fire
>> >>>> events:
>> >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>> >>>> I suppose the question is: what do you want to have happen when you
>> fire an
>> >>>> event? What will receive this event?
>> >>>> -MC
>> >>>>
>> >>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>
>>  wrote:
>> >>>>> Hello!
>> >>>>>
>> >>>>>       May I fire custom cdr event for a LUA script? I've read
>> >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr
>> events.
>> >>>>> Would You please point me to proper documentation or give an
>> example?
>> >>>>>
>> >>>>> --
>> >>>>> 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 list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at 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
>> >>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>
>
> _______________________________________________
> 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
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +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 boris at tagnet.ru  Thu May 19 20:20:04 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 19 May 2011 22:20:04 +0600
Subject: [Freeswitch-users] CDR event
In-Reply-To: <F3CEB067-D2A0-41D8-BC61-4265277AF106@ipeva.fr>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>	<4DD4CCCB.2080700@tagnet.ru>
	<F3CEB067-D2A0-41D8-BC61-4265277AF106@ipeva.fr>
Message-ID: <4DD54334.6010501@tagnet.ru>

Yes, you are right. No, I don't bill for incoming calls for clients. My 
billing solution is based on A-leg CDR records, and the record about 
original source of call is needed for inter-operator billing.
> Boris,
>
> so basically, you just need the second CDR to bill the customer, don't 
> you ?
> I suppose you don't bill for incoming calls, so the redirected call is 
> the only one you care about, or am I missing something ?
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 19/05/2011 ? 09:54, Boris Kovalenko a ?crit :
>
>> Hello!
>>
>>     It is the way I try to do manual redirect. May be I do it wrong, 
>> so may be somebody tell me another way. So... I use only A-leg 
>> accounting records. One of my users set the redirect from its phone 
>> to cellular phone (PAP2T device). So, when call is arrived from PBX
>> and redirected by user, in A-leg CDR record I see only _original_ 
>> information: pbx_caller_id_number and local_destination_number. And 
>> this is wrong for billing. With help of CDR I want to emulate two 
>> calls scenario:
>> First CDR record - PBX -> user number
>> Second CDR record - user number -> redirected number
>>
>>> I don't understand why you would need this.
>>>
>>> What would the difference between the variables be?
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru 
>>> <mailto:boris at tagnet.ru>> wrote:
>>>
>>>     Hello!
>>>
>>>         I know :) But with one call I get only one CDR record for A
>>>     leg and
>>>     one record for B leg. I need to get 2 records for A leg with two
>>>     different sets of variables. Is this possible?
>>>
>>>     > There's no 'standard' CDR. The XML CDR will contain everything,
>>>     > including all variables. If you mean the default mod_cdr_csv,
>>>     you can
>>>     > customise which variables are saved in its configuration.
>>>     >
>>>     > -Steve
>>>     >
>>>     >
>>>     >
>>>     > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru
>>>     <mailto:boris at tagnet.ru>>  wrote:
>>>     >> Hello!
>>>     >>
>>>     >> I don't know what is asterisk fork_cdr :) I want to bridge
>>>     two calls and
>>>     >> get 2 cdr records for A leg: one the standard cdr, and
>>>     another with
>>>     >> variables I need.
>>>     >>
>>>     >>> I suppose he's asking for asterisk analog of fork_cdr
>>>     >>>
>>>     >>> 2011/5/17 Michael Collins<msc at freeswitch.org
>>>     <mailto:msc at freeswitch.org>>:
>>>     >>>> You want your Lua script to *send* an event? It most
>>>     certainly can fire
>>>     >>>> events:
>>>     >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>>     >>>> I suppose the question is: what do you want to have happen
>>>     when you fire an
>>>     >>>> event? What will receive this event?
>>>     >>>> -MC
>>>     >>>>
>>>     >>>> On Sat, May 14, 2011 at 12:59 AM, Boris
>>>     Kovalenko<boris at tagnet.ru <mailto:boris at tagnet.ru>>    wrote:
>>>     >>>>> Hello!
>>>     >>>>>
>>>     >>>>>       May I fire custom cdr event for a LUA script? I've read
>>>     >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't
>>>     found cdr events.
>>>     >>>>> Would You please point me to proper documentation or give
>>>     an example?
>>>     >>>>>
>>>     >>>>> --
>>>     >>>>> 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 <http://www.freeswitch.org/>
>>>     >>>> _______________________________________________
>>>     >>>> FreeSWITCH-users mailing list
>>>     >>>> FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     >>>>
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     >>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>     >>>>
>>>     >>>>
>>>     >>> _______________________________________________
>>>     >>> FreeSWITCH-users mailing list
>>>     >>> FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     >>>
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     >>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>     >>
>>>     >> --
>>>     >> ? ?????????,
>>>     >>    ????? ?????????
>>>     >>    ??? "??????"
>>>     >>    ???. +7 (3435) 230001
>>>     >>    ???? +7 (3435) 230005
>>>     >>
>>>     >>
>>>     >> _______________________________________________
>>>     >> FreeSWITCH-users mailing list
>>>     >> FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     >>
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>     >>
>>>     > _______________________________________________
>>>     > FreeSWITCH-users mailing list
>>>     > FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     >
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>     --
>>>     ? ?????????,
>>>       ????? ?????????
>>>       ??? "??????"
>>>       ???. +7 (3435) 230001
>>>       ???? +7 (3435) 230005
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> -- 
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    ???. +7 (3435) 230001
>>    ???? +7 (3435) 230005
>> _______________________________________________
>> 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


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

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From boris at tagnet.ru  Thu May 19 20:24:02 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 19 May 2011 22:24:02 +0600
Subject: [Freeswitch-users] CDR event
In-Reply-To: <BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>	<4DD4CCCB.2080700@tagnet.ru>
	<BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>
Message-ID: <4DD54422.7010507@tagnet.ru>

So there is not solution to fire additional custom CDR event? Ok, I'll 
look at XML CDR.

> The XML CDR contains a history of the call, which might be more suited 
> to you. One CDR but it'll contain everything you need. You can then 
> choose to create 2 DB records if that's what you wish to do.
>
> Otherwise write all relevant variables to a CDR (if you're overwriting 
> one in the dialplan set it to another variable first otherwise it 
> won't be available any more when mod_cdr_csv writes it).
>
> -Steve
>
>
> On 19 May 2011 08:54, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         It is the way I try to do manual redirect. May be I do it
>     wrong, so may be somebody tell me another way. So... I use only
>     A-leg accounting records. One of my users set the redirect from
>     its phone to cellular phone (PAP2T device). So, when call is
>     arrived from PBX
>     and redirected by user, in A-leg CDR record I see only _original_
>     information: pbx_caller_id_number and local_destination_number.
>     And this is wrong for billing. With help of CDR I want to emulate
>     two calls scenario:
>     First CDR record - PBX -> user number
>     Second CDR record - user number -> redirected number
>
>
>>     I don't understand why you would need this.
>>
>>     What would the difference between the variables be?
>>
>>     -Steve
>>
>>
>>
>>     On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru
>>     <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>             I know :) But with one call I get only one CDR record for
>>         A leg and
>>         one record for B leg. I need to get 2 records for A leg with two
>>         different sets of variables. Is this possible?
>>
>>         > There's no 'standard' CDR. The XML CDR will contain everything,
>>         > including all variables. If you mean the default
>>         mod_cdr_csv, you can
>>         > customise which variables are saved in its configuration.
>>         >
>>         > -Steve
>>         >
>>         >
>>         >
>>         > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru
>>         <mailto:boris at tagnet.ru>>  wrote:
>>         >> Hello!
>>         >>
>>         >> I don't know what is asterisk fork_cdr :) I want to bridge
>>         two calls and
>>         >> get 2 cdr records for A leg: one the standard cdr, and
>>         another with
>>         >> variables I need.
>>         >>
>>         >>> I suppose he's asking for asterisk analog of fork_cdr
>>         >>>
>>         >>> 2011/5/17 Michael Collins<msc at freeswitch.org
>>         <mailto:msc at freeswitch.org>>:
>>         >>>> You want your Lua script to *send* an event? It most
>>         certainly can fire
>>         >>>> events:
>>         >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>         >>>> I suppose the question is: what do you want to have
>>         happen when you fire an
>>         >>>> event? What will receive this event?
>>         >>>> -MC
>>         >>>>
>>         >>>> On Sat, May 14, 2011 at 12:59 AM, Boris
>>         Kovalenko<boris at tagnet.ru <mailto:boris at tagnet.ru>>    wrote:
>>         >>>>> Hello!
>>         >>>>>
>>         >>>>>       May I fire custom cdr event for a LUA script?
>>         I've read
>>         >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't
>>         found cdr events.
>>         >>>>> Would You please point me to proper documentation or
>>         give an example?
>>         >>>>>
>>         >>>>> --
>>         >>>>> 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
>>         >>>>
>>         >>>>
>>         >>> _______________________________________________
>>         >>> 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
>>         >>
>>         >> --
>>         >> ? ?????????,
>>         >>    ????? ?????????
>>         >>    ??? "??????"
>>         >>    ???. +7 (3435) 230001
>>         >>    ???? +7 (3435) 230005
>>         >>
>>         >>
>>         >> _______________________________________________
>>         >> 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
>>
>>
>>         --
>>         ? ?????????,
>>           ????? ?????????
>>           ??? "??????"
>>           ???. +7 (3435) 230001
>>           ???? +7 (3435) 230005
>>
>>
>>         _______________________________________________
>>         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
>
>
>     -- 
>     ? ?????????,
>        ????? ?????????
>        ??? "??????"
>        ???. +7 (3435) 230001
>        ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     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


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

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From david.ponzone at ipeva.fr  Thu May 19 20:26:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 19 May 2011 18:26:28 +0200
Subject: [Freeswitch-users] Spoofed CID?
In-Reply-To: <1305770468329-6380373.post@n2.nabble.com>
References: <1305770468329-6380373.post@n2.nabble.com>
Message-ID: <6F1038EA-63B4-470B-85F3-8390D63F1AED@ipeva.fr>

Mazilo,

well, as anyone said, you can't know that for sure.

Here, in France, and I think this may be applicable in most western countries, the end user of a telephony service will always receive only one CLI.
If the call comes from an analog line on the PSTN, the CLI is provided by the telco.
If the call comes from an ISDN line on the PSTN, the CLI is the one sent by the caller's PBX.
So for calls coming from ISDN, the telco will always add another CLI, the real CLI of the customer. We call it NDI in France (meaning Installation Number).
For SIP, you can guess what applies: no rules...
So basically, if you are a end user (not a telco), you will receive a CLI which can be fake.
If you are a telco and are interconnected in SIP or SS7 in the right way, you may (meaning "you are allowed to") receive the 2 CLIs for calls coming from ISDN.
For SIP calls, the issue is still there anyway: the reliability of the CLI depends on the seriousness of the ITSPs out there.

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

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

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




Le 19/05/2011 ? 04:01, mazilo a ?crit :

> Today, after my phone rang with a CID of (443)366-9222, I took a look at
> /var/log/freeswitch/freeswitch.log file on my Seagate DockStar (excerpted
> below). What caught my attention is the last line which shows two numbers,
> namely 4433669222 and 4016488312. The question I have is if 4433669222 is a
> spoofed number?
> 
> 2011-05-18 12:54:03.258849 [NOTICE] switch_channel.c:812 New Channel
> sofia/external/4433669222 at 64.154.
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4744 Channel
> sofia/external/4433669222 at 64.154.41.150 enteri
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4755 Remote SDP:                                           
> v=0^M                                                                                                 
> o=root 241964608 241964608 IN IP4 64.154.41.150^M                                                     
> s=Asterisk PBX 1.6.2.13^M                                                                             
> c=IN IP4 64.154.41.150^M                                                                              
> t=0 0^M                                                                                               
> m=audio 10730 RTP/AVP 0 8 3 18 101^M                                                                  
> a=rtpmap:0 PCMU/8000^M                                                                                
> a=rtpmap:8 PCMA/8000^M                                                                                
> a=rtpmap:3 GSM/8000^M                                                                                 
> a=rtpmap:18 G729/8000^M                                                                               
> a=fmtp:18 annexb=no^M                                                                                 
> a=rtpmap:101 telephone-event/8000^M                                                                   
> a=fmtp:101 0-16^M                                                                                     
> a=ptime:20^M                                                                                          
> 
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G722:
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:2760 Set Codec
> sofia/external/4433669222 at 64.154.41.150
> 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4751 Set 2833 dtmf send/recv
> payload to 101           
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
> (sofia/external/4433669222 at 64.154.41.150) State Change
> 
> 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922
> (sofia/external/4433669222 at 64.154.41.150) State Change
> 2011-05-18 12:54:03.258849 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/4433669222 at 64
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:84
> sofia/external/4433669222 at 64.154.41.150 SOFIA INIT  
> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:124
> (sofia/external/4433669222 at 64.154.41.150) State Cha
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/4433669222 at 64
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] switch_channel.c:1668
> (sofia/external/4433669222 at 64.154.41.150) Cal
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/4433669222 at 64.154.4
> 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:147
> sofia/external/4433669222 at 64.154.41.150 SOFIA ROUTI
> 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:77
> sofia/external/4433669222 at 64.154.41.
> 2011-05-18 12:54:03.265631 [INFO] mod_dialplan_xml.c:331 Processing
> 4433669222 <4433669222>->401648831
> 
> 
> -----
> 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/Spoofed-CID-tp6380373p6380373.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Thu May 19 20:29:06 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 19 May 2011 18:29:06 +0200
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD54422.7010507@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>	<4DD4CCCB.2080700@tagnet.ru>
	<BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>
	<4DD54422.7010507@tagnet.ru>
Message-ID: <D80E7467-7E54-4387-AB94-CD4FD3AD0577@ipeva.fr>

Boris,

I think we still wonder what additional custom CDR you need.
You already know that one redirected call will fire 2 CDRs.
One for the incoming call you need for inter-operator billing, and the one for the outgoing redirected call you need to bill the customer.

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

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

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




Le 19/05/2011 ? 18:24, Boris Kovalenko a ?crit :

> So there is not solution to fire additional custom CDR event? Ok, I'll look at XML CDR.
> 
>> The XML CDR contains a history of the call, which might be more suited to you. One CDR but it'll contain everything you need. You can then choose to create 2 DB records if that's what you wish to do.
>> 
>> Otherwise write all relevant variables to a CDR (if you're overwriting one in the dialplan set it to another variable first otherwise it won't be available any more when mod_cdr_csv writes it).
>> 
>> -Steve
>> 
>> 
>> On 19 May 2011 08:54, Boris Kovalenko <boris at tagnet.ru> wrote:
>> Hello!
>> 
>>     It is the way I try to do manual redirect. May be I do it wrong, so may be somebody tell me another way. So... I use only A-leg accounting records. One of my users set the redirect from its phone to cellular phone (PAP2T device). So, when call is arrived from PBX
>> and redirected by user, in A-leg CDR record I see only _original_ information: pbx_caller_id_number and local_destination_number. And this is wrong for billing. With help of CDR I want to emulate two calls scenario:
>> First CDR record - PBX -> user number
>> Second CDR record - user number -> redirected number
>> 
>>  
>>> I don't understand why you would need this.
>>> 
>>> What would the difference between the variables be?
>>> 
>>> -Steve
>>> 
>>> 
>>> 
>>> On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru> wrote:
>>> Hello!
>>> 
>>>     I know :) But with one call I get only one CDR record for A leg and
>>> one record for B leg. I need to get 2 records for A leg with two
>>> different sets of variables. Is this possible?
>>> 
>>> > There's no 'standard' CDR. The XML CDR will contain everything,
>>> > including all variables. If you mean the default mod_cdr_csv, you can
>>> > customise which variables are saved in its configuration.
>>> >
>>> > -Steve
>>> >
>>> >
>>> >
>>> > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> >> Hello!
>>> >>
>>> >> I don't know what is asterisk fork_cdr :) I want to bridge two calls and
>>> >> get 2 cdr records for A leg: one the standard cdr, and another with
>>> >> variables I need.
>>> >>
>>> >>> I suppose he's asking for asterisk analog of fork_cdr
>>> >>>
>>> >>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>> >>>> You want your Lua script to *send* an event? It most certainly can fire
>>> >>>> events:
>>> >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>> >>>> I suppose the question is: what do you want to have happen when you fire an
>>> >>>> event? What will receive this event?
>>> >>>> -MC
>>> >>>>
>>> >>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>> >>>>> Hello!
>>> >>>>>
>>> >>>>>       May I fire custom cdr event for a LUA script? I've read
>>> >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>>> >>>>> Would You please point me to proper documentation or give an example?
>>> >>>>>
>>> >>>>> --
>>> >>>>> 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 list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at 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
>>> >>
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at 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
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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

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From Nabble at slickdeals.endjunk.com  Thu May 19 20:34:30 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 19 May 2011 09:34:30 -0700 (PDT)
Subject: [Freeswitch-users] Spoofed CID?
In-Reply-To: <6F1038EA-63B4-470B-85F3-8390D63F1AED@ipeva.fr>
References: <1305770468329-6380373.post@n2.nabble.com>
	<6F1038EA-63B4-470B-85F3-8390D63F1AED@ipeva.fr>
Message-ID: <1305822870318-6382763.post@n2.nabble.com>

Thanks David + everyone for the explanation and I appreciated that.

-----
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/Spoofed-CID-tp6380373p6382763.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rhuddleston at gmail.com  Thu May 19 20:34:13 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Thu, 19 May 2011 12:34:13 -0400
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD54422.7010507@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>	<4DD4CCCB.2080700@tagnet.ru>
	<BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>
	<4DD54422.7010507@tagnet.ru>
Message-ID: <20d501cc1642$9742ece0$c5c8c6a0$@com>

I use XML CDR and writing business logic on my php server side that consumes the XML CDR post.

 

The nice thing about the XML CDR is that it is very verbose

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Boris Kovalenko
Sent: Thursday, May 19, 2011 12:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] CDR event

 

So there is not solution to fire additional custom CDR event? Ok, I'll look at XML CDR.




The XML CDR contains a history of the call, which might be more suited to you. One CDR but it'll contain everything you need. You can then choose to create 2 DB records if that's what you wish to do.

Otherwise write all relevant variables to a CDR (if you're overwriting one in the dialplan set it to another variable first otherwise it won't be available any more when mod_cdr_csv writes it).

-Steve



On 19 May 2011 08:54, Boris Kovalenko <boris at tagnet.ru> wrote:

Hello!

    It is the way I try to do manual redirect. May be I do it wrong, so may be somebody tell me another way. So... I use only A-leg accounting records. One of my users set the redirect from its phone to cellular phone (PAP2T device). So, when call is arrived from PBX
and redirected by user, in A-leg CDR record I see only _original_ information: pbx_caller_id_number and local_destination_number. And this is wrong for billing. With help of CDR I want to emulate two calls scenario:
First CDR record - PBX -> user number
Second CDR record - user number -> redirected number 


 



I don't understand why you would need this.

What would the difference between the variables be?

-Steve




On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru> wrote:



Hello!

    I know :) But with one call I get only one CDR record for A leg and
one record for B leg. I need to get 2 records for A leg with two
different sets of variables. Is this possible?


> There's no 'standard' CDR. The XML CDR will contain everything,
> including all variables. If you mean the default mod_cdr_csv, you can
> customise which variables are saved in its configuration.
>
> -Steve
>
>
>
> On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>> I don't know what is asterisk fork_cdr :) I want to bridge two calls and
>> get 2 cdr records for A leg: one the standard cdr, and another with
>> variables I need.
>>
>>> I suppose he's asking for asterisk analog of fork_cdr
>>>
>>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>>> You want your Lua script to *send* an event? It most certainly can fire
>>>> events:
>>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>>> I suppose the question is: what do you want to have happen when you fire an
>>>> event? What will receive this event?
>>>> -MC
>>>>
>>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>>>> Hello!
>>>>>
>>>>>       May I fire custom cdr event for a LUA script? I've read
>>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr events.
>>>>> Would You please point me to proper documentation or give an example?
>>>>>
>>>>> --
>>>>> 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 list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

 

 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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

 

 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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
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From m.sobkow at marketelsystems.com  Thu May 19 20:35:24 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 19 May 2011 10:35:24 -0600
Subject: [Freeswitch-users] Sound quality issues
Message-ID: <4DD546CC.7090209@marketelsystems.com>

We've got a FreeSwitch box using a SIP trunk to an Asterisk box, which 
connects to our T1.

We're having significant voice quality issues, and I'm not sure what to 
look into.  CPU load is under 10% on the Freeswitch box.

Even playing back 8kHz audio files shows significant breakup, so I think 
I've been on the wrong track thinking it was a SIP issue.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From andre.rosowski at omigos.de  Thu May 19 17:38:53 2011
From: andre.rosowski at omigos.de (realdoe)
Date: Thu, 19 May 2011 06:38:53 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch with Sipgate
Message-ID: <1305812333066-6381973.post@n2.nabble.com>

Hi all,

I try to get Freeswitch running with Sipgate as a Gateway but when I enter
"sofia status" I get the following error:

2011-05-19 15:24:06.897915 [ERR] sofia_reg.c:1611 sipconnect.sipgate.de
Registration Failed with status Forbidden (client configuration error)
[403]. failure #4


I created a file called sipgate.xml under
/opt/freeswitch/conf/sip_profiles/external/ and filled it with the
following:

include
  gateway name="sipgate.de"
    param name="proxy" value="sipconnect.live.sipgate.de"/
    param name="from-domain" value="sipgate.de"/
    param name="username" value="MY_NAME"/
    param name="password" value="MY_PASSWORD"/
  /gateway
/include
(Removed the brackets...)

Does anyone know what I might be missing or what went wrong? Thanks for your
time...

Greetings

Realdoe



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-with-Sipgate-tp6381973p6381973.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mkopacki at gmail.com  Thu May 19 20:40:21 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Thu, 19 May 2011 18:40:21 +0200
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
Message-ID: <4DD547F5.4080504@gmail.com>

     Sorry for interrupt, but is there any way to filter messages from 
sip trace ? For example show only one ext/host/uri etc

-- 
Michal

---------------
> By default it only goes to console. 'sofia tracelevel debug' will let 
> you set a log level which'll let you log it to the logfile (you can 
> choose any of the log files).
>
> -Steve
>
>
> On 19 May 2011 15:57, Eric Beard <eric at loopfx.com 
> <mailto:eric at loopfx.com>> wrote:
>
>     Hello,
>
>     In fs_cli, I can issue this command:
>
>     sofia global siptrace on
>
>     And I see all SIP traffic.  But the SIP does not get logged to the
>     actual log file.  How do I get the same output to be sent to the
>     log?  I?ve searched through the Wiki but I?m not seeing a clear
>     way to do it.
>
>     Thanks!
>
>     -----------------------
>
>     *Eric Z. Beard, CTO*
>
>     Loop LLC
>
>     w (877) 850-2010 x9249
>
>     m (727) 776-2768
>
>     eric at loopfx.com <mailto:eric at loopfx.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
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From avi at avimarcus.net  Thu May 19 20:45:07 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 19 May 2011 19:45:07 +0300
Subject: [Freeswitch-users] Sound quality issues
In-Reply-To: <4DD546CC.7090209@marketelsystems.com>
References: <4DD546CC.7090209@marketelsystems.com>
Message-ID: <BANLkTi=hUgmxn8g+DmVDmSer0ffOpNGJbA@mail.gmail.com>

Are you using some sort of virtualization?
-Avi

On Thu, May 19, 2011 at 7:35 PM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> We've got a FreeSwitch box using a SIP trunk to an Asterisk box, which
> connects to our T1.
>
> We're having significant voice quality issues, and I'm not sure what to
> look into.  CPU load is under 10% on the Freeswitch box.
>
> Even playing back 8kHz audio files shows significant breakup, so I think
> I've been on the wrong track thinking it was a SIP issue.
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
> Visit our Blog for industry related information.
> http://marketel-systems.blogspot.com/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eric at loopfx.com  Thu May 19 20:47:59 2011
From: eric at loopfx.com (Eric Beard)
Date: Thu, 19 May 2011 12:47:59 -0400
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFB8D48B@doctorbob.loop.local>

That didn't work for me.

What I really want is for the exact output I see in fs_cli to go to the log file.  At one point in the past I managed to get Sip messages into the log file but I did it on accident and I can't figure out what I did to make it happen.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Thursday, May 19, 2011 12:16 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP Trace

By default it only goes to console. 'sofia tracelevel debug' will let you set a log level which'll let you log it to the logfile (you can choose any of the log files).

-Steve

On 19 May 2011 15:57, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
Hello,

In fs_cli, I can issue this command:

sofia global siptrace on

And I see all SIP traffic.  But the SIP does not get logged to the actual log file.  How do I get the same output to be sent to the log?  I've searched through the Wiki but I'm not seeing a clear way to do it.

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>


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

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From steveayre at gmail.com  Thu May 19 20:48:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 17:48:36 +0100
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <4DD547F5.4080504@gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
Message-ID: <BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>

No.

If you need that it's best to use tshark, wireshark, ngrep or some similar
tool.

-Steve


On 19 May 2011 17:40, Michal Kopacki <mkopacki at gmail.com> wrote:

>      Sorry for interrupt, but is there any way to filter messages from sip
> trace ? For example show only one ext/host/uri etc
>
> --
> Michal
>
> ---------------
>
>  By default it only goes to console. 'sofia tracelevel debug' will let you
> set a log level which'll let you log it to the logfile (you can choose any
> of the log files).
>
> -Steve
>
>
>  On 19 May 2011 15:57, Eric Beard <eric at loopfx.com> wrote:
>
>>  Hello,
>>
>>
>>
>> In fs_cli, I can issue this command:
>>
>>
>>
>> sofia global siptrace on
>>
>>
>>
>> And I see all SIP traffic.  But the SIP does not get logged to the actual
>> log file.  How do I get the same output to be sent to the log?  I?ve
>> searched through the Wiki but I?m not seeing a clear way to do it.
>>
>>
>>
>> Thanks!
>>
>>
>>
>> -----------------------
>>
>> *Eric Z. Beard, CTO*
>>
>> Loop LLC
>>
>> w (877) 850-2010 x9249
>>
>> m (727) 776-2768
>>
>> eric at loopfx.com
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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 michal.bielicki at seventhsignal.de  Thu May 19 20:52:34 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Thu, 19 May 2011 18:52:34 +0200
Subject: [Freeswitch-users] Freeswitch with Sipgate
In-Reply-To: <1305812333066-6381973.post@n2.nabble.com>
References: <1305812333066-6381973.post@n2.nabble.com>
Message-ID: <5951E53C-DA31-4E7E-A8DC-442CDDF16D13@seventhsignal.de>

You have to distinguish between standard sipgate accounts and sipgate trunk, which you are trying to connect to.

Sipgate trunk requires the settings to be the following way:

<gateway name="sipgate_trunk_1">
 <param name="proxy" value="sipconnect.sipgate.de"/>
 <param name="outbound-proxy" value="sipconnect.live.sipgate.de"/>
 <param name="register-proxy" value="sipconnect.live.sipgate.de"/>
 <param name="username" value="YOUR_SIPGATE_TRUNK_USER"/>
 <param name="auth-username" value="YOUR_SIPGATE_TRUNK_USER"/>
 <param name="from-user" value="YOUR_SIPGATE_TRUNK_USER"/>
 <param name="from-domain" value="sipconnect.sipgate.de"/>
 <param name="password" value="YOUR_SIPGATE_TRUNK_PASSWORD"/>
 <param name="extension" value="WHERE_YOU_WANT_INCOMING_CALLS_TO_GO"/>
 <param name="ping" value="25"/>
 <param name="context" value="CONTEXT_INCOMING:_CALLS_SHOULD_GO"/>
 <param name="register" value="true"/>
 <param name="register-transport" value="udp"/>
 <param name="cid-type" value="pid"/>
 <param name="extension-in-contact" value="false"/>
 <param name="caller-id-in-from" value="false"/>
</gateway>

cheers

cypromis

Am 19.05.2011 um 15:38 schrieb realdoe:

> Hi all,
> 
> I try to get Freeswitch running with Sipgate as a Gateway but when I enter
> "sofia status" I get the following error:
> 
> 2011-05-19 15:24:06.897915 [ERR] sofia_reg.c:1611 sipconnect.sipgate.de
> Registration Failed with status Forbidden (client configuration error)
> [403]. failure #4
> 
> 
> I created a file called sipgate.xml under
> /opt/freeswitch/conf/sip_profiles/external/ and filled it with the
> following:
> 
> include
>  gateway name="sipgate.de"
>    param name="proxy" value="sipconnect.live.sipgate.de"/
>    param name="from-domain" value="sipgate.de"/
>    param name="username" value="MY_NAME"/
>    param name="password" value="MY_PASSWORD"/
>  /gateway
> /include
> (Removed the brackets...)
> 
> Does anyone know what I might be missing or what went wrong? Thanks for your
> time...
> 
> Greetings
> 
> Realdoe
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-with-Sipgate-tp6381973p6381973.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

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From anthony.minessale at gmail.com  Thu May 19 21:37:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 May 2011 12:37:55 -0500
Subject: [Freeswitch-users] ClueCon Registration Open
Message-ID: <BANLkTin4BtWVp4ceJq_Lux2jKNPYYvwHpQ@mail.gmail.com>

Please register for ClueCon 2011!

FreeSWITCH Solutions LLC, the collective who maintains FreeSWITCH
hosts the annual ClueCon conference every August in Chicago.
This year its August 9-11th and registration is open now.  This is one
way to give back to FreeSWITCH community but joining us to welcome
other open source projects and collaborate every summer.

Anyone who registers now gets 4 extra tickets for the big raffle
contest that is giving away several prizes including an engraved
MacBook and iPAD.
You can also save $300 just by staying at the Sofitel where the event
will be held.

Register today while there is still room!  Why wait when you can
increase your chances of winning!

Call 877 742 CLUE or go to http://www.cluecon.com


-- 
Anthony Minessale II

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  Thu May 19 23:00:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 May 2011 12:00:15 -0700
Subject: [Freeswitch-users] Problem ringback does not working by
	orignatecalls
In-Reply-To: <Pine.LNX.4.64.1105180345430.9120@debian>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180345430.9120@debian>
Message-ID: <BANLkTi=LS+9PWnNbwtpYOfz8K1nbCxAZ4Q@mail.gmail.com>

You can't answer the call and then later on send a "ring_ready". Once the
call has been answered there is no reason to have the calling party hear
ringing. I'm not sure what you are trying to do but on thing you can try is
to use "pre_answer" instead of "answer" application.

-MC

On Tue, May 17, 2011 at 6:58 PM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi all, i generage a callback via originate.
> the orinatecommand:
> /usr/local/freeswitch/bin/fs_cli -x "bgapi originate
> {ignore_early_media=true,bypass_media=true,originate_retries=10,origination_caller_id_name=Callback,originate_retry_sleep_ms=60000,originate_timeout=900}loopback/XXXXXXX/callback_only_out
> &bridge(loopback/callback_only_sendex1/callback_only_send)"
> My phonenumber is ringing. here the prompt. I don.t here a transferrington
> or music.
> My extension:
> <extension name="callback_only_sendex1">
> <condition field="destination_number" expression="^callback_only_sendex1$">
>  <action application="answer" />
>  <action application="playback" data="silence_stream://10"/>
> <action application="set" data="ringback=${de-ring}"/>
> <action application="set" data="transfer_ringback=$${hold_music}"/>
> <action application="ring_ready"/>
> <action application="set" data="call_timeout=30"/>
>  <action application="set" data="hangup_after_bridge=true"/>
>
>  <action application="set" data="ignore_early_media=true" />
>
> <action application="bridge"
> data="loopback/0174XXXXX/disa"/>
> </condition>
> </extension>
>
>
> Can your help please thank you
>
>
> ---------------
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu May 19 23:03:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 May 2011 12:03:30 -0700
Subject: [Freeswitch-users] G.729A licencing validation error
In-Reply-To: <BANLkTi=42gOrfQqvKdsT4-4R-refv1D=Eg@mail.gmail.com>
References: <BANLkTi=42gOrfQqvKdsT4-4R-refv1D=Eg@mail.gmail.com>
Message-ID: <BANLkTimpEcJyNxJA-Nno0gFyU13PO0MMEA@mail.gmail.com>

Can you try validating again and let us know if it is still not working?
-MC

On Tue, May 17, 2011 at 7:52 AM, Tim Tobin <ttobin at ubq.thrupoint.net> wrote:

> Hi,
>
> I've just purchased a license for the G.729 codec module. I've uploaded
> fsg729-194-installer to my freeswitch machine and have executed it. When i
> try to validate i get the following response:
>
> Failed with error code 6
>
>
> Does anybody know why this might be?
>
> Cheers,
>
> --------------------
> Note: The information contained in this message may be privileged and confidential
> and protected from disclosure. If the reader of this message is not the intended
> recipient, or an employee or agent responsible for delivering this message to the
> intended recipient, you are hereby notified that any dissemination, distribution or
> copying of this communication is strictly prohibited. If you have received this
> communication in error, please notify us immediately by replying to the message and
> deleting it from your computer. Thank you. ThruPoint, Inc. - U5MJHZGXCFFDOYV
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu May 19 23:08:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 May 2011 12:08:06 -0700
Subject: [Freeswitch-users] Ptime and Aastra phone
In-Reply-To: <assp.012027e0a5.92097A6A775D5147B1078E3F15430B9252BF0D@prato.activenetwerx.local>
References: <assp.012027e0a5.92097A6A775D5147B1078E3F15430B9252BF0D@prato.activenetwerx.local>
Message-ID: <BANLkTimtPYZL3C_tUPtq5uqXrTwGP90Saw@mail.gmail.com>

Are you experiencing any audio issues? I believe this is indeed just a
warning.
-MC

On Wed, May 18, 2011 at 7:19 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> I have a few 6730i's that have codec choice set to All, as per the Aastra
> doc
> ptime gets defaulted to 30 and won't accept 20 which seems to be what fs
> is using. I am seeing the following lines in the log:
> [WARNING] sofia_glue.c:213 Codec PCMU payload 0 added to sdp wanting ptime
> 20 but it's already 30 (PCMU:0:30), disabling ptime.
>
> How does one correct this? This is on a recent trunk build.
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jcasale at activenetwerx.com  Thu May 19 23:49:33 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 19 May 2011 19:49:33 +0000
Subject: [Freeswitch-users] Ptime and Aastra phone
In-Reply-To: <BANLkTimtPYZL3C_tUPtq5uqXrTwGP90Saw@mail.gmail.com>
References: <92097A6A775D5147B1078E3F15430B9252BF0D@prato.activenetwerx.local>
	<BANLkTimtPYZL3C_tUPtq5uqXrTwGP90Saw@mail.gmail.com>
Message-ID: <assp.0120e6c2a3.92097A6A775D5147B1078E3F15430B92530586@prato.activenetwerx.local>

>Are you experiencing any audio issues? I believe this is indeed just a warning.?

Hi Michael,
No audio issues, just addressing any and all items I am made of aware of and those
I see in the log while addressing other issues. I wasn't sure if it would be indicative
of anything, half the phones are 480's which don't exhibit the warning, the other half
are these 6730's that do.

I'll ignore it...
Thanks,
jlc


From msc at freeswitch.org  Fri May 20 00:15:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 May 2011 13:15:36 -0700
Subject: [Freeswitch-users] Ptime and Aastra phone
In-Reply-To: <assp.0120e6c2a3.92097A6A775D5147B1078E3F15430B92530586@prato.activenetwerx.local>
References: <92097A6A775D5147B1078E3F15430B9252BF0D@prato.activenetwerx.local>
	<BANLkTimtPYZL3C_tUPtq5uqXrTwGP90Saw@mail.gmail.com>
	<assp.0120e6c2a3.92097A6A775D5147B1078E3F15430B92530586@prato.activenetwerx.local>
Message-ID: <BANLkTinWvdwH=bRDYTGSpn1RNMch9t27dA@mail.gmail.com>

On Thu, May 19, 2011 at 12:49 PM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> >Are you experiencing any audio issues? I believe this is indeed just a
> warning.
>
> Hi Michael,
> No audio issues, just addressing any and all items I am made of aware of
> and those
> I see in the log while addressing other issues. I wasn't sure if it would
> be indicative
> of anything, half the phones are 480's which don't exhibit the warning, the
> other half
> are these 6730's that do.
>
> I'll bet the 6730 is doing something wrong or stupid (or both). Ignoring it
is fine, replacing the 6730's is better. :P

-MC
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From jcasale at activenetwerx.com  Fri May 20 00:24:38 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 19 May 2011 20:24:38 +0000
Subject: [Freeswitch-users] Ptime and Aastra phone
In-Reply-To: <BANLkTinWvdwH=bRDYTGSpn1RNMch9t27dA@mail.gmail.com>
References: <92097A6A775D5147B1078E3F15430B9252BF0D@prato.activenetwerx.local>
	<BANLkTimtPYZL3C_tUPtq5uqXrTwGP90Saw@mail.gmail.com>
	<assp.0120e6c2a3.92097A6A775D5147B1078E3F15430B92530586@prato.activenetwerx.local>
	<BANLkTinWvdwH=bRDYTGSpn1RNMch9t27dA@mail.gmail.com>
Message-ID: <assp.0120442191.92097A6A775D5147B1078E3F15430B92530748@prato.activenetwerx.local>

>I'll bet the 6730 is doing something wrong or stupid (or both). Ignoring it is fine, replacing the 6730's is better. :P

Lol, now that made me laugh...
I have to say, I am firm believer in you get what you pay for, but for a $100.00 those
phones are doing great. We have some Aastra's we paid almost 3 times for that don't
do any better. And the http conf server versus tftp is welcome!

jlc


From admin at blindi.net  Fri May 20 01:26:54 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 19 May 2011 23:26:54 +0200 (CEST)
Subject: [Freeswitch-users] Problem Registrationerrors accessdata and server
	correct.
In-Reply-To: <201105121532017963789@asiainfo-linkage.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<201105121532017963789@asiainfo-linkage.com>
Message-ID: <Pine.LNX.4.64.1105192204540.6224@debian>

Hi all, i have a problem.
I have a Trunk.
  I can use from any other softphone these data, but not by fs.
Fs give a 403 errormessage.
I Send later with my recording.
My Gatewaysection is:
<gateway name="thomas1">
         <param name="username" value="thomas1"/>
         <param name="from-domain" value="strato-iphone.de"/>
         <param name="password" value="meinpasswd"/>
         <param name="caller-id-in-from" value="false"/>
         <param name="proxy" value="strato-iphone.de"/>
         <param name="expire-seconds" value="120"/>
         <param name="register" value="true"/>
         <param name="retry-seconds" value="60"/>
         <param name="extension" value="thomas1"/>
</gateway>
</include>

My trace now:
Script started on Do 19 Mai 2011 22:24:50 CEST
ubuntu:/tmp# /etc/init.d/freeswitch restart
ubuntu:/tmp# fs_cli
             _____ ____     ____ _     ___
            |  ___/ ___|   / ___| |   |_ _|
            | |_  \___ \  | |   | |    | |
            |  _|  ___) | | |___| |___ | |
            |_|   |____/   \____|_____|___|

*******************************************************
* Anthony Minessale II, Ken Rice, Michael Jerris      *
* FreeSWITCH (http://www.freeswitch.org)              *
* Paypal Donations Appreciated: paypal at freeswitch.org *
* Brought to you by ClueCon http://www.cluecon.com/   *
*******************************************************

Type /help <enter> to see a list of commands


+OK log level  [7]
freeswitch at internal> sofia loglevel all 9
Sofia log level for component [all] has been set to [9]
freeswitch at internal> 2011-05-19 22:26:54.797919 [NOTICE] sofia_reg.c:367 Registering mobile1
nua: nua_register: entering
nua(0x7fa78000d970): sent signal r_register
nua(0x7fa78000d970): recv signal r_register
nua: nua_stack_set_params: entering
soa_set_params(static::0x7fa78802c0f0, ...) called
nua(0x7fa78000d970): adding register usage
nta_leg_tcreate(0x7fa788057100)
auth_digest_a1() has A1 = MD5(thomas1:strato-iphone.de:meinpasswd) = c6c958107e411451fe8880d3e42330b1
A2 = MD5(REGISTER:sip:strato-iphone.de;transport=udp)
auth_response: ca9d6bf51bcbd256e8be6d139097a85d = MD5(c6c958107e411451fe8880d3e42330b1:4dd57c75aa62e69ded38f11ed8f922eaed95f8a8:a9fc154a8935968d2d7aca6dbd10faa6) (qop=NONE)
nta: selecting scheme sip
sres_cache_get(0x14d52e0, SRV, "_sip._udp.strato-iphone.de.") called
sres_cache_get(0x14d52e0, SRV, "_sip._udp.strato-iphone.de.") returned 2 entries
nta: for "strato-iphone.de" query "_sip._udp.strato-iphone.de" SRV (cached)
nta: _sip._udp.strato-iphone.de IN SRV 0 0  5060 strato-iphone-1.sip.1und1.de. (udp)
nta: _sip._udp.strato-iphone.de IN SRV 0 0  5060 strato-iphone-2.sip.1und1.de. (udp)
sres_cache_get(0x14d52e0, A, "strato-iphone-1.sip.1und1.de.") called
nta: for "strato-iphone.de" query "strato-iphone-1.sip.1und1.de." A
sres_query(0x14da550, 0x7fa7880435f0, A, "strato-iphone-1.sip.1und1.de.") called
sres_send_dns_query(0x14da550, 0x7fa78805f990) called
sres_send_dns_query(0x14da550, 0x7fa78805f990) id=36318 A strato-iphone-1.sip.1und1.de. (to [127.0.0.1]:53)
sres_resolver_receive(0x14da550, 37) called
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
ANSWER RR received strato-iphone-1.sip.1und1.de. A IN 300 rdlen=4
AUTHORITY RR received 1und1.de. NS IN 84886 rdlen=15
AUTHORITY RR received 1und1.de. NS IN 84886 rdlen=6
sres_resolver_receive(0x14da550, 0x7fa78805f990) id=36318 (from [127.0.0.1]:53)
nta: strato-iphone-1.sip.1und1.de. IN A 212.227.18.136
nta(0x7fa7880435f0):  A 212.227.18.137
nta(0x7fa7880435f0):  A 212.227.18.197
nta(0x7fa7880435f0):  A 212.227.18.205
nta(0x7fa7880435f0):  A 212.227.18.206
nta(0x7fa7880435f0):  A 212.227.67.131
nta(0x7fa7880435f0):  A 212.227.67.132
nta(0x7fa7880435f0):  A 212.227.67.134
nta(0x7fa7880435f0):  A 212.227.67.201
nta(0x7fa7880435f0):  A 212.227.67.202
nta(0x7fa7880435f0):  A 212.227.67.204
nta(0x7fa7880435f0):  A 212.227.18.135
tport_tsend(0x7fa788034fa0) tpn = udp/212.227.18.136:5060
tport_resolve addrinfo = 212.227.18.136:5060
tport_by_addrinfo(0x7fa788034fa0): not found by name udp/212.227.18.136:5060
tport_vsend(0x7fa788034fa0): 872 bytes of 872 to udp/212.227.18.136:5060
tport_vsend returned 872
nta: sent REGISTER (12582343) to udp/212.227.18.136:5060
tport_pend(0x7fa788034fa0): pending 0x7fa7880aaae0 for udp/217.172.171.13:5080 (already 0)
nta: timer set to 32000 ms
nta: timer shortened to 1000 ms
tport_wakeup_pri(0x7fa788034fa0): events IN
tport_recv_event(0x7fa788034fa0)
tport_recv_iovec(0x7fa788034fa0) msg 0x7fa78001fd10 from (udp/217.172.171.13:5080) has 528 bytes, veclen = 1
tport_deliver(0x7fa788034fa0): msg 0x7fa78001fd10 (528 bytes) from udp/212.227.18.136:5080/sip next=(nil)
nta: received 401 Unauthorized for REGISTER (12582343)
nta: 401 Unauthorized is going to a transaction
nta_outgoing: RTT is 10.162 ms
tport_release(0x7fa788034fa0): 0x7fa7880aaae0 by 0x7fa7880435f0 with 0x7fa78001fd10
auth_digest_challenge_get(): got 3
auth_digest_a1() has A1 = MD5(thomas1:strato-iphone.de:meinpasswd) = c6c958107e411451fe8880d3e42330b1
A2 = MD5(REGISTER:sip:strato-iphone.de;transport=udp)
auth_response: 66a53a7552eb95e388d9c88413d6f4da = MD5(c6c958107e411451fe8880d3e42330b1:4dd57d2cf24de219e3a4b67ee77b6fb7091d4fd9:a9fc154a8935968d2d7aca6dbd10faa6) (qop=NONE)
nta: selecting scheme sip
sres_cache_get(0x14d52e0, SRV, "_sip._udp.strato-iphone.de.") called
sres_cache_get(0x14d52e0, SRV, "_sip._udp.strato-iphone.de.") returned 2 entries
nta: for "strato-iphone.de" query "_sip._udp.strato-iphone.de" SRV (cached)
nta: _sip._udp.strato-iphone.de IN SRV 0 0  5060 strato-iphone-1.sip.1und1.de. (udp)
nta: _sip._udp.strato-iphone.de IN SRV 0 0  5060 strato-iphone-2.sip.1und1.de. (udp)
sres_cache_get(0x14d52e0, A, "strato-iphone-1.sip.1und1.de.") called
sres_cache_get(0x14d52e0, A, "strato-iphone-1.sip.1und1.de.") returned 12 entries
nta: for "strato-iphone.de" query "strato-iphone-1.sip.1und1.de." A (cached)
nta: strato-iphone-1.sip.1und1.de. IN A 212.227.18.137
nta(0x7fa780020130):  A 212.227.18.197
nta(0x7fa780020130):  A 212.227.18.205
nta(0x7fa780020130):  A 212.227.18.206
nta(0x7fa780020130):  A 212.227.67.131
nta(0x7fa780020130):  A 212.227.67.132
nta(0x7fa780020130):  A 212.227.67.134
nta(0x7fa780020130):  A 212.227.67.201
nta(0x7fa780020130):  A 212.227.67.202
nta(0x7fa780020130):  A 212.227.67.204
nta(0x7fa780020130):  A 212.227.18.135
nta(0x7fa780020130):  A 212.227.18.136
tport_tsend(0x7fa788034fa0) tpn = udp/212.227.18.137:5060
tport_resolve addrinfo = 212.227.18.137:5060
tport_by_addrinfo(0x7fa788034fa0): not found by name udp/212.227.18.137:5060
tport_vsend(0x7fa788034fa0): 872 bytes of 872 to udp/212.227.18.137:5060
tport_vsend returned 872
nta: sent REGISTER (12582344) to udp/212.227.18.137:5060
tport_pend(0x7fa788034fa0): pending 0x152b3e0 for udp/217.172.171.13:5080 (already 0)
nua(0x7fa78000d970): event r_register 100 Request Authorized by Cache
nua: nua_application_event: entering
nua: nua_handle_magic: entering
tport_wakeup_pri(0x7fa788034fa0): events IN
tport_recv_event(0x7fa788034fa0)
tport_recv_iovec(0x7fa788034fa0) msg 0x7fa78001e570 from (udp/217.172.171.13:5080) has 445 bytes, veclen = 1
tport_deliver(0x7fa788034fa0): msg 0x7fa78001e570 (445 bytes) from udp/212.227.18.137:5080/sip next=(nil)
nta: received 403 Contact User und Anrufernummer verschieden for REGISTER (12582344)
nta: 403 Contact User und Anrufernummer verschieden is going to a transaction
nta_outgoing: RTT is 9.805 ms
tport_release(0x7fa788034fa0): 0x152b3e0 by 0x7fa780020130 with 0x7fa78001e570
nua(0x7fa78000d970): event r_register 403 Contact User und Anrufernummer verschieden
nua(0x7fa78000d970): removing register usage
nta_leg_destroy(0x7fa788057100)
nua: nua_application_event: entering
2011-05-19 22:26:54.837925 [ERR] sofia_reg.c:1771 mobile1 Registration Failed with status Contact User und Anrufernummer verschieden [403]. failure #3
nua: nua_handle_magic: entering
2011-05-19 22:26:55.797927 [WARNING] sofia_reg.c:425 mobile1 Failed Registration [0], setting retry to 240 seconds.
nta: timer set next to 4009 ms
nta: timer K fired, terminate REGISTER (12582343)
outgoing_reclaim_all((nil), (nil), 0x7fa78c8d7d00)
nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta: timer set next to 9 ms
nta: timer K fired, terminate REGISTER (12582344)
outgoing_reclaim_all((nil), (nil), 0x7fa78c8d7d00)
nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set

freeswitch at internal> /exit

ubuntu:/tmp# exit

Script done on Do 19 Mai 2011 22:28:40 CEST

Can your help please?

---------------
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 admin at blindi.net  Fri May 20 02:17:27 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 20 May 2011 00:17:27 +0200 (CEST)
Subject: [Freeswitch-users] Problem ringback does not working by
 orignatecalls
In-Reply-To: <BANLkTi=LS+9PWnNbwtpYOfz8K1nbCxAZ4Q@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180345430.9120@debian>
	<BANLkTi=LS+9PWnNbwtpYOfz8K1nbCxAZ4Q@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105200008440.6224@debian>

Hi Michael, i have a Ivr set.
The prompt: Press 1 to call me. Press 2 to call my girlfriend.
If the caller makes his selection, it is music after the transfer or a 
riningback.
For example: Please hold!! transfer the call, give music befor answering.

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 steveayre at gmail.com  Fri May 20 02:20:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 23:20:30 +0100
Subject: [Freeswitch-users] Problem ringback does not working by
	orignatecalls
In-Reply-To: <BANLkTi=LS+9PWnNbwtpYOfz8K1nbCxAZ4Q@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
	<B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>
	<BANLkTikgXO63tz38zujC-fZPttV0s9MjLQ@mail.gmail.com>
	<BANLkTimfxXFkBZr1WvspWWmv0BerLXs74A@mail.gmail.com>
	<BANLkTimKUGviDis8NPtnZm=6EqmvN7KGsg@mail.gmail.com>
	<Pine.LNX.4.64.1105180345430.9120@debian>
	<BANLkTi=LS+9PWnNbwtpYOfz8K1nbCxAZ4Q@mail.gmail.com>
Message-ID: <F810E155-3049-4543-A0E9-2765292C0BF3@gmail.com>

I'm not sure they bypass_media is applicable here either.

Steve on iPhone

On 19 May 2011, at 20:00, Michael Collins <msc at freeswitch.org> wrote:

> You can't answer the call and then later on send a "ring_ready". Once the call has been answered there is no reason to have the calling party hear ringing. I'm not sure what you are trying to do but on thing you can try is to use "pre_answer" instead of "answer" application.
> 
> -MC
> 
> On Tue, May 17, 2011 at 6:58 PM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Hi all, i generage a callback via originate.
> the orinatecommand:
> /usr/local/freeswitch/bin/fs_cli -x "bgapi originate {ignore_early_media=true,bypass_media=true,originate_retries=10,origination_caller_id_name=Callback,originate_retry_sleep_ms=60000,originate_timeout=900}loopback/XXXXXXX/callback_only_out &bridge(loopback/callback_only_sendex1/callback_only_send)"
> My phonenumber is ringing. here the prompt. I don.t here a transferrington or music.
> My extension:
> <extension name="callback_only_sendex1">
> <condition field="destination_number" expression="^callback_only_sendex1$">
>  <action application="answer" />
>  <action application="playback" data="silence_stream://10"/>
> <action application="set" data="ringback=${de-ring}"/>
> <action application="set" data="transfer_ringback=$${hold_music}"/>
> <action application="ring_ready"/>
> <action application="set" data="call_timeout=30"/>
>  <action application="set" data="hangup_after_bridge=true"/>
> 
>  <action application="set" data="ignore_early_media=true" />
> 
> <action application="bridge"
> data="loopback/0174XXXXX/disa"/>
> </condition>
> </extension>
> 
> 
> Can your help please thank you
> 
> 
> ---------------
> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing 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 nico at clickfono.com  Fri May 20 03:39:13 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Thu, 19 May 2011 19:39:13 -0400
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
Message-ID: <BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>

Actually, I get the exact same output from the console in my logfile
(log/freeswitch.log), I just do:

fsctl loglevel 3
sofia profile_name siptrace on
sofia tracelevel 3

I think you may use any loglevel, but that works fine, and I don't get all
the debugging output.



On Thu, May 19, 2011 at 12:48 PM, Steven Ayre <steveayre at gmail.com> wrote:

> No.
>
> If you need that it's best to use tshark, wireshark, ngrep or some similar
> tool.
>
> -Steve
>
>
>
> On 19 May 2011 17:40, Michal Kopacki <mkopacki at gmail.com> wrote:
>
>>      Sorry for interrupt, but is there any way to filter messages from sip
>> trace ? For example show only one ext/host/uri etc
>>
>> --
>> Michal
>>
>> ---------------
>>
>>  By default it only goes to console. 'sofia tracelevel debug' will let you
>> set a log level which'll let you log it to the logfile (you can choose any
>> of the log files).
>>
>> -Steve
>>
>>
>>  On 19 May 2011 15:57, Eric Beard <eric at loopfx.com> wrote:
>>
>>>  Hello,
>>>
>>>
>>>
>>> In fs_cli, I can issue this command:
>>>
>>>
>>>
>>> sofia global siptrace on
>>>
>>>
>>>
>>> And I see all SIP traffic.  But the SIP does not get logged to the actual
>>> log file.  How do I get the same output to be sent to the log?  I?ve
>>> searched through the Wiki but I?m not seeing a clear way to do it.
>>>
>>>
>>>
>>> Thanks!
>>>
>>>
>>>
>>> -----------------------
>>>
>>> *Eric Z. Beard, CTO*
>>>
>>> Loop LLC
>>>
>>> w (877) 850-2010 x9249
>>>
>>> m (727) 776-2768
>>>
>>> eric at loopfx.com
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> 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 list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 20 07:20:16 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 20 May 2011 09:20:16 +0600
Subject: [Freeswitch-users] CDR event
In-Reply-To: <D80E7467-7E54-4387-AB94-CD4FD3AD0577@ipeva.fr>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>	<4DD4CCCB.2080700@tagnet.ru>	<BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>	<4DD54422.7010507@tagnet.ru>
	<D80E7467-7E54-4387-AB94-CD4FD3AD0577@ipeva.fr>
Message-ID: <4DD5DDF0.9090903@tagnet.ru>

Hello!

     David, as I wrote I collect only A-leg records. This is the way my 
bill logic works. And I asked how to write custom A-leg CDR record while 
it is not available in normal way. And now I understood this is 
impossible so I need to change my bill logic :)

> Boris,
>
> I think we still wonder what additional custom CDR you need.
> You already know that one redirected call will fire 2 CDRs.
> One for the incoming call you need for inter-operator billing, and the 
> one for the outgoing redirected call you need to bill the customer.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 19/05/2011 ? 18:24, Boris Kovalenko a ?crit :
>
>> So there is not solution to fire additional custom CDR event? Ok, 
>> I'll look at XML CDR.
>>
>>> The XML CDR contains a history of the call, which might be more 
>>> suited to you. One CDR but it'll contain everything you need. You 
>>> can then choose to create 2 DB records if that's what you wish to do.
>>>
>>> Otherwise write all relevant variables to a CDR (if you're 
>>> overwriting one in the dialplan set it to another variable first 
>>> otherwise it won't be available any more when mod_cdr_csv writes it).
>>>
>>> -Steve
>>>
>>>
>>> On 19 May 2011 08:54, Boris Kovalenko <boris at tagnet.ru 
>>> <mailto:boris at tagnet.ru>> wrote:
>>>
>>>     Hello!
>>>
>>>         It is the way I try to do manual redirect. May be I do it
>>>     wrong, so may be somebody tell me another way. So... I use only
>>>     A-leg accounting records. One of my users set the redirect from
>>>     its phone to cellular phone (PAP2T device). So, when call is
>>>     arrived from PBX
>>>     and redirected by user, in A-leg CDR record I see only
>>>     _original_ information: pbx_caller_id_number and
>>>     local_destination_number. And this is wrong for billing. With
>>>     help of CDR I want to emulate two calls scenario:
>>>     First CDR record - PBX -> user number
>>>     Second CDR record - user number -> redirected number
>>>
>>>
>>>>     I don't understand why you would need this.
>>>>
>>>>     What would the difference between the variables be?
>>>>
>>>>     -Steve
>>>>
>>>>
>>>>
>>>>     On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru
>>>>     <mailto:boris at tagnet.ru>> wrote:
>>>>
>>>>         Hello!
>>>>
>>>>             I know :) But with one call I get only one CDR record
>>>>         for A leg and
>>>>         one record for B leg. I need to get 2 records for A leg
>>>>         with two
>>>>         different sets of variables. Is this possible?
>>>>
>>>>         > There's no 'standard' CDR. The XML CDR will contain
>>>>         everything,
>>>>         > including all variables. If you mean the default
>>>>         mod_cdr_csv, you can
>>>>         > customise which variables are saved in its configuration.
>>>>         >
>>>>         > -Steve
>>>>         >
>>>>         >
>>>>         >
>>>>         > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru
>>>>         <mailto:boris at tagnet.ru>>  wrote:
>>>>         >> Hello!
>>>>         >>
>>>>         >> I don't know what is asterisk fork_cdr :) I want to
>>>>         bridge two calls and
>>>>         >> get 2 cdr records for A leg: one the standard cdr, and
>>>>         another with
>>>>         >> variables I need.
>>>>         >>
>>>>         >>> I suppose he's asking for asterisk analog of fork_cdr
>>>>         >>>
>>>>         >>> 2011/5/17 Michael Collins<msc at freeswitch.org
>>>>         <mailto:msc at freeswitch.org>>:
>>>>         >>>> You want your Lua script to *send* an event? It most
>>>>         certainly can fire
>>>>         >>>> events:
>>>>         >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>>>         >>>> I suppose the question is: what do you want to have
>>>>         happen when you fire an
>>>>         >>>> event? What will receive this event?
>>>>         >>>> -MC
>>>>         >>>>
>>>>         >>>> On Sat, May 14, 2011 at 12:59 AM, Boris
>>>>         Kovalenko<boris at tagnet.ru <mailto:boris at tagnet.ru>>    wrote:
>>>>         >>>>> Hello!
>>>>         >>>>>
>>>>         >>>>>       May I fire custom cdr event for a LUA script?
>>>>         I've read
>>>>         >>>>> http://wiki.freeswitch.org/wiki/Event_list but
>>>>         haven't found cdr events.
>>>>         >>>>> Would You please point me to proper documentation or
>>>>         give an example?
>>>>         >>>>>
>>>>         >>>>> --
>>>>         >>>>> 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 <http://www.freeswitch.org/>
>>>>         >>>> _______________________________________________
>>>>         >>>> FreeSWITCH-users mailing list
>>>>         >>>> FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         >>>>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         >>>>
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         >>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>         >>>>
>>>>         >>>>
>>>>         >>> _______________________________________________
>>>>         >>> FreeSWITCH-users mailing list
>>>>         >>> FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         >>>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         >>>
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         >>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>         >>
>>>>         >> --
>>>>         >> ? ?????????,
>>>>         >>    ????? ?????????
>>>>         >>    ??? "??????"
>>>>         >>    ???. +7 (3435) 230001
>>>>         >>    ???? +7 (3435) 230005
>>>>         >>
>>>>         >>
>>>>         >> _______________________________________________
>>>>         >> FreeSWITCH-users mailing list
>>>>         >> FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         >>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         >>
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         >> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>         >>
>>>>         > _______________________________________________
>>>>         > FreeSWITCH-users mailing list
>>>>         > FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         >
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         > http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>>         --
>>>>         ? ?????????,
>>>>           ????? ?????????
>>>>           ??? "??????"
>>>>           ???. +7 (3435) 230001
>>>>           ???? +7 (3435) 230005
>>>>
>>>>
>>>>         _______________________________________________
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>>
>>>>     _______________________________________________
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>
>>>
>>>     -- 
>>>     ? ?????????,
>>>        ????? ?????????
>>>        ??? "??????"
>>>        ???. +7 (3435) 230001
>>>        ???? +7 (3435) 230005
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> -- 
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    ???. +7 (3435) 230001
>>    ???? +7 (3435) 230005
>> _______________________________________________
>> 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


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

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From msc at freeswitch.org  Fri May 20 07:24:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 May 2011 20:24:21 -0700
Subject: [Freeswitch-users] call somebody into conference after
	conferenceestablished
In-Reply-To: <BANLkTikPq5GteE1Y5w56oL_=oAQ3ieVC1w@mail.gmail.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<201105121532017963789@asiainfo-linkage.com>
	<BANLkTi=wbd8ivSA4Gom3P2QsJ9AY7GRJUA@mail.gmail.com>
	<BANLkTikPq5GteE1Y5w56oL_=oAQ3ieVC1w@mail.gmail.com>
Message-ID: <BANLkTimMVoF3RDjo1Zi2FG0K=ZwOXUyZKg@mail.gmail.com>

Okay, I did something 100% in dialplan with some conference.conf.xml
additions. It's an example of how crazy powerful the dialplan really is.
First, you need to add these to your conference.conf.xml:

<caller-controls>
...
    <group name="plain">
      <control action="mute" digits="0"/>
      <control action="energy up" digits="9"/>
      <control action="energy equ" digits="8"/>
      <control action="energy dn" digits="7"/>
      <control action="vol talk up" digits="3"/>
      <control action="vol talk zero" digits="2"/>
      <control action="vol talk dn" digits="1"/>
      <control action="vol listen up" digits="6"/>
      <control action="vol listen zero" digits="5"/>
      <control action="vol listen dn" digits="4"/>
    </group>
...
</caller-controls>

And add a new conference profile:

<profiles>
...
    <profile name="simple">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="16000"/>
      <param name="interval" value="10"/>
      <param name="energy-level" value="300"/>
      <param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>
      <param name="muted-sound" value="conference/conf-muted.wav"/>
      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
      <param name="alone-sound" value="conference/conf-alone.wav"/>
      <param name="moh-sound" value="$${hold_music}"/>
      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
      <param name="exit-sound"
value="tone_stream://%(500,0,300,200,100,50,25)"/>
      <param name="kicked-sound" value="conference/conf-kicked.wav"/>
      <param name="locked-sound" value="conference/conf-locked.wav"/>
      <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
      <param name="is-unlocked-sound"
value="conference/conf-is-unlocked.wav"/>
      <param name="pin-sound" value="conference/conf-pin.wav"/>
      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
      <param name="caller-id-name" value="$${outbound_caller_name}"/>
      <param name="caller-id-number" value="$${outbound_caller_id}"/>
      <param name="caller-controls" value="plain"/>
      <param name="comfort-noise" value="true"/>
    </profile>
...
</profiles>

Add these extensions to your dialplan. I just put them in
conf/dialplan/default/01_Conf_Add_Caller.xml:

<include>
  <!-- Sample dialplan for sending member or moderator to a conference where
he/she can add/remove calls -->
  <!-- Dial 46xx for member, *46xx for moderator -->
    <extension name="Simple add-caller conf: member">
      <condition field="destination_number" expression="^(46\d{2})$">
<action application="answer"/>
<action application="conference" data="$1 at simple"/>
     </condition>
    </extension>

    <extension name="Simple add-caller conf: moderator">
      <condition field="destination_number" expression="^\*(46\d{2}$)">
<!-- set up a few bind_digit_action (BDA) bindings for the moderator -->
        <action application="bind_digit_action"
data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__$1 XML default"/>
<action application="bind_digit_action"
data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__$1 XML
default"/>
<action application="digit_action_set_realm" data="moderator"/>
<action application="conference" data="$1 at simple"/>
      </condition>
    </extension>

    <extension name="Add new OB call to conference">
      <condition field="destination_number"
expression="^ASK_FOR_NUMBER__(\d+)$">
<!-- ask caller for a number + #, collect into ${target_num} variable -->
<action application="play_and_get_digits" data="4 11 1 5000 #
ivr/ivr-enter_destination_telephone_number.wav
ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
<!-- add this call to the conference -->
<action application="execute_extension"
data="ADD_CALL_TO_CONF__${target_num}"/>
      </condition>
    </extension>

    <extension name="Remove last OB call added to conference">
      <condition field="destination_number"
expression="^CANCEL_LAST_CALL__(\d+)$">
<!-- remove this call from the conference -->
<action application="set" data="res=${uuid_kill ${new_uuid}}"/>
      </condition>
    </extension>

<extension name="add that call">
<!-- if we have a four-digit number then attempt to dial it as a user ...
-->
<condition field="destination_number"
expression="^ADD_CALL_TO_CONF__(\d\d\d\d)$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="set" data="res=${bgapi originate
{origination_uuid=${new_uuid}}user/$1 &conference(${conference_name})}"/>
</condition>
<!-- if we have a five+ digit number then attempt to dial it as a gw
connection ... -->
<condition field="destination_number"
expression="^ADD_CALL_TO_CONF__(\d\d\d\d)$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="set" data="res=${bgapi originate
{origination_uuid=${new_uuid}}sofia/gateway/mygw/$1
&conference(${conference_name})}"/>
<!-- Alternatively, you can just use loopback, but that creates three call
legs instead of one, so be warned -->
<!--
<action application="set" data="res=${bgapi originate
{origination_uuid=${new_uuid}}loopback/$1
&conference(${conference_name})}"/>
-->
</condition>
<!-- ... otherwise inform moderator that the operation was not exactly
successful -->
<condition field="destination_number" expression="^ADD_CALL_TO_CONF__$">
<action application="playback" data="ivr/ivr-dude_you_suck.wav"/>
</condition>
</extension>
</include>

How it works:
Moderator dials *46xx to go into a conference. ("Normal" users dial 46xx and
they cannot add a new call.) Moderator dials *1 and gets a dialog asking to
enter the destination number and # key. Moderator keys in a phone number (or
extension number) and presses #. System makes call and drops it into the
conference. As a bonus feature I added *2 which will disconnect the most
recently added phone call.

I did this mostly as a proof of concept, but there's no reason you can't
take this code and put it into production. In fact, you could add a PIN or
user id (or both) checker so that just anyone can't add a call to a
conference. If you guys actually like this then I'll add it to the wiki or
something so everyone else can reference it.

Enjoy,
MC


2011/5/16 fieldpeak <fieldpeak at gmail.com>

> Hi Gurus,
>
> Is there anybody can help this issue? thanks!
>
> Regards,
> Charles
>
> ? 2011?5?14? ??3:54?fieldpeak <fieldpeak at gmail.com>???
>
> Hi Michael,
>>
>> Thanks for your detailed suggestion.
>>
>> i test it as below steps, it works fine, however, there is a limiation
>> that i have to fill fixed callee number in conference.conf.xml
>> (data="execute_extension 5001"), while actually i what to dial arbitrary
>> number after "*" dynamically(e.g *1234, *2345), i can think out it could
>> realize by a IVR script like "data="execute_extension IVR_scripts" and after
>> IVR give some voice prompt (pls input the callee number) then ... however,
>> considering simplicity, can i convey the DTMF into the dialplan (just
>> directly press *1234, and then 1234 ring...)? could you please provide any
>> hints or any suggestion... Thanks a lot!
>>
>> 1. in conference.conf.xml, set as below,
>> <control action="execute_application" digits="*" data="execute_extension
>> 5001"/>
>>
>> 2. in dial plan,
>>
>> <!-- for confernce named conf100 -->
>> <extension name="conf100" >
>>    <condition field="destination_number" expression="^666$">
>>        <action application="answer"/>
>>        <action application="set" data="conference_user_list=|"/>
>>        <action application="conference" data="conf100 at default"/>
>>    </condition>
>> </extension>
>>
>> and
>>
>> <!-- for dial out 5001 from the conference conf100 -->
>>
>> <extension name="5_to_conf_out" >
>>    <condition field="destination_number" expression="^(5\d+)$">
>>        <!-- <action application="conference" data="conf100 dial
>> sofia/internal/5001 at 192.168.200.201 100 Conference_Caller_Name"/> -->
>>      <!-- <action application="bridge" data="sofia/internal/$
>> 1 at 192.168.200.201"/> -->
>>      <action application="set" data="api_result=${conference(conf100 dial
>> sofia/internal/$1 at 192.168.200.201 100 Conference_Caller_Name)}"/>
>>    </condition>
>> </extension>
>>
>> 3. register a extension 3001, call 666 and join a conference, and then
>> press "*", the FS will call 5001 on a IAD.
>> then 5001 and 3001 join the same conference.
>>
>>
>>  2011/5/12 liuyp2 <liuyp2 at asiainfo-linkage.com>
>>
>>>   conference Your-Conf-Name dial user/1002
>>>
>>>  ------------------------------
>>>  liuyp2
>>> 2011-05-12
>>>  ------------------------------
>>>  *????* fieldpeak
>>> *?????* 2011-05-12 11:11:41
>>> *????* FreeSWITCH-users
>>> *???*
>>> *???* [Freeswitch-users] call somebody into conference after
>>> conferenceestablished
>>>
>>> Hi Gurus,
>>>
>>> i dial 666 and enter a conference, then i need call some body's phone
>>> number to join him into this conference...
>>> is there anyone can advise how can realize this scenario?
>>>
>>> thanks.
>>>
>>> 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
>
>
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From darknesslabs at gmail.com  Fri May 20 07:44:03 2011
From: darknesslabs at gmail.com (darknesslabs at gmail.com)
Date: Fri, 20 May 2011 03:44:03 +0000
Subject: [Freeswitch-users] call somebody into conference
	afterconferenceestablished
In-Reply-To: <BANLkTimMVoF3RDjo1Zi2FG0K=ZwOXUyZKg@mail.gmail.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com><201105121532017963789@asiainfo-linkage.com><BANLkTi=wbd8ivSA4Gom3P2QsJ9AY7GRJUA@mail.gmail.com><BANLkTikPq5GteE1Y5w56oL_=oAQ3ieVC1w@mail.gmail.com><BANLkTimMVoF3RDjo1Zi2FG0K=ZwOXUyZKg@mail.gmail.com>
Message-ID: <1412527818-1305863043-cardhu_decombobulator_blackberry.rim.net-1696936367-@b5.c17.bise6.blackberry>

Super useful, please do add it to the wiki. 
Sent via BlackBerry

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Thu, 19 May 2011 20:24:21 
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] call somebody into conference after
	conferenceestablished

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


From u2nsam at gmail.com  Fri May 20 09:52:21 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 20 May 2011 11:22:21 +0530
Subject: [Freeswitch-users] VM
Message-ID: <BANLkTinQfmN0J012e9xrad76uCLMkD8bnA@mail.gmail.com>

Hi,


Is there any method to create voicemail box on the fly even though there is
no such user on FS.

FS will just create a mailbox with the user and stores the voicemail in to
the desired mailbox.


Regards
Sam
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From anton.vazir at gmail.com  Fri May 20 11:08:17 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 20 May 2011 12:08:17 +0500
Subject: [Freeswitch-users] CDR event
In-Reply-To: <4DD5DDF0.9090903@tagnet.ru>
References: <4DCE3654.7080002@tagnet.ru>
	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
	<4DD3849F.40409@tagnet.ru>
	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>
	<4DD48A99.4000700@tagnet.ru>
	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>
	<4DD4CCCB.2080700@tagnet.ru>
	<BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>
	<4DD54422.7010507@tagnet.ru>
	<D80E7467-7E54-4387-AB94-CD4FD3AD0577@ipeva.fr>
	<4DD5DDF0.9090903@tagnet.ru>
Message-ID: <BANLkTinNF6Xh1kwe_niegb9j9DbVXUwUiQ@mail.gmail.com>

Actually, for proper call-forward accounting you need second CDR for
SECOND A-Leg (emulated)
I have an example, how I do that in Asterisk - there is a
Dial(LOCAL/XXX) option, so Dial acts as second Dial, firing one more
CDR. Like you dial out originating from user, which have been dialed
in (B-Leg than one more A-Leg)
I will also need to do the give in FS - will see what mod_local gives.

Borya, i trust you should describe your task more clear, maybe with
examples, as I see, from the above there was no understanding of what
exactly you try to achieve, and why you need second CDR, that why you
don't get a specific answer...

2011/5/20 Boris Kovalenko <boris at tagnet.ru>:
> Hello!
>
> ??? David, as I wrote I collect only A-leg records. This is the way my bill
> logic works. And I asked how to write custom A-leg CDR record while it is
> not available in normal way. And now I understood this is impossible so I
> need to change my bill logic :)
>
> Boris,
> I think we still wonder what additional custom CDR you need.
> You already know that one redirected call will fire 2 CDRs.
> One for the incoming call you need for inter-operator billing, and the one
> for the outgoing redirected call you need to bill the customer.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 19/05/2011 ? 18:24, Boris Kovalenko a ?crit :
>
> So there is not solution to fire additional custom CDR event? Ok, I'll look
> at XML CDR.
>
> The XML CDR contains a history of the call, which might be more suited to
> you. One CDR but it'll contain everything you need. You can then choose to
> create 2 DB records if that's what you wish to do.
>
> Otherwise write all relevant variables to a CDR (if you're overwriting one
> in the dialplan set it to another variable first otherwise it won't be
> available any more when mod_cdr_csv writes it).
>
> -Steve
>
>
> On 19 May 2011 08:54, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>> Hello!
>>
>> ??? It is the way I try to do manual redirect. May be I do it wrong, so
>> may be somebody tell me another way. So... I use only A-leg accounting
>> records. One of my users set the redirect from its phone to cellular phone
>> (PAP2T device). So, when call is arrived from PBX
>> and redirected by user, in A-leg CDR record I see only _original_
>> information: pbx_caller_id_number and local_destination_number. And this is
>> wrong for billing. With help of CDR I want to emulate two calls scenario:
>> First CDR record - PBX -> user number
>> Second CDR record - user number -> redirected number
>>
>>
>> I don't understand why you would need this.
>>
>> What would the difference between the variables be?
>>
>> -Steve
>>
>>
>>
>> On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>
>>> Hello!
>>>
>>> ? ? I know :) But with one call I get only one CDR record for A leg and
>>> one record for B leg. I need to get 2 records for A leg with two
>>> different sets of variables. Is this possible?
>>>
>>> > There's no 'standard' CDR. The XML CDR will contain everything,
>>> > including all variables. If you mean the default mod_cdr_csv, you can
>>> > customise which variables are saved in its configuration.
>>> >
>>> > -Steve
>>> >
>>> >
>>> >
>>> > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>> >> Hello!
>>> >>
>>> >> I don't know what is asterisk fork_cdr :) I want to bridge two calls
>>> >> and
>>> >> get 2 cdr records for A leg: one the standard cdr, and another with
>>> >> variables I need.
>>> >>
>>> >>> I suppose he's asking for asterisk analog of fork_cdr
>>> >>>
>>> >>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>> >>>> You want your Lua script to *send* an event? It most certainly can
>>> >>>> fire
>>> >>>> events:
>>> >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>> >>>> I suppose the question is: what do you want to have happen when you
>>> >>>> fire an
>>> >>>> event? What will receive this event?
>>> >>>> -MC
>>> >>>>
>>> >>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>
>>> >>>> ?wrote:
>>> >>>>> Hello!
>>> >>>>>
>>> >>>>> ? ? ? May I fire custom cdr event for a LUA script? I've read
>>> >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr
>>> >>>>> events.
>>> >>>>> Would You please point me to proper documentation or give an
>>> >>>>> example?
>>> >>>>>
>>> >>>>> --
>>> >>>>> 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 list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>
>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at 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
>>> >>
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Fri May 20 11:48:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 20 May 2011 00:48:57 -0700
Subject: [Freeswitch-users] VM
In-Reply-To: <BANLkTinQfmN0J012e9xrad76uCLMkD8bnA@mail.gmail.com>
References: <BANLkTinQfmN0J012e9xrad76uCLMkD8bnA@mail.gmail.com>
Message-ID: <BANLkTimrsxWPYHQjBFAVb0_1LNPp2XbQdQ@mail.gmail.com>

You'll need a directory entry in order to have a mailbox. You can achieve
that with static XML or dynamically with mod_xml_curl. You probably need to
use the latter and have some means of telling FS that a particular "user"
exists (even if it really does not) so that you can emulate the behavior you
want.

-MC

On Thu, May 19, 2011 at 10:52 PM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
>
> Is there any method to create voicemail box on the fly even though there is
> no such user on FS.
>
> FS will just create a mailbox with the user and stores the voicemail in to
> the desired mailbox.
>
>
> Regards
> Sam
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anton.vazir at gmail.com  Fri May 20 12:39:45 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 20 May 2011 13:39:45 +0500
Subject: [Freeswitch-users] CDR event
In-Reply-To: <BANLkTinNF6Xh1kwe_niegb9j9DbVXUwUiQ@mail.gmail.com>
References: <4DCE3654.7080002@tagnet.ru>
	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>
	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>
	<4DD3849F.40409@tagnet.ru>
	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>
	<4DD48A99.4000700@tagnet.ru>
	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>
	<4DD4CCCB.2080700@tagnet.ru>
	<BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>
	<4DD54422.7010507@tagnet.ru>
	<D80E7467-7E54-4387-AB94-CD4FD3AD0577@ipeva.fr>
	<4DD5DDF0.9090903@tagnet.ru>
	<BANLkTinNF6Xh1kwe_niegb9j9DbVXUwUiQ@mail.gmail.com>
Message-ID: <BANLkTik-DfKkXGty4pyuP0PNfNq0UDEDkw@mail.gmail.com>

i meant mod_loopback, not mod_local

2011/5/20 Anton VG <anton.vazir at gmail.com>:
> Actually, for proper call-forward accounting you need second CDR for
> SECOND A-Leg (emulated)
> I have an example, how I do that in Asterisk - there is a
> Dial(LOCAL/XXX) option, so Dial acts as second Dial, firing one more
> CDR. Like you dial out originating from user, which have been dialed
> in (B-Leg than one more A-Leg)
> I will also need to do the give in FS - will see what mod_local gives.
>
> Borya, i trust you should describe your task more clear, maybe with
> examples, as I see, from the above there was no understanding of what
> exactly you try to achieve, and why you need second CDR, that why you
> don't get a specific answer...
>
> 2011/5/20 Boris Kovalenko <boris at tagnet.ru>:
>> Hello!
>>
>> ??? David, as I wrote I collect only A-leg records. This is the way my bill
>> logic works. And I asked how to write custom A-leg CDR record while it is
>> not available in normal way. And now I understood this is impossible so I
>> need to change my bill logic :)
>>
>> Boris,
>> I think we still wonder what additional custom CDR you need.
>> You already know that one redirected call will fire 2 CDRs.
>> One for the incoming call you need for inter-operator billing, and the one
>> for the outgoing redirected call you need to bill the customer.
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 19/05/2011 ? 18:24, Boris Kovalenko a ?crit :
>>
>> So there is not solution to fire additional custom CDR event? Ok, I'll look
>> at XML CDR.
>>
>> The XML CDR contains a history of the call, which might be more suited to
>> you. One CDR but it'll contain everything you need. You can then choose to
>> create 2 DB records if that's what you wish to do.
>>
>> Otherwise write all relevant variables to a CDR (if you're overwriting one
>> in the dialplan set it to another variable first otherwise it won't be
>> available any more when mod_cdr_csv writes it).
>>
>> -Steve
>>
>>
>> On 19 May 2011 08:54, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>
>>> Hello!
>>>
>>> ??? It is the way I try to do manual redirect. May be I do it wrong, so
>>> may be somebody tell me another way. So... I use only A-leg accounting
>>> records. One of my users set the redirect from its phone to cellular phone
>>> (PAP2T device). So, when call is arrived from PBX
>>> and redirected by user, in A-leg CDR record I see only _original_
>>> information: pbx_caller_id_number and local_destination_number. And this is
>>> wrong for billing. With help of CDR I want to emulate two calls scenario:
>>> First CDR record - PBX -> user number
>>> Second CDR record - user number -> redirected number
>>>
>>>
>>> I don't understand why you would need this.
>>>
>>> What would the difference between the variables be?
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 19 May 2011 04:12, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>>
>>>> Hello!
>>>>
>>>> ? ? I know :) But with one call I get only one CDR record for A leg and
>>>> one record for B leg. I need to get 2 records for A leg with two
>>>> different sets of variables. Is this possible?
>>>>
>>>> > There's no 'standard' CDR. The XML CDR will contain everything,
>>>> > including all variables. If you mean the default mod_cdr_csv, you can
>>>> > customise which variables are saved in its configuration.
>>>> >
>>>> > -Steve
>>>> >
>>>> >
>>>> >
>>>> > On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>>> >> Hello!
>>>> >>
>>>> >> I don't know what is asterisk fork_cdr :) I want to bridge two calls
>>>> >> and
>>>> >> get 2 cdr records for A leg: one the standard cdr, and another with
>>>> >> variables I need.
>>>> >>
>>>> >>> I suppose he's asking for asterisk analog of fork_cdr
>>>> >>>
>>>> >>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>>> >>>> You want your Lua script to *send* an event? It most certainly can
>>>> >>>> fire
>>>> >>>> events:
>>>> >>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>>> >>>> I suppose the question is: what do you want to have happen when you
>>>> >>>> fire an
>>>> >>>> event? What will receive this event?
>>>> >>>> -MC
>>>> >>>>
>>>> >>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>
>>>> >>>> ?wrote:
>>>> >>>>> Hello!
>>>> >>>>>
>>>> >>>>> ? ? ? May I fire custom cdr event for a LUA script? I've read
>>>> >>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr
>>>> >>>>> events.
>>>> >>>>> Would You please point me to proper documentation or give an
>>>> >>>>> example?
>>>> >>>>>
>>>> >>>>> --
>>>> >>>>> 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 list
>>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>>
>>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>> http://www.freeswitch.org
>>>> >>>>
>>>> >>>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at 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
>>>> >>
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at 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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 ttobin at ubq.thrupoint.net  Fri May 20 16:20:48 2011
From: ttobin at ubq.thrupoint.net (Tim Tobin)
Date: Fri, 20 May 2011 13:20:48 +0100
Subject: [Freeswitch-users] G.729A licencing validation error
In-Reply-To: <BANLkTimpEcJyNxJA-Nno0gFyU13PO0MMEA@mail.gmail.com>
References: <BANLkTi=42gOrfQqvKdsT4-4R-refv1D=Eg@mail.gmail.com>
	<BANLkTimpEcJyNxJA-Nno0gFyU13PO0MMEA@mail.gmail.com>
Message-ID: <BANLkTikLDHyT5osc-W0_L0cxy19_jsp8bA@mail.gmail.com>

Hi Michael,

Thanks for the response. Validation is still failing. The issue seems to be
that the machine on which FreeSwitch is running has no direct access to the
internet. This isn't a machine i control and so can't change this. As i
understand things we could potentially resolve this issue by tunnelling HTTP
and DNS traffic. I've tried to set up DNS tunnelling but have so far been
unable to get this to work. I guess another approach would be to simply add
entries to the hosts file.

Also, am i right in assuming that, as far as licensing is concerned, once
the license has been validated we no longer need internet access?

Tim

On 19 May 2011 20:03, Michael Collins <msc at freeswitch.org> wrote:

> Can you try validating again and let us know if it is still not working?
> -MC
>
> On Tue, May 17, 2011 at 7:52 AM, Tim Tobin <ttobin at ubq.thrupoint.net>wrote:
>
>> Hi,
>>
>> I've just purchased a license for the G.729 codec module. I've uploaded
>> fsg729-194-installer to my freeswitch machine and have executed it. When i
>> try to validate i get the following response:
>>
>> Failed with error code 6
>>
>>
>> Does anybody know why this might be?
>>
>> Cheers,
>>
>> --------------------
>> Note: The information contained in this message may be privileged and confidential
>> and protected from disclosure. If the reader of this message is not the intended
>> recipient, or an employee or agent responsible for delivering this message to the
>> intended recipient, you are hereby notified that any dissemination, distribution or
>> copying of this communication is strictly prohibited. If you have received this
>> communication in error, please notify us immediately by replying to the message and
>> deleting it from your computer. Thank you. ThruPoint, Inc. - U5MJHZGXCFFDOYV
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

--------------------
Note: The information contained in this message may be privileged and confidential 
and protected from disclosure. If the reader of this message is not the intended 
recipient, or an employee or agent responsible for delivering this message to the 
intended recipient, you are hereby notified that any dissemination, distribution or 
copying of this communication is strictly prohibited. If you have received this 
communication in error, please notify us immediately by replying to the message and 
deleting it from your computer. Thank you. ThruPoint, Inc. - U5MJHZGXCFFDOYV
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From boris at tagnet.ru  Fri May 20 16:45:14 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 20 May 2011 18:45:14 +0600
Subject: [Freeswitch-users] CDR event
In-Reply-To: <BANLkTik-DfKkXGty4pyuP0PNfNq0UDEDkw@mail.gmail.com>
References: <4DCE3654.7080002@tagnet.ru>	<BANLkTikM=t=0XqEoOYdEzsS7TLB=KVZdOg@mail.gmail.com>	<BANLkTi=J_D0nCZ3yvPqM_hCVWUqAqiPNSQ@mail.gmail.com>	<4DD3849F.40409@tagnet.ru>	<BANLkTinuWT-6SYCxHR1BW7jVNYG_9h_1KA@mail.gmail.com>	<4DD48A99.4000700@tagnet.ru>	<BANLkTimPS=J5LdTWnrqDe7xqBBNvdo+53Q@mail.gmail.com>	<4DD4CCCB.2080700@tagnet.ru>	<BANLkTi=gmgg-CJVJVNrK75QYWHWd7Md+rw@mail.gmail.com>	<4DD54422.7010507@tagnet.ru>	<D80E7467-7E54-4387-AB94-CD4FD3AD0577@ipeva.fr>	<4DD5DDF0.9090903@tagnet.ru>	<BANLkTinNF6Xh1kwe_niegb9j9DbVXUwUiQ@mail.gmail.com>
	<BANLkTik-DfKkXGty4pyuP0PNfNq0UDEDkw@mail.gmail.com>
Message-ID: <4DD6625A.6020304@tagnet.ru>

Hello!

     Yes, you're right. This is my poor English. I will try again.

I have a user with PAP2T which set uncoditional call forward. To be more 
concretic, calling number is 1234, called number is 1111 and forwarded 
number is 2222, manual-redirect=true. I use mod_cdr_cvs for CDR logging 
and log only A-legs for billing purposes. So, when there is a call 1234 
-> 1111 it is redirected to my redirect context and there I do a bridge 
1234 -> 2222. After that I will have one A-leg CDR record where 
caller_id_number is 1234 and destination_number is 2222. IFAIK I may use 
mod_loopback, but I still will have only one A-leg CDR and two (???) 
B-leg CDRs. With my current billing logic this is not what I need. I 
need two A-leg CDRs, first with 1234 -> 1111 and second with 1111 -> 
2222. Hope I was concretic and more clear now :)

> i meant mod_loopback, not mod_local
>
> 2011/5/20 Anton VG<anton.vazir at gmail.com>:
>> Actually, for proper call-forward accounting you need second CDR for
>> SECOND A-Leg (emulated)
>> I have an example, how I do that in Asterisk - there is a
>> Dial(LOCAL/XXX) option, so Dial acts as second Dial, firing one more
>> CDR. Like you dial out originating from user, which have been dialed
>> in (B-Leg than one more A-Leg)
>> I will also need to do the give in FS - will see what mod_local gives.
>>
>> Borya, i trust you should describe your task more clear, maybe with
>> examples, as I see, from the above there was no understanding of what
>> exactly you try to achieve, and why you need second CDR, that why you
>> don't get a specific answer...
>>
>> 2011/5/20 Boris Kovalenko<boris at tagnet.ru>:
>>> Hello!
>>>
>>>      David, as I wrote I collect only A-leg records. This is the way my bill
>>> logic works. And I asked how to write custom A-leg CDR record while it is
>>> not available in normal way. And now I understood this is impossible so I
>>> need to change my bill logic :)
>>>
>>> Boris,
>>> I think we still wonder what additional custom CDR you need.
>>> You already know that one redirected call will fire 2 CDRs.
>>> One for the incoming call you need for inter-operator billing, and the one
>>> for the outgoing redirected call you need to bill the customer.
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 19/05/2011 ? 18:24, Boris Kovalenko a ?crit :
>>>
>>> So there is not solution to fire additional custom CDR event? Ok, I'll look
>>> at XML CDR.
>>>
>>> The XML CDR contains a history of the call, which might be more suited to
>>> you. One CDR but it'll contain everything you need. You can then choose to
>>> create 2 DB records if that's what you wish to do.
>>>
>>> Otherwise write all relevant variables to a CDR (if you're overwriting one
>>> in the dialplan set it to another variable first otherwise it won't be
>>> available any more when mod_cdr_csv writes it).
>>>
>>> -Steve
>>>
>>>
>>> On 19 May 2011 08:54, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>>> Hello!
>>>>
>>>>      It is the way I try to do manual redirect. May be I do it wrong, so
>>>> may be somebody tell me another way. So... I use only A-leg accounting
>>>> records. One of my users set the redirect from its phone to cellular phone
>>>> (PAP2T device). So, when call is arrived from PBX
>>>> and redirected by user, in A-leg CDR record I see only _original_
>>>> information: pbx_caller_id_number and local_destination_number. And this is
>>>> wrong for billing. With help of CDR I want to emulate two calls scenario:
>>>> First CDR record - PBX ->  user number
>>>> Second CDR record - user number ->  redirected number
>>>>
>>>>
>>>> I don't understand why you would need this.
>>>>
>>>> What would the difference between the variables be?
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 19 May 2011 04:12, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>>>> Hello!
>>>>>
>>>>>      I know :) But with one call I get only one CDR record for A leg and
>>>>> one record for B leg. I need to get 2 records for A leg with two
>>>>> different sets of variables. Is this possible?
>>>>>
>>>>>> There's no 'standard' CDR. The XML CDR will contain everything,
>>>>>> including all variables. If you mean the default mod_cdr_csv, you can
>>>>>> customise which variables are saved in its configuration.
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 18 May 2011 09:34, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>>>>>> Hello!
>>>>>>>
>>>>>>> I don't know what is asterisk fork_cdr :) I want to bridge two calls
>>>>>>> and
>>>>>>> get 2 cdr records for A leg: one the standard cdr, and another with
>>>>>>> variables I need.
>>>>>>>
>>>>>>>> I suppose he's asking for asterisk analog of fork_cdr
>>>>>>>>
>>>>>>>> 2011/5/17 Michael Collins<msc at freeswitch.org>:
>>>>>>>>> You want your Lua script to *send* an event? It most certainly can
>>>>>>>>> fire
>>>>>>>>> events:
>>>>>>>>> http://wiki.freeswitch.org/wiki/Lua#Sending_an_Event
>>>>>>>>> I suppose the question is: what do you want to have happen when you
>>>>>>>>> fire an
>>>>>>>>> event? What will receive this event?
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>> On Sat, May 14, 2011 at 12:59 AM, Boris Kovalenko<boris at tagnet.ru>
>>>>>>>>>   wrote:
>>>>>>>>>> Hello!
>>>>>>>>>>
>>>>>>>>>>        May I fire custom cdr event for a LUA script? I've read
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Event_list but haven't found cdr
>>>>>>>>>> events.
>>>>>>>>>> Would You please point me to proper documentation or give an
>>>>>>>>>> example?
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> 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 list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>>
>>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 eric at loopfx.com  Fri May 20 17:25:35 2011
From: eric at loopfx.com (Eric Beard)
Date: Fri, 20 May 2011 09:25:35 -0400
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFB8D8FD@doctorbob.loop.local>

The only one of those 3 that sends SIP to the log file for me is "sofia tracelevel 3".  But where do I put that in a configuration file so it is enabled by default?

I tried setting this in the profile:

   <param name="sip-trace" value="yes"/>

But it didn't do anything.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nicolas Brenner
Sent: Thursday, May 19, 2011 7:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP Trace

Actually, I get the exact same output from the console in my logfile (log/freeswitch.log), I just do:

fsctl loglevel 3
sofia profile_name siptrace on
sofia tracelevel 3

I think you may use any loglevel, but that works fine, and I don't get all the debugging output.


On Thu, May 19, 2011 at 12:48 PM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
No.

If you need that it's best to use tshark, wireshark, ngrep or some similar tool.

-Steve


On 19 May 2011 17:40, Michal Kopacki <mkopacki at gmail.com<mailto:mkopacki at gmail.com>> wrote:
    Sorry for interrupt, but is there any way to filter messages from sip trace ? For example show only one ext/host/uri etc

--
Michal

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


By default it only goes to console. 'sofia tracelevel debug' will let you set a log level which'll let you log it to the logfile (you can choose any of the log files).

-Steve

On 19 May 2011 15:57, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
Hello,

In fs_cli, I can issue this command:

sofia global siptrace on

And I see all SIP traffic.  But the SIP does not get logged to the actual log file.  How do I get the same output to be sent to the log?  I've searched through the Wiki but I'm not seeing a clear way to do it.

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249<tel:%28877%29%20850-2010%20x9249>
m (727) 776-2768<tel:%28727%29%20776-2768>
eric at loopfx.com<mailto:eric at loopfx.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<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 kashif at kashifbukhari.com  Fri May 20 15:13:19 2011
From: kashif at kashifbukhari.com (Kashif Ali)
Date: Fri, 20 May 2011 16:13:19 +0500
Subject: [Freeswitch-users] Failed to activate LibSngSS7!
Message-ID: <BANLkTikyQdThHw0d+dKf1nXKoQt4X6=b9g@mail.gmail.com>

i am getting following error when i run ftdm start 1 what does this mean?


freeswitch at billing> ftdm start 1


2011-05-20 16:10:23.637209 [INFO] ftmod_sangoma_ss7_main.c:1380 Starting
span wp1:1.
2011-05-20 16:10:23.637209 [ERR] ftmod_sangoma_ss7_logger.c:87 sng_ss7->

date: 05/20/2011 time: 16:10:23

2011-05-20 16:10:23.637209 [ERR] ftmod_sangoma_ss7_logger.c:87 sng_ss7->
mtss(posix): sw error:  ent: 001  inst: 000  proc id: 512
file: ../ss/ss_task.c line: 3477 errcode: 00515 errcls: ERRCLS_DEBUG
errval: 00000  errdesc: Unknown task

2011-05-20 16:10:23.837205 [ERR] ftmod_sangoma_ss7_support.c:1772 Failed to
get status of ISUP intf 1
2011-05-20 16:10:23.837205 [ERR] ftmod_sangoma_ss7_logger.c:87 sng_ss7->

date: 05/20/2011 time: 16:10:23

2011-05-20 16:10:23.837205 [ERR] ftmod_sangoma_ss7_logger.c:87 sng_ss7->
mtss(posix): sw error:  ent: 001  inst: 000  proc id: 512
file: ../ss/ss_task.c line: 3477 errcode: 00515 errcls: ERRCLS_DEBUG
errval: 00000  errdesc: Unknown task

2011-05-20 16:10:24.037197 [CRIT] ftmod_sangoma_ss7_cntrl.c:94 ISAP 1
Enable: NOT OK

-ERR failure

2011-05-20 16:10:24.037197 [CRIT] ftmod_sangoma_ss7_main.c:1443 Failed to
activate LibSngSS7!
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From revmichael at bethelightchapel.com  Fri May 20 16:18:25 2011
From: revmichael at bethelightchapel.com (Rev Michael Carbone)
Date: Fri, 20 May 2011 08:18:25 -0400
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
	record to Shoutcast
Message-ID: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>

I have things setup and working and am using fusionpbx and am totally  
new to all this.

I am setting things up to have an online call in talk show.  What I  
want to be able to do is this:

Have me and a guest on and the call broadcasting to a  
shoutcast/icecast server (which I have already) and want to be able to  
broadcast this as MOH, how can I also have people in a FIFO cue  
listening in on the show and be able to bring them in one at a time.

Broken down:

2 of us on a call
broadcasting to shoutcast
other people call in and are put in cue with conversation as MOH
bring people on air one at a time while others are still muted

I'm sure this can be done, but how?

Please remember I'm new to all this so instructions would be helpful.

Thank you,

Michael




From infos at madovsky.org  Fri May 20 18:25:31 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 20 May 2011 10:25:31 -0400
Subject: [Freeswitch-users] SIP Trace
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local><BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com><4DD547F5.4080504@gmail.com><BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com><BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFB8D8FD@doctorbob.loop.local>
Message-ID: <E2694C5E2AB74C4081843E81437EC433@e1705>

in sofia.conf.xml
  ----- Original Message ----- 
  From: Eric Beard 
  To: FreeSWITCH Users Help 
  Sent: Friday, May 20, 2011 9:25 AM
  Subject: Re: [Freeswitch-users] SIP Trace


  The only one of those 3 that sends SIP to the log file for me is "sofia tracelevel 3".  But where do I put that in a configuration file so it is enabled by default?

   

  I tried setting this in the profile:

   

     <param name="sip-trace" value="yes"/>

   

  But it didn't do anything.

   

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

  Eric Z. Beard, CTO

  Loop LLC

  w (877) 850-2010 x9249

  m (727) 776-2768

  eric at loopfx.com

   

  From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nicolas Brenner
  Sent: Thursday, May 19, 2011 7:39 PM
  To: FreeSWITCH Users Help
  Subject: Re: [Freeswitch-users] SIP Trace

   

  Actually, I get the exact same output from the console in my logfile (log/freeswitch.log), I just do:

   

  fsctl loglevel 3

  sofia profile_name siptrace on

  sofia tracelevel 3

   

  I think you may use any loglevel, but that works fine, and I don't get all the debugging output.

   

   

  On Thu, May 19, 2011 at 12:48 PM, Steven Ayre <steveayre at gmail.com> wrote:

  No.

  If you need that it's best to use tshark, wireshark, ngrep or some similar tool.

  -Steve





  On 19 May 2011 17:40, Michal Kopacki <mkopacki at gmail.com> wrote:

      Sorry for interrupt, but is there any way to filter messages from sip trace ? For example show only one ext/host/uri etc

  -- 
  Michal

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





  By default it only goes to console. 'sofia tracelevel debug' will let you set a log level which'll let you log it to the logfile (you can choose any of the log files).

  -Steve



  On 19 May 2011 15:57, Eric Beard <eric at loopfx.com> wrote:

  Hello,

   

  In fs_cli, I can issue this command:

   

  sofia global siptrace on

   

  And I see all SIP traffic.  But the SIP does not get logged to the actual log file.  How do I get the same output to be sent to the log?  I've searched through the Wiki but I'm not seeing a clear way to do it.

   

  Thanks!

   

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

  Eric Z. Beard, CTO

  Loop LLC

  w (877) 850-2010 x9249

  m (727) 776-2768

  eric at loopfx.com

   


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


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

   


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

   



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.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 at slickdeals.endjunk.com  Fri May 20 18:51:45 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 20 May 2011 07:51:45 -0700 (PDT)
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
 record to Shoutcast
In-Reply-To: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
References: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
Message-ID: <1305903105560-6386372.post@n2.nabble.com>


Rev Michael Carbone wrote:
> Broken down:
> 
> 2 of us on a call
> broadcasting to shoutcast
> other people call in and are put in cue with conversation as MOH
> bring people on air one at a time while others are still muted
> 
> I'm sure this can be done, but how?
FS comes with a built-in un-muted conference room by default on extension
3000. If you reconfigure with all muted by default, then you can select
which participant to un-mute while the rests on the conference room are
still muted and be able to listen to the conversation.

-----
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/Setup-help-Conference-3-way-call-FIFO-MOH-record-to-Shoutcast-tp6386123p6386372.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Fri May 20 18:53:57 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 20 May 2011 10:53:57 -0400
Subject: [Freeswitch-users] Unset but no unexport?
Message-ID: <BANLkTinvV0QSeHKxv=03OEH6qNaLuphJbg@mail.gmail.com>

Hello everyone,

  There is unset:

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

  Why no unexport?

-- 
Kristian Kielhofner


From Nabble at slickdeals.endjunk.com  Fri May 20 19:07:18 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 20 May 2011 08:07:18 -0700 (PDT)
Subject: [Freeswitch-users] mod_zmq with zeromq-2.1.7.tar.gz?
Message-ID: <1305904038989-6386441.post@n2.nabble.com>

Is there a reason not to upgrade mod_zmq with  http://download.zeromq.org
zeromq-2.1.7.tar.gz  released on 2011/05/12?

-----
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-zmq-with-zeromq-2-1-7-tar-gz-tp6386441p6386441.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From revmichael at bethelightchapel.com  Fri May 20 19:05:17 2011
From: revmichael at bethelightchapel.com (Rev Michael Carbone)
Date: Fri, 20 May 2011 11:05:17 -0400
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
 record to Shoutcast
In-Reply-To: <1305903105560-6386372.post@n2.nabble.com>
References: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
	<1305903105560-6386372.post@n2.nabble.com>
Message-ID: <20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>

Like I said I'm new to all this, I know where to find the extensions,  
but I'm not seeing extension 3000 anywhere, am I missing something?   
I'm using fusionpbx as well.

thank you for your help

Michael

>
> Rev Michael Carbone wrote:
>> Broken down:
>>
>> 2 of us on a call
>> broadcasting to shoutcast
>> other people call in and are put in cue with conversation as MOH
>> bring people on air one at a time while others are still muted
>>
>> I'm sure this can be done, but how?
> FS comes with a built-in un-muted conference room by default on extension
> 3000. If you reconfigure with all muted by default, then you can select
> which participant to un-mute while the rests on the conference room are
> still muted and be able to listen to the conversation.
>
> -----
> 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/Setup-help-Conference-3-way-call-FIFO-MOH-record-to-Shoutcast-tp6386123p6386372.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 steveayre at gmail.com  Fri May 20 19:08:16 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 20 May 2011 16:08:16 +0100
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFB8D8FD@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFB8D8FD@doctorbob.loop.local>
Message-ID: <BANLkTi=xT1-n36woRytVSjeB0fL+60gAEA@mail.gmail.com>

That's because it the combination of the three:

fsctl loglevel 3

- Sets the logging level to 3

sofia profile_name siptrace on

- Start tracing SIP packets (console level by default, which won't go to the
logfile)

sofia tracelevel 3

- Start logging SIP traces with loglevel 3. Because you just set the log
level to 3 it'll show up.



>From the sofia config you'll need to set a tracelevel other than the console
default as well as enabling sip-trace:


<global_settings>
  ...
  <param name="tracelevel" value="DEBUG"/>
  ...
</global_settings>
<profiles>
  ...
  <profile name="...">
    ...
    <param name="sip-trace" value="yes"/>
    ...
  </profile>
  ...
</profiles>



-Steve


On 20 May 2011 14:25, Eric Beard <eric at loopfx.com> wrote:

> The only one of those 3 that sends SIP to the log file for me is ?sofia
> tracelevel 3?.  But where do I put that in a configuration file so it is
> enabled by default?
>
>
>
> I tried setting this in the profile:
>
>
>
>    <param name="sip-trace" value="yes"/>
>
>
>
> But it didn?t do anything.
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nicolas
> Brenner
> *Sent:* Thursday, May 19, 2011 7:39 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP Trace
>
>
>
> Actually, I get the exact same output from the console in my logfile
> (log/freeswitch.log), I just do:
>
>
>
> fsctl loglevel 3
>
> sofia profile_name siptrace on
>
> sofia tracelevel 3
>
>
>
> I think you may use any loglevel, but that works fine, and I don't get all
> the debugging output.
>
>
>
>
>
> On Thu, May 19, 2011 at 12:48 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
> No.
>
> If you need that it's best to use tshark, wireshark, ngrep or some similar
> tool.
>
> -Steve
>
>
>
> On 19 May 2011 17:40, Michal Kopacki <mkopacki at gmail.com> wrote:
>
>     Sorry for interrupt, but is there any way to filter messages from sip
> trace ? For example show only one ext/host/uri etc
>
> --
> Michal
>
> ---------------
>
>
>
> By default it only goes to console. 'sofia tracelevel debug' will let you
> set a log level which'll let you log it to the logfile (you can choose any
> of the log files).
>
> -Steve
>
> On 19 May 2011 15:57, Eric Beard <eric at loopfx.com> wrote:
>
> Hello,
>
>
>
> In fs_cli, I can issue this command:
>
>
>
> sofia global siptrace on
>
>
>
> And I see all SIP traffic.  But the SIP does not get logged to the actual
> log file.  How do I get the same output to be sent to the log?  I?ve
> searched through the Wiki but I?m not seeing a clear way to do it.
>
>
>
> Thanks!
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.sobkow at marketelsystems.com  Fri May 20 19:12:52 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 20 May 2011 09:12:52 -0600
Subject: [Freeswitch-users] Sound quality issues
In-Reply-To: <BANLkTi=hUgmxn8g+DmVDmSer0ffOpNGJbA@mail.gmail.com>
References: <4DD546CC.7090209@marketelsystems.com>
	<BANLkTi=hUgmxn8g+DmVDmSer0ffOpNGJbA@mail.gmail.com>
Message-ID: <4DD684F4.1040504@marketelsystems.com>

No, that's what's puzzling.  If anything, the box is overpowered for 
it's duties (PostgreSQL server, our application services, and 
Freeswitch.)  Over 6GB free memory, less than 10% CPU load, and a 
gigabit network link on a switch that's got less than 5% utilization.

On 19/05/2011 10:45 AM, Avi Marcus wrote:
> Are you using some sort of virtualization?
> -Avi
>
> On Thu, May 19, 2011 at 7:35 PM, Mark Sobkow 
> <m.sobkow at marketelsystems.com <mailto:m.sobkow at marketelsystems.com>> 
> wrote:
>
>     We've got a FreeSwitch box using a SIP trunk to an Asterisk box, which
>     connects to our T1.
>
>     We're having significant voice quality issues, and I'm not sure
>     what to
>     look into.  CPU load is under 10% on the Freeswitch box.
>
>     Even playing back 8kHz audio files shows significant breakup, so I
>     think
>     I've been on the wrong track thinking it was a SIP issue.
>

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/

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From avi at avimarcus.net  Fri May 20 19:15:21 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 20 May 2011 18:15:21 +0300
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
 record to Shoutcast
In-Reply-To: <20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>
References: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
	<1305903105560-6386372.post@n2.nabble.com>
	<20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>
Message-ID: <BANLkTinRyRn3BW1bPVM+FwDZRB3BC+v72Q@mail.gmail.com>

The 3000-3099 extension range is in default.xml, which FusionPBX mostly
leaves unmodified.
-Avi

On Fri, May 20, 2011 at 6:05 PM, Rev Michael Carbone <
revmichael at bethelightchapel.com> wrote:

> Like I said I'm new to all this, I know where to find the extensions,
> but I'm not seeing extension 3000 anywhere, am I missing something?
> I'm using fusionpbx as well.
>
> thank you for your help
>
> Michael
>
> >
> > Rev Michael Carbone wrote:
> >> Broken down:
> >>
> >> 2 of us on a call
> >> broadcasting to shoutcast
> >> other people call in and are put in cue with conversation as MOH
> >> bring people on air one at a time while others are still muted
> >>
> >> I'm sure this can be done, but how?
> > FS comes with a built-in un-muted conference room by default on extension
> > 3000. If you reconfigure with all muted by default, then you can select
> > which participant to un-mute while the rests on the conference room are
> > still muted and be able to listen to the conversation.
> >
> > -----
> > 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/Setup-help-Conference-3-way-call-FIFO-MOH-record-to-Shoutcast-tp6386123p6386372.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 rhuddleston at gmail.com  Fri May 20 19:20:06 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Fri, 20 May 2011 11:20:06 -0400
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
	record to Shoutcast
In-Reply-To: <20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>
References: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
	<1305903105560-6386372.post@n2.nabble.com>
	<20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>
Message-ID: <25a801cc1701$6740d510$35c27f30$@com>

If you feel that you are not getting the response you expect - try jumping
over to the IRC channel and asking there.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rev
Michael Carbone
Sent: Friday, May 20, 2011 11:05 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Setup help Conference, 3 way call, FIFO,
MOH, record to Shoutcast

Like I said I'm new to all this, I know where to find the extensions,  
but I'm not seeing extension 3000 anywhere, am I missing something?   
I'm using fusionpbx as well.

thank you for your help

Michael

>
> Rev Michael Carbone wrote:
>> Broken down:
>>
>> 2 of us on a call
>> broadcasting to shoutcast
>> other people call in and are put in cue with conversation as MOH
>> bring people on air one at a time while others are still muted
>>
>> I'm sure this can be done, but how?
> FS comes with a built-in un-muted conference room by default on extension
> 3000. If you reconfigure with all muted by default, then you can select
> which participant to un-mute while the rests on the conference room are
> still muted and be able to listen to the conversation.
>
> -----
> 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/Setup-help-Conference-3-way-ca
ll-FIFO-MOH-record-to-Shoutcast-tp6386123p6386372.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 eric at loopfx.com  Fri May 20 20:39:52 2011
From: eric at loopfx.com (Eric Beard)
Date: Fri, 20 May 2011 12:39:52 -0400
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <BANLkTi=xT1-n36woRytVSjeB0fL+60gAEA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFB8D8FD@doctorbob.loop.local>
	<BANLkTi=xT1-n36woRytVSjeB0fL+60gAEA@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFB8DB15@doctorbob.loop.local>

That's what I was missing, thanks! I was searching the source code for "trace-level" (with a dash).

I'm still not entirely clear on the difference between "loglevel" and "tracelevel", but at least it's logging the SIP to my log file automatically.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Friday, May 20, 2011 11:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP Trace

That's because it the combination of the three:
fsctl loglevel 3
- Sets the logging level to 3
sofia profile_name siptrace on
- Start tracing SIP packets (console level by default, which won't go to the logfile)
sofia tracelevel 3
- Start logging SIP traces with loglevel 3. Because you just set the log level to 3 it'll show up.


>From the sofia config you'll need to set a tracelevel other than the console default as well as enabling sip-trace:

<global_settings>
  ...
  <param name="tracelevel" value="DEBUG"/>
  ...
</global_settings>
<profiles>
  ...
  <profile name="...">
    ...
    <param name="sip-trace" value="yes"/>
    ...
  </profile>
  ...
</profiles>


-Steve

On 20 May 2011 14:25, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
The only one of those 3 that sends SIP to the log file for me is "sofia tracelevel 3".  But where do I put that in a configuration file so it is enabled by default?

I tried setting this in the profile:

   <param name="sip-trace" value="yes"/>

But it didn't do anything.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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 Nicolas Brenner
Sent: Thursday, May 19, 2011 7:39 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP Trace

Actually, I get the exact same output from the console in my logfile (log/freeswitch.log), I just do:

fsctl loglevel 3
sofia profile_name siptrace on
sofia tracelevel 3

I think you may use any loglevel, but that works fine, and I don't get all the debugging output.


On Thu, May 19, 2011 at 12:48 PM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
No.

If you need that it's best to use tshark, wireshark, ngrep or some similar tool.

-Steve

On 19 May 2011 17:40, Michal Kopacki <mkopacki at gmail.com<mailto:mkopacki at gmail.com>> wrote:
    Sorry for interrupt, but is there any way to filter messages from sip trace ? For example show only one ext/host/uri etc

--
Michal

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

By default it only goes to console. 'sofia tracelevel debug' will let you set a log level which'll let you log it to the logfile (you can choose any of the log files).

-Steve
On 19 May 2011 15:57, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
Hello,

In fs_cli, I can issue this command:

sofia global siptrace on

And I see all SIP traffic.  But the SIP does not get logged to the actual log file.  How do I get the same output to be sent to the log?  I've searched through the Wiki but I'm not seeing a clear way to do it.

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249<tel:%28877%29%20850-2010%20x9249>
m (727) 776-2768<tel:%28727%29%20776-2768>
eric at loopfx.com<mailto:eric at loopfx.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

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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
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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
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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
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From msc at freeswitch.org  Fri May 20 20:39:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 20 May 2011 09:39:58 -0700
Subject: [Freeswitch-users] G.729A licencing validation error
In-Reply-To: <BANLkTikLDHyT5osc-W0_L0cxy19_jsp8bA@mail.gmail.com>
References: <BANLkTi=42gOrfQqvKdsT4-4R-refv1D=Eg@mail.gmail.com>
	<BANLkTimpEcJyNxJA-Nno0gFyU13PO0MMEA@mail.gmail.com>
	<BANLkTikLDHyT5osc-W0_L0cxy19_jsp8bA@mail.gmail.com>
Message-ID: <BANLkTikjZZv8PzE1VppqpKTW=8Gd+V+cug@mail.gmail.com>

On Fri, May 20, 2011 at 5:20 AM, Tim Tobin <ttobin at ubq.thrupoint.net> wrote:

> Hi Michael,
>
> Thanks for the response. Validation is still failing. The issue seems to be
> that the machine on which FreeSwitch is running has no direct access to the
> internet. This isn't a machine i control and so can't change this. As i
> understand things we could potentially resolve this issue by tunnelling HTTP
> and DNS traffic. I've tried to set up DNS tunnelling but have so far been
> unable to get this to work. I guess another approach would be to simply add
> entries to the hosts file.
>
> Also, am i right in assuming that, as far as licensing is concerned, once
> the license has been validated we no longer need internet access?
>

That is correct. You only need the Internet for the actual validation and
the download of the license file. After that you're good, so if you can open
a connection for a few minutes while running the validation then you should
be good.
-MC
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From msc at freeswitch.org  Fri May 20 20:44:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 20 May 2011 09:44:29 -0700
Subject: [Freeswitch-users] Sound quality issues
In-Reply-To: <4DD684F4.1040504@marketelsystems.com>
References: <4DD546CC.7090209@marketelsystems.com>
	<BANLkTi=hUgmxn8g+DmVDmSer0ffOpNGJbA@mail.gmail.com>
	<4DD684F4.1040504@marketelsystems.com>
Message-ID: <BANLkTin3BO7T8ZiLe=VZ-LKrbKp0uv-t5Q@mail.gmail.com>

I'm sorry but I don't think I heard the whole story. What OS is this and
what kernel timing?
-MC

On Fri, May 20, 2011 at 8:12 AM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

>  No, that's what's puzzling.  If anything, the box is overpowered for it's
> duties (PostgreSQL server, our application services, and Freeswitch.)  Over
> 6GB free memory, less than 10% CPU load, and a gigabit network link on a
> switch that's got less than 5% utilization.
>
> On 19/05/2011 10:45 AM, Avi Marcus wrote:
>
> Are you using some sort of virtualization?
> -Avi
>
> On Thu, May 19, 2011 at 7:35 PM, Mark Sobkow <m.sobkow at marketelsystems.com
> > wrote:
>
>> We've got a FreeSwitch box using a SIP trunk to an Asterisk box, which
>> connects to our T1.
>>
>> We're having significant voice quality issues, and I'm not sure what to
>> look into.  CPU load is under 10% on the Freeswitch box.
>>
>> Even playing back 8kHz audio files shows significant breakup, so I think
>> I've been on the wrong track thinking it was a SIP issue.
>>
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
> Visit our Blog for industry related information.http://marketel-systems.blogspot.com/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri May 20 22:47:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 20 May 2011 19:47:38 +0100
Subject: [Freeswitch-users] SIP Trace
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFB8DB15@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFB8D8FD@doctorbob.loop.local>
	<BANLkTi=xT1-n36woRytVSjeB0fL+60gAEA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFB8DB15@doctorbob.loop.local>
Message-ID: <BANLkTik_F_TCY6Aih+L25nrSKJKv_4TqqA@mail.gmail.com>

loglevel is the debug level of components of the sofia stack, the same as
the 'sofia loglevel x' command. You shouldn't normally need it - it'll give
you extremely ugly and verbose low level information about the processing of
the packets

-Steve

On 20 May 2011 17:39, Eric Beard <eric at loopfx.com> wrote:

> That?s what I was missing, thanks! I was searching the source code for
> ?trace-level? (with a dash).
>
>
>
> I?m still not entirely clear on the difference between ?loglevel? and
> ?tracelevel?, but at least it?s logging the SIP to my log file
> automatically.
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Friday, May 20, 2011 11:08 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP Trace
>
>
>
> That's because it the combination of the three:
>
> fsctl loglevel 3
>
> - Sets the logging level to 3
>
> sofia profile_name siptrace on
>
> - Start tracing SIP packets (console level by default, which won't go to
> the logfile)
>
> sofia tracelevel 3
>
> - Start logging SIP traces with loglevel 3. Because you just set the log
> level to 3 it'll show up.
>
>
>
>
>
> From the sofia config you'll need to set a tracelevel other than the
> console default as well as enabling sip-trace:
>
>
>
> <global_settings>
>   ...
>   <param name="tracelevel" value="DEBUG"/>
>   ...
> </global_settings>
> <profiles>
>   ...
>   <profile name="...">
>     ...
>     <param name="sip-trace" value="yes"/>
>     ...
>   </profile>
>   ...
> </profiles>
>
>
>
>
>
> -Steve
>
>
>
> On 20 May 2011 14:25, Eric Beard <eric at loopfx.com> wrote:
>
> The only one of those 3 that sends SIP to the log file for me is ?sofia
> tracelevel 3?.  But where do I put that in a configuration file so it is
> enabled by default?
>
>
>
> I tried setting this in the profile:
>
>
>
>    <param name="sip-trace" value="yes"/>
>
>
>
> But it didn?t do anything.
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nicolas
> Brenner
> *Sent:* Thursday, May 19, 2011 7:39 PM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP Trace
>
>
>
> Actually, I get the exact same output from the console in my logfile
> (log/freeswitch.log), I just do:
>
>
>
> fsctl loglevel 3
>
> sofia profile_name siptrace on
>
> sofia tracelevel 3
>
>
>
> I think you may use any loglevel, but that works fine, and I don't get all
> the debugging output.
>
>
>
>
>
> On Thu, May 19, 2011 at 12:48 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
> No.
>
> If you need that it's best to use tshark, wireshark, ngrep or some similar
> tool.
>
> -Steve
>
>
>
> On 19 May 2011 17:40, Michal Kopacki <mkopacki at gmail.com> wrote:
>
>     Sorry for interrupt, but is there any way to filter messages from sip
> trace ? For example show only one ext/host/uri etc
>
> --
> Michal
>
> ---------------
>
>
>
> By default it only goes to console. 'sofia tracelevel debug' will let you
> set a log level which'll let you log it to the logfile (you can choose any
> of the log files).
>
> -Steve
>
> On 19 May 2011 15:57, Eric Beard <eric at loopfx.com> wrote:
>
> Hello,
>
>
>
> In fs_cli, I can issue this command:
>
>
>
> sofia global siptrace on
>
>
>
> And I see all SIP traffic.  But the SIP does not get logged to the actual
> log file.  How do I get the same output to be sent to the log?  I?ve
> searched through the Wiki but I?m not seeing a clear way to do it.
>
>
>
> Thanks!
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri May 20 23:12:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 20 May 2011 12:12:19 -0700
Subject: [Freeswitch-users] call somebody into conference
	afterconferenceestablished
In-Reply-To: <1412527818-1305863043-cardhu_decombobulator_blackberry.rim.net-1696936367-@b5.c17.bise6.blackberry>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<201105121532017963789@asiainfo-linkage.com>
	<BANLkTi=wbd8ivSA4Gom3P2QsJ9AY7GRJUA@mail.gmail.com>
	<BANLkTikPq5GteE1Y5w56oL_=oAQ3ieVC1w@mail.gmail.com>
	<BANLkTimMVoF3RDjo1Zi2FG0K=ZwOXUyZKg@mail.gmail.com>
	<1412527818-1305863043-cardhu_decombobulator_blackberry.rim.net-1696936367-@b5.c17.bise6.blackberry>
Message-ID: <BANLkTinMBuhpje-ZVbutGdwMWqV2qF75tg@mail.gmail.com>

On Thu, May 19, 2011 at 8:44 PM, <darknesslabs at gmail.com> wrote:

> Super useful, please do add it to the wiki.
> Sent via BlackBerry
>
Done:
http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example

-MC
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From seichhorn at gci.com  Fri May 20 22:54:21 2011
From: seichhorn at gci.com (Sean Eichhorn)
Date: Fri, 20 May 2011 10:54:21 -0800
Subject: [Freeswitch-users] rtpmap line missing on answer
Message-ID: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>

I have a freeswitch system that (for the most part) works flawlessly for me.  However, I need to have an additional rtpmap line in the SDP when a call is answered.
My client sends the following SDP upon answering the call :
   m=audio 17214 RTP/AVP 0 19 101 100
   c=IN IP4 192.168.98.79
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:100 X-NSE/8000
   a=fmtp:100 192-194

Freeswitch forwards the following SDP to the external gateway :
   m=audio 20654 RTP/AVP 0 19 101
   c=IN IP4 66.223.187.208
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16

The other information appears fine.  Everything works, but my client-side is not receiving the a=rtpmap:100 line.
This issue appears regardless of whether or not I'm in proxy mode, bypass mode, or neither.

Using "sip_append_audio_sdp" has no effect on the answering SDP, only the initial offer.

Any ideas?

Thanks in advance,
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From david.ponzone at ipeva.fr  Sat May 21 01:02:52 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 20 May 2011 23:02:52 +0200
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
Message-ID: <E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>

Sean,

X-NSE is the Clear Channel codec from Cisco.
I suspect FreeSWITCH does not support it, and as such, it can't be offered to leg B.
Perhaps there is a way to enable in outbound codecs as bypass, but I really doubt so.
Though, you could try to enable late-negotiation where client and gateway will negotiate together.

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

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

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




Le 20/05/2011 ? 20:54, Sean Eichhorn a ?crit :

> I have a freeswitch system that (for the most part) works flawlessly for me.  However, I need to have an additional rtpmap line in the SDP when a call is answered.
> My client sends the following SDP upon answering the call :
>    m=audio 17214 RTP/AVP 0 19 101 100
>    c=IN IP4 192.168.98.79
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:19 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:100 X-NSE/8000
>    a=fmtp:100 192-194
>  
> Freeswitch forwards the following SDP to the external gateway :
>    m=audio 20654 RTP/AVP 0 19 101
>    c=IN IP4 66.223.187.208
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:19 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>  
> The other information appears fine.  Everything works, but my client-side is not receiving the a=rtpmap:100 line.
> This issue appears regardless of whether or not I?m in proxy mode, bypass mode, or neither.
>  
> Using "sip_append_audio_sdp? has no effect on the answering SDP, only the initial offer.
>  
> Any ideas?
>  
> 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 revmichael at bethelightchapel.com  Sat May 21 02:30:37 2011
From: revmichael at bethelightchapel.com (Rev Michael Carbone)
Date: Fri, 20 May 2011 18:30:37 -0400
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
 record to Shoutcast
In-Reply-To: <25a801cc1701$6740d510$35c27f30$@com>
References: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
	<1305903105560-6386372.post@n2.nabble.com>
	<20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>
	<25a801cc1701$6740d510$35c27f30$@com>
Message-ID: <20110520183037.83314gw215dgz5og@psychicawakeningschool.com>

Been trying to get to the IRC channel
If anyone can help off list please e-mail me at revmichael at  
bethelightchapel dot com

thank you


Quoting Robert Huddleston <rhuddleston at gmail.com>:

> If you feel that you are not getting the response you expect - try jumping
> over to the IRC channel and asking there.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rev
> Michael Carbone
> Sent: Friday, May 20, 2011 11:05 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Setup help Conference, 3 way call, FIFO,
> MOH, record to Shoutcast
>
> Like I said I'm new to all this, I know where to find the extensions,
> but I'm not seeing extension 3000 anywhere, am I missing something?
> I'm using fusionpbx as well.
>
> thank you for your help
>
> Michael
>
>>
>> Rev Michael Carbone wrote:
>>> Broken down:
>>>
>>> 2 of us on a call
>>> broadcasting to shoutcast
>>> other people call in and are put in cue with conversation as MOH
>>> bring people on air one at a time while others are still muted
>>>
>>> I'm sure this can be done, but how?
>> FS comes with a built-in un-muted conference room by default on extension
>> 3000. If you reconfigure with all muted by default, then you can select
>> which participant to un-mute while the rests on the conference room are
>> still muted and be able to listen to the conversation.
>>
>> -----
>> 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/Setup-help-Conference-3-way-ca
> ll-FIFO-MOH-record-to-Shoutcast-tp6386123p6386372.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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>





From jaybinks at gmail.com  Sat May 21 05:20:12 2011
From: jaybinks at gmail.com (Jay Binks)
Date: Sat, 21 May 2011 11:20:12 +1000
Subject: [Freeswitch-users] Sound quality issues
In-Reply-To: <BANLkTin3BO7T8ZiLe=VZ-LKrbKp0uv-t5Q@mail.gmail.com>
References: <4DD546CC.7090209@marketelsystems.com>
	<BANLkTi=hUgmxn8g+DmVDmSer0ffOpNGJbA@mail.gmail.com>
	<4DD684F4.1040504@marketelsystems.com>
	<BANLkTin3BO7T8ZiLe=VZ-LKrbKp0uv-t5Q@mail.gmail.com>
Message-ID: <AA52884A-560D-4A39-9C30-2E9A4AABB907@gmail.com>

Is it a kernel with a 100hz timer ?? ( like some / most ubuntu etc )

J



On 21/05/2011, at 2:44 AM, Michael Collins <msc at freeswitch.org> wrote:

> I'm sorry but I don't think I heard the whole story. What OS is this and what kernel timing? 
> -MC
> 
> On Fri, May 20, 2011 at 8:12 AM, Mark Sobkow <m.sobkow at marketelsystems.com> wrote:
> No, that's what's puzzling.  If anything, the box is overpowered for it's duties (PostgreSQL server, our application services, and Freeswitch.)  Over 6GB free memory, less than 10% CPU load, and a gigabit network link on a switch that's got less than 5% utilization.
> 
> On 19/05/2011 10:45 AM, Avi Marcus wrote:
>> 
>> Are you using some sort of virtualization?
>> -Avi
>> 
>> On Thu, May 19, 2011 at 7:35 PM, Mark Sobkow <m.sobkow at marketelsystems.com> wrote:
>> We've got a FreeSwitch box using a SIP trunk to an Asterisk box, which
>> connects to our T1.
>> 
>> We're having significant voice quality issues, and I'm not sure what to
>> look into.  CPU load is under 10% on the Freeswitch box.
>> 
>> Even playing back 8kHz audio files shows significant breakup, so I think
>> I've been on the wrong track thinking it was a SIP issue.
> 
> -- 
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
> 
> Visit our Blog for industry related information.
> http://marketel-systems.blogspot.com/
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sid.kshatriya at gmail.com  Sat May 21 12:11:36 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Sat, 21 May 2011 13:41:36 +0530
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
	rfc2833 not working!
Message-ID: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>

Dear Friends,

I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
list). While I'm generally happy with their service there seems to be one
fatal problem: My IVR does not recognize DTMF!

I have set <param name="dtmf-type" value="rfc2833"/> in both
sip_profile/internal.xml and sip_profile/external.xml

The symptom of the problem is that making a call via skype will
*always*make the IVR recognize the DTMF while using something like a
mobile phone
*almost always* won't!

I've tried in-band detection too. I'm making international calls into my IVR
and the reliability of the in-band detection is not so good, possibly
because of the quality of the call.

Can someone please help me?

Thanks,

Sidharth

-- 
Sidharth Kshatriya
www.sidk.info
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From covici at ccs.covici.com  Sat May 21 16:27:23 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 21 May 2011 08:27:23 -0400
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
	rfc2833 not working!
In-Reply-To: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
Message-ID: <17177.1305980843@ccs.covici.com>

Try listening to the dtmf from the caller in question and see if you can
hear the sound.

Sidharth Kshatriya <sid.kshatriya at gmail.com> wrote:

> Dear Friends,
> 
> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
> list). While I'm generally happy with their service there seems to be one
> fatal problem: My IVR does not recognize DTMF!
> 
> I have set <param name="dtmf-type" value="rfc2833"/> in both
> sip_profile/internal.xml and sip_profile/external.xml
> 
> The symptom of the problem is that making a call via skype will
> *always*make the IVR recognize the DTMF while using something like a
> mobile phone
> *almost always* won't!
> 
> I've tried in-band detection too. I'm making international calls into my IVR
> and the reliability of the in-band detection is not so good, possibly
> because of the quality of the call.
> 
> Can someone please help me?
> 
> Thanks,
> 
> Sidharth
> 
> -- 
> Sidharth Kshatriya
> www.sidk.info
> 
> ----------------------------------------------------
> 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


From cmcureau at gmail.com  Sat May 21 23:17:32 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Sat, 21 May 2011 14:17:32 -0500
Subject: [Freeswitch-users] Question about faxing between FreeSwitch servers
Message-ID: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>

Hi, everyone!

I am trying to implement a workable fax solution between two FreeSwitch
servers.  This is what I had in mind so far:

T1 PRI <--> Sangoma A102 <--> FreeSwitch (host) <--> T.38 SIP (internet)
<--> FreeSwitch (remote) <--> Sangoma B600D <--> FAX machine

I am assuming here that the T.38 protocol between FreeSwitch servers will
ensure that the data transfers between the two sites successfully.  Would
this actually be a two step process, where one server receives the fax (say,
from the PRI) on one call, then sends to the second server, and the second
server initiates a second call to the fax machine?

The object here is to successfully pass faxes using SIP.  Is there a better
way to handle this?

Thanks in advance,
Chris
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From tayeb.meftah at gmail.com  Sun May 22 01:01:27 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 21 May 2011 23:01:27 +0200
Subject: [Freeswitch-users] Question about faxing between FreeSwitch
	servers
In-Reply-To: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
References: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
Message-ID: <4DD82827.9080405@gmail.com>

T.38 is good but is a pain.
i strongly suggest you to pass it through PCMU (G.711U) codec, sunse 
you're willing to use E1 boards
no transcoding required and less ressources
thank you
On 21/05/2011 21:17, Chris Cureau wrote:
> Hi, everyone!
>
> I am trying to implement a workable fax solution between two 
> FreeSwitch servers.  This is what I had in mind so far:
>
> T1 PRI <--> Sangoma A102 <--> FreeSwitch (host) <--> T.38 SIP 
> (internet) <--> FreeSwitch (remote) <--> Sangoma B600D <--> FAX machine
>
> I am assuming here that the T.38 protocol between FreeSwitch servers 
> will ensure that the data transfers between the two sites 
> successfully.  Would this actually be a two step process, where one 
> server receives the fax (say, from the PRI) on one call, then sends to 
> the second server, and the second server initiates a second call to 
> the fax machine?
>
> The object here is to successfully pass faxes using SIP.  Is there a 
> better way to handle this?
>
> Thanks in advance,
> 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
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883

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From bwibowo at gmail.com  Sun May 22 02:42:47 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 22 May 2011 05:42:47 +0700
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
	<BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
Message-ID: <BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>

failed with some glib issue on source code compiling, then i use yum as
suggested.
mono installed, thx
Mono JIT compiler version 2.10.2 (tarball Mon Apr 18 19:06:50 UTC 2011)
Copyright (C) 2002-2011 Novell, Inc and Contributors. www.mono-project.com
        TLS:           __thread
        SIGSEGV:       altstack
        Notifications: epoll
        Architecture:  amd64
        Disabled:      none
        Misc:          debugger softdebug
        LLVM:          supported, not enabled.
        GC:            Included Boehm (with typed GC and Parallel Mark)

when i try to install mod_managed i found this error, may be some missing
config in my linux box

making all mod_managed
Compiling freeswitch_managed.cpp...
g++ -I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
-DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
In file included from freeswitch_managed.cpp:35:
freeswitch_managed.h:43:18: error: glib.h: No such file or directory
freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or directory
freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such file or
directory
freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such file
or directory
freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such file
or directory
freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such file or
directory
freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No such
file or directory


thx
budi wibowo

On Wed, May 18, 2011 at 9:06 AM, Yitzchok <adminjew at gmail.com> wrote:

> Make sure mono is installed and check which version using "mono -V" and
> follow the instructions on http://wiki.freeswitch.org/wiki/Mod_managed let
> us know if you get into any errors.
>
> This helped me for installing mono
> http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
>
>
> <http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
> Yitzchok
>
>
> On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com> wrote:
>
>> hi
>> i have fs running on centos 5.5, any body knows the combination version of
>> software need to be installed like mono etc.
>> have tried few combination but always failed compiling
>>
>>
>> thx
>> budi
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sireeps at gmail.com  Sun May 22 03:16:31 2011
From: sireeps at gmail.com (Kamen)
Date: Sat, 21 May 2011 16:16:31 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
Message-ID: <1306019791787-6390451.post@n2.nabble.com>

Hi Michael,

Frankly I am not sure what exactly you referring to.  I am completely new to
freeswitch.  If you are talking about running command (I found it in online
doc):

TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch

then there is a problem,  Like I said I am running it in Windows. Running
FreeSWITCH\FreeSwitchConsole.exe. Besides I have no idea what pastebin is. 
Only can guess it preserves the colors of debug printouts.

I have instead prepared below a  debug printout I manage to get in the
console.  Also added my comment.  Please, could anyone have a look at it. 
Briefly, I want a simple functionality.  If call comes in to my SIP provider
I want it to be forwarded to another fixed external number.  That is it! 
Seems like a straight forward task for freeswitch. The call seems to be
forwarded (according to the log), but something stops it and it gets paused
for 10 seconds, until busy signal and call is disconnected afterwards.

Thanks in advance.

Sergei Kamen

==============================================================

public.xml has extension:

 <extension name="Forward All">
  	<condition field="caller_id_number" expression="^(\d+)$">
 		<action application="transfer" data="3472 XML default"/>
 	</condition>
 </extension>

and default.xml has:

    <extension name="Forward All">
        <condition field="destination_number" expression="3472">
	<action application="bridge"
data="sofia/gateway/freephoneline/15191212121"/>
        </condition>
    </extension>

The following printout in my console is as follows from the beginning:
------------------------------

nta: bind([::1]:5070;transport=*): No such file or directory
nua: initializing SIP stack failed

[SK:] could it be the error above?

2011-05-21 17:59:38.187500 [ERR] sofia.c:1533 Error Creating SIP UA for
profile: internal-ipv6
2011-05-21 17:59:40.875000 [DEBUG] switch_nat.c:502 mapped public port 5080
protocol TCP to localport 5080
2011-05-21 17:59:40.875000 [DEBUG] sofia.c:1487 Created TCP nat mapping for
external port 5080
2011-05-21 17:59:40.875000 [DEBUG] switch_nat.c:502 mapped public port 5070
protocol TCP to localport 5070
2011-05-21 17:59:40.875000 [DEBUG] sofia.c:1487 Created TCP nat mapping for
internal port 5070
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1539 Created agent for external
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1576 Set params for external
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1599 Activated db for external
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1626 Starting thread for external
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1539 Created agent for internal
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1576 Set params for internal
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1599 Activated db for internal
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1626 Starting thread for internal
2011-05-21 17:59:41.921875 [NOTICE] sofia_reg.c:367 Registering
mysipprovider
2011-05-21 17:59:44.062500 [DEBUG] sofia_reg.c:1922 Authenticating
'16472222222' with 'Digest:"voip.mysipprovider.ca":16472222222:mypassword'.
2011-05-21 17:59:44.921875 [DEBUG] sofia_reg.c:331 Registered mysipprovider
2011-05-21 18:00:00.984375 [WARNING] sofia.c:4065 Ping succeeded
mysipprovider with code 501 - count -1/1/1, state UP

[SK:]I am calling here to my SIP provider number (16472222222) from another
phone (9051212121).  I expect it to forward to a third number (15191212121)
with updated Caller ID of my provider.

2011-05-21 18:00:34.609375 [NOTICE] switch_channel.c:812 New Channel
sofia/external/9051212121 at 218.165.240.142
[bc74d840-a951-458e-9eec-c70941fdd05a]
2011-05-21 18:00:34.609375 [DEBUG] sofia.c:4744 Channel
sofia/external/9051212121 at 218.165.240.142 entering state [received][100]
2011-05-21 18:00:34.609375 [DEBUG] sofia.c:4755 Remote SDP:
v=0
o=Sippy 383374256 0 IN IP4 218.165.240.142
s=-
t=0 0
m=audio 35774 RTP/AVP 0
c=IN IP4 208.72.120.90

2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_NEW
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:338
(sofia/external/9051212121 at 218.165.240.142) State NEW
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:2760 Set Codec
sofia/external/9051212121 at 218.165.240.142 PCMU/8000 20 ms 160 samples 64000
bits
2011-05-21 18:00:34.609375 [DEBUG] sofia.c:4922
(sofia/external/9051212121 at 218.165.240.142) State Change CS_NEW -> CS_INIT
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_INIT
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:356
(sofia/external/9051212121 at 218.165.240.142) State INIT
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:84
sofia/external/9051212121 at 218.165.240.142 SOFIA INIT
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:124
(sofia/external/9051212121 at 218.165.240.142) State Change CS_INIT ->
CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:356
(sofia/external/9051212121 at 218.165.240.142) State INIT going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_channel.c:1668
(sofia/external/9051212121 at 218.165.240.142) Callstate Change DOWN -> RINGING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:147
sofia/external/9051212121 at 218.165.240.142 SOFIA ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:77
sofia/external/9051212121 at 218.165.240.142 Standard ROUTING
2011-05-21 18:00:34.609375 [INFO] mod_dialplan_xml.c:331 Processing MICHAEL
S <9051212121>->3472 in context public 
Dialplan: sofia/external/9051212121 at 218.165.240.142 parsing [public->Forward
All] continue=false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Regex (PASS) [Forward
All] caller_id_number(9051212121) =~ /^(\d+)$/ break=on-false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Action transfer(3472 XML
default)
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:119
(sofia/external/9051212121 at 218.165.240.142) State Change CS_ROUTING ->
CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:240
sofia/external/9051212121 at 218.165.240.142 SOFIA EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:157
sofia/external/9051212121 at 218.165.240.142 Standard EXECUTE
EXECUTE sofia/external/9051212121 at 218.165.240.142 transfer(3472 XML default)
2011-05-21 18:00:34.609375 [DEBUG] switch_ivr.c:1600
(sofia/external/9051212121 at 218.165.240.142) State Change CS_EXECUTE ->
CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:709 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [NOTICE] switch_ivr.c:1606 Transfer
sofia/external/9051212121 at 218.165.240.142 to XML[3472 at default]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:147
sofia/external/9051212121 at 218.165.240.142 SOFIA ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:77
sofia/external/9051212121 at 218.165.240.142 Standard ROUTING
2011-05-21 18:00:34.609375 [INFO] mod_dialplan_xml.c:331 Processing SIDOROV
S <9051212121>->3472 in context default
Dialplan: sofia/external/9051212121 at 218.165.240.142 parsing
[default->Forward All] continue=false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Regex (PASS) [Forward
All] destination_number(3472) =~ /3472/ break=on-false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Action log(INFO dialing
destination number)
Dialplan: sofia/external/9051212121 at 218.165.240.142 Action
bridge(sofia/gateway/mysipprovider/15191212121)
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:119
(sofia/external/9051212121 at 218.165.240.142) State Change CS_ROUTING ->
CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:240
sofia/external/9051212121 at 218.165.240.142 SOFIA EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:157
sofia/external/9051212121 at 218.165.240.142 Standard EXECUTE
EXECUTE sofia/external/9051212121 at 218.165.240.142 log(INFO dialing
destination number)
2011-05-21 18:00:34.609375 [INFO] mod_dptools.c:1183 dialing destination
number
EXECUTE sofia/external/9051212121 at 218.165.240.142
bridge(sofia/gateway/mysipprovider/15191212121)
2011-05-21 18:00:34.625000 [NOTICE] switch_channel.c:812 New Channel
sofia/external/15191212121 [5a8e5141-6d37-4db3-b5d4-5c3d1f952bbe]
2011-05-21 18:00:34.625000 [DEBUG] mod_sofia.c:4286
(sofia/external/15191212121)
 State Change CS_NEW -> CS_INIT
2011-05-21 18:00:34.625000 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:34.625000 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_INIT
2011-05-21 18:00:34.625000 [DEBUG] switch_core_state_machine.c:356
(sofia/external/15191212121) State INIT
2011-05-21 18:00:34.625000 [DEBUG] mod_sofia.c:84 sofia/external/15191212121
SOFIA INIT

[SK:]Pause starts here

2011-05-21 18:00:39.625000 [DEBUG] switch_nat.c:502 mapped public port 17090
protocol UDP to localport 17090

[SK:]I get short "busy" signal here, and hang up.  The rest is "hanging up"
process messages.

2011-05-21 18:00:44.421875 [DEBUG] sofia.c:4744 Channel
sofia/external/9051212121 at 218.165.240.142 entering state [terminated][487]
2011-05-21 18:00:44.421875 [DEBUG] switch_channel.c:2563
(sofia/external/9051212121 at 218.165.240.142) Callstate Change RINGING ->
HANGUP
2011-05-21 18:00:44.421875 [NOTICE] sofia.c:5384 Hangup
sofia/external/9051212121 at 218.165.240.142 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2011-05-21 18:00:44.421875 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/9051212121 at 218.165.240.142 [KILL]
2011-05-21 18:00:44.421875 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:44.437500 [DEBUG] switch_channel.c:2563
(sofia/external/15191212121) Callstate Change DOWN -> HANGUP
2011-05-21 18:00:44.437500 [NOTICE] switch_ivr_originate.c:3343 Hangup
sofia/external/15191212121 [CS_INIT] [ORIGINATOR_CANCEL]
2011-05-21 18:00:44.437500 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/15191212121 [KILL]
2011-05-21 18:00:44.437500 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:44.437500 [DEBUG] switch_ivr_originate.c:3500 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2011-05-21 18:00:44.437500 [INFO] mod_dptools.c:2623 Originate Failed. 
Cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.437500 [DEBUG] switch_core_session.c:2060
sofia/external/9051212121 at 218.165.240.142 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE going to sleep
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_HANGUP
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:560
(sofia/external/9051212121 at 218.165.240.142) State HANGUP
2011-05-21 18:00:44.437500 [DEBUG] mod_sofia.c:451
sofia/external/9051212121 at 218.165.240.142 Overriding SIP cause 487 with 487
from the other leg
2011-05-21 18:00:44.437500 [DEBUG] mod_sofia.c:457 Channel
sofia/external/9051212121 at 218.165.240.142 hanging up, cause:
ORIGINATOR_CANCEL
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:46
sofia/external/9051212121 at 218.165.240.142 Standard HANGUP, cause:
ORIGINATOR_CANCEL
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:560
(sofia/external/9051212121 at 218.165.240.142) State HANGUP going to sleep
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:351
(sofia/external/9051212121 at 218.165.240.142) State Change CS_HANGUP ->
CS_REPORTING
2011-05-21 18:00:44.437500 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change
CS_REPORTING
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:620
(sofia/external/9051212121 at 218.165.240.142) State REPORTING
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:53
sofia/external/9051212121 at 218.165.240.142 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:620
(sofia/external/9051212121 at 218.165.240.142) State REPORTING going to sleep
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:345
(sofia/external/9051212121 at 218.165.240.142) State Change CS_REPORTING ->
CS_DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:44.453125 [DEBUG] switch_core_session.c:1288 Session 1
(sofia/external/9051212121 at 218.165.240.142) Locked, Waiting on external
entities
2011-05-21 18:00:44.453125 [NOTICE] switch_core_session.c:1306 Session 1
(sofia/external/9051212121 at 218.165.240.142) Ended
2011-05-21 18:00:44.453125 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/9051212121 at 218.165.240.142 [CS_DESTROY]
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:449
(sofia/external/9051212121 at 218.165.240.142) Callstate Change HANGUP -> DOWN
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:452
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:462
(sofia/external/9051212121 at 218.165.240.142) State DESTROY
2011-05-21 18:00:44.453125 [DEBUG] mod_sofia.c:362
sofia/external/9051212121 at 218.165.240.142 SOFIA DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:60
sofia/external/9051212121 at 218.165.240.142 Standard DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:462
(sofia/external/9051212121 at 218.165.240.142) State DESTROY going to sleep
2011-05-21 18:00:44.625000 [DEBUG] switch_nat.c:502 mapped public port 17091
protocol UDP to localport 17091
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:356
(sofia/external/15191212121) State INIT going to 
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_HANGUP
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:560
(sofia/external/15191212121) State HANGUP
2011-05-21 18:00:44.625000 [DEBUG] mod_sofia.c:451
sofia/external/15191212121 Overriding SIP cause 487 with 487 from the other
leg
2011-05-21 18:00:44.625000 [DEBUG] mod_sofia.c:457 Channel
sofia/external/15191212121 hanging up, cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.625000 [DEBUG] mod_sofia.c:510 Sending CANCEL to
sofia/external/15191212121
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:46
sofia/external/15191212121 Standard HANGUP, cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:560
(sofia/external/15191212121) State HANGUP going to sleep
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:351
(sofia/external/15191212121) State Change CS_HANGUP -> CS_REPORTING
2011-05-21 18:00:44.625000 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_REPORTING
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:620
(sofia/external/15191212121) State REPORTING
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:53
sofia/external/15191212121 Standard REPORTING, cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:620
(sofia/external/15191212121) State REPORTING going to sleep
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:345
(sofia/external/15191212121) State Change CS_REPORTING -> CS_DESTROY
2011-05-21 18:00:44.625000 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:44.625000 [DEBUG] switch_core_session.c:1288 Session 2
(sofia/external/15191212121) Locked, Waiting on external entities
2011-05-21 18:00:44.625000 [NOTICE] switch_core_session.c:1306 Session 2
(sofia/external/15191212121) Ended
2011-05-21 18:00:44.625000 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/15191212121 [CS_DESTROY]
2011-05-21 18:00:44.640625 [DEBUG] switch_core_state_machine.c:449
(sofia/external/15191212121) Callstate Change HANGUP -> DOWN
2011-05-21 18:00:44.640625 [DEBUG] switch_core_state_machine.c:452
(sofia/external/15191212121) Running State Change CS_DESTROY
2011-05-21 18:00:44.640625 [DEBUG] switch_core_state_machine.c:462
(sofia/external/15191212121) State DESTROY
2011-05-21 18:00:44.640625 [DEBUG] mod_sofia.c:362
sofia/external/15191212121 SOFIA DESTROY
2011-05-21 18:00:49.640625 [DEBUG] switch_nat.c:562 unmapped public port
17090 protocol UDP to localport 17090
2011-05-21 18:00:54.640625 [DEBUG] switch_nat.c:562 unmapped public port
17091 protocol UDP to localport 17091
2011-05-21 18:00:54.640625 [DEBUG] switch_core_state_machine.c:60
sofia/external/15191212121 Standard DESTROY
2011-05-21 18:00:54.640625 [DEBUG] switch_core_state_machine.c:462
(sofia/external/15191212121) State DESTROY going to sleep


Thanks,

Sergei Kamen

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6390451.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gabe at gundy.org  Sun May 22 08:14:09 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 21 May 2011 22:14:09 -0600
Subject: [Freeswitch-users] Question about faxing between FreeSwitch
	servers
In-Reply-To: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
References: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
Message-ID: <BANLkTi=7U-rA-oLd7oexE7rpYxLWpmmcRA@mail.gmail.com>

On Sat, May 21, 2011 at 1:17 PM, Chris Cureau <cmcureau at gmail.com> wrote:
> I am trying to implement a workable fax solution between two FreeSwitch
> servers.

This might be crazy, but are you FAXing others with these two server
also, or is it just these two servers FAXing back and forth? Anyway,
if it's just these two and you own both side, forget FAX and figure
out another way to get docs back and forth :)

If you have to FAX others as well, sorry for the noise and good luck.

Best,
Gabe


From curriegrad2004 at gmail.com  Sun May 22 08:44:25 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 21 May 2011 21:44:25 -0700
Subject: [Freeswitch-users] Question about faxing between FreeSwitch
	servers
In-Reply-To: <BANLkTi=7U-rA-oLd7oexE7rpYxLWpmmcRA@mail.gmail.com>
References: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
	<BANLkTi=7U-rA-oLd7oexE7rpYxLWpmmcRA@mail.gmail.com>
Message-ID: <BANLkTimur3c_2oCTqJSEgmMwyWiUe+pOBw@mail.gmail.com>

Actually an encrypted e-mail solution would work better for faxing
between 2 FreeSwitch boxes. Can you do this in POST with HTTP over
SSL?

On Sat, May 21, 2011 at 9:14 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Sat, May 21, 2011 at 1:17 PM, Chris Cureau <cmcureau at gmail.com> wrote:
>> I am trying to implement a workable fax solution between two FreeSwitch
>> servers.
>
> This might be crazy, but are you FAXing others with these two server
> also, or is it just these two servers FAXing back and forth? Anyway,
> if it's just these two and you own both side, forget FAX and figure
> out another way to get docs back and forth :)
>
> If you have to FAX others as well, sorry for the noise and good luck.
>
> Best,
> 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 chat2jesse at gmail.com  Sun May 22 11:13:55 2011
From: chat2jesse at gmail.com (jesse)
Date: Sun, 22 May 2011 00:13:55 -0700
Subject: [Freeswitch-users] Project Announcement and Request for
	Direction
In-Reply-To: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>
References: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>
Message-ID: <BANLkTimKS+B1HH=oWdn+azaugVQqZJ-NhA@mail.gmail.com>

Leveraging freeswitch event socket & customized module can accomplish all
your goals.
On May 16, 2011 7:51 AM, "Kurtis Heimerl" <kheimerl at cs.berkeley.edu> wrote:
> Hello Freeswitch Users!
>
> My name is Kurtis Heimerl, and I'm a graduate researcher at the
> University of California, Berkeley in the Technology and
> Infrastructure for Emerging Regions (TIER) group under Eric Brewer.
> We're currently investigating the use of OpenBTS
> (http://openbts.sourceforge.net/) for providing cellular coverage via
> low-cost base stations in low-density parts of the world. This project
> is called The Village Base Station. In support of that project, we're
> also investigating the use of Freeswitch to support OpenBTS and
> provide a flexible, extensible, and simple platform for deploying
> voice/sms/data applications on the basestation itself. Towards this
> end, I'm asking you, the freeswitch community, for advice and
> direction on some of our research goals.
>
> The basic story is simple. OpenBTS uses a software radio and generic
> PC to provide cellular service. As a byproduct of this architecture,
> we also gain the ability to run freeswitch applications "locally"
> (concurrently with OpenBTS), and take advantage of the benefits this
> tight coupling provides us. These benefits are numerous; calls/SMS
> between BTS users can be much cheaper, as they do not use any backhaul
> bandwidth. Applications can query the system for more information
> about users, such as location or status. As an example, unlike
> traditional GSM telephony applications, we are able to query if users
> are currently available on the network. This could be used to create
> voice "chat lists", which tell participants which of their friends are
> currently within cellular range.
>
> We foresee 6 features required to support these "local" freeswitch
> applications on our OpenBTS system. I'm very curious how the
> freeswitch community feels about these possible additions, as well how
> they might be implemented.
>
> There are a few that are already available in freeswitch, but may be
> rougher than we would like. These include:
>
> 1) Identity: The ability to query for user's status, numbers, etc.
> This seems simple enough in the existing system. However, we'd like to
> provide hooks for applications to act on these sign-ons or offs. For
> instance, an app may hold messages until a phone logs onto the system
> and push them then. My understanding is that this should be simple,
> probably hooking onto "SIP presence" events?
> 2) Storage: Freeswitch currently seems to support only per-application
> storage, with limited support for cross-application storage (mostly
> user directories). This is occasionally problematic: one issue we've
> heard is that it is difficult to place messages into voice mailboxes
> from other apps. We'd like a more unified storage framework. Or...
> 3) "Pipes": This is the ability to pass messages between freeswitch
> apps. This seems pretty well supported though simple dialplan
> interactions, though the modules themselves may not provide enough
> functionality. Is there a way to do this inside of apps? I consider
> this an alternative to the storage framework discussed in #2
>
> Lastly, there are three functions we don't believe are well-supported
> in freeswitch. These are...
> 1) Privacy: We expect our BTS to be used in politically sensitive
> areas. Given this, freeswitch could provide an anonymity layer,
> providing short term phone/SMS numbers, or directing communications
> through more secure layers (e.g., Tor).
> 2) Asynchrony: While freeswitch seems to support basic asynchrony
> though its event system, I couldn't find any way to delay events for
> indeterminate times. For instance, we may want to schedule a traffic
> warning for 1PM every Wednesday to every phone currently on the
> system. Is there a way to do that currently?
> 3) SMS: Freeswitch seems to currently support SIP chat messages (using
> SIMPLE?). We need to either extend OpenBTS to speak SIMPLE, or extend
> freeswitch to speak OpenBTS's SMS protocol. Neither seems particularly
> difficult. This will allow our apps to send and receive both voice and
> SMS messages from users.
>
> We believe these core functions will enable a wide variety of BTS
> applications. I have a laundry list of those, but I'll omit them for
> sake of space.
>
> If any members of the community (that means you!) have any directions,
> ideas, projects, or thoughts, please pass them on! We're just
> beginning this part of the project, and getting the lay of the land.
> Feedback is critical at this point.
>
> 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 peter.olsson at visionutveckling.se  Sun May 22 13:21:31 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 22 May 2011 11:21:31 +0200
Subject: [Freeswitch-users] setting up forward incoming calls
	to	external gateway
In-Reply-To: <1306019791787-6390451.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>,
	<1306019791787-6390451.post@n2.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>

You will need to enable siptrace also - "sofia global siptrace on".

My guess is this a NAT issue.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Kamen [sireeps at gmail.com]
Skickat: den 22 maj 2011 01:16
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] setting up forward incoming calls to       external gateway

Hi Michael,

Frankly I am not sure what exactly you referring to.  I am completely new to
freeswitch.  If you are talking about running command (I found it in online
doc):

TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch

then there is a problem,  Like I said I am running it in Windows. Running
FreeSWITCH\FreeSwitchConsole.exe. Besides I have no idea what pastebin is.
Only can guess it preserves the colors of debug printouts.

I have instead prepared below a  debug printout I manage to get in the
console.  Also added my comment.  Please, could anyone have a look at it.
Briefly, I want a simple functionality.  If call comes in to my SIP provider
I want it to be forwarded to another fixed external number.  That is it!
Seems like a straight forward task for freeswitch. The call seems to be
forwarded (according to the log), but something stops it and it gets paused
for 10 seconds, until busy signal and call is disconnected afterwards.

Thanks in advance.

Sergei Kamen

==============================================================

public.xml has extension:

 <extension name="Forward All">
        <condition field="caller_id_number" expression="^(\d+)$">
                <action application="transfer" data="3472 XML default"/>
        </condition>
 </extension>

and default.xml has:

    <extension name="Forward All">
        <condition field="destination_number" expression="3472">
        <action application="bridge"
data="sofia/gateway/freephoneline/15191212121"/>
        </condition>
    </extension>

The following printout in my console is as follows from the beginning:
------------------------------

nta: bind([::1]:5070;transport=*): No such file or directory
nua: initializing SIP stack failed

[SK:] could it be the error above?

2011-05-21 17:59:38.187500 [ERR] sofia.c:1533 Error Creating SIP UA for
profile: internal-ipv6
2011-05-21 17:59:40.875000 [DEBUG] switch_nat.c:502 mapped public port 5080
protocol TCP to localport 5080
2011-05-21 17:59:40.875000 [DEBUG] sofia.c:1487 Created TCP nat mapping for
external port 5080
2011-05-21 17:59:40.875000 [DEBUG] switch_nat.c:502 mapped public port 5070
protocol TCP to localport 5070
2011-05-21 17:59:40.875000 [DEBUG] sofia.c:1487 Created TCP nat mapping for
internal port 5070
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1539 Created agent for external
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1576 Set params for external
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1599 Activated db for external
2011-05-21 17:59:40.906250 [DEBUG] sofia.c:1626 Starting thread for external
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1539 Created agent for internal
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1576 Set params for internal
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1599 Activated db for internal
2011-05-21 17:59:40.921875 [DEBUG] sofia.c:1626 Starting thread for internal
2011-05-21 17:59:41.921875 [NOTICE] sofia_reg.c:367 Registering
mysipprovider
2011-05-21 17:59:44.062500 [DEBUG] sofia_reg.c:1922 Authenticating
'16472222222' with 'Digest:"voip.mysipprovider.ca":16472222222:mypassword'.
2011-05-21 17:59:44.921875 [DEBUG] sofia_reg.c:331 Registered mysipprovider
2011-05-21 18:00:00.984375 [WARNING] sofia.c:4065 Ping succeeded
mysipprovider with code 501 - count -1/1/1, state UP

[SK:]I am calling here to my SIP provider number (16472222222) from another
phone (9051212121).  I expect it to forward to a third number (15191212121)
with updated Caller ID of my provider.

2011-05-21 18:00:34.609375 [NOTICE] switch_channel.c:812 New Channel
sofia/external/9051212121 at 218.165.240.142
[bc74d840-a951-458e-9eec-c70941fdd05a]
2011-05-21 18:00:34.609375 [DEBUG] sofia.c:4744 Channel
sofia/external/9051212121 at 218.165.240.142 entering state [received][100]
2011-05-21 18:00:34.609375 [DEBUG] sofia.c:4755 Remote SDP:
v=0
o=Sippy 383374256 0 IN IP4 218.165.240.142
s=-
t=0 0
m=audio 35774 RTP/AVP 0
c=IN IP4 208.72.120.90

2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_NEW
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:338
(sofia/external/9051212121 at 218.165.240.142) State NEW
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:4637 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-05-21 18:00:34.609375 [DEBUG] sofia_glue.c:2760 Set Codec
sofia/external/9051212121 at 218.165.240.142 PCMU/8000 20 ms 160 samples 64000
bits
2011-05-21 18:00:34.609375 [DEBUG] sofia.c:4922
(sofia/external/9051212121 at 218.165.240.142) State Change CS_NEW -> CS_INIT
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_INIT
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:356
(sofia/external/9051212121 at 218.165.240.142) State INIT
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:84
sofia/external/9051212121 at 218.165.240.142 SOFIA INIT
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:124
(sofia/external/9051212121 at 218.165.240.142) State Change CS_INIT ->
CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:356
(sofia/external/9051212121 at 218.165.240.142) State INIT going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_channel.c:1668
(sofia/external/9051212121 at 218.165.240.142) Callstate Change DOWN -> RINGING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:147
sofia/external/9051212121 at 218.165.240.142 SOFIA ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:77
sofia/external/9051212121 at 218.165.240.142 Standard ROUTING
2011-05-21 18:00:34.609375 [INFO] mod_dialplan_xml.c:331 Processing MICHAEL
S <9051212121>->3472 in context public
Dialplan: sofia/external/9051212121 at 218.165.240.142 parsing [public->Forward
All] continue=false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Regex (PASS) [Forward
All] caller_id_number(9051212121) =~ /^(\d+)$/ break=on-false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Action transfer(3472 XML
default)
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:119
(sofia/external/9051212121 at 218.165.240.142) State Change CS_ROUTING ->
CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:240
sofia/external/9051212121 at 218.165.240.142 SOFIA EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:157
sofia/external/9051212121 at 218.165.240.142 Standard EXECUTE
EXECUTE sofia/external/9051212121 at 218.165.240.142 transfer(3472 XML default)
2011-05-21 18:00:34.609375 [DEBUG] switch_ivr.c:1600
(sofia/external/9051212121 at 218.165.240.142) State Change CS_EXECUTE ->
CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:709 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [NOTICE] switch_ivr.c:1606 Transfer
sofia/external/9051212121 at 218.165.240.142 to XML[3472 at default]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:147
sofia/external/9051212121 at 218.165.240.142 SOFIA ROUTING
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:77
sofia/external/9051212121 at 218.165.240.142 Standard ROUTING
2011-05-21 18:00:34.609375 [INFO] mod_dialplan_xml.c:331 Processing SIDOROV
S <9051212121>->3472 in context default
Dialplan: sofia/external/9051212121 at 218.165.240.142 parsing
[default->Forward All] continue=false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Regex (PASS) [Forward
All] destination_number(3472) =~ /3472/ break=on-false
Dialplan: sofia/external/9051212121 at 218.165.240.142 Action log(INFO dialing
destination number)
Dialplan: sofia/external/9051212121 at 218.165.240.142 Action
bridge(sofia/gateway/mysipprovider/15191212121)
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:119
(sofia/external/9051212121 at 218.165.240.142) State Change CS_ROUTING ->
CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:359
(sofia/external/9051212121 at 218.165.240.142) State ROUTING going to sleep
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] mod_sofia.c:240
sofia/external/9051212121 at 218.165.240.142 SOFIA EXECUTE
2011-05-21 18:00:34.609375 [DEBUG] switch_core_state_machine.c:157
sofia/external/9051212121 at 218.165.240.142 Standard EXECUTE
EXECUTE sofia/external/9051212121 at 218.165.240.142 log(INFO dialing
destination number)
2011-05-21 18:00:34.609375 [INFO] mod_dptools.c:1183 dialing destination
number
EXECUTE sofia/external/9051212121 at 218.165.240.142
bridge(sofia/gateway/mysipprovider/15191212121)
2011-05-21 18:00:34.625000 [NOTICE] switch_channel.c:812 New Channel
sofia/external/15191212121 [5a8e5141-6d37-4db3-b5d4-5c3d1f952bbe]
2011-05-21 18:00:34.625000 [DEBUG] mod_sofia.c:4286
(sofia/external/15191212121)
 State Change CS_NEW -> CS_INIT
2011-05-21 18:00:34.625000 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:34.625000 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_INIT
2011-05-21 18:00:34.625000 [DEBUG] switch_core_state_machine.c:356
(sofia/external/15191212121) State INIT
2011-05-21 18:00:34.625000 [DEBUG] mod_sofia.c:84 sofia/external/15191212121
SOFIA INIT

[SK:]Pause starts here

2011-05-21 18:00:39.625000 [DEBUG] switch_nat.c:502 mapped public port 17090
protocol UDP to localport 17090

[SK:]I get short "busy" signal here, and hang up.  The rest is "hanging up"
process messages.

2011-05-21 18:00:44.421875 [DEBUG] sofia.c:4744 Channel
sofia/external/9051212121 at 218.165.240.142 entering state [terminated][487]
2011-05-21 18:00:44.421875 [DEBUG] switch_channel.c:2563
(sofia/external/9051212121 at 218.165.240.142) Callstate Change RINGING ->
HANGUP
2011-05-21 18:00:44.421875 [NOTICE] sofia.c:5384 Hangup
sofia/external/9051212121 at 218.165.240.142 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2011-05-21 18:00:44.421875 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/9051212121 at 218.165.240.142 [KILL]
2011-05-21 18:00:44.421875 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:44.437500 [DEBUG] switch_channel.c:2563
(sofia/external/15191212121) Callstate Change DOWN -> HANGUP
2011-05-21 18:00:44.437500 [NOTICE] switch_ivr_originate.c:3343 Hangup
sofia/external/15191212121 [CS_INIT] [ORIGINATOR_CANCEL]
2011-05-21 18:00:44.437500 [DEBUG] switch_channel.c:2579 Send signal
sofia/external/15191212121 [KILL]
2011-05-21 18:00:44.437500 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:44.437500 [DEBUG] switch_ivr_originate.c:3500 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2011-05-21 18:00:44.437500 [INFO] mod_dptools.c:2623 Originate Failed.
Cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.437500 [DEBUG] switch_core_session.c:2060
sofia/external/9051212121 at 218.165.240.142 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:366
(sofia/external/9051212121 at 218.165.240.142) State EXECUTE going to sleep
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_HANGUP
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:560
(sofia/external/9051212121 at 218.165.240.142) State HANGUP
2011-05-21 18:00:44.437500 [DEBUG] mod_sofia.c:451
sofia/external/9051212121 at 218.165.240.142 Overriding SIP cause 487 with 487
from the other leg
2011-05-21 18:00:44.437500 [DEBUG] mod_sofia.c:457 Channel
sofia/external/9051212121 at 218.165.240.142 hanging up, cause:
ORIGINATOR_CANCEL
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:46
sofia/external/9051212121 at 218.165.240.142 Standard HANGUP, cause:
ORIGINATOR_CANCEL
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:560
(sofia/external/9051212121 at 218.165.240.142) State HANGUP going to sleep
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:351
(sofia/external/9051212121 at 218.165.240.142) State Change CS_HANGUP ->
CS_REPORTING
2011-05-21 18:00:44.437500 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:320
(sofia/external/9051212121 at 218.165.240.142) Running State Change
CS_REPORTING
2011-05-21 18:00:44.437500 [DEBUG] switch_core_state_machine.c:620
(sofia/external/9051212121 at 218.165.240.142) State REPORTING
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:53
sofia/external/9051212121 at 218.165.240.142 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:620
(sofia/external/9051212121 at 218.165.240.142) State REPORTING going to sleep
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:345
(sofia/external/9051212121 at 218.165.240.142) State Change CS_REPORTING ->
CS_DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/9051212121 at 218.165.240.142 [BREAK]
2011-05-21 18:00:44.453125 [DEBUG] switch_core_session.c:1288 Session 1
(sofia/external/9051212121 at 218.165.240.142) Locked, Waiting on external
entities
2011-05-21 18:00:44.453125 [NOTICE] switch_core_session.c:1306 Session 1
(sofia/external/9051212121 at 218.165.240.142) Ended
2011-05-21 18:00:44.453125 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/9051212121 at 218.165.240.142 [CS_DESTROY]
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:449
(sofia/external/9051212121 at 218.165.240.142) Callstate Change HANGUP -> DOWN
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:452
(sofia/external/9051212121 at 218.165.240.142) Running State Change CS_DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:462
(sofia/external/9051212121 at 218.165.240.142) State DESTROY
2011-05-21 18:00:44.453125 [DEBUG] mod_sofia.c:362
sofia/external/9051212121 at 218.165.240.142 SOFIA DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:60
sofia/external/9051212121 at 218.165.240.142 Standard DESTROY
2011-05-21 18:00:44.453125 [DEBUG] switch_core_state_machine.c:462
(sofia/external/9051212121 at 218.165.240.142) State DESTROY going to sleep
2011-05-21 18:00:44.625000 [DEBUG] switch_nat.c:502 mapped public port 17091
protocol UDP to localport 17091
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:356
(sofia/external/15191212121) State INIT going to
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_HANGUP
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:560
(sofia/external/15191212121) State HANGUP
2011-05-21 18:00:44.625000 [DEBUG] mod_sofia.c:451
sofia/external/15191212121 Overriding SIP cause 487 with 487 from the other
leg
2011-05-21 18:00:44.625000 [DEBUG] mod_sofia.c:457 Channel
sofia/external/15191212121 hanging up, cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.625000 [DEBUG] mod_sofia.c:510 Sending CANCEL to
sofia/external/15191212121
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:46
sofia/external/15191212121 Standard HANGUP, cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:560
(sofia/external/15191212121) State HANGUP going to sleep
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:351
(sofia/external/15191212121) State Change CS_HANGUP -> CS_REPORTING
2011-05-21 18:00:44.625000 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:320
(sofia/external/15191212121) Running State Change CS_REPORTING
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:620
(sofia/external/15191212121) State REPORTING
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:53
sofia/external/15191212121 Standard REPORTING, cause: ORIGINATOR_CANCEL
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:620
(sofia/external/15191212121) State REPORTING going to sleep
2011-05-21 18:00:44.625000 [DEBUG] switch_core_state_machine.c:345
(sofia/external/15191212121) State Change CS_REPORTING -> CS_DESTROY
2011-05-21 18:00:44.625000 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/15191212121 [BREAK]
2011-05-21 18:00:44.625000 [DEBUG] switch_core_session.c:1288 Session 2
(sofia/external/15191212121) Locked, Waiting on external entities
2011-05-21 18:00:44.625000 [NOTICE] switch_core_session.c:1306 Session 2
(sofia/external/15191212121) Ended
2011-05-21 18:00:44.625000 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/15191212121 [CS_DESTROY]
2011-05-21 18:00:44.640625 [DEBUG] switch_core_state_machine.c:449
(sofia/external/15191212121) Callstate Change HANGUP -> DOWN
2011-05-21 18:00:44.640625 [DEBUG] switch_core_state_machine.c:452
(sofia/external/15191212121) Running State Change CS_DESTROY
2011-05-21 18:00:44.640625 [DEBUG] switch_core_state_machine.c:462
(sofia/external/15191212121) State DESTROY
2011-05-21 18:00:44.640625 [DEBUG] mod_sofia.c:362
sofia/external/15191212121 SOFIA DESTROY
2011-05-21 18:00:49.640625 [DEBUG] switch_nat.c:562 unmapped public port
17090 protocol UDP to localport 17090
2011-05-21 18:00:54.640625 [DEBUG] switch_nat.c:562 unmapped public port
17091 protocol UDP to localport 17091
2011-05-21 18:00:54.640625 [DEBUG] switch_core_state_machine.c:60
sofia/external/15191212121 Standard DESTROY
2011-05-21 18:00:54.640625 [DEBUG] switch_core_state_machine.c:462
(sofia/external/15191212121) State DESTROY going to sleep


Thanks,

Sergei Kamen

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6390451.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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!DSPAM:4dd8496432761158895848!



From adminjew at gmail.com  Sun May 22 15:10:03 2011
From: adminjew at gmail.com (Yitzchok)
Date: Sun, 22 May 2011 07:10:03 -0400
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
	<BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
	<BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>
Message-ID: <BANLkTikkSWcge7f0XxzwL-iV2ehj=R7kUA@mail.gmail.com>

Try this

cd /usr/local/src/freeswitch/src/mod/languages/mod_managed/
git apply mono28.patch
make reswig
make



Yitzchok


On Sat, May 21, 2011 at 6:42 PM, budi wibowo <bwibowo at gmail.com> wrote:

> failed with some glib issue on source code compiling, then i use yum as
> suggested.
> mono installed, thx
> Mono JIT compiler version 2.10.2 (tarball Mon Apr 18 19:06:50 UTC 2011)
> Copyright (C) 2002-2011 Novell, Inc and Contributors. www.mono-project.com
>         TLS:           __thread
>         SIGSEGV:       altstack
>         Notifications: epoll
>         Architecture:  amd64
>         Disabled:      none
>         Misc:          debugger softdebug
>         LLVM:          supported, not enabled.
>         GC:            Included Boehm (with typed GC and Parallel Mark)
>
> when i try to install mod_managed i found this error, may be some missing
> config in my linux box
>
> making all mod_managed
> Compiling freeswitch_managed.cpp...
> g++ -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
> In file included from freeswitch_managed.cpp:35:
> freeswitch_managed.h:43:18: error: glib.h: No such file or directory
> freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or
> directory
> freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such file
> or directory
> freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such
> file or directory
> freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such
> file or directory
> freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such file or
> directory
> freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No such
> file or directory
>
>
> thx
> budi wibowo
>
> On Wed, May 18, 2011 at 9:06 AM, Yitzchok <adminjew at gmail.com> wrote:
>
>> Make sure mono is installed and check which version using "mono -V" and
>> follow the instructions on http://wiki.freeswitch.org/wiki/Mod_managed let
>> us know if you get into any errors.
>>
>> This helped me for installing mono
>> http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
>>
>>
>> <http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
>> Yitzchok
>>
>>
>> On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>
>>> hi
>>> i have fs running on centos 5.5, any body knows the combination version
>>> of software need to be installed like mono etc.
>>> have tried few combination but always failed compiling
>>>
>>>
>>> thx
>>> budi
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Sun May 22 21:52:25 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 22 May 2011 19:52:25 +0200
Subject: [Freeswitch-users] Question about faxing between FreeSwitch
	servers
In-Reply-To: <4DD82827.9080405@gmail.com>
References: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
	<4DD82827.9080405@gmail.com>
Message-ID: <5F2C8090-52CD-414F-BE90-1F8BE126BDA0@ipeva.fr>

G711 is absolutely unreliable to transmit fax.
RTT has to be perfect, no jitter at all, so basically, LAN recommended.
T38 was invented for that reason.

I personally have more and more success with T38, with FS doing passthrough only:
-from Patton endpoint (SN4961 and others) to Dialogic SS7 gateway
-from Mediatrix endpoints (4102) to Dialogic SS7 gateway
-from Zoiper softphone to Dialogic SS7 gateway

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

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

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




Le 21/05/2011 ? 23:01, Meftah Tayeb a ?crit :

> T.38 is good but is a pain.
> i strongly suggest you to pass it through PCMU (G.711U) codec, sunse you're willing to use E1 boards
> no transcoding required and less ressources
> thank you
> On 21/05/2011 21:17, Chris Cureau wrote:
>> 
>> Hi, everyone!
>> 
>> I am trying to implement a workable fax solution between two FreeSwitch servers.  This is what I had in mind so far:
>> 
>> T1 PRI <--> Sangoma A102 <--> FreeSwitch (host) <--> T.38 SIP (internet) <--> FreeSwitch (remote) <--> Sangoma B600D <--> FAX machine
>> 
>> I am assuming here that the T.38 protocol between FreeSwitch servers will ensure that the data transfers between the two sites successfully.  Would this actually be a two step process, where one server receives the fax (say, from the PRI) on one call, then sends to the second server, and the second server initiates a second call to the fax machine?
>> 
>> The object here is to successfully pass faxes using SIP.  Is there a better way to handle this?
>> 
>> Thanks in advance,
>> 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
>>   
> 
> 
> -- 
> Meftah Tayeb
> inum: +883510001288000
> phone: +13477595883
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sireeps at gmail.com  Sun May 22 22:08:38 2011
From: sireeps at gmail.com (Kamen)
Date: Sun, 22 May 2011 11:08:38 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls
	to	external gateway
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
Message-ID: <1306087718406-6392049.post@n2.nabble.com>

Hi Peter, Michael

Thanks for your suggestion.  I enabled the sip trace and got the following
traces while calling inbound:

=====================================
freeswitch at mars> sofia global siptrace on

+OK Global siptrace on

freeswitch at mars> recv 869 bytes from udp/[218.165.240.142]:5060 at
16:23:07.562500:
   ------------------------------------------------------------------------
   INVITE
sip:gw+mysipprovider at 192.168.123.100:5080;transport=udp;gw=mysipprovider
SIP/2.0
   Via: SIP/2.0/UDP
218.165.240.142:5060;branch=z9hG4bK-d8754z-03f1d4147d39e75a-1---d8754z-;rport
   Via: SIP/2.0/UDP
218.165.240.142:5061;branch=z9hG4bK-6yit3js32dookuqy;rport=5061
   Max-Forwards: 69
   Record-Route: &lt;sip:218.165.240.142;lr&gt;
   Contact: "Anonymous"<sip:218.165.240.142:5061>
   To: &lt;sip:16472222222 at 218.165.240.142&gt;
   From: "MICHAEL
S"&lt;sip:9051212121 at 218.165.240.142&gt;;tag=uvgya25q6jekqd53.o
   Call-ID: 80741E4A at 208.72.120.66~o
   CSeq: 120 INVITE
   Expires: 300
   Content-Disposition: session
   Content-Type: application/sdp
   User-Agent: Sippy
   cisco-GUID: 51683594-2224034272-2239627291-559912524
   h323-conf-id: 51683594-2224034272-2239627291-559912524
   Content-Length: 109
   
   v=0
   o=Sippy 450031128 0 IN IP4 218.165.240.142
   s=-
   t=0 0
   m=audio 34260 RTP/AVP 0
   c=IN IP4 208.72.120.90
   ------------------------------------------------------------------------
send 506 bytes to udp/[218.165.240.142]:5060 at 16:23:07.562500:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
218.165.240.142:5060;branch=z9hG4bK-d8754z-03f1d4147d39e75a-1---d8754z-;rport=5060
   Via: SIP/2.0/UDP
218.165.240.142:5061;branch=z9hG4bK-6yit3js32dookuqy;rport=5061
   Record-Route: &lt;sip:218.165.240.142;lr&gt;
   From: "MICHAEL
S"&lt;sip:9051212121 at 218.165.240.142&gt;;tag=uvgya25q6jekqd53.o
   To: &lt;sip:16472222222 at 218.165.240.142&gt;
   Call-ID: 80741E4A at 208.72.120.66~o
   CSeq: 120 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-57b6255 2011-03-22 15-15-09
-0500
   Content-Length: 0
   
   ------------------------------------------------------------------------
2011-05-22 12:23:07.562500 [NOTICE] switch_channel.c:812 New Channel
sofia/external/9051212121 at 218.165.240.142
[2b6b8394-faab-4704-a3be-158396d08477]
2011-05-22 12:23:07.562500 [DEBUG] sofia.c:4744 Channel
sofia/external/9051212121 at 218.165.240.142 entering state [received][100]
2011-05-22 12:23:07.562500 [DEBUG] sofia.c:4755 Remote SDP:
v=0

o=Sippy 450031128 0 IN IP4 218.165.240.142

s=-

t=0 0

m=audio 34260 RTP/AVP 0

c=IN IP4 208.72.120.90

============================================================

I am not sure how to interpret above, but I hope this will give you some
clue.  I also have the printout file, but I do not see how I can attach it
here.  Do not think putting over 2000 lines into this message would be
polite :) 

After you suggestion that it could be a NAT issue, I checked the log more
closely and found the following at the very startup:
============================================================


2011-05-22 12:21:40.312500 [INFO] switch_event.c:615 Activate Eventing
Engine.
2011-05-22 12:21:40.343750 [DEBUG] switch_event.c:594 Create event dispatch
thread 0
2011-05-22 12:21:41.328125 [INFO] switch_nat.c:411 Scanning for NAT
2011-05-22 12:21:41.343750 [DEBUG] switch_nat.c:168 Checking for PMP 1/5
2011-05-22 12:21:41.343750 [ERR] switch_nat.c:199 Error checking for PMP
[general error]
2011-05-22 12:21:41.343750 [DEBUG] switch_nat.c:416 Checking for UPnP
2011-05-22 12:21:47.046875 [INFO] switch_nat.c:425 NAT detected type: upnp,
ExtIP: '173.248.202.119'
2011-05-22 12:21:47.046875 [ERR] switch_nat.c:251 Bind Error
2011-05-22 12:21:47.046875 [ERR] switch_nat.c:358 Unable to initialize NAT
thread

==================================================================

Those are the very first lines of the traces. The last line of this quote
kind of points to the NAT problem. What do you think, could it be related? 
Although I did an experiment and connected my PC over DMZ on my router.  The
result was exactly the same.

What bugs me is that I can see communication in the trace. The call for sure
gets to FS, where it being processed and kind of forwarded to the defined
number.  Where it gets stalled.  If there was a problem with SIP, would it
even get to FS from outside?

Anyway, thanks for looking into it.  Appreciate it. It helps me to look at
the problem from other angles. Hope to hear more of your suggestions. 
Meanwhile I will keep banging my head against it "from different angles". 
:)  

Regards,

Sergei Kamen


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6392049.html
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From david.ponzone at ipeva.fr  Sun May 22 22:15:39 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 22 May 2011 20:15:39 +0200
Subject: [Freeswitch-users] Question about faxing between FreeSwitch
	servers
In-Reply-To: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
References: <BANLkTi=nn-CQLxgMFwiops+onuLjrzVpug@mail.gmail.com>
Message-ID: <96D84DFC-5BE8-4992-A5DF-8491E0A2ACC7@ipeva.fr>

Chris,

With T38, the simplest seems to be a one-step process:
the call received from PRI is routed to the fax machine, with T38 enabled
The 2 FS should not have any issue to talk T38.

or to avoid T38 at all, as the others proposed, just find a way to transmit the fax as data (rsync, scp, ....)

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

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

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




Le 21/05/2011 ? 21:17, Chris Cureau a ?crit :

> Hi, everyone!
> 
> I am trying to implement a workable fax solution between two FreeSwitch servers.  This is what I had in mind so far:
> 
> T1 PRI <--> Sangoma A102 <--> FreeSwitch (host) <--> T.38 SIP (internet) <--> FreeSwitch (remote) <--> Sangoma B600D <--> FAX machine
> 
> I am assuming here that the T.38 protocol between FreeSwitch servers will ensure that the data transfers between the two sites successfully.  Would this actually be a two step process, where one server receives the fax (say, from the PRI) on one call, then sends to the second server, and the second server initiates a second call to the fax machine?
> 
> The object here is to successfully pass faxes using SIP.  Is there a better way to handle this?
> 
> Thanks in advance,
> Chris 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Mon May 23 00:33:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 22 May 2011 21:33:33 +0100
Subject: [Freeswitch-users] setting up forward incoming calls
	to	external gateway
In-Reply-To: <1306087718406-6392049.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
Message-ID: <E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>

"Do not think putting over 2000 lines into this message would be
polite :) "

You can always paste it into http://pastebin.freeswitch.org and then put the URL in the email :)

Steve on iPhone

On 22 May 2011, at 19:08, Kamen <sireeps at gmail.com> wrote:

> Do not think putting over 2000 lines into this message would be
> polite :)
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From sireeps at gmail.com  Mon May 23 02:34:56 2011
From: sireeps at gmail.com (Kamen)
Date: Sun, 22 May 2011 15:34:56 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls
	to	external gateway
In-Reply-To: <E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
Message-ID: <1306103696780-6392576.post@n2.nabble.com>


Steven Ayre wrote:
> 
> You can always paste it into http://pastebin.freeswitch.org and then put
> the URL in the email :)
> 

Oh, great!  Thanks Steven!  I was wondering what that pastebin means.

I am confused, though.  While trying to get into pastebin, I was asked for a
login. Do I need to sign up to use it?  I tried to login with my credentials
for this forum/mailing list and got rejected. Please help me out here.

Thanks,

Sergei Kamen



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6392576.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Mon May 23 02:47:21 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Sun, 22 May 2011 18:47:21 -0400
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
Message-ID: <BANLkTinCpQMjATO_pKCZkog7gO3pNz2vLg@mail.gmail.com>

You'll need to use proxy media to pass NSE across a bridge:

http://wiki.freeswitch.org/wiki/Proxy_Media

Please keep in mind there are several advantages and disadvantages to
using this media mode.

On Fri, May 20, 2011 at 2:54 PM, Sean Eichhorn <seichhorn at gci.com> wrote:
> I have a freeswitch system that (for the most part) works flawlessly for
> me.? However, I need to have an additional rtpmap line in the SDP when a
> call is answered.
>
> My client sends the following SDP upon answering the call :
>
> ?? m=audio 17214 RTP/AVP 0 19 101 100
>
> ?? c=IN IP4 192.168.98.79
>
> ?? a=rtpmap:0 PCMU/8000
>
> ?? a=rtpmap:19 CN/8000
>
> ?? a=rtpmap:101 telephone-event/8000
>
> ?? a=fmtp:101 0-16
>
> ?? a=rtpmap:100 X-NSE/8000
>
> ?? a=fmtp:100 192-194
>
>
>
> Freeswitch forwards the following SDP to the external gateway :
>
> ?? m=audio 20654 RTP/AVP 0 19 101
>
> ?? c=IN IP4 66.223.187.208
>
> ?? a=rtpmap:0 PCMU/8000
>
> ?? a=rtpmap:19 CN/8000
>
> ?? a=rtpmap:101 telephone-event/8000
>
> ?? a=fmtp:101 0-16
>
>
>
> The other information appears fine.? Everything works, but my client-side is
> not receiving the a=rtpmap:100 line.
>
> This issue appears regardless of whether or not I?m in proxy mode, bypass
> mode, or neither.
>
>
>
> Using "sip_append_audio_sdp? has no effect on the answering SDP, only the
> initial offer.
>
>
>
> Any ideas?
>
>
>
> 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
>
>



-- 
Kristian Kielhofner


From bwibowo at gmail.com  Mon May 23 02:49:21 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Mon, 23 May 2011 05:49:21 +0700
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTikkSWcge7f0XxzwL-iV2ehj=R7kUA@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
	<BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
	<BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>
	<BANLkTikkSWcge7f0XxzwL-iV2ehj=R7kUA@mail.gmail.com>
Message-ID: <BANLkTimvVKAhvxF9wKyanzx8qZXEVRu=uA@mail.gmail.com>

hi
still got this error, even though already sert PKG_CONFIG_PATH
making all mod_managed
Package mono-2 was not found in the pkg-config search path.
Perhaps you should add the directory containing `mono-2.pc'
to the PKG_CONFIG_PATH environment variable
No package 'mono-2' found
Package mono-2 was not found in the pkg-config search path.
Perhaps you should add the directory containing `mono-2.pc'
to the PKG_CONFIG_PATH environment variable
No package 'mono-2' found
Compiling freeswitch_managed.cpp...
g++ -I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
-DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
In file included from freeswitch_managed.cpp:35:
freeswitch_managed.h:43:26: error: mono/jit/jit.h: No such file or directory
freeswitch_managed.h:44:36: error: mono/metadata/assembly.h: No such file or
directory
freeswitch_managed.h:45:39: error: mono/metadata/environment.h: No such file
or directory
freeswitch_managed.h:46:39: error: mono/metadata/mono-config.h: No such file
or directory
freeswitch_managed.h:47:35: error: mono/metadata/threads.h: No such file or
directory
freeswitch_managed.h:48:41: error: mono/metadata/debug-helpers.h: No such
file or directory
freeswitch_managed.h:55: error: ISO C++ forbids declaration of ?MonoDomain?
with no type


thx
budi

On Sun, May 22, 2011 at 6:10 PM, Yitzchok <adminjew at gmail.com> wrote:

> Try this
>
> cd /usr/local/src/freeswitch/src/mod/languages/mod_managed/
> git apply mono28.patch
> make reswig
> make
>
>
>
> Yitzchok
>
>
>
> On Sat, May 21, 2011 at 6:42 PM, budi wibowo <bwibowo at gmail.com> wrote:
>
>> failed with some glib issue on source code compiling, then i use yum as
>> suggested.
>> mono installed, thx
>> Mono JIT compiler version 2.10.2 (tarball Mon Apr 18 19:06:50 UTC 2011)
>> Copyright (C) 2002-2011 Novell, Inc and Contributors.
>> www.mono-project.com
>>         TLS:           __thread
>>         SIGSEGV:       altstack
>>         Notifications: epoll
>>         Architecture:  amd64
>>         Disabled:      none
>>         Misc:          debugger softdebug
>>         LLVM:          supported, not enabled.
>>         GC:            Included Boehm (with typed GC and Parallel Mark)
>>
>> when i try to install mod_managed i found this error, may be some missing
>> config in my linux box
>>
>> making all mod_managed
>> Compiling freeswitch_managed.cpp...
>> g++ -I/usr/local/src/freeswitch/src/include
>> -I/usr/local/src/freeswitch/src/include
>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>> In file included from freeswitch_managed.cpp:35:
>> freeswitch_managed.h:43:18: error: glib.h: No such file or directory
>> freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or
>> directory
>> freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such file
>> or directory
>> freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such
>> file or directory
>> freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such
>> file or directory
>> freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such file
>> or directory
>> freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No such
>> file or directory
>>
>>
>> thx
>> budi wibowo
>>
>> On Wed, May 18, 2011 at 9:06 AM, Yitzchok <adminjew at gmail.com> wrote:
>>
>>> Make sure mono is installed and check which version using "mono -V" and
>>> follow the instructions on http://wiki.freeswitch.org/wiki/Mod_managed let
>>> us know if you get into any errors.
>>>
>>> This helped me for installing mono
>>> http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
>>>
>>>
>>> <http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
>>> Yitzchok
>>>
>>>
>>> On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>>
>>>> hi
>>>> i have fs running on centos 5.5, any body knows the combination version
>>>> of software need to be installed like mono etc.
>>>> have tried few combination but always failed compiling
>>>>
>>>>
>>>> thx
>>>> budi
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Mon May 23 02:58:06 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 22 May 2011 22:58:06 +0000
Subject: [Freeswitch-users] setting up forward incoming
	calls	to	external gateway
In-Reply-To: <1306103696780-6392576.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306103696780-6392576.post@n2.nabble.com>
Message-ID: <assp.012349e92e.92097A6A775D5147B1078E3F15430B9255A3A9@prato.activenetwerx.local>

>I am confused, though.  While trying to get into pastebin, I was asked for a
>login. Do I need to sign up to use it?  I tried to login with my credentials
>for this forum/mailing list and got rejected. Please help me out here.

Are you sure you were asked for a login, because you _read_ what it asked you:)
Try again, but have a quick read of the prompt...


From ayhkor at gmail.com  Mon May 23 04:27:24 2011
From: ayhkor at gmail.com (deniro)
Date: Sun, 22 May 2011 20:27:24 -0400
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
Message-ID: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>

Hi All
I am trying to get mod_xml_cdr working

My webserver is like  www.xxx.com  at port 80
the test web site is working fine with perl
http://www.xxx.com/PERL/test3.pl

I have mod_xml_cdr and mod_xml_curl modules loaded.

several questions;
1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
trying to figure out if my problem is with cdr or with curl

2-- when dialing into conference, getting following from fs_cli
[ERR] mod_xml_cdr.c:365 Got error [502] posting to web server [
http://www.xxx.com/PERL/cdr.pl]

3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
 <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>

4-- if I need mod_xml_curl (question 1)
what should this line be like in xml_curl.conf.xml?
<param name="gateway-url" value="http://www.xxx.com:80 <http://www.xxx.com/>"
bindings="dialplan"/>

again my webserver  www.xxx.com  at port 80

Thx
deiro
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From fieldpeak at gmail.com  Mon May 23 05:57:28 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 23 May 2011 09:57:28 +0800
Subject: [Freeswitch-users] call somebody into conference
	afterconferenceestablished
In-Reply-To: <BANLkTinMBuhpje-ZVbutGdwMWqV2qF75tg@mail.gmail.com>
References: <BANLkTinO7E7D9gFms2XhOKztY1baYkoz9g@mail.gmail.com>
	<201105121532017963789@asiainfo-linkage.com>
	<BANLkTi=wbd8ivSA4Gom3P2QsJ9AY7GRJUA@mail.gmail.com>
	<BANLkTikPq5GteE1Y5w56oL_=oAQ3ieVC1w@mail.gmail.com>
	<BANLkTimMVoF3RDjo1Zi2FG0K=ZwOXUyZKg@mail.gmail.com>
	<1412527818-1305863043-cardhu_decombobulator_blackberry.rim.net-1696936367-@b5.c17.bise6.blackberry>
	<BANLkTinMBuhpje-ZVbutGdwMWqV2qF75tg@mail.gmail.com>
Message-ID: <BANLkTik3dGC9kt2q2avC06hRO9keWA364A@mail.gmail.com>

Michael,

it works great! perfect! thanks for your elaborated solution.

2011/5/21 Michael Collins <msc at freeswitch.org>

>
>
> On Thu, May 19, 2011 at 8:44 PM, <darknesslabs at gmail.com> wrote:
>
>> Super useful, please do add it to the wiki.
>> Sent via BlackBerry
>>
> Done:
> http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example
>
> -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 adminjew at gmail.com  Mon May 23 08:28:29 2011
From: adminjew at gmail.com (Yitzchok)
Date: Mon, 23 May 2011 00:28:29 -0400
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTimvVKAhvxF9wKyanzx8qZXEVRu=uA@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
	<BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
	<BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>
	<BANLkTikkSWcge7f0XxzwL-iV2ehj=R7kUA@mail.gmail.com>
	<BANLkTimvVKAhvxF9wKyanzx8qZXEVRu=uA@mail.gmail.com>
Message-ID: <BANLkTinkdQOamMnt9WHs8BMgHBdOqGhB+w@mail.gmail.com>

Your missing the mono development package that doesn't get installed with
the monotools-addon-server.

run
*yum install mono-addon-devel.x86_64*
or
*yum install mono-addon-devel.i386*
depending on your cpu/os


Yitzchok


On Sun, May 22, 2011 at 6:49 PM, budi wibowo <bwibowo at gmail.com> wrote:

> hi
> still got this error, even though already sert PKG_CONFIG_PATH
> making all mod_managed
> Package mono-2 was not found in the pkg-config search path.
> Perhaps you should add the directory containing `mono-2.pc'
> to the PKG_CONFIG_PATH environment variable
> No package 'mono-2' found
> Package mono-2 was not found in the pkg-config search path.
> Perhaps you should add the directory containing `mono-2.pc'
> to the PKG_CONFIG_PATH environment variable
> No package 'mono-2' found
> Compiling freeswitch_managed.cpp...
> g++ -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
> In file included from freeswitch_managed.cpp:35:
> freeswitch_managed.h:43:26: error: mono/jit/jit.h: No such file or
> directory
> freeswitch_managed.h:44:36: error: mono/metadata/assembly.h: No such file
> or directory
> freeswitch_managed.h:45:39: error: mono/metadata/environment.h: No such
> file or directory
> freeswitch_managed.h:46:39: error: mono/metadata/mono-config.h: No such
> file or directory
> freeswitch_managed.h:47:35: error: mono/metadata/threads.h: No such file or
> directory
> freeswitch_managed.h:48:41: error: mono/metadata/debug-helpers.h: No such
> file or directory
> freeswitch_managed.h:55: error: ISO C++ forbids declaration of ?MonoDomain?
> with no type
>
>
> thx
> budi
>
> On Sun, May 22, 2011 at 6:10 PM, Yitzchok <adminjew at gmail.com> wrote:
>
>> Try this
>>
>> cd /usr/local/src/freeswitch/src/mod/languages/mod_managed/
>> git apply mono28.patch
>> make reswig
>> make
>>
>>
>>
>> Yitzchok
>>
>>
>>
>> On Sat, May 21, 2011 at 6:42 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>
>>> failed with some glib issue on source code compiling, then i use yum as
>>> suggested.
>>> mono installed, thx
>>> Mono JIT compiler version 2.10.2 (tarball Mon Apr 18 19:06:50 UTC 2011)
>>> Copyright (C) 2002-2011 Novell, Inc and Contributors.
>>> www.mono-project.com
>>>         TLS:           __thread
>>>         SIGSEGV:       altstack
>>>         Notifications: epoll
>>>         Architecture:  amd64
>>>         Disabled:      none
>>>         Misc:          debugger softdebug
>>>         LLVM:          supported, not enabled.
>>>         GC:            Included Boehm (with typed GC and Parallel Mark)
>>>
>>> when i try to install mod_managed i found this error, may be some missing
>>> config in my linux box
>>>
>>> making all mod_managed
>>> Compiling freeswitch_managed.cpp...
>>> g++ -I/usr/local/src/freeswitch/src/include
>>> -I/usr/local/src/freeswitch/src/include
>>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>>> In file included from freeswitch_managed.cpp:35:
>>> freeswitch_managed.h:43:18: error: glib.h: No such file or directory
>>> freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or
>>> directory
>>> freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such file
>>> or directory
>>> freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such
>>> file or directory
>>> freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such
>>> file or directory
>>> freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such file
>>> or directory
>>> freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No such
>>> file or directory
>>>
>>>
>>> thx
>>> budi wibowo
>>>
>>> On Wed, May 18, 2011 at 9:06 AM, Yitzchok <adminjew at gmail.com> wrote:
>>>
>>>> Make sure mono is installed and check which version using "mono -V" and
>>>> follow the instructions on http://wiki.freeswitch.org/wiki/Mod_managed let
>>>> us know if you get into any errors.
>>>>
>>>> This helped me for installing mono
>>>> http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
>>>>
>>>>
>>>> <http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
>>>> Yitzchok
>>>>
>>>>
>>>> On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>>>
>>>>> hi
>>>>> i have fs running on centos 5.5, any body knows the combination version
>>>>> of software need to be installed like mono etc.
>>>>> have tried few combination but always failed compiling
>>>>>
>>>>>
>>>>> thx
>>>>> budi
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Mon May 23 09:13:38 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Mon, 23 May 2011 12:13:38 +0700
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTinkdQOamMnt9WHs8BMgHBdOqGhB+w@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
	<BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
	<BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>
	<BANLkTikkSWcge7f0XxzwL-iV2ehj=R7kUA@mail.gmail.com>
	<BANLkTimvVKAhvxF9wKyanzx8qZXEVRu=uA@mail.gmail.com>
	<BANLkTinkdQOamMnt9WHs8BMgHBdOqGhB+w@mail.gmail.com>
Message-ID: <BANLkTim-5F7kV_PmJQ+SB4BGJroZTj61ig@mail.gmail.com>

thx for your suggestion
another error occured

making all mod_managed
Compiling freeswitch_wrap.cpp...
g++ -I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
-D_REENTRANT -I/opt/novell/mono/lib64/pkgconfig/../../include/mono-2.0
-DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp
freeswitch_wrap.cpp: In function ?void
SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))?:
freeswitch_wrap.cpp:274: error: redefinition of ?void
SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))?
freeswitch_wrap.cpp:248: error: ?void
SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))? previously
defined here
freeswitch_wrap.cpp: In function ?char* SWIG_csharp_string_callback(const
char*)?:
freeswitch_wrap.cpp:278: error: ?char* SWIG_csharp_string_callback(const
char*)? redeclared as different kind of symbol
freeswitch_wrap.cpp:242: error: previous declaration of ?char* (*
SWIG_csharp_string_callback)(const char*)?
make[4]: *** [freeswitch_wrap.o] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_managed-all] Error 1
make[1]: *** [mod_managed] Error 2
make: *** [mod_managed] Error 2


regards

budi

On Mon, May 23, 2011 at 11:28 AM, Yitzchok <adminjew at gmail.com> wrote:

> Your missing the mono development package that doesn't get installed with
> the monotools-addon-server.
>
> run
> *yum install mono-addon-devel.x86_64*
> or
> *yum install mono-addon-devel.i386*
> depending on your cpu/os
>
>
> Yitzchok
>
>
>
> On Sun, May 22, 2011 at 6:49 PM, budi wibowo <bwibowo at gmail.com> wrote:
>
>> hi
>> still got this error, even though already sert PKG_CONFIG_PATH
>> making all mod_managed
>> Package mono-2 was not found in the pkg-config search path.
>> Perhaps you should add the directory containing `mono-2.pc'
>> to the PKG_CONFIG_PATH environment variable
>> No package 'mono-2' found
>> Package mono-2 was not found in the pkg-config search path.
>> Perhaps you should add the directory containing `mono-2.pc'
>> to the PKG_CONFIG_PATH environment variable
>> No package 'mono-2' found
>> Compiling freeswitch_managed.cpp...
>> g++ -I/usr/local/src/freeswitch/src/include
>> -I/usr/local/src/freeswitch/src/include
>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>> In file included from freeswitch_managed.cpp:35:
>> freeswitch_managed.h:43:26: error: mono/jit/jit.h: No such file or
>> directory
>> freeswitch_managed.h:44:36: error: mono/metadata/assembly.h: No such file
>> or directory
>> freeswitch_managed.h:45:39: error: mono/metadata/environment.h: No such
>> file or directory
>> freeswitch_managed.h:46:39: error: mono/metadata/mono-config.h: No such
>> file or directory
>> freeswitch_managed.h:47:35: error: mono/metadata/threads.h: No such file
>> or directory
>> freeswitch_managed.h:48:41: error: mono/metadata/debug-helpers.h: No such
>> file or directory
>> freeswitch_managed.h:55: error: ISO C++ forbids declaration of
>> ?MonoDomain? with no type
>>
>>
>> thx
>> budi
>>
>> On Sun, May 22, 2011 at 6:10 PM, Yitzchok <adminjew at gmail.com> wrote:
>>
>>> Try this
>>>
>>> cd /usr/local/src/freeswitch/src/mod/languages/mod_managed/
>>> git apply mono28.patch
>>> make reswig
>>> make
>>>
>>>
>>>
>>> Yitzchok
>>>
>>>
>>>
>>> On Sat, May 21, 2011 at 6:42 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>>
>>>> failed with some glib issue on source code compiling, then i use yum as
>>>> suggested.
>>>> mono installed, thx
>>>> Mono JIT compiler version 2.10.2 (tarball Mon Apr 18 19:06:50 UTC 2011)
>>>> Copyright (C) 2002-2011 Novell, Inc and Contributors.
>>>> www.mono-project.com
>>>>         TLS:           __thread
>>>>         SIGSEGV:       altstack
>>>>         Notifications: epoll
>>>>         Architecture:  amd64
>>>>         Disabled:      none
>>>>         Misc:          debugger softdebug
>>>>         LLVM:          supported, not enabled.
>>>>         GC:            Included Boehm (with typed GC and Parallel Mark)
>>>>
>>>> when i try to install mod_managed i found this error, may be some
>>>> missing config in my linux box
>>>>
>>>> making all mod_managed
>>>> Compiling freeswitch_managed.cpp...
>>>> g++ -I/usr/local/src/freeswitch/src/include
>>>> -I/usr/local/src/freeswitch/src/include
>>>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>>>> In file included from freeswitch_managed.cpp:35:
>>>> freeswitch_managed.h:43:18: error: glib.h: No such file or directory
>>>> freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or
>>>> directory
>>>> freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such
>>>> file or directory
>>>> freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such
>>>> file or directory
>>>> freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such
>>>> file or directory
>>>> freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such file
>>>> or directory
>>>> freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No
>>>> such file or directory
>>>>
>>>>
>>>> thx
>>>> budi wibowo
>>>>
>>>> On Wed, May 18, 2011 at 9:06 AM, Yitzchok <adminjew at gmail.com> wrote:
>>>>
>>>>> Make sure mono is installed and check which version using "mono -V" and
>>>>> follow the instructions on http://wiki.freeswitch.org/wiki/Mod_managed let
>>>>> us know if you get into any errors.
>>>>>
>>>>> This helped me for installing mono
>>>>> http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
>>>>>
>>>>>
>>>>> <http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
>>>>> Yitzchok
>>>>>
>>>>>
>>>>> On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com>wrote:
>>>>>
>>>>>> hi
>>>>>> i have fs running on centos 5.5, any body knows the combination
>>>>>> version of software need to be installed like mono etc.
>>>>>> have tried few combination but always failed compiling
>>>>>>
>>>>>>
>>>>>> thx
>>>>>> budi
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gabe at gundy.org  Mon May 23 10:24:24 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 23 May 2011 00:24:24 -0600
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <assp.012349e92e.92097A6A775D5147B1078E3F15430B9255A3A9@prato.activenetwerx.local>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306103696780-6392576.post@n2.nabble.com>
	<assp.012349e92e.92097A6A775D5147B1078E3F15430B9255A3A9@prato.activenetwerx.local>
Message-ID: <BANLkTim_+xuWAysUKfXwAMSG9-6xQX9ZNg@mail.gmail.com>

On Sun, May 22, 2011 at 4:58 PM, Joseph L. Casale
<jcasale at activenetwerx.com> wrote:
> Are you sure you were asked for a login, because you _read_ what it asked you:)
> Try again, but have a quick read of the prompt...

This is true, but if you've seen it in Chrome, it's not very clear
what's expected. I hate to include attachments to the list, but this
isn't the first time it's come up.  Anyway, see the attached dialogs
if you're interested in seeing the difference yourselves.

Is there a better way of letting users know how to get in? Is this
even an issue?

Best,
Gabe
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From gabe at gundy.org  Mon May 23 10:38:27 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 23 May 2011 00:38:27 -0600
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
Message-ID: <BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>

On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
> several questions;
> 1-- do I really need to? load mod_xml_curl for?mod_xml_cdr?
> trying to figure out if my problem is with cdr or with curl

No, they are not functionally related... easy to get mixed up because
they both use HTTP and XML. One does not depend on the other. However,
you *can* configure mod_xml_cdr (or any other module) with
mod_xml_curl.



> 2-- when dialing into conference, getting following from fs_cli
> [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
> [http://www.xxx.com/PERL/cdr.pl]

There isn't a lot of info to work on here, but I'd look at your web
setup and not FS for the mix-up (particularly true as "HTTP Error 502"
means you have a bad gateway).



> 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
> ?<param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>

Looks good to me, but check it with the wiki:
http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration



> 4-- if I need mod_xml_curl (question 1)
> what should this line be like in xml_curl.conf.xml?
> <param name="gateway-url" value="http://www.xxx.com:80"
> bindings="dialplan"/>

This one is a little tricker to get right.  Again, the wiki is might
be the best place to reference:
http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring



Good luck,
Gabe


From steveayre at gmail.com  Mon May 23 11:45:28 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 23 May 2011 08:45:28 +0100
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <BANLkTim_+xuWAysUKfXwAMSG9-6xQX9ZNg@mail.gmail.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306103696780-6392576.post@n2.nabble.com>
	<assp.012349e92e.92097A6A775D5147B1078E3F15430B9255A3A9@prato.activenetwerx.local>
	<BANLkTim_+xuWAysUKfXwAMSG9-6xQX9ZNg@mail.gmail.com>
Message-ID: <BANLkTinb4K7oMQr2=L=EyABUaoi2Fd-MrA@mail.gmail.com>

Yuck, you're right that is a little confusing. :)



On 23 May 2011 07:24, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, May 22, 2011 at 4:58 PM, Joseph L. Casale
> <jcasale at activenetwerx.com> wrote:
> > Are you sure you were asked for a login, because you _read_ what it asked
> you:)
> > Try again, but have a quick read of the prompt...
>
> This is true, but if you've seen it in Chrome, it's not very clear
> what's expected. I hate to include attachments to the list, but this
> isn't the first time it's come up.  Anyway, see the attached dialogs
> if you're interested in seeing the difference yourselves.
>
> Is there a better way of letting users know how to get in? Is this
> even an issue?
>
> Best,
> 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
>
>
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From steveayre at gmail.com  Mon May 23 11:50:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 23 May 2011 08:50:09 +0100
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
	<BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
Message-ID: <BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>

>
> > 2-- when dialing into conference, getting following from fs_cli
> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
> > [http://www.xxx.com/PERL/cdr.pl]
>
> There isn't a lot of info to work on here, but I'd look at your web
> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
> means you have a bad gateway).
>

Check your web server logs - there may be some useful message. It sounds
like the cgi-bin/perl handler hasn't been configured correctly, the script
isn't executable, or it's not returning some headers. Try visiting the URL
in your webbrowser (you
won't be submitting a CDR but you can at least check the script runs without
an (unexpected) error.




On 23 May 2011 07:38, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
> > several questions;
> > 1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
> > trying to figure out if my problem is with cdr or with curl
>
> No, they are not functionally related... easy to get mixed up because
> they both use HTTP and XML. One does not depend on the other. However,
> you *can* configure mod_xml_cdr (or any other module) with
> mod_xml_curl.
>
>
>
> > 2-- when dialing into conference, getting following from fs_cli
> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
> > [http://www.xxx.com/PERL/cdr.pl]
>
> There isn't a lot of info to work on here, but I'd look at your web
> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
> means you have a bad gateway).
>
>
>
> > 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
> >  <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>
>
> Looks good to me, but check it with the wiki:
> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration
>
>
>
> > 4-- if I need mod_xml_curl (question 1)
> > what should this line be like in xml_curl.conf.xml?
> > <param name="gateway-url" value="http://www.xxx.com:80"
> > bindings="dialplan"/>
>
> This one is a little tricker to get right.  Again, the wiki is might
> be the best place to reference:
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring
>
>
>
> 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
>
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From admin at blindi.net  Mon May 23 11:54:48 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 23 May 2011 09:54:48 +0200 (CEST)
Subject: [Freeswitch-users] How can i drop the outcallcontext from the
	originate callerid?
In-Reply-To: <BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1105230943190.3447@debian>

Hi all,
I set up a call screening.
The problem:
  When I try Originate a call trigger the Outcallcontext is appended to
  the Callerid.
for example:
caller_id_number_=089XXXX/callback_out.wav
I lie to set:
caller_id_number=089XXXX.
  This problem only occurs when I leave Originate call about a subscriber.
I don.t fine a originate-variable to overwrite the context after the 
calleridnumber.
  Record stops because the directory is not at all natural.
Someone has an idea 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 awais-nazeer at hotmail.com  Mon May 23 13:36:39 2011
From: awais-nazeer at hotmail.com (awais nazir)
Date: Mon, 23 May 2011 15:36:39 +0600
Subject: [Freeswitch-users] Freeswitch commands customization
Message-ID: <BAY156-w46ADFBA8861EB91BC3526C88720@phx.gbl>


Hi

I am using a webpanel to run freeswitch commands (for example show calls). Can we do some customization like if we want certain colums to be eliminated from "show calls" resultset?

Regards


 		 	   		  
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From krice at freeswitch.org  Mon May 23 15:06:22 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 23 May 2011 06:06:22 -0500
Subject: [Freeswitch-users] Freeswitch commands customization
In-Reply-To: <BAY156-w46ADFBA8861EB91BC3526C88720@phx.gbl>
Message-ID: <C9FFA9DE.18839%krice@freeswitch.org>

If you are doing a webpanel just filter them in your display code else you
have to have to freeswitch code itself them you are stuck maintaining your
own set up patches to update every time you update freeswitch


On 5/23/11 4:36 AM, "awais nazir" <awais-nazeer at hotmail.com> wrote:

> Hi
> 
> I am using a webpanel to run freeswitch commands (for example show calls). Can
> we do some customization like if we want certain colums to be eliminated from
> "show calls" resultset?
> 
> 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 steveayre at gmail.com  Mon May 23 15:49:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 23 May 2011 12:49:45 +0100
Subject: [Freeswitch-users] Freeswitch commands customization
In-Reply-To: <BAY156-w46ADFBA8861EB91BC3526C88720@phx.gbl>
References: <BAY156-w46ADFBA8861EB91BC3526C88720@phx.gbl>
Message-ID: <BANLkTi=w_2j-09+AGm7uXnBQubs1+9y+gQ@mail.gmail.com>

You're best off using "show calls as xml", parsing the XML, and then
choosing which to display yourself.

Using XML will be less error prone than using the text display in case the
formatting/columns change in later FreeSWITCH versions... the XML element
names should remain the same.

-Steve


On 23 May 2011 10:36, awais nazir <awais-nazeer at hotmail.com> wrote:

>  Hi
>
> I am using a webpanel to run freeswitch commands (for example show calls).
> Can we do some customization like if we want certain colums to be eliminated
> from "show calls" resultset?
>
> 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 david.ponzone at ipeva.fr  Mon May 23 16:13:45 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 23 May 2011 14:13:45 +0200
Subject: [Freeswitch-users] How can i drop the outcallcontext from the
	originate callerid?
In-Reply-To: <Pine.LNX.4.64.1105230943190.3447@debian>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
	<Pine.LNX.4.64.1105230943190.3447@debian>
Message-ID: <C8355FEB-5D0C-4C7B-867A-764FB49D1CAD@ipeva.fr>

There should not be a outcallcontext (what is that by the way ?) appended to the caller_id_number.
Something must be wrong in your config.

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

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

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




Le 23/05/2011 ? 09:54, Thomas Hoellriegel a ?crit :

> Hi all,
> I set up a call screening.
> The problem:
> When I try Originate a call trigger the Outcallcontext is appended to
> the Callerid.
> for example:
> caller_id_number_=089XXXX/callback_out.wav
> I lie to set:
> caller_id_number=089XXXX.
> This problem only occurs when I leave Originate call about a subscriber.
> I don.t fine a originate-variable to overwrite the context after the calleridnumber.
> Record stops because the directory is not at all natural.
> Someone has an idea 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
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mgende at gendesign.com  Mon May 23 18:33:23 2011
From: mgende at gendesign.com (Michael Gende)
Date: Mon, 23 May 2011 09:33:23 -0500
Subject: [Freeswitch-users] 30 Seconds - About 6 Rings
Message-ID: <BANLkTinU-owGNW1uTFnYWWPpY4Z+v_8F2Q@mail.gmail.com>

 Hello,

We just had an issue come up. Short version: we want to allow for about 6
rings for incoming calls.

We had been using the simple logic below (from our default.xml) allowing 15
seconds (see "call_timeout" value). That gave time for about three rings.

Need 6 rings? Just call the VoIP provider (Urbancom in our case) make sure
they will cooperate and "up" the timeout to 30. Right? Hasn't worked out
that way. Two things happen:

Turns out most calls (some land line, some cellular) ring for about 20
seconds, then the caller gets an error tone and our FS returns 847,
originator cancel.

Strangely, when I call in from Sprint (near Chicago), the logic just goes
"round and round", ringing forever. It just seems to "re-set" and rings
another 20 seconds, then resets again, ringing forever.

I'm thinking that Urbancom's PSTN provider may not wait for 30 seconds
before canceling the call. Sprint must "come in" to Urbancom another way.
But that doesn't explain not going to VM.

Those are my thoughts. Anything I'm missing?

Thanks,

Mike

<!-- Mike Added: Incoming Urbancom Calls. Pretty straightforward, ring a
call group for a time, if no one answers, go to ext 1000's VM -->

    <extension name="UrbancomIn">
      <condition field="destination_number" expression="^(8201)$">
         <action application="set" data="*call_timeout=15*"/>
         <action application="set" data="domain_name=192.168.0.198"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="bridge" data="group/support@${domain_name}"/>
         <action application="answer"/>
         <action application="sleep" data="1000"/>
         <action application="voicemail" data="default 192.168.0.198 1000"/>
      </condition>
    </extension>
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From mtaylor at employees.org  Mon May 23 06:07:59 2011
From: mtaylor at employees.org (Mike Taylor)
Date: Mon, 23 May 2011 14:07:59 +1200
Subject: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
Message-ID: <CA001ABE.2BD10%mtaylor@employees.org>

Hi,

This is driving me nuts.
I have a Cisco CME (a few actually) where I cannot get DTMF to work.

Does someone happen to have the magic recipe of SIP/Dialplan profiles and
Cisco config to make DTMF <relay> work?

I spent several very very painful days trying to get RFC2833 to work, using
FS to try and generate the tones itself. I ended up with all sorts of codec
mismatches, delay,distortion and some really mangled DTMF tones?

So, now I'm back to;

PSTN----CiscoGW----<external>----FS----<internal>----<CUST PABX>---Phone

Where <CUST PABX> is 3 CMEs that won't talk DTMF
30+ other vendor's gateway's that WILL do DTMF?

Context External and internal are both set to dtmf-type info (or dtmf_type
info) 

DTMF works FROM the Cisco GW towards our network, doesn't


So, does anyone have a magic solution?
 Because Cisco TAC say they see DTMF being sent (as rtp-nte), when I need
INFO or in-band.



Thanks,
Mike


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From sireeps at gmail.com  Mon May 23 19:01:30 2011
From: sireeps at gmail.com (Kamen)
Date: Mon, 23 May 2011 08:01:30 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <BANLkTinb4K7oMQr2=L=EyABUaoi2Fd-MrA@mail.gmail.com>
References: <1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306103696780-6392576.post@n2.nabble.com>
	<assp.012349e92e.92097A6A775D5147B1078E3F15430B9255A3A9@prato.activenetwerx.local>
	<BANLkTim_+xuWAysUKfXwAMSG9-6xQX9ZNg@mail.gmail.com>
	<BANLkTinb4K7oMQr2=L=EyABUaoi2Fd-MrA@mail.gmail.com>
Message-ID: <1306162890766-6394825.post@n2.nabble.com>

Gabriel!

I really appreciate your clarification post. I am using Chrome as my default
browser and the message confused me.  Following you suggestion I opened it
in Firefox and finally got what was needed.  Yeah, may be I am not that
quick thinking chap, but hay, thats why I am here looking for your advise. 

Cheers

Sergei Kamen

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6394825.html
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From msc at freeswitch.org  Mon May 23 19:24:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 May 2011 08:24:39 -0700
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
Message-ID: <BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>

Sidharth,

A mobile phone will always send DTMFs in-band, so you need to be ready for
that scenario. I recommend you add this to your dialplan for inbound calls:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf

Test various scenarios and make sure they work - call from Skype, from
mobile phone, from a land line, etc. Let us know what happens.

-MC

On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya <sid.kshatriya at gmail.com
> wrote:

> Dear Friends,
>
> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
> list). While I'm generally happy with their service there seems to be one
> fatal problem: My IVR does not recognize DTMF!
>
> I have set <param name="dtmf-type" value="rfc2833"/> in both
> sip_profile/internal.xml and sip_profile/external.xml
>
> The symptom of the problem is that making a call via skype will *always*make the IVR recognize the DTMF while using something like a mobile phone
> *almost always* won't!
>
> I've tried in-band detection too. I'm making international calls into my
> IVR and the reliability of the in-band detection is not so good, possibly
> because of the quality of the call.
>
> Can someone please help me?
>
> Thanks,
>
> Sidharth
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sireeps at gmail.com  Mon May 23 19:25:31 2011
From: sireeps at gmail.com (Kamen)
Date: Mon, 23 May 2011 08:25:31 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls
	to	external gateway
In-Reply-To: <E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
Message-ID: <1306164331568-6394908.post@n2.nabble.com>

Hello,

I have posted the trace with sip trace on at pastebin.  Thanks everyone for
helping me out with that.  The only thing is that I do not know how to get
the trace with freeswitch highlights on.  So for now it is plain text (I
will appreciate any hints on how to get the highlights, though).  I added
some comments in there.

http://pastebin.freeswitch.org/16358 http://pastebin.freeswitch.org/16358 

Could anyone please have a look and let me know what I have wrong in my
setup. I'd like to repeat that this is really a simple case of forwarding
incoming external call to outgoing external.

Thanks a lot!

Sergei Kamen

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6394908.html
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From adminjew at gmail.com  Mon May 23 19:37:52 2011
From: adminjew at gmail.com (Yitzchok)
Date: Mon, 23 May 2011 11:37:52 -0400
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTim-5F7kV_PmJQ+SB4BGJroZTj61ig@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
	<BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
	<BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>
	<BANLkTikkSWcge7f0XxzwL-iV2ehj=R7kUA@mail.gmail.com>
	<BANLkTimvVKAhvxF9wKyanzx8qZXEVRu=uA@mail.gmail.com>
	<BANLkTinkdQOamMnt9WHs8BMgHBdOqGhB+w@mail.gmail.com>
	<BANLkTim-5F7kV_PmJQ+SB4BGJroZTj61ig@mail.gmail.com>
Message-ID: <BANLkTik3tVRpAQmrzpLvBFD=aAXEgFu-eg@mail.gmail.com>

Do you have SWIG installed?

http://www.swig.org/download.html

<http://www.swig.org/download.html>install swig and run *make reswig*


Yitzchok


On Mon, May 23, 2011 at 1:13 AM, budi wibowo <bwibowo at gmail.com> wrote:

> thx for your suggestion
> another error occured
>
> making all mod_managed
> Compiling freeswitch_wrap.cpp...
> g++ -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
> -D_REENTRANT -I/opt/novell/mono/lib64/pkgconfig/../../include/mono-2.0
> -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp
> freeswitch_wrap.cpp: In function ?void
> SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))?:
> freeswitch_wrap.cpp:274: error: redefinition of ?void
> SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))?
> freeswitch_wrap.cpp:248: error: ?void
> SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))? previously
> defined here
> freeswitch_wrap.cpp: In function ?char* SWIG_csharp_string_callback(const
> char*)?:
> freeswitch_wrap.cpp:278: error: ?char* SWIG_csharp_string_callback(const
> char*)? redeclared as different kind of symbol
> freeswitch_wrap.cpp:242: error: previous declaration of ?char* (*
> SWIG_csharp_string_callback)(const char*)?
> make[4]: *** [freeswitch_wrap.o] Error 1
> make[3]: *** [all] Error 1
> make[2]: *** [mod_managed-all] Error 1
> make[1]: *** [mod_managed] Error 2
> make: *** [mod_managed] Error 2
>
>
> regards
>
> budi
>
> On Mon, May 23, 2011 at 11:28 AM, Yitzchok <adminjew at gmail.com> wrote:
>
>> Your missing the mono development package that doesn't get installed with
>> the monotools-addon-server.
>>
>> run
>> *yum install mono-addon-devel.x86_64*
>> or
>> *yum install mono-addon-devel.i386*
>> depending on your cpu/os
>>
>>
>> Yitzchok
>>
>>
>>
>> On Sun, May 22, 2011 at 6:49 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>
>>> hi
>>> still got this error, even though already sert PKG_CONFIG_PATH
>>> making all mod_managed
>>> Package mono-2 was not found in the pkg-config search path.
>>> Perhaps you should add the directory containing `mono-2.pc'
>>> to the PKG_CONFIG_PATH environment variable
>>> No package 'mono-2' found
>>> Package mono-2 was not found in the pkg-config search path.
>>> Perhaps you should add the directory containing `mono-2.pc'
>>> to the PKG_CONFIG_PATH environment variable
>>> No package 'mono-2' found
>>> Compiling freeswitch_managed.cpp...
>>> g++ -I/usr/local/src/freeswitch/src/include
>>> -I/usr/local/src/freeswitch/src/include
>>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>>> In file included from freeswitch_managed.cpp:35:
>>> freeswitch_managed.h:43:26: error: mono/jit/jit.h: No such file or
>>> directory
>>> freeswitch_managed.h:44:36: error: mono/metadata/assembly.h: No such file
>>> or directory
>>> freeswitch_managed.h:45:39: error: mono/metadata/environment.h: No such
>>> file or directory
>>> freeswitch_managed.h:46:39: error: mono/metadata/mono-config.h: No such
>>> file or directory
>>> freeswitch_managed.h:47:35: error: mono/metadata/threads.h: No such file
>>> or directory
>>> freeswitch_managed.h:48:41: error: mono/metadata/debug-helpers.h: No such
>>> file or directory
>>> freeswitch_managed.h:55: error: ISO C++ forbids declaration of
>>> ?MonoDomain? with no type
>>>
>>>
>>> thx
>>> budi
>>>
>>> On Sun, May 22, 2011 at 6:10 PM, Yitzchok <adminjew at gmail.com> wrote:
>>>
>>>> Try this
>>>>
>>>> cd /usr/local/src/freeswitch/src/mod/languages/mod_managed/
>>>> git apply mono28.patch
>>>> make reswig
>>>> make
>>>>
>>>>
>>>>
>>>> Yitzchok
>>>>
>>>>
>>>>
>>>> On Sat, May 21, 2011 at 6:42 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>>>
>>>>> failed with some glib issue on source code compiling, then i use yum as
>>>>> suggested.
>>>>> mono installed, thx
>>>>> Mono JIT compiler version 2.10.2 (tarball Mon Apr 18 19:06:50 UTC 2011)
>>>>> Copyright (C) 2002-2011 Novell, Inc and Contributors.
>>>>> www.mono-project.com
>>>>>         TLS:           __thread
>>>>>         SIGSEGV:       altstack
>>>>>         Notifications: epoll
>>>>>         Architecture:  amd64
>>>>>         Disabled:      none
>>>>>         Misc:          debugger softdebug
>>>>>         LLVM:          supported, not enabled.
>>>>>         GC:            Included Boehm (with typed GC and Parallel Mark)
>>>>>
>>>>> when i try to install mod_managed i found this error, may be some
>>>>> missing config in my linux box
>>>>>
>>>>> making all mod_managed
>>>>> Compiling freeswitch_managed.cpp...
>>>>> g++ -I/usr/local/src/freeswitch/src/include
>>>>> -I/usr/local/src/freeswitch/src/include
>>>>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>>>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>>>>> In file included from freeswitch_managed.cpp:35:
>>>>> freeswitch_managed.h:43:18: error: glib.h: No such file or directory
>>>>> freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or
>>>>> directory
>>>>> freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such
>>>>> file or directory
>>>>> freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such
>>>>> file or directory
>>>>> freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such
>>>>> file or directory
>>>>> freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such
>>>>> file or directory
>>>>> freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No
>>>>> such file or directory
>>>>>
>>>>>
>>>>> thx
>>>>> budi wibowo
>>>>>
>>>>> On Wed, May 18, 2011 at 9:06 AM, Yitzchok <adminjew at gmail.com> wrote:
>>>>>
>>>>>> Make sure mono is installed and check which version using "mono -V"
>>>>>> and follow the instructions on
>>>>>> http://wiki.freeswitch.org/wiki/Mod_managed let us know if you get
>>>>>> into any errors.
>>>>>>
>>>>>> This helped me for installing mono
>>>>>> http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
>>>>>>
>>>>>>
>>>>>> <http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
>>>>>> Yitzchok
>>>>>>
>>>>>>
>>>>>> On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com>wrote:
>>>>>>
>>>>>>> hi
>>>>>>> i have fs running on centos 5.5, any body knows the combination
>>>>>>> version of software need to be installed like mono etc.
>>>>>>> have tried few combination but always failed compiling
>>>>>>>
>>>>>>>
>>>>>>> thx
>>>>>>> budi
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From seichhorn at gci.com  Mon May 23 19:52:29 2011
From: seichhorn at gci.com (Sean Eichhorn)
Date: Mon, 23 May 2011 07:52:29 -0800
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
Message-ID: <7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>

Yes, I understand that this is the Cisco proprietary codec.  The problem is that between two Cisco systems the SIP endpoints will not use this protocol if it is not sent in the SDP.  The *original* SDP does contain the codec, but when the call is answered, another SDP is sent.  This SDP is altered by Freeswitch and all codecs other than the selected codec, NTE, and comfort noise are removed.
It might not be Freeswitch doing it.  I?m digging into the source and it looks like it might be the Sofia-SIP library.
The funny thing is that it works in one direction, but not the other.
FS subscriber -> CIDR defined static SIP endpoint (WORKS!)
CIDR defined SIP endpoint -> FS Subscriber (Doesn?t work)

It seems that one of the major differences is that the subscriber answers with a 200 OK, whereas the CIDR endpoint answers with a 183.

I?m currently searching to see if there is a way to change the response type, but so far no luck.

Sean Eichhorn
General Communications Inc.
IP Telephony Engineer
(907) 868-6902

-No telephony products were harmed in the making of the message-
From: David Ponzone [mailto:david.ponzone at ipeva.fr]
Sent: Friday, May 20, 2011 01:03
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] rtpmap line missing on answer

Sean,

X-NSE is the Clear Channel codec from Cisco.
I suspect FreeSWITCH does not support it, and as such, it can't be offered to leg B.
Perhaps there is a way to enable in outbound codecs as bypass, but I really doubt so.
Though, you could try to enable late-negotiation where client and gateway will negotiate together.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 20/05/2011 ? 20:54, Sean Eichhorn a ?crit :


I have a freeswitch system that (for the most part) works flawlessly for me.  However, I need to have an additional rtpmap line in the SDP when a call is answered.
My client sends the following SDP upon answering the call :
   m=audio 17214 RTP/AVP 0 19 101 100
   c=IN IP4 192.168.98.79
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:100 X-NSE/8000
   a=fmtp:100 192-194

Freeswitch forwards the following SDP to the external gateway :
   m=audio 20654 RTP/AVP 0 19 101
   c=IN IP4 66.223.187.208
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16

The other information appears fine.  Everything works, but my client-side is not receiving the a=rtpmap:100 line.
This issue appears regardless of whether or not I?m in proxy mode, bypass mode, or neither.

Using "sip_append_audio_sdp? has no effect on the answering SDP, only the initial offer.

Any ideas?

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

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From steveayre at gmail.com  Mon May 23 19:59:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 23 May 2011 16:59:27 +0100
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
Message-ID: <BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>

Who is answering the call? Are you sure they're not choosing a different
codec from the ones offered in the INVITE?

-Steve


On 23 May 2011 16:52, Sean Eichhorn <seichhorn at gci.com> wrote:

>  Yes, I understand that this is the Cisco proprietary codec.  The problem
> is that between two Cisco systems the SIP endpoints will not use this
> protocol if it is not sent in the SDP.  The **original** SDP does contain
> the codec, but when the call is answered, another SDP is sent.  This SDP is
> altered by Freeswitch and all codecs other than the selected codec, NTE, and
> comfort noise are removed.
>
> It might not be Freeswitch doing it.  I?m digging into the source and it
> looks like it might be the Sofia-SIP library.
>
> The funny thing is that it works in one direction, but not the other.
>
> FS subscriber -> CIDR defined static SIP endpoint (WORKS!)
>
> CIDR defined SIP endpoint -> FS Subscriber (Doesn?t work)
>
>
>
> It seems that one of the major differences is that the subscriber answers
> with a 200 OK, whereas the CIDR endpoint answers with a 183.
>
>
>
> I?m currently searching to see if there is a way to change the response
> type, but so far no luck.
>
>
>
> Sean Eichhorn
>
> General Communications Inc.
>
> IP Telephony Engineer
>
> (907) 868-6902
>
>
>
> -No telephony products were harmed in the making of the message-
>
> *From:* David Ponzone [mailto:david.ponzone at ipeva.fr]
> *Sent:* Friday, May 20, 2011 01:03
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] rtpmap line missing on answer
>
>
>
> Sean,e
>
>
>
> X-NSE is the Clear Channel codec from Cisco.
>
> I suspect FreeSWITCH does not support it, and as such, it can't be offered
> to leg B.
>
> Perhaps there is a way to enable in outbound codecs as bypass, but I really
> doubt so.
>
> Though, you could try to enable late-negotiation where client and gateway
> will negotiate together.
>
>
>
> David Ponzone  Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel:      01 74 03 18 97
>
> gsm:   06 66 98 76 34
>
>
>
> Service Client IPeva
>
> tel:      0811 46 26 26
>
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
>
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
>
> * *
>
>
>
>
>
> Le 20/05/2011 ? 20:54, Sean Eichhorn a ?crit :
>
>
>
>   I have a freeswitch system that (for the most part) works flawlessly for
> me.  However, I need to have an additional rtpmap line in the SDP when a
> call is answered.
>
> My client sends the following SDP upon answering the call :
>
>    m=audio 17214 RTP/AVP 0 19 101 100
>
>    c=IN IP4 192.168.98.79
>
>    a=rtpmap:0 PCMU/8000
>
>    a=rtpmap:19 CN/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>    a=rtpmap:100 X-NSE/8000
>
>    a=fmtp:100 192-194
>
>
>
> Freeswitch forwards the following SDP to the external gateway :
>
>    m=audio 20654 RTP/AVP 0 19 101
>
>    c=IN IP4 66.223.187.208
>
>    a=rtpmap:0 PCMU/8000
>
>    a=rtpmap:19 CN/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>
>
> The other information appears fine.  Everything works, but my client-side
> is not receiving the a=rtpmap:100 line.
>
> This issue appears regardless of whether or not I?m in proxy mode, bypass
> mode, or neither.
>
>
>
> Using "sip_append_audio_sdp? has no effect on the answering SDP, only the
> initial offer.
>
>
>
> Any ideas?
>
>
>
> 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 msc at freeswitch.org  Mon May 23 22:44:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 May 2011 11:44:11 -0700
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <1306164331568-6394908.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306164331568-6394908.post@n2.nabble.com>
Message-ID: <BANLkTinvAzxVVTBGi-c3D8_hci6i2maBpA@mail.gmail.com>

When you do the pastebin there is a drop-down box for "Syntax Highlighting"
- choose "FreeSWITCH Log" and it will do the colorizing for you.
-MC

On Mon, May 23, 2011 at 8:25 AM, Kamen <sireeps at gmail.com> wrote:

> Hello,
>
> I have posted the trace with sip trace on at pastebin.  Thanks everyone for
> helping me out with that.  The only thing is that I do not know how to get
> the trace with freeswitch highlights on.  So for now it is plain text (I
> will appreciate any hints on how to get the highlights, though).  I added
> some comments in there.
>
> http://pastebin.freeswitch.org/16358 http://pastebin.freeswitch.org/16358
>
> Could anyone please have a look and let me know what I have wrong in my
> setup. I'd like to repeat that this is really a simple case of forwarding
> incoming external call to outgoing external.
>
> Thanks a lot!
>
> Sergei Kamen
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6394908.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 oseslija at gmail.com  Mon May 23 22:46:09 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 23 May 2011 20:46:09 +0200
Subject: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
In-Reply-To: <CA001ABE.2BD10%mtaylor@employees.org>
References: <CA001ABE.2BD10%mtaylor@employees.org>
Message-ID: <BANLkTin7a61Nw-L6qRhGbwYanHO9LGVDGg@mail.gmail.com>

Why use INFO when it's only method I know Cisco won't speak (2833, kpml,
sub/notify)?

Here's the CME snippet for FS trunk I have. FS sends 2833 tones to CME btw.

dial-peer voice 100 voip
 description ** TO FS **
 destination-pattern .T
 b2bua
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad




On Mon, May 23, 2011 at 4:07 AM, Mike Taylor <mtaylor at employees.org> wrote:

> Hi,
>
> This is driving me nuts.
> I have a Cisco CME (a few actually) where I cannot get DTMF to work.
>
> Does someone happen to have the magic recipe of SIP/Dialplan profiles and
> Cisco config to make DTMF <relay> work?
>
> I spent several very very painful days trying to get RFC2833 to work, using
> FS to try and generate the tones itself. I ended up with all sorts of codec
> mismatches, delay,distortion and some really mangled DTMF tones?
>
> So, now I'm back to;
>
> PSTN----CiscoGW----<external>----FS----<internal>----<CUST PABX>---Phone
>
> Where <CUST PABX> is 3 CMEs that won't talk DTMF
> 30+ other vendor's gateway's that WILL do DTMF?
>
> Context External and internal are both set to dtmf-type info (or dtmf_type
> info)
>
> DTMF works FROM the Cisco GW towards our network, doesn't
>
>
> So, does anyone have a magic solution?
>  Because Cisco TAC say they see DTMF being sent (as rtp-nte), when I need
> INFO or in-band.
>
>
>
> Thanks,
> 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
>
>
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From jonyoung111 at gmail.com  Mon May 23 22:47:46 2011
From: jonyoung111 at gmail.com (Jon Young)
Date: Mon, 23 May 2011 11:47:46 -0700
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
Message-ID: <BANLkTi=UsLW4b9crvGHdfVt6Mxp49Doxqg@mail.gmail.com>

Would this also be an issue/resolution with Google Voice as well?
Using Dingaling I am routing my calls to an IVR to intercept the call
from going to Google Voice voicemail.  However, the IVR won't accept
DTMF.

On Mon, May 23, 2011 at 8:24 AM, Michael Collins <msc at freeswitch.org> wrote:
> Sidharth,
> A mobile phone will always send DTMFs in-band, so you need to be ready for
> that scenario. I recommend you add this to your dialplan for inbound calls:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> Test various scenarios and make sure they work - call from Skype, from
> mobile phone, from a land line, etc. Let us know what happens.
> -MC
> On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya
> <sid.kshatriya at gmail.com> wrote:
>>
>> Dear Friends,
>>
>> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
>> list). While I'm generally happy with their service there seems to be one
>> fatal problem: My IVR does not recognize DTMF!
>>
>> I have set <param name="dtmf-type" value="rfc2833"/> in both
>> sip_profile/internal.xml and sip_profile/external.xml
>>
>> The symptom of the problem is that making a call via skype will always
>> make the IVR recognize the DTMF while using something like a mobile phone
>> almost always won't!
>>
>> I've tried in-band detection too. I'm making international calls into my
>> IVR and the reliability of the in-band detection is not so good, possibly
>> because of the quality of the call.
>>
>> Can someone please help me?
>>
>> Thanks,
>>
>> Sidharth
>>
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From rmorin at blie-ent.com  Mon May 23 22:51:00 2011
From: rmorin at blie-ent.com (Rob Morin)
Date: Mon, 23 May 2011 14:51:00 -0400
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
Message-ID: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>

I?m having problems faxing if the B leg offers T.38.  Several scenarios, but
always the same result.

 

First, the architecture

 

FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another
carrier (PSTN) and the fax machine)

 

On my ATA, T.38 is enabled (Auto Detect).

 

If the destination supports T.38, the Carrier will offer it to Freeswitch.
If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
faxes go through.

 

I?ve set Freeswitch as: 

      <action application="set" data="fax_enable_t38=true"/>

      <action application="export" data="t38_passthru=true"/>  

When that?s the case, and the B leg offers T.38, Freeswitch passes through
the offer and the ATA responds, 3 times. But the response never gets sent to
the B leg, so it terminates the call after about 30 seconds.

 

When I set Freeswitch as: 

      <action application="set" data="fax_enable_t38=false"/>

The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t
pass the offer through and the B leg terminates the call after about 30
seconds.

 

I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
2011-05-17 22-51-47 -0500) ).

 

I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
?passthrough? has had the most success, mostly when the other ends aren?t
T.38 capable and it stays in G711u.

 

Is there something else I need to do to enable T.38 in passthrough mode?  

 

Thank you,
Rob Morin

 

PS ? I can provide tcpdumps of this, or whatever else is necessary.

 

 

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From yungwei at resolvity.com  Mon May 23 23:40:00 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 23 May 2011 15:40:00 -0400
Subject: [Freeswitch-users] Getting the pizza demo to work
Message-ID: <33095823FD21DF429B481B5163264B7950AC3ABF25@VMBX102.ihostexchange.net>

Hi,

I just installed the latest snapshot on CentOS 5. 
I want to try the pizza demo associated with 00_pizza_demo.xml, but I'm having trouble getting the pizza demo to work. 

According to freeswitch.log, it looks like mod_pocketsphinx doesn't exist.
...skipped...
Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (PASS) [pizza_demo] destination_number(74992) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (PASS) [pizza_demo] ${module_exists(mod_spidermonkey)}(true) =~ /true/ break=on-false
Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (FAIL) [pizza_demo] ${module_exists(mod_pocketsphinx)}(false) =~ /true/ break=on-false
Dialplan: sofia/internal/1000 at 192.168.16.3 parsing [default->Talking Clock Time] continue=false
Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (FAIL) [Talking Clock Time] destination_number(74992) =~ /^9170$/ break=on-false
...skipped...
Dialplan: sofia/internal/1000 at 192.168.16.3 Action transfer(74992 enum)

But I do see mod_pocketsphinx.so in /usr/local/freeswitch/mod folder.
[root at templ0 mod]# ls /usr/local/freeswitch/mod/mod_pocketsphinx.so -l
-rwxr-xr-x 1 root root 1353092 May 23 13:23 /usr/local/freeswitch/mod/mod_pocketsphinx.so

I also went through http://wiki.freeswitch.org/wiki/Mod_pocketsphinx, but the problem is still there.
How am I missing here? Thanks.


From p2.freeswitch-list at coleinternet.com  Mon May 23 23:31:42 2011
From: p2.freeswitch-list at coleinternet.com (Jay)
Date: Mon, 23 May 2011 12:31:42 -0700
Subject: [Freeswitch-users] Voicemail recordings being cut off on the front
	end.
Message-ID: <4DDAB61E.3010400@coleinternet.com>

I need some direction on tracking down why some voicemail recordings are being cut off at the front end. There are instances where the start of the .wav file is a good 5 seconds into the intended message.

what FS components could be responsible clipped voicemail recordings?    is there silence detection going on? if so, are there any controls such as automatic gain control, fixed volume settings for recordings or other normalizing going on?

any advice?

thx.



From bwibowo at gmail.com  Tue May 24 00:39:05 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Tue, 24 May 2011 03:39:05 +0700
Subject: [Freeswitch-users] mod_manage install
In-Reply-To: <BANLkTik3tVRpAQmrzpLvBFD=aAXEgFu-eg@mail.gmail.com>
References: <BANLkTi=C374KYYjz3G0_1dWAXgw75p-dvg@mail.gmail.com>
	<BANLkTikMeYjiLnwCZBL0WaeEEVDWLd72cA@mail.gmail.com>
	<BANLkTimKGMLd6Xu5wSK835m3vBYzNAEVgg@mail.gmail.com>
	<BANLkTikkSWcge7f0XxzwL-iV2ehj=R7kUA@mail.gmail.com>
	<BANLkTimvVKAhvxF9wKyanzx8qZXEVRu=uA@mail.gmail.com>
	<BANLkTinkdQOamMnt9WHs8BMgHBdOqGhB+w@mail.gmail.com>
	<BANLkTim-5F7kV_PmJQ+SB4BGJroZTj61ig@mail.gmail.com>
	<BANLkTik3tVRpAQmrzpLvBFD=aAXEgFu-eg@mail.gmail.com>
Message-ID: <BANLkTikKn4S=TT7GLY3hsN24Qyt9B_qP1w@mail.gmail.com>

many thx yitzchok, works now :)

On Mon, May 23, 2011 at 10:37 PM, Yitzchok <adminjew at gmail.com> wrote:

> Do you have SWIG installed?
>
> http://www.swig.org/download.html
>
> <http://www.swig.org/download.html>install swig and run *make reswig*
>
>
> Yitzchok
>
>
>
> On Mon, May 23, 2011 at 1:13 AM, budi wibowo <bwibowo at gmail.com> wrote:
>
>> thx for your suggestion
>> another error occured
>>
>> making all mod_managed
>> Compiling freeswitch_wrap.cpp...
>> g++ -I/usr/local/src/freeswitch/src/include
>> -I/usr/local/src/freeswitch/src/include
>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>> -D_REENTRANT -I/opt/novell/mono/lib64/pkgconfig/../../include/mono-2.0
>> -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp
>> freeswitch_wrap.cpp: In function ?void
>> SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))?:
>> freeswitch_wrap.cpp:274: error: redefinition of ?void
>> SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))?
>> freeswitch_wrap.cpp:248: error: ?void
>> SWIGRegisterStringCallback_freeswitch(char* (*)(const char*))? previously
>> defined here
>> freeswitch_wrap.cpp: In function ?char* SWIG_csharp_string_callback(const
>> char*)?:
>> freeswitch_wrap.cpp:278: error: ?char* SWIG_csharp_string_callback(const
>> char*)? redeclared as different kind of symbol
>> freeswitch_wrap.cpp:242: error: previous declaration of ?char* (*
>> SWIG_csharp_string_callback)(const char*)?
>> make[4]: *** [freeswitch_wrap.o] Error 1
>> make[3]: *** [all] Error 1
>> make[2]: *** [mod_managed-all] Error 1
>> make[1]: *** [mod_managed] Error 2
>> make: *** [mod_managed] Error 2
>>
>>
>> regards
>>
>> budi
>>
>> On Mon, May 23, 2011 at 11:28 AM, Yitzchok <adminjew at gmail.com> wrote:
>>
>>> Your missing the mono development package that doesn't get installed with
>>> the monotools-addon-server.
>>>
>>> run
>>> *yum install mono-addon-devel.x86_64*
>>> or
>>> *yum install mono-addon-devel.i386*
>>> depending on your cpu/os
>>>
>>>
>>> Yitzchok
>>>
>>>
>>>
>>> On Sun, May 22, 2011 at 6:49 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>>
>>>> hi
>>>> still got this error, even though already sert PKG_CONFIG_PATH
>>>> making all mod_managed
>>>> Package mono-2 was not found in the pkg-config search path.
>>>> Perhaps you should add the directory containing `mono-2.pc'
>>>> to the PKG_CONFIG_PATH environment variable
>>>> No package 'mono-2' found
>>>> Package mono-2 was not found in the pkg-config search path.
>>>> Perhaps you should add the directory containing `mono-2.pc'
>>>> to the PKG_CONFIG_PATH environment variable
>>>> No package 'mono-2' found
>>>> Compiling freeswitch_managed.cpp...
>>>> g++ -I/usr/local/src/freeswitch/src/include
>>>> -I/usr/local/src/freeswitch/src/include
>>>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>>>> In file included from freeswitch_managed.cpp:35:
>>>> freeswitch_managed.h:43:26: error: mono/jit/jit.h: No such file or
>>>> directory
>>>> freeswitch_managed.h:44:36: error: mono/metadata/assembly.h: No such
>>>> file or directory
>>>> freeswitch_managed.h:45:39: error: mono/metadata/environment.h: No such
>>>> file or directory
>>>> freeswitch_managed.h:46:39: error: mono/metadata/mono-config.h: No such
>>>> file or directory
>>>> freeswitch_managed.h:47:35: error: mono/metadata/threads.h: No such file
>>>> or directory
>>>> freeswitch_managed.h:48:41: error: mono/metadata/debug-helpers.h: No
>>>> such file or directory
>>>> freeswitch_managed.h:55: error: ISO C++ forbids declaration of
>>>> ?MonoDomain? with no type
>>>>
>>>>
>>>> thx
>>>> budi
>>>>
>>>> On Sun, May 22, 2011 at 6:10 PM, Yitzchok <adminjew at gmail.com> wrote:
>>>>
>>>>> Try this
>>>>>
>>>>> cd /usr/local/src/freeswitch/src/mod/languages/mod_managed/
>>>>> git apply mono28.patch
>>>>> make reswig
>>>>> make
>>>>>
>>>>>
>>>>>
>>>>> Yitzchok
>>>>>
>>>>>
>>>>>
>>>>> On Sat, May 21, 2011 at 6:42 PM, budi wibowo <bwibowo at gmail.com>wrote:
>>>>>
>>>>>> failed with some glib issue on source code compiling, then i use yum
>>>>>> as suggested.
>>>>>> mono installed, thx
>>>>>> Mono JIT compiler version 2.10.2 (tarball Mon Apr 18 19:06:50 UTC
>>>>>> 2011)
>>>>>> Copyright (C) 2002-2011 Novell, Inc and Contributors.
>>>>>> www.mono-project.com
>>>>>>         TLS:           __thread
>>>>>>         SIGSEGV:       altstack
>>>>>>         Notifications: epoll
>>>>>>         Architecture:  amd64
>>>>>>         Disabled:      none
>>>>>>         Misc:          debugger softdebug
>>>>>>         LLVM:          supported, not enabled.
>>>>>>         GC:            Included Boehm (with typed GC and Parallel
>>>>>> Mark)
>>>>>>
>>>>>> when i try to install mod_managed i found this error, may be some
>>>>>> missing config in my linux box
>>>>>>
>>>>>> making all mod_managed
>>>>>> Compiling freeswitch_managed.cpp...
>>>>>> g++ -I/usr/local/src/freeswitch/src/include
>>>>>> -I/usr/local/src/freeswitch/src/include
>>>>>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>>>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>>>>> -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp
>>>>>> In file included from freeswitch_managed.cpp:35:
>>>>>> freeswitch_managed.h:43:18: error: glib.h: No such file or directory
>>>>>> freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or
>>>>>> directory
>>>>>> freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such
>>>>>> file or directory
>>>>>> freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No
>>>>>> such file or directory
>>>>>> freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No
>>>>>> such file or directory
>>>>>> freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such
>>>>>> file or directory
>>>>>> freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No
>>>>>> such file or directory
>>>>>>
>>>>>>
>>>>>> thx
>>>>>> budi wibowo
>>>>>>
>>>>>> On Wed, May 18, 2011 at 9:06 AM, Yitzchok <adminjew at gmail.com> wrote:
>>>>>>
>>>>>>> Make sure mono is installed and check which version using "mono -V"
>>>>>>> and follow the instructions on
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_managed let us know if you get
>>>>>>> into any errors.
>>>>>>>
>>>>>>> This helped me for installing mono
>>>>>>> http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum
>>>>>>>
>>>>>>>
>>>>>>> <http://stackoverflow.com/questions/3510320/install-mono-on-centos5-5-using-yum>
>>>>>>> Yitzchok
>>>>>>>
>>>>>>>
>>>>>>> On Tue, May 17, 2011 at 8:02 PM, budi wibowo <bwibowo at gmail.com>wrote:
>>>>>>>
>>>>>>>> hi
>>>>>>>> i have fs running on centos 5.5, any body knows the combination
>>>>>>>> version of software need to be installed like mono etc.
>>>>>>>> have tried few combination but always failed compiling
>>>>>>>>
>>>>>>>>
>>>>>>>> thx
>>>>>>>> budi
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eric at loopfx.com  Tue May 24 00:58:30 2011
From: eric at loopfx.com (Eric Beard)
Date: Mon, 23 May 2011 16:58:30 -0400
Subject: [Freeswitch-users] Read SIP Response Headers from Dialplan or Lua
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFC39AB0@doctorbob.loop.local>

Hello,

I'm wondering if it's possible to read SIP response headers from the dial plan, or from Lua.

One of my gateways returns a generic 500 for all kinds of different error conditions, and expects me to read variables in the sip response.

e.g.

SIP/2.0 500 Service Unavailable
Via:..
From:...
Etc:
X-Custom-Var:val

The value in the custom variable might be "busy", "bad number", etc.

If I add something to the INVITE, I know that I can read it with channel variables "sip_h_X-Name", but I don't get the same thing for the response.

Thanks,

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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From Prometheus001 at gmx.net  Tue May 24 01:21:32 2011
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 23 May 2011 23:21:32 +0200
Subject: [Freeswitch-users] mod_event_zmq - sending commands?
Message-ID: <4DDACFDC.4040103@gmx.net>

Hello,

we have been successfully testing mod_event_zmq and we are quite pleased
with this module so far.

My question: In order for being able to use the same architecture for
sending events: Are ther any plans for sending API commands to
Freeswitch by zmq? This may be an alternate method for XML-RPC.
I understand this my be a security issue, but we could manage this by
some firewall rules.

Best regards
Peter
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From steveayre at gmail.com  Tue May 24 01:25:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 23 May 2011 22:25:02 +0100
Subject: [Freeswitch-users] mod_event_zmq - sending commands?
In-Reply-To: <4DDACFDC.4040103@gmx.net>
References: <4DDACFDC.4040103@gmx.net>
Message-ID: <BANLkTikmAju=GK3mXcVm1+3Us0nDf5G9Vw@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_event_zmq lists it as a ToDo.


On 23 May 2011 22:21, Peter P GMX <Prometheus001 at gmx.net> wrote:

>  Hello,
>
> we have been successfully testing mod_event_zmq and we are quite pleased
> with this module so far.
>
> My question: In order for being able to use the same architecture for
> sending events: Are ther any plans for sending API commands to Freeswitch by
> zmq? This may be an alternate method for XML-RPC.
> I understand this my be a security issue, but we could manage this by some
> firewall rules.
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jonyoung111 at gmail.com  Tue May 24 01:28:15 2011
From: jonyoung111 at gmail.com (Jon Young)
Date: Mon, 23 May 2011 14:28:15 -0700
Subject: [Freeswitch-users] Voicemail recordings being cut off on the
 front end.
In-Reply-To: <4DDAB61E.3010400@coleinternet.com>
References: <4DDAB61E.3010400@coleinternet.com>
Message-ID: <BANLkTingw-_e+jpN7C588kewvqpq3CKDRw@mail.gmail.com>

If it is the voicemail recording being left by the caller, it is
possible that your gateway / phones / SIP provider are using silence
suppression.  You would need to look at the RTP stream coming in to
FS.  You can use Wireshark (if G.711) to replay it and see if you are
receiving it correctly.

On Mon, May 23, 2011 at 12:31 PM, Jay
<p2.freeswitch-list at coleinternet.com> wrote:
> I need some direction on tracking down why some voicemail recordings are being cut off at the front end. There are instances where the start of the .wav file is a good 5 seconds into the intended message.
>
> what FS components could be responsible clipped voicemail recordings? ? ?is there silence detection going on? if so, are there any controls such as automatic gain control, fixed volume settings for recordings or other normalizing going on?
>
> any advice?
>
> thx.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Tue May 24 01:28:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 May 2011 14:28:29 -0700
Subject: [Freeswitch-users] How can i drop the outcallcontext from the
 originate callerid?
In-Reply-To: <C8355FEB-5D0C-4C7B-867A-764FB49D1CAD@ipeva.fr>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
	<Pine.LNX.4.64.1105230943190.3447@debian>
	<C8355FEB-5D0C-4C7B-867A-764FB49D1CAD@ipeva.fr>
Message-ID: <BANLkTi=ChOvEP5BbHFFxSD1ke0jM7RR7aQ@mail.gmail.com>

On Mon, May 23, 2011 at 5:13 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> There should not be a outcallcontext (what is that by the way ?) appended
> to the caller_id_number.
> Something must be wrong in your config.
>

I tend to agree with David on this. Please use pastebin.freeswitch.org and
drop us a debug log of a call that has this symptom. Use "FreeSWITCH Log"
for the syntax highlighting and reply to this thread with the pb link...

-MC
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From msc at freeswitch.org  Tue May 24 01:30:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 May 2011 14:30:47 -0700
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTi=UsLW4b9crvGHdfVt6Mxp49Doxqg@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTi=UsLW4b9crvGHdfVt6Mxp49Doxqg@mail.gmail.com>
Message-ID: <BANLkTinjx_8pnJu-T8BCL03km+H7Q=sRNA@mail.gmail.com>

On Mon, May 23, 2011 at 11:47 AM, Jon Young <jonyoung111 at gmail.com> wrote:

> Would this also be an issue/resolution with Google Voice as well?
> Using Dingaling I am routing my calls to an IVR to intercept the call
> from going to Google Voice voicemail.  However, the IVR won't accept
> DTMF.
>
> That would not surprise me. The best way to know 100% for certain on your
DTMFs is to get a pcap of the call and analyze it in wireshark.
-MC
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From sireeps at gmail.com  Tue May 24 01:36:36 2011
From: sireeps at gmail.com (Kamen)
Date: Mon, 23 May 2011 14:36:36 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <BANLkTinvAzxVVTBGi-c3D8_hci6i2maBpA@mail.gmail.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306164331568-6394908.post@n2.nabble.com>
	<BANLkTinvAzxVVTBGi-c3D8_hci6i2maBpA@mail.gmail.com>
Message-ID: <1306186596224-6396358.post@n2.nabble.com>

Yeah, it does!  Thanks, MC!

Here is the colored trace:

http://pastebin.freeswitch.org/16360 http://pastebin.freeswitch.org/16360 

Please have a look. Could it be really NAT?  Seems like everything else is
working fine over my router.  Besides I was trying to test it directly over
DMZ connection with the same results. Thanks in advance!

Regards,

Sergei Kamen


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6396358.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue May 24 01:33:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 May 2011 14:33:40 -0700
Subject: [Freeswitch-users] Read SIP Response Headers from Dialplan or
	Lua
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFC39AB0@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DFC39AB0@doctorbob.loop.local>
Message-ID: <BANLkTi=-5f2vNLhQa9RG_sO6KFXMkoSP5A@mail.gmail.com>

I would turn on mod_xml_cdr and look at a failed call's XML output. Look in
the <variables> section and see if any of those have the data you're looking
for.

-MC

On Mon, May 23, 2011 at 1:58 PM, Eric Beard <eric at loopfx.com> wrote:

> Hello,
>
>
>
> I?m wondering if it?s possible to read SIP response headers from the dial
> plan, or from Lua.
>
>
>
> One of my gateways returns a generic 500 for all kinds of different error
> conditions, and expects me to read variables in the sip response.
>
>
>
> e.g.
>
>
>
> SIP/2.0 500 Service Unavailable
>
> Via:..
>
> From:?
>
> Etc:
>
> X-Custom-Var:val
>
>
>
> The value in the custom variable might be ?busy?, ?bad number?, etc.
>
>
>
> If I add something to the INVITE, I know that I can read it with channel
> variables ?sip_h_X-Name?, but I don?t get the same thing for the response.
>
>
>
> Thanks,
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mtaylor at employees.org  Tue May 24 02:11:36 2011
From: mtaylor at employees.org (Mike Taylor)
Date: Tue, 24 May 2011 10:11:36 +1200
Subject: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
In-Reply-To: <BANLkTin7a61Nw-L6qRhGbwYanHO9LGVDGg@mail.gmail.com>
Message-ID: <CA013499.2C1FE%mike.taylor@staff.snap.net.nz>

Good point, thanks,

But, RFC2833 I've found to be unreliable, broken, patchy, scratchy?.

I've got it set to RFC2833 now, and it just ain't working.

Will get more traces, trial and error until I stumble across a good combo,
or bug(s)?

Thanks again,
Mike

From:  Ognjen Seslija <oseslija at gmail.com>
Reply-To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date:  Mon, 23 May 2011 20:46:09 +0200
To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] FS - DTMF and Cisco CME/UCE

Why use INFO when it's only method I know Cisco won't speak (2833, kpml,
sub/notify)?

Here's the CME snippet for FS trunk I have. FS sends 2833 tones to CME btw.

dial-peer voice 100 voip
 description ** TO FS **
 destination-pattern .T
 b2bua
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad




On Mon, May 23, 2011 at 4:07 AM, Mike Taylor <mtaylor at employees.org> wrote:
> Hi,
> 
> This is driving me nuts.
> I have a Cisco CME (a few actually) where I cannot get DTMF to work.
> 
> Does someone happen to have the magic recipe of SIP/Dialplan profiles and
> Cisco config to make DTMF <relay> work?
> 
> I spent several very very painful days trying to get RFC2833 to work, using FS
> to try and generate the tones itself. I ended up with all sorts of codec
> mismatches, delay,distortion and some really mangled DTMF tones?
> 
> So, now I'm back to;
> 
> PSTN----CiscoGW----<external>----FS----<internal>----<CUST PABX>---Phone
> 
> Where <CUST PABX> is 3 CMEs that won't talk DTMF
> 30+ other vendor's gateway's that WILL do DTMF?
> 
> Context External and internal are both set to dtmf-type info (or dtmf_type
> info) 
> 
> DTMF works FROM the Cisco GW towards our network, doesn't
> 
> 
> So, does anyone have a magic solution?
>  Because Cisco TAC say they see DTMF being sent (as rtp-nte), when I need INFO
> or in-band.
> 
> 
> 
> Thanks,
> 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
> 

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

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From david.ponzone at ipeva.fr  Tue May 24 02:11:27 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 24 May 2011 00:11:27 +0200
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>
Message-ID: <FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>

Rob,

perhaps you should not consider T38 is 100% interoperable.
You may tell us what ATA is that, because some of them are nice piece of junk.

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

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

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




Le 23/05/2011 ? 20:51, Rob Morin a ?crit :

> I?m having problems faxing if the B leg offers T.38.  Several scenarios, but always the same result.
>  
> First, the architecture
>  
> FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another carrier (PSTN) and the fax machine)
>  
> On my ATA, T.38 is enabled (Auto Detect).
>  
> If the destination supports T.38, the Carrier will offer it to Freeswitch. If the Destination doesn?t support T.38, the B leg doesn?t offer it and the faxes go through.
>  
> I?ve set Freeswitch as:
>       <action application="set" data="fax_enable_t38=true"/>
>       <action application="export" data="t38_passthru=true"/> 
> When that?s the case, and the B leg offers T.38, Freeswitch passes through the offer and the ATA responds, 3 times. But the response never gets sent to the B leg, so it terminates the call after about 30 seconds.
>  
> When I set Freeswitch as:
>       <action application="set" data="fax_enable_t38=false"/>
> The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t pass the offer through and the B leg terminates the call after about 30 seconds.
>  
> I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d 2011-05-17 22-51-47 -0500) ).
>  
> I?ve had no success attempting to configure FS as a gateway. Frankly, T.38 ?passthrough? has had the most success, mostly when the other ends aren?t T.38 capable and it stays in G711u.
>  
> Is there something else I need to do to enable T.38 in passthrough mode? 
>  
> Thank you,
> Rob Morin
>  
> PS ? I can provide tcpdumps of this, or whatever else is necessary.
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue May 24 01:50:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 May 2011 14:50:59 -0700
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <1306186596224-6396358.post@n2.nabble.com>
References: <1305407454535-6363912.post@n2.nabble.com>
	<1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306164331568-6394908.post@n2.nabble.com>
	<BANLkTinvAzxVVTBGi-c3D8_hci6i2maBpA@mail.gmail.com>
	<1306186596224-6396358.post@n2.nabble.com>
Message-ID: <BANLkTikdyLNY-PSwpd1As9aAEU833bXOVw@mail.gmail.com>

Okay, let me make a suggestion: only include the actual call. You have all
of the startup and shutdown stuff in there and it's confusing. Capture only
the part where you make the call and include it from start to finish,
including the siptrace.
-MC

On Mon, May 23, 2011 at 2:36 PM, Kamen <sireeps at gmail.com> wrote:

> Yeah, it does!  Thanks, MC!
>
> Here is the colored trace:
>
> http://pastebin.freeswitch.org/16360 http://pastebin.freeswitch.org/16360
>
> Please have a look. Could it be really NAT?  Seems like everything else is
> working fine over my router.  Besides I was trying to test it directly over
> DMZ connection with the same results. Thanks in advance!
>
> Regards,
>
> Sergei Kamen
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6396358.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 chris at fowler.cc  Tue May 24 02:49:46 2011
From: chris at fowler.cc (Chris Fowler)
Date: Mon, 23 May 2011 18:49:46 -0400
Subject: [Freeswitch-users] How to troubleshoot "503 Maximum Calls In
	Progress"?
Message-ID: <7454A296C7EDE34EA57199FAA401E2F1220CB5293C@VMBX113.ihostexchange.net>

Hi,

On our Production FreeSWITCH box:

FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500)

UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, 751 microseconds
12519 session(s) since startup
2 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/100.00

At 14:30:37 a reloadxml command was issued.
At 14:30:43 the box started rejecting all calls with 503 error.
At 14:38 it started working again.


Looking at the code I see three conditions can trigger this behavior:

              if (sess_count >= sess_max || !sofia_test_pflag(profile, PFLAG_RUNNING) || !switch_core_ready()) {
                        nua_respond(nh, 503, "Maximum Calls In Progress", SIPTAG_RETRY_AFTER_STR("300"), TAG_END());

                        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n");

                        goto done;
               }

Logs show it wasn't lack of available sessions.  How can sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and busy the system?

Thoughts? Thx, Chris.

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From yungwei at resolvity.com  Tue May 24 03:45:35 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 23 May 2011 19:45:35 -0400
Subject: [Freeswitch-users] Error playing TTS using flite
Message-ID: <33095823FD21DF429B481B5163264B7950AC3ABFC7@VMBX102.ihostexchange.net>

Hi,

I am testing tts using flite against freeswitch (snapshot downloaded last week) on CentOS 5.
I'm getting "Invalid speech module [cepstral]!" and "Invalid TTS module" errors when dialing 9911 from my SIP client.
What am I missing here? Thanks.

...
Dialplan: sofia/internal/1000 at 192.168.216.18 parsing [default->test1] continue=false
Dialplan: sofia/internal/1000 at 192.168.216.18 Regex (PASS) [test1] destination_number(9911) =~ /^9911$/ break=on-false
Dialplan: sofia/internal/1000 at 192.168.216.18 Action javascript(dtmf_test.js)  --> Freeswitch figured out dtmf_test.js is the action to perform
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000 at 192.168.216.18) State Change CS_ROUTING -> CS_EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/1000 at 192.168.216.18 [BREAK]
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1000 at 192.168.216.18) State ROUTING going to sleep
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000 at 192.168.216.18) Running State Change CS_EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1000 at 192.168.216.18) State EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] mod_sofia.c:240 sofia/internal/1000 at 192.168.216.18 SOFIA EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 192.168.216.18 Standard EXECUTE
EXECUTE sofia/internal/1000 at 192.168.216.18 set(open=true)
2011-05-23 18:13:09.859007 [DEBUG] mod_dptools.c:1060 sofia/internal/1000 at 192.168.216.18 SET [open]=[true]
EXECUTE sofia/internal/1000 at 192.168.216.18 hash(insert/192.168.216.18-spymap/1000/399178b8-8592-11e0-ae8d-e9a6940f4ae1)
EXECUTE sofia/internal/1000 at 192.168.216.18 hash(insert/192.168.216.18-last_dial/1000/9911)
EXECUTE sofia/internal/1000 at 192.168.216.18 hash(insert/192.168.216.18-last_dial/global/399178b8-8592-11e0-ae8d-e9a6940f4ae1)
EXECUTE sofia/internal/1000 at 192.168.216.18 set(RFC2822_DATE=Mon, 23 May 2011 18:13:09 -0500)
2011-05-23 18:13:09.859007 [DEBUG] mod_dptools.c:1060 sofia/internal/1000 at 192.168.216.18 SET [RFC2822_DATE]=[Mon, 23 May 2011 18:13:09 -0500]
EXECUTE sofia/internal/1000 at 192.168.216.18 javascript(dtmf_test.js)
2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3034 AUDIO RTP [sofia/internal/1000 at 192.168.216.18] 192.168.216.18 port 31886 -> 192.168.216.9 port 5062 codec: 8 ms: 20
2011-05-23 18:13:09.879013 [DEBUG] switch_rtp.c:1636 Starting timer [soft] 160 bytes per 20ms
2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3296 Set 2833 dtmf send payload to 101
2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3301 Set 2833 dtmf receive payload to 101
2011-05-23 18:13:09.879013 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/1000 at 192.168.216.18:
v=0
o=FreeSWITCH 1306160503 1306160504 IN IP4 192.168.216.18
s=FreeSWITCH
c=IN IP4 192.168.216.18
t=0 0
m=audio 31886 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
a=sendrecv

2011-05-23 18:13:09.879013 [DEBUG] switch_core_session.c:707 Send signal sofia/internal/1000 at 192.168.216.18 [BREAK]
2011-05-23 18:13:09.879013 [DEBUG] switch_channel.c:2859 (sofia/internal/1000 at 192.168.216.18) Callstate Change RINGING -> ACTIVE
2011-05-23 18:13:09.879013 [NOTICE] mod_spidermonkey.c:2068 Channel [sofia/internal/1000 at 192.168.216.18] has been answered
2011-05-23 18:13:09.879013 [DEBUG] dtmf_test.js:34 sayivrmenu: menu=[mainmenu] validdigits=[0123#]
2011-05-23 18:13:09.879013 [DEBUG] sofia.c:4770 Channel sofia/internal/1000 at 192.168.216.18 entering state [completed][200]
2011-05-23 18:13:09.879013 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Press 0 for the Operator.] (en:en)
2011-05-23 18:13:09.879013 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]! -->cepstral is never installed. Why did Freeswitch try to use cepstral?
2011-05-23 18:13:09.879013 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module!
2011-05-23 18:13:09.879013 [DEBUG] sofia.c:4770 Channel sofia/internal/1000 at 192.168.216.18 entering state [ready][200]
2011-05-23 18:13:09.939014 [DEBUG] switch_rtp.c:3104 Correct ip/port confirmed.
2011-05-23 18:13:15.979391 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Press 0 for the Operator.] (en:en)
2011-05-23 18:13:15.979391 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2011-05-23 18:13:15.979391 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module!
2011-05-23 18:13:22.079772 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Press 0 for the Operator.] (en:en)
2011-05-23 18:13:22.079772 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2011-05-23 18:13:22.079772 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module!

During installatio, I uncommented the following in modules.conf so mod_flite is installed. mod_cepstral remains commented.
asr_tts/mod_unimrcp
asr_tts/mod_flite
asr_tts/mod_pocketsphinx

After installation, I uncommented mod_flite in /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml
    <!-- ASR /TTS -->
    <load module="mod_flite"/>
    <!-- <load module="mod_pocketsphinx"/> -->
    <!-- <load module="mod_cepstral"/> -->
    <!-- <load module="mod_tts_commandline"/> -->
    <!-- <load module="mod_rss"/> -->

Here's my dial-plan
  <extension name="test1">
        <condition field="destination_number" expression="^9911$">
                <action application="set" data="tts_engine=flite"/>
                <action application="set" data="tts_voice=kal"/>
                <action application="javascript" data="dtmf_test.js" />
        </condition>
  </extension>

dtmf_test.js is copied from http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js to /usr/local/freeswitch/scripts folder.
The phrase macro is also copied from that link to /usr/local/freeswitch/conf/lang/en/demo folder.




From seichhorn at gci.com  Tue May 24 03:53:08 2011
From: seichhorn at gci.com (Sean Eichhorn)
Date: Mon, 23 May 2011 15:53:08 -0800
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
	<BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>
Message-ID: <7EB00560C8AFF348A7EA76EFD3DC5598332C035F@dtn1mbx01.gci.com>

Yeah, they're negotiating the same codec on both sides.
The problem is the way Cisco handles the negotiation.  Here's an example of what I see in Freeswitch:

Received from CALLED :

  SIP/2.0 200 OK
   Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bKXZZ18p8tD03cS
   From: "pending" <sip:xxxxxxxx02 at A.B.C.D>;tag=NH1HrNe744HSB
   To: <sip:sean at E.F.G.H:5060>;tag=622555CC-408
   Date: Mon, 23 May 2011 23:50:56 GMT
   Call-ID: 69258f8c-0038-122f-19ba-000c29c18d38
   CSeq: 12760849 INVITE
   Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
   Allow-Events: telephone-event
   Contact: <sip:xxxxxxxx75 at E.F.G.H:5060>
   Supported: replaces
   Supported: sdp-anat
   Server: Cisco-SIPGateway/IOS-12.x
   Content-Type: application/sdp
   Content-Disposition: session;handling=required
   Content-Length: 311

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 4080 7999 IN IP4 E.F.G.H
   s=SIP Call
   c=IN IP4 E.F.G.H
   t=0 0
   m=audio 17122 RTP/AVP 0 19 101 100
   c=IN IP4 E.F.G.H
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:100 X-NSE/8000
   a=fmtp:100 192-194

Sent to CALLING :
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP I.J.K.L:5060;branch=z9hG4bK4C2B6C64
   From: <sip:xxxxxxxx02@ I.J.K.L >;tag=9020153C-21B3
   To: <sip:xxxxxxxx75 at A.B.C.D>;tag=m87rptX37UU6F
   Call-ID: AFADEF2E-84CC11E0-B1D4F23A-27804614@ I.J.K.L
   CSeq: 101 INVITE
   Contact: <sip:xxxxxxxx75@ A.B.C.D:5060;transport=udp>
   User-Agent: GCI
   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
   Min-SE: 1800
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 261
   Remote-Party-ID: "Outbound Call" <sip:sean@ A.B.C.D >;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1995227603 1995227604 IN IP4 A.B.C.D
   s=FreeSWITCH
   c=IN IP4 A.B.C.D
   t=0 0
   m=audio 24636 RTP/AVP 0 19 101
   c=IN IP4 A.B.C.D
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16

This is not the case when the call direction is reversed.  Like I said, the only difference appears to be that one direction is responding with a 183 while this one is responding with a 200.

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From gcd at i.ph  Tue May 24 04:14:02 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 24 May 2011 08:14:02 +0800
Subject: [Freeswitch-users] Getting the pizza demo to work
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3ABF25@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3ABF25@VMBX102.ihostexchange.net>
Message-ID: <BANLkTi=Mj0t-YpG0w+c8Umqp6BJj57COmQ@mail.gmail.com>

perhaps mod_pocketsphinx is not loaded in
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml. you can check
if the module works by loading the module in the CLI before dialing the
pizza demo.
-nandy

On Tue, May 24, 2011 at 3:40 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I just installed the latest snapshot on CentOS 5.
> I want to try the pizza demo associated with 00_pizza_demo.xml, but I'm
> having trouble getting the pizza demo to work.
>
> According to freeswitch.log, it looks like mod_pocketsphinx doesn't exist.
> ...skipped...
> Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (PASS) [pizza_demo]
> destination_number(74992) =~ /^(pizza|74992)$/ break=on-false
> Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (PASS) [pizza_demo]
> ${module_exists(mod_spidermonkey)}(true) =~ /true/ break=on-false
> Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (FAIL) [pizza_demo]
> ${module_exists(mod_pocketsphinx)}(false) =~ /true/ break=on-false
> Dialplan: sofia/internal/1000 at 192.168.16.3 parsing [default->Talking Clock
> Time] continue=false
> Dialplan: sofia/internal/1000 at 192.168.16.3 Regex (FAIL) [Talking Clock
> Time] destination_number(74992) =~ /^9170$/ break=on-false
> ...skipped...
> Dialplan: sofia/internal/1000 at 192.168.16.3 Action transfer(74992 enum)
>
> But I do see mod_pocketsphinx.so in /usr/local/freeswitch/mod folder.
> [root at templ0 mod]# ls /usr/local/freeswitch/mod/mod_pocketsphinx.so -l
> -rwxr-xr-x 1 root root 1353092 May 23 13:23
> /usr/local/freeswitch/mod/mod_pocketsphinx.so
>
> I also went through http://wiki.freeswitch.org/wiki/Mod_pocketsphinx, but
> the problem is still there.
> How am I missing here? 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 sireeps at gmail.com  Tue May 24 04:22:00 2011
From: sireeps at gmail.com (Kamen)
Date: Mon, 23 May 2011 17:22:00 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <BANLkTikdyLNY-PSwpd1As9aAEU833bXOVw@mail.gmail.com>
References: <1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306164331568-6394908.post@n2.nabble.com>
	<BANLkTinvAzxVVTBGi-c3D8_hci6i2maBpA@mail.gmail.com>
	<1306186596224-6396358.post@n2.nabble.com>
	<BANLkTikdyLNY-PSwpd1As9aAEU833bXOVw@mail.gmail.com>
Message-ID: <1306196520259-6396818.post@n2.nabble.com>

Sure, no problem!  Sorry about that.  I am not sure what is important and
what is not, that's why. 

The trimmed down trace is here:
http://pastebin.freeswitch.org/16362 http://pastebin.freeswitch.org/16362 

Cheers,



Sergei Kamen



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-calls-to-external-gateway-tp6363912p6396818.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rmorin at blie-ent.com  Tue May 24 04:59:24 2011
From: rmorin at blie-ent.com (Rob Morin)
Date: Mon, 23 May 2011 20:59:24 -0400
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>
	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>
Message-ID: <02e001cc19ad$d3d26780$7b773680$@blie-ent.com>

David,

 

It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and
Cisco/Linksys SPA 2102. Same story in both cases.

 

I had my carrier help me troubleshoot and we can see the SIP request from
the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but
the response from the A leg isn?t forwarded back to the B leg. I can produce
traces that show this. 

 

When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t
respond or forward it. So the connection dies.  I have every reason to
believe that I?d be able to send the fax if FS would respond with a 488
indicating T.38 is not acceptable. But instead, all I get is silence. Again,
I can produce traces that demonstrate this.

 

There?s another similar email  trail out there,
http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821
994.html, but I?m not certain that they?re related.

 

Thank you for your help!
Rob 

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Monday, May 23, 2011 6:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

perhaps you should not consider T38 is 100% interoperable.

You may tell us what ATA is that, because some of them are nice piece of
junk.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 





 

Le 23/05/2011 ? 20:51, Rob Morin a ?crit :





I?m having problems faxing if the B leg offers T.38.  Several scenarios, but
always the same result.

 

First, the architecture

 

FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another
carrier (PSTN) and the fax machine)

 

On my ATA, T.38 is enabled (Auto Detect).

 

If the destination supports T.38, the Carrier will offer it to Freeswitch.
If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
faxes go through.

 

I?ve set Freeswitch as:

      <action application="set" data="fax_enable_t38=true"/>

      <action application="export" data="t38_passthru=true"/> 

When that?s the case, and the B leg offers T.38, Freeswitch passes through
the offer and the ATA responds, 3 times. But the response never gets sent to
the B leg, so it terminates the call after about 30 seconds.

 

When I set Freeswitch as:

      <action application="set" data="fax_enable_t38=false"/>

The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t
pass the offer through and the B leg terminates the call after about 30
seconds.

 

I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
2011-05-17 22-51-47 -0500) ).

 

I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
?passthrough? has had the most success, mostly when the other ends aren?t
T.38 capable and it stays in G711u.

 

Is there something else I need to do to enable T.38 in passthrough mode? 

 

Thank you,
Rob Morin

 

PS ? I can provide tcpdumps of this, or whatever else is necessary.

 

 

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

 

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From kheimerl at cs.berkeley.edu  Tue May 24 06:56:46 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 23 May 2011 19:56:46 -0700
Subject: [Freeswitch-users] Removing SIP Auth
Message-ID: <BANLkTim+kovEEk0ZeKCWcsObBYWvVdq+LQ@mail.gmail.com>

Hello!

I've been trying to get my SIP clients to register and authenticate
with freeswitch without passwords. I've been following a few wiki
pages, primarily:

http://wiki.freeswitch.org/wiki/Acl
and
http://gnuradio.org/redmine/wiki/gnuradio/OpenBTSSettingUpFreeSWITCH

Most of the configs are stock, except my directory which has one new
user in the "default" folder:

<include>
  <user id="IMSI641104378340883" cidr="192.168.1.0/24,127.0.0.1/32"
number-alias="1301">
    <params>
      <param name="vm-password" value="1000"/>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
    </params>
...

This config is working in one one way, I do not need to register to
make a call. The phone is currently able to call into FS. However, any
attempts to register ()REGISTER messages) are met with a 401
authorization required response (as shown by wireshark). This means FS
is unable to route calls to my handsets. My understanding was that the
line,

<param name="accept-blind-auth" value="true"/>

when coupled with the cidr value, would remove that need. It's clearly
not working. I've tried a variety of other config options (e.g.,
<param name="apply-register-acl" value="<acl_list|cidr>"/>), and found
no joy. ACLs are complicated business, and I could take a few more
stabs in the dark (using localnet.auto?). However, I thought I'd ask
this list.

What else do I need to do to let my users connect without a password?

Thanks!


From yehavi.bourvine at gmail.com  Tue May 24 07:54:52 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 24 May 2011 06:54:52 +0300
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <02e001cc19ad$d3d26780$7b773680$@blie-ent.com>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>
	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>
	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>
Message-ID: <BANLkTimoStTFR7+Oi4vCmkP0q0StbLHtkg@mail.gmail.com>

It might be related to JIRA 3274 which I've (re)opened recently.
When Freeswitch receives a RE-Invite with T.38 parameters inside it responds
with "Trying" but no "OK" following it. The originator then times out and
disconnects the call.

                     Regards, __Yehavi:

2011/5/24 Rob Morin <rmorin at blie-ent.com>

> David,
>
>
>
> It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and
> Cisco/Linksys SPA 2102. Same story in both cases.
>
>
>
> I had my carrier help me troubleshoot and we can see the SIP request from
> the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but
> the response from the A leg isn?t forwarded back to the B leg. I can produce
> traces that show this.
>
>
>
> When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t
> respond or forward it. So the connection dies.  I have every reason to
> believe that I?d be able to send the fax if FS would respond with a 488
> indicating T.38 is not acceptable. But instead, all I get is silence. Again,
> I can produce traces that demonstrate this.
>
>
>
> There?s another similar email  trail out there,
> http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821994.html,
> but I?m not certain that they?re related.
>
>
>
> Thank you for your help!
> Rob
>
>
>
> *From:* David Ponzone [mailto:david.ponzone at ipeva.fr]
> *Sent:* Monday, May 23, 2011 6:11 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>
>
>
> Rob,
>
>
>
> perhaps you should not consider T38 is 100% interoperable.
>
> You may tell us what ATA is that, because some of them are nice piece of
> junk.
>
>
>
> David Ponzone  Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel:      01 74 03 18 97
>
> gsm:   06 66 98 76 34
>
>
>
> Service Client IPeva
>
> tel:      0811 46 26 26
>
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
>
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
>
> * *
>
>
>
>
>
> Le 23/05/2011 ? 20:51, Rob Morin a ?crit :
>
>
>
> I?m having problems faxing if the B leg offers T.38.  Several scenarios,
> but always the same result.
>
>
>
> First, the architecture
>
>
>
> FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another
> carrier (PSTN) and the fax machine)
>
>
>
> On my ATA, T.38 is enabled (Auto Detect).
>
>
>
> If the destination supports T.38, the Carrier will offer it to Freeswitch.
> If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
> faxes go through.
>
>
>
> I?ve set Freeswitch as:
>
>       <action application="set" data="fax_enable_t38=true"/>
>
>       <action application="export" data="t38_passthru=true"/>
>
> When that?s the case, and the B leg offers T.38, Freeswitch passes through
> the offer and the ATA responds, 3 times. But the response never gets sent to
> the B leg, so it terminates the call after about 30 seconds.
>
>
>
> When I set Freeswitch as:
>
>       <action application="set" data="fax_enable_t38=false"/>
>
> The B leg still offers T.38 if its other side is capable. Freeswitch
> doesn?t pass the offer through and the B leg terminates the call after about
> 30 seconds.
>
>
>
> I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
> 2011-05-17 22-51-47 -0500) ).
>
>
>
> I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
> ?passthrough? has had the most success, mostly when the other ends aren?t
> T.38 capable and it stays in G711u.
>
>
>
> Is there something else I need to do to enable T.38 in passthrough mode?
>
>
>
> Thank you,
> Rob Morin
>
>
>
> PS ? I can provide tcpdumps of this, or whatever else is necessary.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Tue May 24 08:01:16 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 24 May 2011 07:01:16 +0300
Subject: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
In-Reply-To: <CA013499.2C1FE%mike.taylor@staff.snap.net.nz>
References: <BANLkTin7a61Nw-L6qRhGbwYanHO9LGVDGg@mail.gmail.com>
	<CA013499.2C1FE%mike.taylor@staff.snap.net.nz>
Message-ID: <BANLkTik5zdomwsTWHGZs-Lpochth+ptkmQ@mail.gmail.com>

Hello,

  I had for a long time a Cisco SIP-E1(with Q.sig) gateway which caused me
much grief with DTMF; it works partially OK with INFO but not with RFC-2833.
I filed a support call with Cisco, as its debug shows that it receives and
detects the RFC2833 but nothing is sent to the E1 side.

The last conslusion of Cisco's support is that I have to enable transcoding
but they coouldn't find how to enable it in my configuration. I now have an
Audioodes Mediant-1000 on loan which does RFC-2833 without any problems...

The only problem I've found so far with the AudioCodes is name passing via
E1/Q.sig at the ringing phase (we use it to connect our Freeswitch to our
Nortel PbX).

                        Regards, __Yehavi:



2011/5/24 Mike Taylor <mtaylor at employees.org>

> Good point, thanks,
>
> But, RFC2833 I've found to be unreliable, broken, patchy, scratchy?.
>
> I've got it set to RFC2833 now, and it just ain't working.
>
> Will get more traces, trial and error until I stumble across a good combo,
> or bug(s)?
>
> Thanks again,
> Mike
>
> From: Ognjen Seslija <oseslija at gmail.com>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 23 May 2011 20:46:09 +0200
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
>
> Why use INFO when it's only method I know Cisco won't speak (2833, kpml,
> sub/notify)?
>
> Here's the CME snippet for FS trunk I have. FS sends 2833 tones to CME btw.
>
> dial-peer voice 100 voip
>  description ** TO FS **
>  destination-pattern .T
>  b2bua
>  session protocol sipv2
>  session target sip-server
>  session transport udp
>  dtmf-relay rtp-nte
>  codec g711alaw
>  no vad
>
>
>
>
> On Mon, May 23, 2011 at 4:07 AM, Mike Taylor <mtaylor at employees.org>wrote:
>
>> Hi,
>>
>> This is driving me nuts.
>> I have a Cisco CME (a few actually) where I cannot get DTMF to work.
>>
>> Does someone happen to have the magic recipe of SIP/Dialplan profiles and
>> Cisco config to make DTMF <relay> work?
>>
>> I spent several very very painful days trying to get RFC2833 to work,
>> using FS to try and generate the tones itself. I ended up with all sorts of
>> codec mismatches, delay,distortion and some really mangled DTMF tones?
>>
>> So, now I'm back to;
>>
>> PSTN----CiscoGW----<external>----FS----<internal>----<CUST PABX>---Phone
>>
>> Where <CUST PABX> is 3 CMEs that won't talk DTMF
>> 30+ other vendor's gateway's that WILL do DTMF?
>>
>> Context External and internal are both set to dtmf-type info (or dtmf_type
>> info)
>>
>> DTMF works FROM the Cisco GW towards our network, doesn't
>>
>>
>> So, does anyone have a magic solution?
>>  Because Cisco TAC say they see DTMF being sent (as rtp-nte), when I need
>> INFO or in-band.
>>
>>
>>
>> Thanks,
>> 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
>>
>>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Tue May 24 09:42:15 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 24 May 2011 07:42:15 +0200
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <7EB00560C8AFF348A7EA76EFD3DC5598332C035F@dtn1mbx01.gci.com>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
	<BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035F@dtn1mbx01.gci.com>
Message-ID: <DF5E4E14-2338-4871-A73E-2291537069B9@ipeva.fr>

Sean,

can you show us the same packets, but for an incoming call (that works, if I understood you correctly) ?

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

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

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




Le 24/05/2011 ? 01:53, Sean Eichhorn a ?crit :

> Yeah, they?re negotiating the same codec on both sides.
> The problem is the way Cisco handles the negotiation.  Here?s an example of what I see in Freeswitch:
>  
> Received from CALLED :
>  
>   SIP/2.0 200 OK
>    Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bKXZZ18p8tD03cS
>    From: "pending" <sip:xxxxxxxx02 at A.B.C.D>;tag=NH1HrNe744HSB
>    To: <sip:sean at E.F.G.H:5060>;tag=622555CC-408
>    Date: Mon, 23 May 2011 23:50:56 GMT
>    Call-ID: 69258f8c-0038-122f-19ba-000c29c18d38
>    CSeq: 12760849 INVITE
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
>    Allow-Events: telephone-event
>    Contact: <sip:xxxxxxxx75 at E.F.G.H:5060>
>    Supported: replaces
>    Supported: sdp-anat
>    Server: Cisco-SIPGateway/IOS-12.x
>    Content-Type: application/sdp
>    Content-Disposition: session;handling=required
>    Content-Length: 311
>  
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 4080 7999 IN IP4 E.F.G.H
>    s=SIP Call
>    c=IN IP4 E.F.G.H
>    t=0 0
>    m=audio 17122 RTP/AVP 0 19 101 100
>    c=IN IP4 E.F.G.H
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:19 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:100 X-NSE/8000
>    a=fmtp:100 192-194
>  
> Sent to CALLING :
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP I.J.K.L:5060;branch=z9hG4bK4C2B6C64
>    From: <sip:xxxxxxxx02@ I.J.K.L >;tag=9020153C-21B3
>    To: <sip:xxxxxxxx75 at A.B.C.D>;tag=m87rptX37UU6F
>    Call-ID: AFADEF2E-84CC11E0-B1D4F23A-27804614@ I.J.K.L
>    CSeq: 101 INVITE
>    Contact: <sip:xxxxxxxx75@ A.B.C.D:5060;transport=udp>
>    User-Agent: GCI
>    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
>    Min-SE: 1800
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 261
>    Remote-Party-ID: "Outbound Call" <sip:sean@ A.B.C.D >;party=calling;privacy=off;screen=no
>  
>    v=0
>    o=FreeSWITCH 1995227603 1995227604 IN IP4 A.B.C.D
>    s=FreeSWITCH
>    c=IN IP4 A.B.C.D
>    t=0 0
>    m=audio 24636 RTP/AVP 0 19 101
>    c=IN IP4 A.B.C.D
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:19 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>  
> This is not the case when the call direction is reversed.  Like I said, the only difference appears to be that one direction is responding with a 183 while this one is responding with a 200.
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Tue May 24 09:47:18 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 24 May 2011 07:47:18 +0200
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <02e001cc19ad$d3d26780$7b773680$@blie-ent.com>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>
	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>
	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>
Message-ID: <59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>

Rob,

following Yehavi's comment, could you downgrade your GIT version to a previous version, like one month old ?

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

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

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




Le 24/05/2011 ? 02:59, Rob Morin a ?crit :

> David,
>  
> It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and Cisco/Linksys SPA 2102. Same story in both cases.
>  
> I had my carrier help me troubleshoot and we can see the SIP request from the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but the response from the A leg isn?t forwarded back to the B leg. I can produce traces that show this.
>  
> When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t respond or forward it. So the connection dies.  I have every reason to believe that I?d be able to send the fax if FS would respond with a 488 indicating T.38 is not acceptable. But instead, all I get is silence. Again, I can produce traces that demonstrate this.
>  
> There?s another similar email  trail out there, http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821994.html, but I?m not certain that they?re related.
>  
> Thank you for your help!
> Rob
>  
> From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
> Sent: Monday, May 23, 2011 6:11 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>  
> Rob,
>  
> perhaps you should not consider T38 is 100% interoperable.
> You may tell us what ATA is that, because some of them are nice piece of junk.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
>  
> Le 23/05/2011 ? 20:51, Rob Morin a ?crit :
> 
> 
> I?m having problems faxing if the B leg offers T.38.  Several scenarios, but always the same result.
>  
> First, the architecture
>  
> FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another carrier (PSTN) and the fax machine)
>  
> On my ATA, T.38 is enabled (Auto Detect).
>  
> If the destination supports T.38, the Carrier will offer it to Freeswitch. If the Destination doesn?t support T.38, the B leg doesn?t offer it and the faxes go through.
>  
> I?ve set Freeswitch as:
>       <action application="set" data="fax_enable_t38=true"/>
>       <action application="export" data="t38_passthru=true"/> 
> When that?s the case, and the B leg offers T.38, Freeswitch passes through the offer and the ATA responds, 3 times. But the response never gets sent to the B leg, so it terminates the call after about 30 seconds.
>  
> When I set Freeswitch as:
>       <action application="set" data="fax_enable_t38=false"/>
> The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t pass the offer through and the B leg terminates the call after about 30 seconds.
>  
> I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d 2011-05-17 22-51-47 -0500) ).
>  
> I?ve had no success attempting to configure FS as a gateway. Frankly, T.38 ?passthrough? has had the most success, mostly when the other ends aren?t T.38 capable and it stays in G711u.
>  
> Is there something else I need to do to enable T.38 in passthrough mode? 
>  
> Thank you,
> Rob Morin
>  
> PS ? I can provide tcpdumps of this, or whatever else is necessary.
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Tue May 24 09:49:12 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 24 May 2011 07:49:12 +0200
Subject: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
In-Reply-To: <BANLkTik5zdomwsTWHGZs-Lpochth+ptkmQ@mail.gmail.com>
References: <BANLkTin7a61Nw-L6qRhGbwYanHO9LGVDGg@mail.gmail.com>
	<CA013499.2C1FE%mike.taylor@staff.snap.net.nz>
	<BANLkTik5zdomwsTWHGZs-Lpochth+ptkmQ@mail.gmail.com>
Message-ID: <8E8B36E9-30A4-4F2B-99D0-619A0432BCE6@ipeva.fr>

You should consider Patton SN4961.
I am no shareholder of Patton, but those boxes work so great and are quite cheaper than the competition, that it is hard to imagine why anyone would buy something else...

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

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

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




Le 24/05/2011 ? 06:01, Yehavi Bourvine a ?crit :

> Hello,
>  
>   I had for a long time a Cisco SIP-E1(with Q.sig) gateway which caused me much grief with DTMF; it works partially OK with INFO but not with RFC-2833. I filed a support call with Cisco, as its debug shows that it receives and detects the RFC2833 but nothing is sent to the E1 side.
>  
> The last conslusion of Cisco's support is that I have to enable transcoding but they coouldn't find how to enable it in my configuration. I now have an Audioodes Mediant-1000 on loan which does RFC-2833 without any problems...
>  
> The only problem I've found so far with the AudioCodes is name passing via E1/Q.sig at the ringing phase (we use it to connect our Freeswitch to our Nortel PbX).
>  
>                         Regards, __Yehavi:
>  
> 
> 
> 2011/5/24 Mike Taylor <mtaylor at employees.org>
> Good point, thanks,
> 
> But, RFC2833 I've found to be unreliable, broken, patchy, scratchy?. 
> 
> I've got it set to RFC2833 now, and it just ain't working.
> 
> Will get more traces, trial and error until I stumble across a good combo, or bug(s)?
> 
> Thanks again,
> Mike
> 
> From: Ognjen Seslija <oseslija at gmail.com>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 23 May 2011 20:46:09 +0200
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
> 
> Why use INFO when it's only method I know Cisco won't speak (2833, kpml, sub/notify)?
> 
> Here's the CME snippet for FS trunk I have. FS sends 2833 tones to CME btw.
> 
> dial-peer voice 100 voip
>  description ** TO FS **
>  destination-pattern .T
>  b2bua
>  session protocol sipv2
>  session target sip-server
>  session transport udp
>  dtmf-relay rtp-nte
>  codec g711alaw
>  no vad
> 
> 
> 
> 
> On Mon, May 23, 2011 at 4:07 AM, Mike Taylor <mtaylor at employees.org> wrote:
> Hi,
> 
> This is driving me nuts. 
> I have a Cisco CME (a few actually) where I cannot get DTMF to work.
> 
> Does someone happen to have the magic recipe of SIP/Dialplan profiles and Cisco config to make DTMF <relay> work?
> 
> I spent several very very painful days trying to get RFC2833 to work, using FS to try and generate the tones itself. I ended up with all sorts of codec mismatches, delay,distortion and some really mangled DTMF tones?
> 
> So, now I'm back to;
> 
> PSTN----CiscoGW----<external>----FS----<internal>----<CUST PABX>---Phone
> 
> Where <CUST PABX> is 3 CMEs that won't talk DTMF
> 30+ other vendor's gateway's that WILL do DTMF?
> 
> Context External and internal are both set to dtmf-type info (or dtmf_type info) 
> 
> DTMF works FROM the Cisco GW towards our network, doesn't 
> 
> 
> So, does anyone have a magic solution?
>  Because Cisco TAC say they see DTMF being sent (as rtp-nte), when I need INFO or in-band.
> 
> 
> 
> Thanks,
> 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
> 
> 
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dome at tel.co.th  Tue May 24 10:37:35 2011
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 24 May 2011 13:37:35 +0700
Subject: [Freeswitch-users] My Freeswitch status
Message-ID: <BANLkTim3XOUvTdy5Xu4=T7oPf6aTuX5GLw@mail.gmail.com>

10 million calls :)

freeswitch at internal> status
UP 0 years, 98 days, 16 hours, 45 minutes, 30 seconds, 743 milliseconds, 805
microseconds
10131867 session(s) since startup
123 session(s) 0/300
5000 session(s) max
min idle cpu 0.00/70.00



Thank FS

Dome C.
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From mtaylor at employees.org  Tue May 24 11:02:27 2011
From: mtaylor at employees.org (Mike Taylor)
Date: Tue, 24 May 2011 19:02:27 +1200
Subject: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
In-Reply-To: <8E8B36E9-30A4-4F2B-99D0-619A0432BCE6@ipeva.fr>
References: <BANLkTin7a61Nw-L6qRhGbwYanHO9LGVDGg@mail.gmail.com>	<CA013499.2C1FE%mike.taylor@staff.snap.net.nz>	<BANLkTik5zdomwsTWHGZs-Lpochth+ptkmQ@mail.gmail.com>
	<8E8B36E9-30A4-4F2B-99D0-619A0432BCE6@ipeva.fr>
Message-ID: <4DDB5803.9080903@employees.org>

Completely agree, we might be putting an E1 card in the Cisco and 
front-ending it with a Patton yet...

I have an SN4961 30 Channel license in stock.

An update on my testing; At the moment, I have the Cisco set to rfc2833, 
and Wireshark (filter on 'rtpenvents') clearly shows the DTMF being 
originated from the Cisco.


I should also point out that it _looks_ like the PSTN<-->SIP gateway is 
doing DTMF in-band, which is passing transparently through FS, from 
'external' to 'internal' i.e. DTMF works in-bound no matter which 
direction the call is placed.

Next step: capture a trace of a 'good' vs 'bad' call between FS and the 
PSTN GW and verify which type of DTMF  the 'good' call is actually using.
     This will also let me know if FS is converting RFC2833 to in-band 
correctly towards the PSTN.

Also, as Yehavi pointed out, making the Cisco transcode might fix it 
(again, the trace between FS and PSTN GW might show something).

Ahh, just looking at the Wiki I have check to make sure I have this correct;
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate

Thanks everyone,
Mike

On 24/05/2011 5:49 p.m., David Ponzone wrote:
> You should consider Patton SN4961.
> I am no shareholder of Patton, but those boxes work so great and are 
> quite cheaper than the competition, that it is hard to imagine why 
> anyone would buy something else...
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 24/05/2011 ? 06:01, Yehavi Bourvine a ?crit :
>
>> Hello,
>>   I had for a long time a Cisco SIP-E1(with Q.sig) gateway which 
>> caused me much grief with DTMF; it works partially OK with INFO but 
>> not with RFC-2833. I filed a support call with Cisco, as its debug 
>> shows that it receives and detects the RFC2833 but nothing is sent to 
>> the E1 side.
>> The last conslusion of Cisco's support is that I have to enable 
>> transcoding but they coouldn't find how to enable it in my 
>> configuration. I now have an Audioodes Mediant-1000 on loan which 
>> does RFC-2833 without any problems...
>> The only problem I've found so far with the AudioCodes is name 
>> passing via E1/Q.sig at the ringing phase (we use it to connect our 
>> Freeswitch to our Nortel PbX).
>>                         Regards, __Yehavi:
>>
>>
>> 2011/5/24 Mike Taylor <mtaylor at employees.org 
>> <mailto:mtaylor at employees.org>>
>>
>>     Good point, thanks,
>>
>>     But, RFC2833 I've found to be unreliable, broken, patchy, scratchy?.
>>
>>     I've got it set to RFC2833 now, and it just ain't working.
>>
>>     Will get more traces, trial and error until I stumble across a
>>     good combo, or bug(s)?
>>
>>     Thanks again,
>>     Mike
>>
>>     From: Ognjen Seslija <oseslija at gmail.com <mailto:oseslija at gmail.com>>
>>     Reply-To: FreeSWITCH Users Help
>>     <freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>>
>>     Date: Mon, 23 May 2011 20:46:09 +0200
>>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>>
>>     Subject: Re: [Freeswitch-users] FS - DTMF and Cisco CME/UCE
>>
>>     Why use INFO when it's only method I know Cisco won't speak
>>     (2833, kpml, sub/notify)?
>>
>>     Here's the CME snippet for FS trunk I have. FS sends 2833 tones
>>     to CME btw.
>>
>>     dial-peer voice 100 voip
>>      description ** TO FS **
>>      destination-pattern .T
>>      b2bua
>>      session protocol sipv2
>>      session target sip-server
>>      session transport udp
>>      dtmf-relay rtp-nte
>>      codec g711alaw
>>      no vad
>>
>>
>>
>>
>>     On Mon, May 23, 2011 at 4:07 AM, Mike Taylor
>>     <mtaylor at employees.org <mailto:mtaylor at employees.org>> wrote:
>>
>>         Hi,
>>
>>         This is driving me nuts.
>>         I have a Cisco CME (a few actually) where I cannot get DTMF
>>         to work.
>>
>>         Does someone happen to have the magic recipe of SIP/Dialplan
>>         profiles and Cisco config to make DTMF <relay> work?
>>
>>         I spent several very very painful days trying to get RFC2833
>>         to work, using FS to try and generate the tones itself. I
>>         ended up with all sorts of codec mismatches, delay,distortion
>>         and some really mangled DTMF tones?
>>
>>         So, now I'm back to;
>>
>>         PSTN----CiscoGW----<external>----FS----<internal>----<CUST
>>         PABX>---Phone
>>
>>         Where <CUST PABX> is 3 CMEs that won't talk DTMF
>>         30+ other vendor's gateway's that WILL do DTMF?
>>
>>         Context External and internal are both set to dtmf-type info
>>         (or dtmf_type info)
>>
>>         DTMF works FROM the Cisco GW towards our network, doesn't
>>
>>
>>         So, does anyone have a magic solution?
>>          Because Cisco TAC say they see DTMF being sent (as rtp-nte),
>>         when I need INFO or in-band.
>>
>>
>>
>>         Thanks,
>>         Mike
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From a.afzali2003 at gmail.com  Tue May 24 11:44:18 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 24 May 2011 12:14:18 +0430
Subject: [Freeswitch-users] My Freeswitch status
In-Reply-To: <BANLkTim3XOUvTdy5Xu4=T7oPf6aTuX5GLw@mail.gmail.com>
References: <BANLkTim3XOUvTdy5Xu4=T7oPf6aTuX5GLw@mail.gmail.com>
Message-ID: <BANLkTinaw0qH7JarmBOu7cTz5C1F+atYeg@mail.gmail.com>

On Tue, May 24, 2011 at 11:07 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> 10 million calls :)
>
> freeswitch at internal> status
> UP 0 years, 98 days, 16 hours, 45 minutes, 30 seconds, 743 milliseconds,
> 805 microseconds
> 10131867 session(s) since startup
> 123 session(s) 0/300
> 5000 session(s) max
> min idle cpu 0.00/70.00
>
>
>
> Thank FS
>
> Dome C.
>
> Dome C. :

Congrats, Its hardware?
-- 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
>
>
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From brad at tritelcomm.com  Tue May 24 12:55:35 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Tue, 24 May 2011 01:55:35 -0700
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
 record to Shoutcast
In-Reply-To: <20110520183037.83314gw215dgz5og@psychicawakeningschool.com>
References: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
	<1305903105560-6386372.post@n2.nabble.com>
	<20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>
	<25a801cc1701$6740d510$35c27f30$@com>
	<20110520183037.83314gw215dgz5og@psychicawakeningschool.com>
Message-ID: <BANLkTikvOPALgLhTEvNdEnafAgD5fmTimw@mail.gmail.com>

"conference foo-10.10.10.1 record shout://foo.bar.com:8000/foobar.mp3"
That, in conference, will record to a uri !
Thanks to Tayeb.

On Fri, May 20, 2011 at 3:30 PM, Rev Michael Carbone <
revmichael at bethelightchapel.com> wrote:

> Been trying to get to the IRC channel
> If anyone can help off list please e-mail me at revmichael at
> bethelightchapel dot com
>
> thank you
>
>
> Quoting Robert Huddleston <rhuddleston at gmail.com>:
>
> > If you feel that you are not getting the response you expect - try
> jumping
> > over to the IRC channel and asking there.
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rev
> > Michael Carbone
> > Sent: Friday, May 20, 2011 11:05 AM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] Setup help Conference, 3 way call, FIFO,
> > MOH, record to Shoutcast
> >
> > Like I said I'm new to all this, I know where to find the extensions,
> > but I'm not seeing extension 3000 anywhere, am I missing something?
> > I'm using fusionpbx as well.
> >
> > thank you for your help
> >
> > Michael
> >
> >>
> >> Rev Michael Carbone wrote:
> >>> Broken down:
> >>>
> >>> 2 of us on a call
> >>> broadcasting to shoutcast
> >>> other people call in and are put in cue with conversation as MOH
> >>> bring people on air one at a time while others are still muted
> >>>
> >>> I'm sure this can be done, but how?
> >> FS comes with a built-in un-muted conference room by default on
> extension
> >> 3000. If you reconfigure with all muted by default, then you can select
> >> which participant to un-mute while the rests on the conference room are
> >> still muted and be able to listen to the conversation.
> >>
> >> -----
> >> 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/Setup-help-Conference-3-way-ca
> > ll-FIFO-MOH-record-to-Shoutcast-tp6386123p6386372.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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 24 13:33:28 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 09:33:28 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
Message-ID: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>

Hello List , Good Morning.

In my fight to get ESL & Python & Freeswitch all to behave properly i
noticed a possible BUG ( need the veterans to confirm).

Scenario :

Originate & ValetPark () + MOH OR  Just Park() , Both show the same problem.
Codec G729

I Make another call:
originate & park()
codec G729

Now Python ESL Inbound Or FS Console :

uuid_bridge uuid1 uuid2

+OK uuid

Now one of the two things happen :

1) One the call gets connected hurray :) , Audio Perfect  , etc.

2) The call gets dropped :(

In both cases uuid_bridge reports +OK even in the case the call is dropped.
Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
could this be related to the use of G729 ?

I have lots of available licenses.

Regards
Ant?nio Teixeira
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From eagle.antonio at gmail.com  Tue May 24 13:53:52 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 09:53:52 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <A32AFBCD54D14437A199337660F137F4@e1705>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
Message-ID: <BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>

Sorry To Revive A Old Thread.

But this could be interesting for some :

Problem :
I started to get some signaling problems with some calls :

The SIP trunk was passing me 180 Ringing but the local station was not
ringing , after a wire shark sniff i noticed that packets were coming out of
order.
Before calling our ISP , i used wire shark on the FS Public Interface , and
the packets were coming in the correct order but FS would take up to 5
seconds to route them into the private network and would mix the signaling.
So debug started , there was lots of CPU + Memory , IO was not a problem ,
network was also fine with 0 packet loss .

So i turned into FS , i noticed this happened on a specific call volume ,
after reading and reading i noticed a line in the wiki stating that libsofia
profile is single threaded .

Solution :

Start Another Profile on a different port Ex:  5081 and route your calls
trough that profile i noticed an improvement right away.

originate sofia/2_profile/XXXXXX at mygatewayip

Now i'm using profiles to load balance calls inside FS.

P.S - I know this can be pretty obvious for some but some of us are still
new to FS so i hope this can help someone out :D.
P.S2 - Also if a veteran thinks this is the wrong approach please respond so
we can all improve.


Regards
Antonio teixeira



2011/4/27 Madovsky <infos at madovsky.org>

> for info I'm using the old Fedora10 64bits
> and everything is working fine with fq at 1000hz
>
>
> ----- Original Message -----
> From: "Ariel Monaco" <arielmonaco at flylabs.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, April 26, 2011 8:24 PM
> Subject: Re: [Freeswitch-users] Tuning Up Freeswitch
>
>
> We had high CPU utilization peaks in the past, which lead to some audio
> issues (clipping). We were using debian at that time, which was a
> customer-side requirement.
>
> I'm not a kernel guru but I remember this had something to do with kernel
> timer cycles and the issue was address by adding "divider=10" or
> "divider=100" as a kernel's boot loader option.
>
> My 2 cents,
> Ariel
>
> On Apr 21, 2011, at 13:24 , Antonio Teixeira wrote:
>
> > Hello List.
> >
> > I'm currently integrating an IVR in python together with freeswitch using
> > mod_python and ESL and my life has been well until ...
> > The flow of calls went over 80 simultaneous calls.
> > Now freeswitch starts sending packets with huge delays ( even when
> > establishing the call , mainly the 200 ) and firing up the IVR with tons
> > of delay up to 20 seconds.
> >
> > So i searched the wiki forums and mailing list:
> >
> > Put freeswitch on a diet , trimmed modules.conf
> > Played with the ulimit stuff.
> > Played with the IVRS to reduce load to a minimum and i was able to
> squeeze
> > more 5 calls of performance.
> >
> > The problem is :
> >
> > Top shows
> > top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76,
> > 1.78
> > Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
> > Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,
> > 0.0%st
> > Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
> > Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
> >
> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
> > 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
> > freeswitch
> >
> > Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
> >
> >
> > Some basic System Info :
> > Debian 6.0 ( i heard the timming module is affected by Debian , but if
> the
> > CPU % gets lower than 95% everything will be more stable)
> > Python 2.5
> >
> > 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
> > 8 GB of Ram
> >
> > as you can see 94 % of the "Cpu Power" is sleeping :\
> >
> >
> > It appears freeswitch is only capable of using let's say "one cpu"/thread
> > ??
> > Do you guys recommend simply starting more instances or redoing the IVR
> > stuff.
> >
> >
> > Hope you guys can help me out.
> >
> > Thanks
> > Ant?nio Teixeira
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> Ariel Monaco ? Systems Engineer
> Flylabs - Open Source Telecommunications and  IT Consultants
>
> Address: Potos? 4456 C1199ACP - Buenos Aires - Argentina
> Web: http://flylabs.com
> E-Mail: arielmonaco at flylabs.com
> Tel. +54 (11) 4982-2689, +1 (315) 688-7333
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sid.kshatriya at gmail.com  Tue May 24 14:29:25 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Tue, 24 May 2011 15:59:25 +0530
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
Message-ID: <BANLkTimxzhdyq3Jp=wOGviF=JfoW=BPWRg@mail.gmail.com>

Thanks for your reply, Michael. I have tried doing a start dtmf but then
I've noticed:

1.  My skype key presses get unreliable...
2. The dtmf detection is patchy with inband -- works sometimes and doesn't
work sometimes :-)

Any settings to keep in mind apart just adding a start dtmf ?

Thanks,

Sidharth

On Mon, May 23, 2011 at 8:54 PM, Michael Collins <msc at freeswitch.org> wrote:

> Sidharth,
>
> A mobile phone will always send DTMFs in-band, so you need to be ready for
> that scenario. I recommend you add this to your dialplan for inbound calls:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
>
> Test various scenarios and make sure they work - call from Skype, from
> mobile phone, from a land line, etc. Let us know what happens.
>
> -MC
>
>  On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya <
> sid.kshatriya at gmail.com> wrote:
>
>>  Dear Friends,
>>
>> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
>> list). While I'm generally happy with their service there seems to be one
>> fatal problem: My IVR does not recognize DTMF!
>>
>> I have set <param name="dtmf-type" value="rfc2833"/> in both
>> sip_profile/internal.xml and sip_profile/external.xml
>>
>> The symptom of the problem is that making a call via skype will *always*make the IVR recognize the DTMF while using something like a mobile phone
>> *almost always* won't!
>>
>> I've tried in-band detection too. I'm making international calls into my
>> IVR and the reliability of the in-band detection is not so good, possibly
>> because of the quality of the call.
>>
>> Can someone please help me?
>>
>> Thanks,
>>
>> Sidharth
>>
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From anton.vazir at gmail.com  Tue May 24 14:57:42 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 24 May 2011 15:57:42 +0500
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
Message-ID: <BANLkTikDNOXr1+he8KgT2joKhKwdMy9VUA@mail.gmail.com>

Did you put any load to get several CPS? Or just out of several
sequential calls, some of them bridges, some not?

2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>:
> Hello List , Good Morning.
>
> In my fight to get ESL & Python & Freeswitch all to behave properly i
> noticed a possible BUG ( need the veterans to confirm).
>
> Scenario :
>
> Originate & ValetPark () + MOH OR? Just Park() , Both show the same problem.
> Codec G729
>
> I Make another call:
> originate & park()
> codec G729
>
> Now Python ESL Inbound Or FS Console :
>
> uuid_bridge uuid1 uuid2
>
> +OK uuid
>
> Now one of the two things happen :
>
> 1) One the call gets connected hurray :) , Audio Perfect? , etc.
>
> 2) The call gets dropped :(
>
> In both cases uuid_bridge reports +OK even in the case the call is dropped.
> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> SYN/ASYNC Problems)? I still get sometimes ( not always) a dropped call
> could this be related to the use of G729 ?
>
> I have lots of available licenses.
>
> Regards
> Ant?nio Teixeira
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Tue May 24 15:04:21 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 11:04:21 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTikDNOXr1+he8KgT2joKhKwdMy9VUA@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTikDNOXr1+he8KgT2joKhKwdMy9VUA@mail.gmail.com>
Message-ID: <BANLkTik8DNOVeyo1ceTTcC_tDLL9Hu3Asw@mail.gmail.com>

Hello Anton

No , this was made on our test environment so only one call at FS.

Manually started some would bridge and some not using the same endpoints.




2011/5/24 Anton VG <anton.vazir at gmail.com>

> Did you put any load to get several CPS? Or just out of several
> sequential calls, some of them bridges, some not?
>
> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>:
> > Hello List , Good Morning.
> >
> > In my fight to get ESL & Python & Freeswitch all to behave properly i
> > noticed a possible BUG ( need the veterans to confirm).
> >
> > Scenario :
> >
> > Originate & ValetPark () + MOH OR  Just Park() , Both show the same
> problem.
> > Codec G729
> >
> > I Make another call:
> > originate & park()
> > codec G729
> >
> > Now Python ESL Inbound Or FS Console :
> >
> > uuid_bridge uuid1 uuid2
> >
> > +OK uuid
> >
> > Now one of the two things happen :
> >
> > 1) One the call gets connected hurray :) , Audio Perfect  , etc.
> >
> > 2) The call gets dropped :(
> >
> > In both cases uuid_bridge reports +OK even in the case the call is
> dropped.
> > Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> > SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
> > could this be related to the use of G729 ?
> >
> > I have lots of available licenses.
> >
> > Regards
> > Ant?nio Teixeira
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anton.vazir at gmail.com  Tue May 24 15:07:42 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 24 May 2011 16:07:42 +0500
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
Message-ID: <BANLkTikc8wNmiZHk5aR2PQy6GOX3A4-bqw@mail.gmail.com>

Sounds kind of strange, that means:
we should limit the simultaneous call per profile
what about users, by default they do to 'Internal" profile, so to
avoid bottleneck, several profiles should be used for serving large
amount of users?

2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>:
> Sorry To Revive A Old Thread.
>
> But this could be interesting for some :
>
> Problem :
> I started to get some signaling problems with some calls :
>
> The SIP trunk was passing me 180 Ringing but the local station was not
> ringing , after a wire shark sniff i noticed that packets were coming out of
> order.
> Before calling our ISP , i used wire shark on the FS Public Interface , and
> the packets were coming in the correct order but FS would take up to 5
> seconds to route them into the private network and would mix the signaling.
> So debug started , there was lots of CPU + Memory , IO was not a problem ,
> network was also fine with 0 packet loss .
>
> So i turned into FS , i noticed this happened on a specific call volume ,
> after reading and reading i noticed a line in the wiki stating that libsofia
> profile is single threaded .
>
> Solution :
>
> Start Another Profile on a different port Ex:? 5081 and route your calls
> trough that profile i noticed an improvement right away.
>
> originate sofia/2_profile/XXXXXX at mygatewayip
>
> Now i'm using profiles to load balance calls inside FS.
>
> P.S - I know this can be pretty obvious for some but some of us are still
> new to FS so i hope this can help someone out :D.
> P.S2 - Also if a veteran thinks this is the wrong approach please respond so
> we can all improve.
>
>
> Regards
> Antonio teixeira
>
>
>
> 2011/4/27 Madovsky <infos at madovsky.org>
>>
>> for info I'm using the old Fedora10 64bits
>> and everything is working fine with fq at 1000hz
>>
>>
>> ----- Original Message -----
>> From: "Ariel Monaco" <arielmonaco at flylabs.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, April 26, 2011 8:24 PM
>> Subject: Re: [Freeswitch-users] Tuning Up Freeswitch
>>
>>
>> We had high CPU utilization peaks in the past, which lead to some audio
>> issues (clipping). We were using debian at that time, which was a
>> customer-side requirement.
>>
>> I'm not a kernel guru but I remember this had something to do with kernel
>> timer cycles and the issue was address by adding "divider=10" or
>> "divider=100" as a kernel's boot loader option.
>>
>> My 2 cents,
>> Ariel
>>
>> On Apr 21, 2011, at 13:24 , Antonio Teixeira wrote:
>>
>> > Hello List.
>> >
>> > I'm currently integrating an IVR in python together with freeswitch
>> > using
>> > mod_python and ESL and my life has been well until ...
>> > The flow of calls went over 80 simultaneous calls.
>> > Now freeswitch starts sending packets with huge delays ( even when
>> > establishing the call , mainly the 200 ) and firing up the IVR with tons
>> > of delay up to 20 seconds.
>> >
>> > So i searched the wiki forums and mailing list:
>> >
>> > Put freeswitch on a diet , trimmed modules.conf
>> > Played with the ulimit stuff.
>> > Played with the IVRS to reduce load to a minimum and i was able to
>> > squeeze
>> > more 5 calls of performance.
>> >
>> > The problem is :
>> >
>> > Top shows
>> > top - 16:14:33 up 35 days, ?8:15, ?3 users, ?load average: 1.92, 1.76,
>> > 1.78
>> > Tasks: 133 total, ? 1 running, 132 sleeping, ? 0 stopped, ? 0 zombie
>> > Cpu(s): ?1.4%us, ?3.3%sy, ?0.0%ni, 94.6%id, ?0.0%wa, ?0.3%hi, ?0.5%si,
>> > 0.0%st
>> > Mem: ? 8193336k total, ?1639156k used, ?6554180k free, ? 177208k buffers
>> > Swap: 19534904k total, ? ? ? ?0k used, 19534904k free, ?1062272k cached
>> >
>> > ? PID USER ? ? ?PR ?NI ?VIRT ?RES ?SHR S %CPU %MEM ? ?TIME+ ?COMMAND
>> > 31361 yadayada ? ? ?20 ? 0 ?716m 164m 9628 S ? 73 ?2.1 155:17.85
>> > freeswitch
>> >
>> > Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>> >
>> >
>> > Some basic System Info :
>> > Debian 6.0 ( i heard the timming module is affected by Debian , but if
>> > the
>> > CPU % gets lower than 95% everything will be more stable)
>> > Python 2.5
>> >
>> > 2 x Intel(R) Xeon(R) CPU ? ? ? ? ? E5506 ?@ 2.13GHz
>> > 8 GB of Ram
>> >
>> > as you can see 94 % of the "Cpu Power" is sleeping :\
>> >
>> >
>> > It appears freeswitch is only capable of using let's say "one
>> > cpu"/thread
>> > ??
>> > Do you guys recommend simply starting more instances or redoing the IVR
>> > stuff.
>> >
>> >
>> > Hope you guys can help me out.
>> >
>> > Thanks
>> > Ant?nio Teixeira
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> Ariel Monaco ? Systems Engineer
>> Flylabs - Open Source Telecommunications and ?IT Consultants
>>
>> Address: Potos? 4456 C1199ACP - Buenos Aires - Argentina
>> Web: http://flylabs.com
>> E-Mail: arielmonaco at flylabs.com
>> Tel. +54 (11) 4982-2689, +1 (315) 688-7333
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> This message has been scanned for viruses and
>> dangerous content by MailScanner, and is
>> believed to be clean.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From juanito1982 at gmail.com  Tue May 24 15:28:54 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Tue, 24 May 2011 13:28:54 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
Message-ID: <BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>

How do you make that load balancing between profiles?

2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>

> Sorry To Revive A Old Thread.
>
> But this could be interesting for some :
>
> Problem :
> I started to get some signaling problems with some calls :
>
> The SIP trunk was passing me 180 Ringing but the local station was not
> ringing , after a wire shark sniff i noticed that packets were coming out of
> order.
> Before calling our ISP , i used wire shark on the FS Public Interface , and
> the packets were coming in the correct order but FS would take up to 5
> seconds to route them into the private network and would mix the signaling.
> So debug started , there was lots of CPU + Memory , IO was not a problem ,
> network was also fine with 0 packet loss .
>
> So i turned into FS , i noticed this happened on a specific call volume ,
> after reading and reading i noticed a line in the wiki stating that libsofia
> profile is single threaded .
>
> Solution :
>
> Start Another Profile on a different port Ex:  5081 and route your calls
> trough that profile i noticed an improvement right away.
>
> originate sofia/2_profile/XXXXXX at mygatewayip
>
> Now i'm using profiles to load balance calls inside FS.
>
> P.S - I know this can be pretty obvious for some but some of us are still
> new to FS so i hope this can help someone out :D.
> P.S2 - Also if a veteran thinks this is the wrong approach please respond
> so we can all improve.
>
>
> Regards
> Antonio teixeira
>
>
>
>
> 2011/4/27 Madovsky <infos at madovsky.org>
>
>> for info I'm using the old Fedora10 64bits
>> and everything is working fine with fq at 1000hz
>>
>>
>> ----- Original Message -----
>> From: "Ariel Monaco" <arielmonaco at flylabs.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, April 26, 2011 8:24 PM
>> Subject: Re: [Freeswitch-users] Tuning Up Freeswitch
>>
>>
>> We had high CPU utilization peaks in the past, which lead to some audio
>> issues (clipping). We were using debian at that time, which was a
>> customer-side requirement.
>>
>> I'm not a kernel guru but I remember this had something to do with kernel
>> timer cycles and the issue was address by adding "divider=10" or
>> "divider=100" as a kernel's boot loader option.
>>
>> My 2 cents,
>> Ariel
>>
>> On Apr 21, 2011, at 13:24 , Antonio Teixeira wrote:
>>
>> > Hello List.
>> >
>> > I'm currently integrating an IVR in python together with freeswitch
>> using
>> > mod_python and ESL and my life has been well until ...
>> > The flow of calls went over 80 simultaneous calls.
>> > Now freeswitch starts sending packets with huge delays ( even when
>> > establishing the call , mainly the 200 ) and firing up the IVR with tons
>> > of delay up to 20 seconds.
>> >
>> > So i searched the wiki forums and mailing list:
>> >
>> > Put freeswitch on a diet , trimmed modules.conf
>> > Played with the ulimit stuff.
>> > Played with the IVRS to reduce load to a minimum and i was able to
>> squeeze
>> > more 5 calls of performance.
>> >
>> > The problem is :
>> >
>> > Top shows
>> > top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76,
>> > 1.78
>> > Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
>> > Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,
>> > 0.0%st
>> > Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
>> > Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
>> >
>> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>> > 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
>> > freeswitch
>> >
>> > Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>> >
>> >
>> > Some basic System Info :
>> > Debian 6.0 ( i heard the timming module is affected by Debian , but if
>> the
>> > CPU % gets lower than 95% everything will be more stable)
>> > Python 2.5
>> >
>> > 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>> > 8 GB of Ram
>> >
>> > as you can see 94 % of the "Cpu Power" is sleeping :\
>> >
>> >
>> > It appears freeswitch is only capable of using let's say "one
>> cpu"/thread
>> > ??
>> > Do you guys recommend simply starting more instances or redoing the IVR
>> > stuff.
>> >
>> >
>> > Hope you guys can help me out.
>> >
>> > Thanks
>> > Ant?nio Teixeira
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> Ariel Monaco ? Systems Engineer
>> Flylabs - Open Source Telecommunications and  IT Consultants
>>
>> Address: Potos? 4456 C1199ACP - Buenos Aires - Argentina
>> Web: http://flylabs.com
>> E-Mail: arielmonaco at flylabs.com
>> Tel. +54 (11) 4982-2689, +1 (315) 688-7333
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> This message has been scanned for viruses and
>> dangerous content by MailScanner, and is
>> believed to be clean.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue May 24 15:43:20 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 24 May 2011 13:43:20 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
	<BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
Message-ID: <BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>

the only real improvment was to move your DB to tmpfs ... the rest ( if
64bit) makes really small difference from distro to distro..


also set your ulimits accordingly!

cheers.



2011/5/24 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>

> How do you make that load balancing between profiles?
>
> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>
>
>> Sorry To Revive A Old Thread.
>>
>> But this could be interesting for some :
>>
>> Problem :
>> I started to get some signaling problems with some calls :
>>
>> The SIP trunk was passing me 180 Ringing but the local station was not
>> ringing , after a wire shark sniff i noticed that packets were coming out of
>> order.
>> Before calling our ISP , i used wire shark on the FS Public Interface ,
>> and the packets were coming in the correct order but FS would take up to 5
>> seconds to route them into the private network and would mix the signaling.
>> So debug started , there was lots of CPU + Memory , IO was not a problem ,
>> network was also fine with 0 packet loss .
>>
>> So i turned into FS , i noticed this happened on a specific call volume ,
>> after reading and reading i noticed a line in the wiki stating that libsofia
>> profile is single threaded .
>>
>> Solution :
>>
>> Start Another Profile on a different port Ex:  5081 and route your calls
>> trough that profile i noticed an improvement right away.
>>
>> originate sofia/2_profile/XXXXXX at mygatewayip
>>
>> Now i'm using profiles to load balance calls inside FS.
>>
>> P.S - I know this can be pretty obvious for some but some of us are still
>> new to FS so i hope this can help someone out :D.
>> P.S2 - Also if a veteran thinks this is the wrong approach please respond
>> so we can all improve.
>>
>>
>> Regards
>> Antonio teixeira
>>
>>
>>
>>
>> 2011/4/27 Madovsky <infos at madovsky.org>
>>
>>> for info I'm using the old Fedora10 64bits
>>> and everything is working fine with fq at 1000hz
>>>
>>>
>>> ----- Original Message -----
>>> From: "Ariel Monaco" <arielmonaco at flylabs.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, April 26, 2011 8:24 PM
>>> Subject: Re: [Freeswitch-users] Tuning Up Freeswitch
>>>
>>>
>>> We had high CPU utilization peaks in the past, which lead to some audio
>>> issues (clipping). We were using debian at that time, which was a
>>> customer-side requirement.
>>>
>>> I'm not a kernel guru but I remember this had something to do with kernel
>>> timer cycles and the issue was address by adding "divider=10" or
>>> "divider=100" as a kernel's boot loader option.
>>>
>>> My 2 cents,
>>> Ariel
>>>
>>> On Apr 21, 2011, at 13:24 , Antonio Teixeira wrote:
>>>
>>> > Hello List.
>>> >
>>> > I'm currently integrating an IVR in python together with freeswitch
>>> using
>>> > mod_python and ESL and my life has been well until ...
>>> > The flow of calls went over 80 simultaneous calls.
>>> > Now freeswitch starts sending packets with huge delays ( even when
>>> > establishing the call , mainly the 200 ) and firing up the IVR with
>>> tons
>>> > of delay up to 20 seconds.
>>> >
>>> > So i searched the wiki forums and mailing list:
>>> >
>>> > Put freeswitch on a diet , trimmed modules.conf
>>> > Played with the ulimit stuff.
>>> > Played with the IVRS to reduce load to a minimum and i was able to
>>> squeeze
>>> > more 5 calls of performance.
>>> >
>>> > The problem is :
>>> >
>>> > Top shows
>>> > top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76,
>>> > 1.78
>>> > Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
>>> > Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,
>>> > 0.0%st
>>> > Mem:   8193336k total,  1639156k used,  6554180k free,   177208k
>>> buffers
>>> > Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
>>> >
>>> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>>> > 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
>>> > freeswitch
>>> >
>>> > Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>>> >
>>> >
>>> > Some basic System Info :
>>> > Debian 6.0 ( i heard the timming module is affected by Debian , but if
>>> the
>>> > CPU % gets lower than 95% everything will be more stable)
>>> > Python 2.5
>>> >
>>> > 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>>> > 8 GB of Ram
>>> >
>>> > as you can see 94 % of the "Cpu Power" is sleeping :\
>>> >
>>> >
>>> > It appears freeswitch is only capable of using let's say "one
>>> cpu"/thread
>>> > ??
>>> > Do you guys recommend simply starting more instances or redoing the IVR
>>> > stuff.
>>> >
>>> >
>>> > Hope you guys can help me out.
>>> >
>>> > Thanks
>>> > Ant?nio Teixeira
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> Ariel Monaco ? Systems Engineer
>>> Flylabs - Open Source Telecommunications and  IT Consultants
>>>
>>> Address: Potos? 4456 C1199ACP - Buenos Aires - Argentina
>>> Web: http://flylabs.com
>>> E-Mail: arielmonaco at flylabs.com
>>> Tel. +54 (11) 4982-2689, +1 (315) 688-7333
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> This message has been scanned for viruses and
>>> dangerous content by MailScanner, and is
>>> believed to be clean.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anton.vazir at gmail.com  Tue May 24 15:57:46 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 24 May 2011 16:57:46 +0500
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
	<BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
	<BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>
Message-ID: <BANLkTi=VGKi3eiuLz3kMoQ0K6ct5mJrrfA@mail.gmail.com>

do you mean sqlite?

2011/5/24 Tihomir Culjaga <tculjaga at gmail.com>:
> the only real improvment was to move your DB to tmpfs ... the rest ( if
> 64bit) makes really small difference from distro to distro..
>


From boris at tagnet.ru  Tue May 24 16:11:08 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 24 May 2011 18:11:08 +0600
Subject: [Freeswitch-users] DTMF problems
Message-ID: <4DDBA05C.6060204@tagnet.ru>

Hello!

     My network enviroment is: PRI --- Cisco 5350 --- FreeSwitch 
(Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)) --- 
AudioCodes MP-118FXS. There is an IVR on FreeSwitch where calling party 
may press a key for personal client line or wait for a secretary. 
Sometimes (as client said) the call to line 5 is placed to line 3 (as I 
undestand DTMF 5 is recognized as 3). How may I detect and solve the 
problem? Configurations below:
Cisco 5350:
dial-peer voice 4000 voip
  description OUTBOUND-CALLS-FROM-USI
  destination-pattern 50000#.T
  voice-class codec 1
  session protocol sipv2
  session target ipv4:82.193.138.187:5060
  dtmf-relay rtp-nte
  fax-relay ecm disable
  fax rate 9600
  fax protocol t38 ls-redundancy 5 hs-redundancy 0 fallback pass-through 
g711alaw
  no vad

FreeSwitch:
profile:
<param name="dtmf-duration" value="2000"/>
<!-- <param name="dtmf-type" value="rfc2833"/> --> (yes, it is commented)

context:
<extension name="ext_001">
<condition field="destination_number" expression="^(73435230100)$">
<action application="set" data="v_tagnet_ats_dstport=10005"/>
<action application="set" data="v_ats_dstport=10005"/>
<action application="set" data="import=v_tagnet_dstport,v_ats_dstpor
t"/>
<!-- <action application="set" data="import=v_tagnet_ats_dstport"/>
-->
<action application="set" data="hangup_after_bridge=true"/>
<!-- <action application="set" data="continue_on_fail=USER_NOT_REGIS
TERED"/> -->
<action application="set" data="ringback=$${ru-ring}"/>
<action application="javascript" data="${scripts_dir}/common/set_dpe.js 
B ui.tagnet.cc 3 3"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="play_and_get_digits" data="1 1 1 5000 # 
/opt/fs/sounds/ntmjui/welcome.wav silence_stream://250 digits \d* 3000"/>
<action application="execute_extension" data="ext_call_internal"/>
<action application="bridge" data="${sofia_contact(101 at ui.tagnet.cc)}" />
<action application="hangup"/>
</condition>
</extension>


-- 
Regards,
Boris




From jaybinks at gmail.com  Tue May 24 16:19:25 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 24 May 2011 22:19:25 +1000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTi=VGKi3eiuLz3kMoQ0K6ct5mJrrfA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
	<BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
	<BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>
	<BANLkTi=VGKi3eiuLz3kMoQ0K6ct5mJrrfA@mail.gmail.com>
Message-ID: <BANLkTi=DQ+F9S3Ksp8YyvbQ8NU_KwWpnHA@mail.gmail.com>

moving your SQLlite DB to tempFS is the biggest single change you can have
with FS.
that alone would account for the massive increase in calls you were able to
achieve.


On Tue, May 24, 2011 at 9:57 PM, Anton VG <anton.vazir at gmail.com> wrote:

> do you mean sqlite?
>
> 2011/5/24 Tihomir Culjaga <tculjaga at gmail.com>:
> > the only real improvment was to move your DB to tmpfs ... the rest ( if
> > 64bit) makes really small difference from distro to distro..
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From kerem.erciyes at gmail.com  Tue May 24 16:21:51 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Tue, 24 May 2011 15:21:51 +0300
Subject: [Freeswitch-users] DTMF problems
In-Reply-To: <4DDBA05C.6060204@tagnet.ru>
References: <4DDBA05C.6060204@tagnet.ru>
Message-ID: <op.vvzjapbizwj8qf@solidsnake.local>

Hi Boris,

I have had same kind of problems with PRI circuits between Channel Banks  
and VoIP Servers due to static on the POTS line and/or too high gain  
settings on the PRI cicuit. Try playing with the gain settings on the  
Cisco 5350, maybe that can help.

Regards,
Kerem



On Tue, 24 May 2011 15:11:08 +0300, Boris Kovalenko <boris at tagnet.ru>  
wrote:

> Hello!
>
>      My network enviroment is: PRI --- Cisco 5350 --- FreeSwitch
> (Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)) ---
> AudioCodes MP-118FXS. There is an IVR on FreeSwitch where calling party
> may press a key for personal client line or wait for a secretary.
> Sometimes (as client said) the call to line 5 is placed to line 3 (as I
> undestand DTMF 5 is recognized as 3). How may I detect and solve the
> problem? Configurations below:
> Cisco 5350:
> dial-peer voice 4000 voip
>   description OUTBOUND-CALLS-FROM-USI
>   destination-pattern 50000#.T
>   voice-class codec 1
>   session protocol sipv2
>   session target ipv4:82.193.138.187:5060
>   dtmf-relay rtp-nte
>   fax-relay ecm disable
>   fax rate 9600
>   fax protocol t38 ls-redundancy 5 hs-redundancy 0 fallback pass-through
> g711alaw
>   no vad
>
> FreeSwitch:
> profile:
> <param name="dtmf-duration" value="2000"/>
> <!-- <param name="dtmf-type" value="rfc2833"/> --> (yes, it is commented)
>
> context:
> <extension name="ext_001">
> <condition field="destination_number" expression="^(73435230100)$">
> <action application="set" data="v_tagnet_ats_dstport=10005"/>
> <action application="set" data="v_ats_dstport=10005"/>
> <action application="set" data="import=v_tagnet_dstport,v_ats_dstpor
> t"/>
> <!-- <action application="set" data="import=v_tagnet_ats_dstport"/>
> -->
> <action application="set" data="hangup_after_bridge=true"/>
> <!-- <action application="set" data="continue_on_fail=USER_NOT_REGIS
> TERED"/> -->
> <action application="set" data="ringback=$${ru-ring}"/>
> <action application="javascript" data="${scripts_dir}/common/set_dpe.js
> B ui.tagnet.cc 3 3"/>
> <action application="answer"/>
> <action application="sleep" data="1000"/>
> <action application="play_and_get_digits" data="1 1 1 5000 #
> /opt/fs/sounds/ntmjui/welcome.wav silence_stream://250 digits \d* 3000"/>
> <action application="execute_extension" data="ext_call_internal"/>
> <action application="bridge" data="${sofia_contact(101 at ui.tagnet.cc)}" />
> <action application="hangup"/>
> </condition>
> </extension>
>
>


-- 
Kerem Erciyes - Sistem Dan??man?
http://keremerciyes.com


From anton.vazir at gmail.com  Tue May 24 16:45:32 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 24 May 2011 17:45:32 +0500
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTi=DQ+F9S3Ksp8YyvbQ8NU_KwWpnHA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
	<BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
	<BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>
	<BANLkTi=VGKi3eiuLz3kMoQ0K6ct5mJrrfA@mail.gmail.com>
	<BANLkTi=DQ+F9S3Ksp8YyvbQ8NU_KwWpnHA@mail.gmail.com>
Message-ID: <BANLkTi=G=P14_vhaM7JDqpLGqBOJXOotEg@mail.gmail.com>

I use Postgres as Db backend. Any note on performance for the given?

2011/5/24 jay binks <jaybinks at gmail.com>:
> moving your SQLlite DB to tempFS is the biggest single change you can have
> with FS.
> that alone would account for the massive increase in calls you were able to
> achieve.
>
> On Tue, May 24, 2011 at 9:57 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> do you mean sqlite?
>>
>> 2011/5/24 Tihomir Culjaga <tculjaga at gmail.com>:
>> > the only real improvment was to move your DB to tmpfs ... the rest ( if
>> > 64bit) makes really small difference from distro to distro..
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue May 24 16:45:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 24 May 2011 13:45:39 +0100
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
Message-ID: <BANLkTim=uKQBet-V79qszTHGTbXjWaQDQw@mail.gmail.com>

uuid_bridge returning +OK just means the API command executed ok starting
the bridge, it doesn't indicate the bridge is successful. You'd need to
watch the events to verify that.

You say you're using G729, a transcoding codec. Which module are you using
to provide this?

If you try to bridge a parked G729 call to a non-G729 call and you're not
using a transcoding version with available licenses, the call will drop.

-Steve




On 24 May 2011 10:33, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

> Hello List , Good Morning.
>
> In my fight to get ESL & Python & Freeswitch all to behave properly i
> noticed a possible BUG ( need the veterans to confirm).
>
> Scenario :
>
> Originate & ValetPark () + MOH OR  Just Park() , Both show the same
> problem.
> Codec G729
>
> I Make another call:
> originate & park()
> codec G729
>
> Now Python ESL Inbound Or FS Console :
>
> uuid_bridge uuid1 uuid2
>
> +OK uuid
>
> Now one of the two things happen :
>
> 1) One the call gets connected hurray :) , Audio Perfect  , etc.
>
> 2) The call gets dropped :(
>
> In both cases uuid_bridge reports +OK even in the case the call is dropped.
> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
> could this be related to the use of G729 ?
>
> I have lots of available licenses.
>
> Regards
> Ant?nio Teixeira
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anton.vazir at gmail.com  Tue May 24 16:47:52 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 24 May 2011 17:47:52 +0500
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTik8DNOVeyo1ceTTcC_tDLL9Hu3Asw@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTikDNOXr1+he8KgT2joKhKwdMy9VUA@mail.gmail.com>
	<BANLkTik8DNOVeyo1ceTTcC_tDLL9Hu3Asw@mail.gmail.com>
Message-ID: <BANLkTimA-Ve-VM+0v+aa3Dcece0e933o=g@mail.gmail.com>

Kind of strange, I suppose there is something wrong with your code,
since otherwise, there is some huge bug in FS, which looks much less
likely, since it would be already noticed

2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>:
> Hello Anton
>
> No , this was made on our test environment so only one call at FS.
>
> Manually started some would bridge and some not using the same endpoints.
>
>
>
>
> 2011/5/24 Anton VG <anton.vazir at gmail.com>
>>
>> Did you put any load to get several CPS? Or just out of several
>> sequential calls, some of them bridges, some not?
>>
>> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>:
>> > Hello List , Good Morning.
>> >
>> > In my fight to get ESL & Python & Freeswitch all to behave properly i
>> > noticed a possible BUG ( need the veterans to confirm).
>> >
>> > Scenario :
>> >
>> > Originate & ValetPark () + MOH OR? Just Park() , Both show the same
>> > problem.
>> > Codec G729
>> >
>> > I Make another call:
>> > originate & park()
>> > codec G729
>> >
>> > Now Python ESL Inbound Or FS Console :
>> >
>> > uuid_bridge uuid1 uuid2
>> >
>> > +OK uuid
>> >
>> > Now one of the two things happen :
>> >
>> > 1) One the call gets connected hurray :) , Audio Perfect? , etc.
>> >
>> > 2) The call gets dropped :(
>> >
>> > In both cases uuid_bridge reports +OK even in the case the call is
>> > dropped.
>> > Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>> > SYN/ASYNC Problems)? I still get sometimes ( not always) a dropped call
>> > could this be related to the use of G729 ?
>> >
>> > I have lots of available licenses.
>> >
>> > Regards
>> > Ant?nio Teixeira
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Tue May 24 16:48:54 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 12:48:54 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTi=DQ+F9S3Ksp8YyvbQ8NU_KwWpnHA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
	<BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
	<BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>
	<BANLkTi=VGKi3eiuLz3kMoQ0K6ct5mJrrfA@mail.gmail.com>
	<BANLkTi=DQ+F9S3Ksp8YyvbQ8NU_KwWpnHA@mail.gmail.com>
Message-ID: <BANLkTinDwjBBASstmT+CeuMDBf2s_=GjgQ@mail.gmail.com>

Regarding the SIgnaling Problem , the DB was already on tmpfs and you may
have read that this was not an I/O problem.

To load Blance a call is simple in our use case.

First understand we use auto-dialers so alot happens in the backend , the
rest we send a originate command so simply change the profile used imagine :

originate sofia/external1_profile/22222222 at 111.11.11.11 IVR XML

The dialer simply changes
originate sofia/external1_profile/22222222 at 111.11.11.11 IVR XML
originate sofia/external2_profile/22222222 at 111.11.11.11 IVR XML
originate sofia/external3_profile/22222222 at 111.11.11.11 IVR XML
originate sofia/external4_profile/22222222 at 111.11.11.11 IVR XML

We use INB ESL.

Now if you want to use "softphones" i think you can use mod_limit and the
XML dialplan to create a realtime version of this.

Notes :

Keep in mind that i do not know if this is the correct way to handle this
situation but we have it deployed  this way and as i was able to increase
the call volume by a factor of 2 while at the same time increasing quality
and decreasing the signaling issues.
 At the expense of RAM more RAM = Profiles.



2011/5/24 jay binks <jaybinks at gmail.com>

> moving your SQLlite DB to tempFS is the biggest single change you can have
> with FS.
> that alone would account for the massive increase in calls you were able to
> achieve.
>
>
> On Tue, May 24, 2011 at 9:57 PM, Anton VG <anton.vazir at gmail.com> wrote:
>
>> do you mean sqlite?
>>
>> 2011/5/24 Tihomir Culjaga <tculjaga at gmail.com>:
>> > the only real improvment was to move your DB to tmpfs ... the rest ( if
>> > 64bit) makes really small difference from distro to distro..
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anton.vazir at gmail.com  Tue May 24 16:50:30 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 24 May 2011 17:50:30 +0500
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTim=uKQBet-V79qszTHGTbXjWaQDQw@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTim=uKQBet-V79qszTHGTbXjWaQDQw@mail.gmail.com>
Message-ID: <BANLkTi=aKeOr_g8Z79=cBpfOKAGjck3HKw@mail.gmail.com>

Interesting point Steven, while bridging the 2 call legs, with
different codecs, they should be transcoded automatically, if there is
a codec available in the system, isn't it?

2011/5/24 Steven Ayre <steveayre at gmail.com>:
> uuid_bridge returning +OK just means the API command executed ok starting
> the bridge, it doesn't indicate the bridge is successful. You'd need to
> watch the events to verify that.
>
> You say you're using G729, a transcoding codec. Which module are you using
> to provide this?
>
> If you try to bridge a parked G729 call to a non-G729 call and you're not
> using a transcoding version with available licenses, the call will drop.
>
> -Steve
>
>
>
>
> On 24 May 2011 10:33, Antonio Teixeira <eagle.antonio at gmail.com> wrote:
>>
>> Hello List , Good Morning.
>>
>> In my fight to get ESL & Python & Freeswitch all to behave properly i
>> noticed a possible BUG ( need the veterans to confirm).
>>
>> Scenario :
>>
>> Originate & ValetPark () + MOH OR? Just Park() , Both show the same
>> problem.
>> Codec G729
>>
>> I Make another call:
>> originate & park()
>> codec G729
>>
>> Now Python ESL Inbound Or FS Console :
>>
>> uuid_bridge uuid1 uuid2
>>
>> +OK uuid
>>
>> Now one of the two things happen :
>>
>> 1) One the call gets connected hurray :) , Audio Perfect? , etc.
>>
>> 2) The call gets dropped :(
>>
>> In both cases uuid_bridge reports +OK even in the case the call is
>> dropped.
>> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>> SYN/ASYNC Problems)? I still get sometimes ( not always) a dropped call
>> could this be related to the use of G729 ?
>>
>> I have lots of available licenses.
>>
>> Regards
>> Ant?nio Teixeira
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Tue May 24 16:57:43 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 12:57:43 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTimA-Ve-VM+0v+aa3Dcece0e933o=g@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTikDNOXr1+he8KgT2joKhKwdMy9VUA@mail.gmail.com>
	<BANLkTik8DNOVeyo1ceTTcC_tDLL9Hu3Asw@mail.gmail.com>
	<BANLkTimA-Ve-VM+0v+aa3Dcece0e933o=g@mail.gmail.com>
Message-ID: <BANLkTikrVMNZKYdgfm3_Ooo37RzKXjxH_w@mail.gmail.com>

For G729 we use :
Mod com g729All calls are using G729,

I can even do this without ESL just need two phone that are Park() and the
FS Console to do the UUID_Bridge , the results will be the same .

I wil try today with G711 to see how it goes :)

Regards
A/T


2011/5/24 Anton VG <anton.vazir at gmail.com>

> Kind of strange, I suppose there is something wrong with your code,
> since otherwise, there is some huge bug in FS, which looks much less
> likely, since it would be already noticed
>
> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>:
> > Hello Anton
> >
> > No , this was made on our test environment so only one call at FS.
> >
> > Manually started some would bridge and some not using the same endpoints.
> >
> >
> >
> >
> > 2011/5/24 Anton VG <anton.vazir at gmail.com>
> >>
> >> Did you put any load to get several CPS? Or just out of several
> >> sequential calls, some of them bridges, some not?
> >>
> >> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>:
> >> > Hello List , Good Morning.
> >> >
> >> > In my fight to get ESL & Python & Freeswitch all to behave properly i
> >> > noticed a possible BUG ( need the veterans to confirm).
> >> >
> >> > Scenario :
> >> >
> >> > Originate & ValetPark () + MOH OR  Just Park() , Both show the same
> >> > problem.
> >> > Codec G729
> >> >
> >> > I Make another call:
> >> > originate & park()
> >> > codec G729
> >> >
> >> > Now Python ESL Inbound Or FS Console :
> >> >
> >> > uuid_bridge uuid1 uuid2
> >> >
> >> > +OK uuid
> >> >
> >> > Now one of the two things happen :
> >> >
> >> > 1) One the call gets connected hurray :) , Audio Perfect  , etc.
> >> >
> >> > 2) The call gets dropped :(
> >> >
> >> > In both cases uuid_bridge reports +OK even in the case the call is
> >> > dropped.
> >> > Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> >> > SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped
> call
> >> > could this be related to the use of G729 ?
> >> >
> >> > I have lots of available licenses.
> >> >
> >> > Regards
> >> > Ant?nio Teixeira
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Tue May 24 17:00:53 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 24 May 2011 15:00:53 +0200
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTi=aKeOr_g8Z79=cBpfOKAGjck3HKw@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTim=uKQBet-V79qszTHGTbXjWaQDQw@mail.gmail.com>,
	<BANLkTi=aKeOr_g8Z79=cBpfOKAGjck3HKw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62ED@cooper>

Yes, but for G729 you must use a licensed codec. Do you have that installed?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
Skickat: den 24 maj 2011 14:50
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK

Interesting point Steven, while bridging the 2 call legs, with
different codecs, they should be transcoded automatically, if there is
a codec available in the system, isn't it?

2011/5/24 Steven Ayre <steveayre at gmail.com>:
> uuid_bridge returning +OK just means the API command executed ok starting
> the bridge, it doesn't indicate the bridge is successful. You'd need to
> watch the events to verify that.
>
> You say you're using G729, a transcoding codec. Which module are you using
> to provide this?
>
> If you try to bridge a parked G729 call to a non-G729 call and you're not
> using a transcoding version with available licenses, the call will drop.
>
> -Steve
>
>
>
>
> On 24 May 2011 10:33, Antonio Teixeira <eagle.antonio at gmail.com> wrote:
>>
>> Hello List , Good Morning.
>>
>> In my fight to get ESL & Python & Freeswitch all to behave properly i
>> noticed a possible BUG ( need the veterans to confirm).
>>
>> Scenario :
>>
>> Originate & ValetPark () + MOH OR  Just Park() , Both show the same
>> problem.
>> Codec G729
>>
>> I Make another call:
>> originate & park()
>> codec G729
>>
>> Now Python ESL Inbound Or FS Console :
>>
>> uuid_bridge uuid1 uuid2
>>
>> +OK uuid
>>
>> Now one of the two things happen :
>>
>> 1) One the call gets connected hurray :) , Audio Perfect  , etc.
>>
>> 2) The call gets dropped :(
>>
>> In both cases uuid_bridge reports +OK even in the case the call is
>> dropped.
>> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>> SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
>> could this be related to the use of G729 ?
>>
>> I have lots of available licenses.
>>
>> Regards
>> Ant?nio Teixeira
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4ddba9d332761894489277!



From eagle.antonio at gmail.com  Tue May 24 17:03:54 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 13:03:54 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62ED@cooper>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTim=uKQBet-V79qszTHGTbXjWaQDQw@mail.gmail.com>
	<BANLkTi=aKeOr_g8Z79=cBpfOKAGjck3HKw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62ED@cooper>
Message-ID: <BANLkTingna3JbrbL7F15qzg1=FHtXEJRdQ@mail.gmail.com>

120 of it :D



2011/5/24 Peter Olsson <peter.olsson at visionutveckling.se>

> Yes, but for G729 you must use a licensed codec. Do you have that
> installed?
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [
> anton.vazir at gmail.com]
> Skickat: den 24 maj 2011 14:50
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
>
> Interesting point Steven, while bridging the 2 call legs, with
> different codecs, they should be transcoded automatically, if there is
> a codec available in the system, isn't it?
>
> 2011/5/24 Steven Ayre <steveayre at gmail.com>:
> > uuid_bridge returning +OK just means the API command executed ok starting
> > the bridge, it doesn't indicate the bridge is successful. You'd need to
> > watch the events to verify that.
> >
> > You say you're using G729, a transcoding codec. Which module are you
> using
> > to provide this?
> >
> > If you try to bridge a parked G729 call to a non-G729 call and you're not
> > using a transcoding version with available licenses, the call will drop.
> >
> > -Steve
> >
> >
> >
> >
> > On 24 May 2011 10:33, Antonio Teixeira <eagle.antonio at gmail.com> wrote:
> >>
> >> Hello List , Good Morning.
> >>
> >> In my fight to get ESL & Python & Freeswitch all to behave properly i
> >> noticed a possible BUG ( need the veterans to confirm).
> >>
> >> Scenario :
> >>
> >> Originate & ValetPark () + MOH OR  Just Park() , Both show the same
> >> problem.
> >> Codec G729
> >>
> >> I Make another call:
> >> originate & park()
> >> codec G729
> >>
> >> Now Python ESL Inbound Or FS Console :
> >>
> >> uuid_bridge uuid1 uuid2
> >>
> >> +OK uuid
> >>
> >> Now one of the two things happen :
> >>
> >> 1) One the call gets connected hurray :) , Audio Perfect  , etc.
> >>
> >> 2) The call gets dropped :(
> >>
> >> In both cases uuid_bridge reports +OK even in the case the call is
> >> dropped.
> >> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> >> SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
> >> could this be related to the use of G729 ?
> >>
> >> I have lots of available licenses.
> >>
> >> Regards
> >> Ant?nio Teixeira
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4ddba9d332761894489277!
>
>
> _______________________________________________
> 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 boris at tagnet.ru  Tue May 24 17:10:03 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 24 May 2011 19:10:03 +0600
Subject: [Freeswitch-users] DTMF problems
In-Reply-To: <op.vvzjapbizwj8qf@solidsnake.local>
References: <4DDBA05C.6060204@tagnet.ru> <op.vvzjapbizwj8qf@solidsnake.local>
Message-ID: <4DDBAE2B.6080208@tagnet.ru>

Hello!

     Kerem, are You talking about input gain parameter? AFAIK it is 0db 
by default. Would You recomend to set it lower?
ISDN 1/0:D - 1/0:D
  Type of VoicePort is ISDN
  Operation State is DORMANT
  Administrative State is UP
  No Interface Down Failure
  Description is Link to ATSE-41
  Noise Regeneration is disabled
  Non Linear Processing is disabled
  Non Linear Mute is disabled
  Non Linear Threshold is -21 dB
  Music On Hold Threshold is Set to -38 dBm
  In Gain is Set to 0 dB
  Out Attenuation is Set to 0 dB
  Echo Cancellation is enabled
  Echo Cancellation NLP mute is disabled
  Echo Cancellation NLP threshold is -21 dB
  Echo Cancel Coverage is set to 32 ms
  Echo Cancel worst case ERL is set to 6 dB
  Playout-delay Mode is set to adaptive
  Playout-delay Nominal is set to 60 ms
  Playout-delay Maximum is set to 250 ms
  Playout-delay Minimum mode is set to default, value 40 ms
  Playout-delay Fax is set to 300 ms
  Connection Mode is normal
  Connection Number is not set
  Initial Time Out is set to 10 s
  Interdigit Time Out is set to 10 s
  Call Disconnect Time Out is set to 60 s
  Ringing Time Out is set to 180 s
  Wait Release Time Out is set to 30 s
  Spe country is russia
  Region Tone is set for RU
  Station name None, Station number None
  Translation profile (Incoming):
  Translation profile (Outgoing):

> Hi Boris,
>
> I have had same kind of problems with PRI circuits between Channel Banks
> and VoIP Servers due to static on the POTS line and/or too high gain
> settings on the PRI cicuit. Try playing with the gain settings on the
> Cisco 5350, maybe that can help.
>
> Regards,
> Kerem
>
>
>
> On Tue, 24 May 2011 15:11:08 +0300, Boris Kovalenko<boris at tagnet.ru>
> wrote:
>
>> Hello!
>>
>>       My network enviroment is: PRI --- Cisco 5350 --- FreeSwitch
>> (Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)) ---
>> AudioCodes MP-118FXS. There is an IVR on FreeSwitch where calling party
>> may press a key for personal client line or wait for a secretary.
>> Sometimes (as client said) the call to line 5 is placed to line 3 (as I
>> undestand DTMF 5 is recognized as 3). How may I detect and solve the
>> problem? Configurations below:
>> Cisco 5350:
>> dial-peer voice 4000 voip
>>    description OUTBOUND-CALLS-FROM-USI
>>    destination-pattern 50000#.T
>>    voice-class codec 1
>>    session protocol sipv2
>>    session target ipv4:82.193.138.187:5060
>>    dtmf-relay rtp-nte
>>    fax-relay ecm disable
>>    fax rate 9600
>>    fax protocol t38 ls-redundancy 5 hs-redundancy 0 fallback pass-through
>> g711alaw
>>    no vad
>>
>> FreeSwitch:
>> profile:
>> <param name="dtmf-duration" value="2000"/>
>> <!--<param name="dtmf-type" value="rfc2833"/>  -->  (yes, it is commented)
>>
>> context:
>> <extension name="ext_001">
>> <condition field="destination_number" expression="^(73435230100)$">
>> <action application="set" data="v_tagnet_ats_dstport=10005"/>
>> <action application="set" data="v_ats_dstport=10005"/>
>> <action application="set" data="import=v_tagnet_dstport,v_ats_dstpor
>> t"/>
>> <!--<action application="set" data="import=v_tagnet_ats_dstport"/>
>> -->
>> <action application="set" data="hangup_after_bridge=true"/>
>> <!--<action application="set" data="continue_on_fail=USER_NOT_REGIS
>> TERED"/>  -->
>> <action application="set" data="ringback=$${ru-ring}"/>
>> <action application="javascript" data="${scripts_dir}/common/set_dpe.js
>> B ui.tagnet.cc 3 3"/>
>> <action application="answer"/>
>> <action application="sleep" data="1000"/>
>> <action application="play_and_get_digits" data="1 1 1 5000 #
>> /opt/fs/sounds/ntmjui/welcome.wav silence_stream://250 digits \d* 3000"/>
>> <action application="execute_extension" data="ext_call_internal"/>
>> <action application="bridge" data="${sofia_contact(101 at ui.tagnet.cc)}" />
>> <action application="hangup"/>
>> </condition>
>> </extension>
>>
>>
>


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



From peter.olsson at visionutveckling.se  Tue May 24 17:18:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 24 May 2011 15:18:42 +0200
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTingna3JbrbL7F15qzg1=FHtXEJRdQ@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTim=uKQBet-V79qszTHGTbXjWaQDQw@mail.gmail.com>
	<BANLkTi=aKeOr_g8Z79=cBpfOKAGjck3HKw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62ED@cooper>
	<BANLkTingna3JbrbL7F15qzg1=FHtXEJRdQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE28246F@cooper>

When tha call drops, examine the log and find out why the call was disconnected.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Teixeira
Skickat: den 24 maj 2011 15:04
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK

120 of it :D


2011/5/24 Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
Yes, but for G729 you must use a licensed codec. Do you have that installed?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r Anton VG [anton.vazir at gmail.com<mailto:anton.vazir at gmail.com>]
Skickat: den 24 maj 2011 14:50
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK

Interesting point Steven, while bridging the 2 call legs, with
different codecs, they should be transcoded automatically, if there is
a codec available in the system, isn't it?

2011/5/24 Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>>:
> uuid_bridge returning +OK just means the API command executed ok starting
> the bridge, it doesn't indicate the bridge is successful. You'd need to
> watch the events to verify that.
>
> You say you're using G729, a transcoding codec. Which module are you using
> to provide this?
>
> If you try to bridge a parked G729 call to a non-G729 call and you're not
> using a transcoding version with available licenses, the call will drop.
>
> -Steve
>
>
>
>
> On 24 May 2011 10:33, Antonio Teixeira <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>> wrote:
>>
>> Hello List , Good Morning.
>>
>> In my fight to get ESL & Python & Freeswitch all to behave properly i
>> noticed a possible BUG ( need the veterans to confirm).
>>
>> Scenario :
>>
>> Originate & ValetPark () + MOH OR  Just Park() , Both show the same
>> problem.
>> Codec G729
>>
>> I Make another call:
>> originate & park()
>> codec G729
>>
>> Now Python ESL Inbound Or FS Console :
>>
>> uuid_bridge uuid1 uuid2
>>
>> +OK uuid
>>
>> Now one of the two things happen :
>>
>> 1) One the call gets connected hurray :) , Audio Perfect  , etc.
>>
>> 2) The call gets dropped :(
>>
>> In both cases uuid_bridge reports +OK even in the case the call is
>> dropped.
>> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>> SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
>> could this be related to the use of G729 ?
>>
>> I have lots of available licenses.
>>
>> Regards
>> Ant?nio Teixeira
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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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From kerem.erciyes at gmail.com  Tue May 24 17:36:33 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Tue, 24 May 2011 16:36:33 +0300
Subject: [Freeswitch-users] DTMF problems
In-Reply-To: <4DDBAE2B.6080208@tagnet.ru>
References: <4DDBA05C.6060204@tagnet.ru> <op.vvzjapbizwj8qf@solidsnake.local>
	<4DDBAE2B.6080208@tagnet.ru>
Message-ID: <op.vvzmq7lyzwj8qf@solidsnake.local>

Yes Boris, try setting it lower incrementally and see if it helps. There  
might be problems related to gain + echo on the line that confuses DTMF  
detection, so check if the line has echo as well. Also sometimes I have  
seen that the remote calling party has a POTS line with too much static it  
can confuse DTMF detection. On other VoIP systems we solved this problem  
by spying on the channel e.g.  
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be  
helpful in debugging the situation.

On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko <boris at tagnet.ru>  
wrote:

> Hello!
>
>      Kerem, are You talking about input gain parameter? AFAIK it is 0db
> by default. Would You recomend to set it lower?
> ISDN 1/0:D - 1/0:D
>   Type of VoicePort is ISDN
>   Operation State is DORMANT
>   Administrative State is UP
>   No Interface Down Failure
>   Description is Link to ATSE-41
>   Noise Regeneration is disabled
>   Non Linear Processing is disabled
>   Non Linear Mute is disabled
>   Non Linear Threshold is -21 dB
>   Music On Hold Threshold is Set to -38 dBm
>   In Gain is Set to 0 dB
>   Out Attenuation is Set to 0 dB
>   Echo Cancellation is enabled
>   Echo Cancellation NLP mute is disabled
>   Echo Cancellation NLP threshold is -21 dB
>   Echo Cancel Coverage is set to 32 ms
>   Echo Cancel worst case ERL is set to 6 dB
>   Playout-delay Mode is set to adaptive
>   Playout-delay Nominal is set to 60 ms
>   Playout-delay Maximum is set to 250 ms
>   Playout-delay Minimum mode is set to default, value 40 ms
>   Playout-delay Fax is set to 300 ms
>   Connection Mode is normal
>   Connection Number is not set
>   Initial Time Out is set to 10 s
>   Interdigit Time Out is set to 10 s
>   Call Disconnect Time Out is set to 60 s
>   Ringing Time Out is set to 180 s
>   Wait Release Time Out is set to 30 s
>   Spe country is russia
>   Region Tone is set for RU
>   Station name None, Station number None
>   Translation profile (Incoming):
>   Translation profile (Outgoing):
>
>> Hi Boris,
>>
>> I have had same kind of problems with PRI circuits between Channel Banks
>> and VoIP Servers due to static on the POTS line and/or too high gain
>> settings on the PRI cicuit. Try playing with the gain settings on the
>> Cisco 5350, maybe that can help.
>>
>> Regards,
>> Kerem
>>
>>
>>
>> On Tue, 24 May 2011 15:11:08 +0300, Boris Kovalenko<boris at tagnet.ru>
>> wrote:
>>
<snip>
>>
>
>




From eric at loopfx.com  Tue May 24 17:48:08 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 24 May 2011 09:48:08 -0400
Subject: [Freeswitch-users] Read SIP Response Headers from Dialplan
	or	Lua
In-Reply-To: <BANLkTi=-5f2vNLhQa9RG_sO6KFXMkoSP5A@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFC39AB0@doctorbob.loop.local>
	<BANLkTi=-5f2vNLhQa9RG_sO6KFXMkoSP5A@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFC39C7D@doctorbob.loop.local>

I have this in my dial plan:

    <action application="set" data="api_hangup_hook=lua hook.lua"/>

And hook.lua has this:

dat = env:serialize()
freeswitch.consoleLog("INFO","hangup hook env:\n" .. dat .. "\n")

This dumps everything, and I don't see the X- variables I'm looking for.  I'm assuming this means I would actually need to use the event socket if I want to see them.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, May 23, 2011 5:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Read SIP Response Headers from Dialplan or Lua

I would turn on mod_xml_cdr and look at a failed call's XML output. Look in the <variables> section and see if any of those have the data you're looking for.

-MC
On Mon, May 23, 2011 at 1:58 PM, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
Hello,

I'm wondering if it's possible to read SIP response headers from the dial plan, or from Lua.

One of my gateways returns a generic 500 for all kinds of different error conditions, and expects me to read variables in the sip response.

e.g.

SIP/2.0 500 Service Unavailable
Via:..
From:...
Etc:
X-Custom-Var:val

The value in the custom variable might be "busy", "bad number", etc.

If I add something to the INVITE, I know that I can read it with channel variables "sip_h_X-Name", but I don't get the same thing for the response.

Thanks,

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>


_______________________________________________
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From boris at tagnet.ru  Tue May 24 17:57:15 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 24 May 2011 19:57:15 +0600
Subject: [Freeswitch-users] DTMF problems
In-Reply-To: <op.vvzmq7lyzwj8qf@solidsnake.local>
References: <4DDBA05C.6060204@tagnet.ru>
	<op.vvzjapbizwj8qf@solidsnake.local>	<4DDBAE2B.6080208@tagnet.ru>
	<op.vvzmq7lyzwj8qf@solidsnake.local>
Message-ID: <4DDBB93B.2040604@tagnet.ru>

Hello!

     Thank You Kerem, I'll try. The main problem is that there is a 
little percent of wrong DTMF and a very nervous customer.
Also what if not to use dtmf-relay on Cisco at all? What kind of 
problems may I have? I use G711alaw inside a network. Some transit calls 
(ie Internet -> PSTN) may use G729.

> Yes Boris, try setting it lower incrementally and see if it helps. There
> might be problems related to gain + echo on the line that confuses DTMF
> detection, so check if the line has echo as well. Also sometimes I have
> seen that the remote calling party has a POTS line with too much static it
> can confuse DTMF detection. On other VoIP systems we solved this problem
> by spying on the channel e.g.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be
> helpful in debugging the situation.
>
> On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko<boris at tagnet.ru>
> wrote:
>
>> Hello!
>>
>>       Kerem, are You talking about input gain parameter? AFAIK it is 0db
>> by default. Would You recomend to set it lower?
>> ISDN 1/0:D - 1/0:D
>>    Type of VoicePort is ISDN
>>    Operation State is DORMANT
>>    Administrative State is UP
>>    No Interface Down Failure
>>    Description is Link to ATSE-41
>>    Noise Regeneration is disabled
>>    Non Linear Processing is disabled
>>    Non Linear Mute is disabled
>>    Non Linear Threshold is -21 dB
>>    Music On Hold Threshold is Set to -38 dBm
>>    In Gain is Set to 0 dB
>>    Out Attenuation is Set to 0 dB
>>    Echo Cancellation is enabled
>>    Echo Cancellation NLP mute is disabled
>>    Echo Cancellation NLP threshold is -21 dB
>>    Echo Cancel Coverage is set to 32 ms
>>    Echo Cancel worst case ERL is set to 6 dB
>>    Playout-delay Mode is set to adaptive
>>    Playout-delay Nominal is set to 60 ms
>>    Playout-delay Maximum is set to 250 ms
>>    Playout-delay Minimum mode is set to default, value 40 ms
>>    Playout-delay Fax is set to 300 ms
>>    Connection Mode is normal
>>    Connection Number is not set
>>    Initial Time Out is set to 10 s
>>    Interdigit Time Out is set to 10 s
>>    Call Disconnect Time Out is set to 60 s
>>    Ringing Time Out is set to 180 s
>>    Wait Release Time Out is set to 30 s
>>    Spe country is russia
>>    Region Tone is set for RU
>>    Station name None, Station number None
>>    Translation profile (Incoming):
>>    Translation profile (Outgoing):
>>
>>> Hi Boris,
>>>
>>> I have had same kind of problems with PRI circuits between Channel Banks
>>> and VoIP Servers due to static on the POTS line and/or too high gain
>>> settings on the PRI cicuit. Try playing with the gain settings on the
>>> Cisco 5350, maybe that can help.
>>>
>>> Regards,
>>> Kerem
>>>
>>>
>>>
>>> On Tue, 24 May 2011 15:11:08 +0300, Boris Kovalenko<boris at tagnet.ru>
>>> wrote:
>>>
> <snip>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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




From rmorin at blie-ent.com  Tue May 24 18:50:21 2011
From: rmorin at blie-ent.com (Rob Morin)
Date: Tue, 24 May 2011 10:50:21 -0400
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>
	<59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>
Message-ID: <04b201cc1a21$e8825400$b986fc00$@blie-ent.com>

I can, if someone can tell me where to get the snapshot.  I looked at
files.freeswitch.org and didn?t see one.

 

Prior to updating to the 5-17 snapshot, I was having the same problems with
the 3-25 snapshot that I was using.

 

Thank you,

Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Tuesday, May 24, 2011 1:47 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

following Yehavi's comment, could you downgrade your GIT version to a
previous version, like one month old ?

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 





 

Le 24/05/2011 ? 02:59, Rob Morin a ?crit :





David,

 

It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and
Cisco/Linksys SPA 2102. Same story in both cases.

 

I had my carrier help me troubleshoot and we can see the SIP request from
the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but
the response from the A leg isn?t forwarded back to the B leg. I can produce
traces that show this.

 

When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t
respond or forward it. So the connection dies.  I have every reason to
believe that I?d be able to send the fax if FS would respond with a 488
indicating T.38 is not acceptable. But instead, all I get is silence. Again,
I can produce traces that demonstrate this.

 

There?s another similar email  trail out there,
http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821
994.html, but I?m not certain that they?re related.

 

Thank you for your help!
Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Monday, May 23, 2011 6:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

perhaps you should not consider T38 is 100% interoperable.

You may tell us what ATA is that, because some of them are nice piece of
junk.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 






 

Le 23/05/2011 ? 20:51, Rob Morin a ?crit :






I?m having problems faxing if the B leg offers T.38.  Several scenarios, but
always the same result.

 

First, the architecture

 

FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another
carrier (PSTN) and the fax machine)

 

On my ATA, T.38 is enabled (Auto Detect).

 

If the destination supports T.38, the Carrier will offer it to Freeswitch.
If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
faxes go through.

 

I?ve set Freeswitch as:

      <action application="set" data="fax_enable_t38=true"/>

      <action application="export" data="t38_passthru=true"/> 

When that?s the case, and the B leg offers T.38, Freeswitch passes through
the offer and the ATA responds, 3 times. But the response never gets sent to
the B leg, so it terminates the call after about 30 seconds.

 

When I set Freeswitch as:

      <action application="set" data="fax_enable_t38=false"/>

The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t
pass the offer through and the B leg terminates the call after about 30
seconds.

 

I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
2011-05-17 22-51-47 -0500) ).

 

I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
?passthrough? has had the most success, mostly when the other ends aren?t
T.38 capable and it stays in G711u.

 

Is there something else I need to do to enable T.38 in passthrough mode? 

 

Thank you,
Rob Morin

 

PS ? I can provide tcpdumps of this, or whatever else is necessary.

 

 

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

 

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

 

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From anthony.minessale at gmail.com  Tue May 24 19:25:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 24 May 2011 10:25:51 -0500
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
Message-ID: <BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>

The +OK only means the attempt to bridge was successful, if something
else goes wrong after that, you will not know because it happens
later.

As suggested, look at the cause of the hangup on the failed bridge.


On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
<eagle.antonio at gmail.com> wrote:
> Hello List , Good Morning.
>
> In my fight to get ESL & Python & Freeswitch all to behave properly i
> noticed a possible BUG ( need the veterans to confirm).
>
> Scenario :
>
> Originate & ValetPark () + MOH OR? Just Park() , Both show the same problem.
> Codec G729
>
> I Make another call:
> originate & park()
> codec G729
>
> Now Python ESL Inbound Or FS Console :
>
> uuid_bridge uuid1 uuid2
>
> +OK uuid
>
> Now one of the two things happen :
>
> 1) One the call gets connected hurray :) , Audio Perfect? , etc.
>
> 2) The call gets dropped :(
>
> In both cases uuid_bridge reports +OK even in the case the call is dropped.
> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> SYN/ASYNC Problems)? I still get sometimes ( not always) a dropped call
> could this be related to the use of G729 ?
>
> I have lots of available licenses.
>
> Regards
> Ant?nio Teixeira
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 May 24 19:33:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 24 May 2011 10:33:36 -0500
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTinDwjBBASstmT+CeuMDBf2s_=GjgQ@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
	<BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
	<BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>
	<BANLkTi=VGKi3eiuLz3kMoQ0K6ct5mJrrfA@mail.gmail.com>
	<BANLkTi=DQ+F9S3Ksp8YyvbQ8NU_KwWpnHA@mail.gmail.com>
	<BANLkTinDwjBBASstmT+CeuMDBf2s_=GjgQ@mail.gmail.com>
Message-ID: <BANLkTim_KuRZotGZMBAeM-_vN3ija86mOg@mail.gmail.com>

Anything you find useful to get more calls is subjective to the power
of the box.
If sofia starts behaving badly, it indicates a maximum resource
consumption.  More profiles of course help with concurrency.

So I could have a box where I could do 5000 calls before another
profile would help and someone else could get benefits of multiple
profiles at 100 calls.

You have 2 choices:
1) Search for ways to squeeze more calls out of a bad box like we used
to do with config.sys in DOS
2) Use the money you saved by getting a free carrier grade switch
platform and buy a halfway decent  box.



On Tue, May 24, 2011 at 7:48 AM, Antonio Teixeira
<eagle.antonio at gmail.com> wrote:
> Regarding the SIgnaling Problem , the DB was already on tmpfs and you may
> have read that this was not an I/O problem.
>
> To load Blance a call is simple in our use case.
>
> First understand we use auto-dialers so alot happens in the backend , the
> rest we send a originate command so simply change the profile used imagine :
>
> originate sofia/external1_profile/22222222 at 111.11.11.11 IVR XML
>
> The dialer simply changes
> originate sofia/external1_profile/22222222 at 111.11.11.11 IVR XML
> originate sofia/external2_profile/22222222 at 111.11.11.11 IVR XML
> originate sofia/external3_profile/22222222 at 111.11.11.11 IVR XML
> originate sofia/external4_profile/22222222 at 111.11.11.11 IVR XML
>
> We use INB ESL.
>
> Now if you want to use "softphones" i think you can use mod_limit and the
> XML dialplan to create a realtime version of this.
>
> Notes :
>
> Keep in mind that i do not know if this is the correct way to handle this
> situation but we have it deployed? this way and as i was able to increase
> the call volume by a factor of 2 while at the same time increasing quality
> and decreasing the signaling issues.
> ?At the expense of RAM more RAM = Profiles.
>
>
>
> 2011/5/24 jay binks <jaybinks at gmail.com>
>>
>> moving your SQLlite DB to tempFS is the biggest single change you can have
>> with FS.
>> that alone would account for the massive increase in calls you were able
>> to achieve.
>>
>> On Tue, May 24, 2011 at 9:57 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>>
>>> do you mean sqlite?
>>>
>>> 2011/5/24 Tihomir Culjaga <tculjaga at gmail.com>:
>>> > the only real improvment was to move your DB to tmpfs ... the rest ( if
>>> > 64bit) makes really small difference from distro to distro..
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Sincerely
>>
>> Jay
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 eagle.antonio at gmail.com  Tue May 24 20:01:33 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 16:01:33 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTim_KuRZotGZMBAeM-_vN3ija86mOg@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<A32AFBCD54D14437A199337660F137F4@e1705>
	<BANLkTimzBus_QkcfhMg9FY6wb=_C-2OhUQ@mail.gmail.com>
	<BANLkTimPx-ZTjdPY91BUT6TJEgcb3JgKpA@mail.gmail.com>
	<BANLkTikSF91u02USi=MY=WNEnrWrYLpNtQ@mail.gmail.com>
	<BANLkTi=VGKi3eiuLz3kMoQ0K6ct5mJrrfA@mail.gmail.com>
	<BANLkTi=DQ+F9S3Ksp8YyvbQ8NU_KwWpnHA@mail.gmail.com>
	<BANLkTinDwjBBASstmT+CeuMDBf2s_=GjgQ@mail.gmail.com>
	<BANLkTim_KuRZotGZMBAeM-_vN3ija86mOg@mail.gmail.com>
Message-ID: <BANLkTikJ28OsopXSzD9Z_R--uVKcEHkULQ@mail.gmail.com>

Well this is a 2 X E5506
With 8 GB of Ram.

The Main Problem is that we use tons of DTMF + TTS + Dialplan Logic and as
you say Anthony this causes an impact on the total Supported Calls.

I'm building a new IVR server that we can deploy on another server so we can
ease the pain that mod_python is causing on the CPU.

TOP Shows AVG idle CPU at 93 %

Anyway for every performance increment i find i will post it here. Thanks
Anthony for your input i think i got your point , I'm assuming an SBC with
media bypass would handle a lot more calls  per profile but as you may have
guessed my objective is not SipTrunking but a high layer of service.

Anyway thanks alot.

Regards
Antonio Teixeira


2011/5/24 Anthony Minessale <anthony.minessale at gmail.com>

> Anything you find useful to get more calls is subjective to the power
> of the box.
> If sofia starts behaving badly, it indicates a maximum resource
> consumption.  More profiles of course help with concurrency.
>
> So I could have a box where I could do 5000 calls before another
> profile would help and someone else could get benefits of multiple
> profiles at 100 calls.
>
> You have 2 choices:
> 1) Search for ways to squeeze more calls out of a bad box like we used
> to do with config.sys in DOS
> 2) Use the money you saved by getting a free carrier grade switch
> platform and buy a halfway decent  box.
>
>
>
> On Tue, May 24, 2011 at 7:48 AM, Antonio Teixeira
> <eagle.antonio at gmail.com> wrote:
> > Regarding the SIgnaling Problem , the DB was already on tmpfs and you may
> > have read that this was not an I/O problem.
> >
> > To load Blance a call is simple in our use case.
> >
> > First understand we use auto-dialers so alot happens in the backend , the
> > rest we send a originate command so simply change the profile used
> imagine :
> >
> > originate sofia/external1_profile/22222222 at 111.11.11.11 IVR XML
> >
> > The dialer simply changes
> > originate sofia/external1_profile/22222222 at 111.11.11.11 IVR XML
> > originate sofia/external2_profile/22222222 at 111.11.11.11 IVR XML
> > originate sofia/external3_profile/22222222 at 111.11.11.11 IVR XML
> > originate sofia/external4_profile/22222222 at 111.11.11.11 IVR XML
> >
> > We use INB ESL.
> >
> > Now if you want to use "softphones" i think you can use mod_limit and the
> > XML dialplan to create a realtime version of this.
> >
> > Notes :
> >
> > Keep in mind that i do not know if this is the correct way to handle this
> > situation but we have it deployed  this way and as i was able to increase
> > the call volume by a factor of 2 while at the same time increasing
> quality
> > and decreasing the signaling issues.
> >  At the expense of RAM more RAM = Profiles.
> >
> >
> >
> > 2011/5/24 jay binks <jaybinks at gmail.com>
> >>
> >> moving your SQLlite DB to tempFS is the biggest single change you can
> have
> >> with FS.
> >> that alone would account for the massive increase in calls you were able
> >> to achieve.
> >>
> >> On Tue, May 24, 2011 at 9:57 PM, Anton VG <anton.vazir at gmail.com>
> wrote:
> >>>
> >>> do you mean sqlite?
> >>>
> >>> 2011/5/24 Tihomir Culjaga <tculjaga at gmail.com>:
> >>> > the only real improvment was to move your DB to tmpfs ... the rest (
> if
> >>> > 64bit) makes really small difference from distro to distro..
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Sincerely
> >>
> >> Jay
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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 eagle.antonio at gmail.com  Tue May 24 20:03:10 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 24 May 2011 16:03:10 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
Message-ID: <BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>

Ok Anthony.

I will provide you with data regarding the use of G711 and full debug logs.
Will keep an eye on the logs.

Regards
A/T

2011/5/24 Anthony Minessale <anthony.minessale at gmail.com>

> The +OK only means the attempt to bridge was successful, if something
> else goes wrong after that, you will not know because it happens
> later.
>
> As suggested, look at the cause of the hangup on the failed bridge.
>
>
> On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
> <eagle.antonio at gmail.com> wrote:
> > Hello List , Good Morning.
> >
> > In my fight to get ESL & Python & Freeswitch all to behave properly i
> > noticed a possible BUG ( need the veterans to confirm).
> >
> > Scenario :
> >
> > Originate & ValetPark () + MOH OR  Just Park() , Both show the same
> problem.
> > Codec G729
> >
> > I Make another call:
> > originate & park()
> > codec G729
> >
> > Now Python ESL Inbound Or FS Console :
> >
> > uuid_bridge uuid1 uuid2
> >
> > +OK uuid
> >
> > Now one of the two things happen :
> >
> > 1) One the call gets connected hurray :) , Audio Perfect  , etc.
> >
> > 2) The call gets dropped :(
> >
> > In both cases uuid_bridge reports +OK even in the case the call is
> dropped.
> > Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> > SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
> > could this be related to the use of G729 ?
> >
> > I have lots of available licenses.
> >
> > Regards
> > Ant?nio Teixeira
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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 seichhorn at gci.com  Tue May 24 20:31:13 2011
From: seichhorn at gci.com (Sean Eichhorn)
Date: Tue, 24 May 2011 08:31:13 -0800
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <DF5E4E14-2338-4871-A73E-2291537069B9@ipeva.fr>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
	<BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035F@dtn1mbx01.gci.com>
	<DF5E4E14-2338-4871-A73E-2291537069B9@ipeva.fr>
Message-ID: <7EB00560C8AFF348A7EA76EFD3DC5598332C0360@dtn1mbx01.gci.com>

Yep, no problem.
Here it is.  Same endpoints.  The only thing that changed is the call direction.

Received Message :
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bK5Qtv3B04eS46g
   From: "Test 2" <sip:xxxxxxxx75 at A.B.C.D>;tag=yy2XQKXSy7gtN
   To: <sip:xxxxx02 at I.J.K.L>;tag=93BEF648-1A8B
   Date: Tue, 24 May 2011 16:37:23 GMT
   Call-ID: b7967af6-00c4-122f-19ba-000c29c18d38
   CSeq: 12790979 INVITE
   Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
   Allow-Events: telephone-event
   Remote-Party-ID: <sip:xxxxx02 at I.J.K.L>;party=called;screen=no;privacy=off
   Contact: <sip:xxxxx02 at I.J.K.L:5060>
   Supported: replaces
   Supported: sdp-anat
   Server: Cisco-SIPGateway/IOS-12.x
   Content-Type: application/sdp
   Content-Disposition: session;handling=required
   Content-Length: 334

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 6942 3511 IN IP4 I.J.K.L
   s=SIP Call
   c=IN IP4 I.J.K.L
   t=0 0
   m=audio 17046 RTP/AVP 18 19 101 100
   c=IN IP4 I.J.K.L
   a=rtpmap:18 G729/8000
   a=rtpmap:19 CN/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:100 X-NSE/8000
   a=fmtp:100 192-194

Sent Message :

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK9E31DBF
   From: "Test 2" <sip:xxxxxxxx75 at A.B.C.D>;tag=65BCD9B0-115E
   To: <sip:xxxxx02 at A.B.C.D>;tag=XN94Nrcp1yt7S
   Call-ID: A4A3C9E0-855A11E0-8237C79A-B2A2FA42 at E.F.G.H
   CSeq: 102 INVITE
   Contact: <sip:xxxxx02 at A.B.C.D:5060;transport=udp>
   User-Agent: GCI
   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
   Min-SE: 1800
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 333
   Remote-Party-ID: "xxxxx02" <sip:xxxxx02 at A.B.C.D>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 3912792884 3912792885 IN IP4 A.B.C.D
   s=FreeSWITCH
   c=IN IP4 A.B.C.D
   t=0 0
   m=audio 27728 RTP/AVP 18 100 19 101
   c=IN IP4 A.B.C.D
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:100 X-NSE/8000
   a=fmtp:100 192-194
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Monday, May 23, 2011 09:42
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] rtpmap line missing on answer

Sean,

can you show us the same packets, but for an incoming call (that works, if I understood you correctly) ?

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 24/05/2011 ? 01:53, Sean Eichhorn a ?crit :


Yeah, they?re negotiating the same codec on both sides.
The problem is the way Cisco handles the negotiation.  Here?s an example of what I see in Freeswitch:

Received from CALLED :

  SIP/2.0 200 OK
   Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bKXZZ18p8tD03cS
   From: "pending" <sip:xxxxxxxx02 at A.B.C.D>;tag=NH1HrNe744HSB
   To: <sip:sean at E.F.G.H:5060>;tag=622555CC-408
   Date: Mon, 23 May 2011 23:50:56 GMT
   Call-ID: 69258f8c-0038-122f-19ba-000c29c18d38
   CSeq: 12760849 INVITE
   Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
   Allow-Events: telephone-event
   Contact: <sip:xxxxxxxx75 at E.F.G.H:5060>
   Supported: replaces
   Supported: sdp-anat
   Server: Cisco-SIPGateway/IOS-12.x
   Content-Type: application/sdp
   Content-Disposition: session;handling=required
   Content-Length: 311

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 4080 7999 IN IP4 E.F.G.H
   s=SIP Call
   c=IN IP4 E.F.G.H
   t=0 0
   m=audio 17122 RTP/AVP 0 19 101 100
   c=IN IP4 E.F.G.H
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:100 X-NSE/8000
   a=fmtp:100 192-194

Sent to CALLING :
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP I.J.K.L:5060;branch=z9hG4bK4C2B6C64
   From: <sip:xxxxxxxx02@ I.J.K.L >;tag=9020153C-21B3
   To: <sip:xxxxxxxx75 at A.B.C.D>;tag=m87rptX37UU6F
   Call-ID: AFADEF2E-84CC11E0-B1D4F23A-27804614@ I.J.K.L
   CSeq: 101 INVITE
   Contact: <sip:xxxxxxxx75@ A.B.C.D:5060;transport=udp>
   User-Agent: GCI
   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
   Min-SE: 1800
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 261
   Remote-Party-ID: "Outbound Call" <sip:sean@ A.B.C.D >;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1995227603 1995227604 IN IP4 A.B.C.D
   s=FreeSWITCH
   c=IN IP4 A.B.C.D
   t=0 0
   m=audio 24636 RTP/AVP 0 19 101
   c=IN IP4 A.B.C.D
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16

This is not the case when the call direction is reversed.  Like I said, the only difference appears to be that one direction is responding with a 183 while this one is responding with a 200.

_______________________________________________
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  Tue May 24 20:34:00 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 24 May 2011 12:34:00 -0400
Subject: [Freeswitch-users] Handling DTMF noinput and nomatch
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC12B@VMBX102.ihostexchange.net>

Hi,

The following javascript file asks callers to enter their 4 digit PINs.
I am wondering if an event will be fired when a caller doesn't enter anything or enter just 3 digits.
If not, what is the recommended way of handling those cases? Thanks.

   function on_dtmf(session, type, digits, arg)
   {
       dtmf_digits += digits.digit;
       return(false);
   }

   session.answer();

   while (session.ready()) {
       dtmf_digits = "";
       session.flushDigits();
       session.speak("flite", "kal", 'please enter your 4 digit pin', on_dtmf);
       dtmf_digits = session.getDigits(4, "", 5000, 1000, 10000);
       console_log("pin=" + dtmf_digits + "\n");
   }


From boris at tagnet.ru  Tue May 24 20:40:05 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 24 May 2011 22:40:05 +0600
Subject: [Freeswitch-users] Handling DTMF noinput and nomatch
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC12B@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC12B@VMBX102.ihostexchange.net>
Message-ID: <4DDBDF65.2010804@tagnet.ru>

Hello!

     And what is the problem to check length of dtmf_digits variable? 
This is string variable, so You may use dtmf_digits.length .
> Hi,
>
> The following javascript file asks callers to enter their 4 digit PINs.
> I am wondering if an event will be fired when a caller doesn't enter anything or enter just 3 digits.
> If not, what is the recommended way of handling those cases? Thanks.
>
>     function on_dtmf(session, type, digits, arg)
>     {
>         dtmf_digits += digits.digit;
>         return(false);
>     }
>
>     session.answer();
>
>     while (session.ready()) {
>         dtmf_digits = "";
>         session.flushDigits();
>         session.speak("flite", "kal", 'please enter your 4 digit pin', on_dtmf);
>         dtmf_digits = session.getDigits(4, "", 5000, 1000, 10000);
>         console_log("pin=" + dtmf_digits + "\n");
>     }
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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




From liam at intersys.co.nz  Tue May 24 10:17:59 2011
From: liam at intersys.co.nz (Liam Farr)
Date: Tue, 24 May 2011 18:17:59 +1200
Subject: [Freeswitch-users] Setting gateway username dynamically
Message-ID: <da470c84f0fd55fce3ec812529064570@mail.gmail.com>

Hi,



I have a sip provider who requires me to set the username to be the same as
the outbound caller id for the gateway (trunk) to them.



Is there a way to do this dynamically for each call?



e.g.



<include>

    <gateway name="SIP-Provider">

      <param name="username" value="${outbound_caller_id_number}"/>

      <param name="password" value="password"/>

      <param name="realm" value="1.2.3.4"/>

      <param name="proxy" value="1.2.3.4"/>

      <param name="expire-seconds" value="800"/>

      <param name="register" value="false"/>

      <param name="retry-seconds" value="60"/>

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

    </gateway>

</include>





Cheers



Liam
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From jmoran at secureachsystems.com  Tue May 24 18:55:31 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Tue, 24 May 2011 10:55:31 -0400
Subject: [Freeswitch-users] Error playing TTS using flite
References: <33095823FD21DF429B481B5163264B7950AC3ABFC7@VMBX102.ihostexchange.net>
Message-ID: <361E98F99D3CC3439EED59BC1924ED69507A77@SERVER2003.SecuReachSystems.local>

Yungwei, I *think* sayPhrase is not supported by flite - only by other
TTS engines. 
Try calling it in this javascript format:
session.speak(ttsLib,ttsVoice,text);

Jason Moran

-----Original Message-----
From: Yungwei Chen [mailto:yungwei at resolvity.com] 
Sent: Monday, May 23, 2011 7:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Error playing TTS using flite

Hi,

I am testing tts using flite against freeswitch (snapshot downloaded
last week) on CentOS 5.
I'm getting "Invalid speech module [cepstral]!" and "Invalid TTS module"
errors when dialing 9911 from my SIP client.
What am I missing here? Thanks.

...
Dialplan: sofia/internal/1000 at 192.168.216.18 parsing [default->test1]
continue=false
Dialplan: sofia/internal/1000 at 192.168.216.18 Regex (PASS) [test1]
destination_number(9911) =~ /^9911$/ break=on-false
Dialplan: sofia/internal/1000 at 192.168.216.18 Action
javascript(dtmf_test.js)  --> Freeswitch figured out dtmf_test.js is the
action to perform
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/1000 at 192.168.216.18) State Change CS_ROUTING ->
CS_EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] switch_core_session.c:1114 Send
signal sofia/internal/1000 at 192.168.216.18 [BREAK]
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:364
(sofia/internal/1000 at 192.168.216.18) State ROUTING going to sleep
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1000 at 192.168.216.18) Running State Change CS_EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:371
(sofia/internal/1000 at 192.168.216.18) State EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] mod_sofia.c:240
sofia/internal/1000 at 192.168.216.18 SOFIA EXECUTE
2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1000 at 192.168.216.18 Standard EXECUTE
EXECUTE sofia/internal/1000 at 192.168.216.18 set(open=true)
2011-05-23 18:13:09.859007 [DEBUG] mod_dptools.c:1060
sofia/internal/1000 at 192.168.216.18 SET [open]=[true]
EXECUTE sofia/internal/1000 at 192.168.216.18
hash(insert/192.168.216.18-spymap/1000/399178b8-8592-11e0-ae8d-e9a6940f4
ae1)
EXECUTE sofia/internal/1000 at 192.168.216.18
hash(insert/192.168.216.18-last_dial/1000/9911)
EXECUTE sofia/internal/1000 at 192.168.216.18
hash(insert/192.168.216.18-last_dial/global/399178b8-8592-11e0-ae8d-e9a6
940f4ae1)
EXECUTE sofia/internal/1000 at 192.168.216.18 set(RFC2822_DATE=Mon, 23 May
2011 18:13:09 -0500)
2011-05-23 18:13:09.859007 [DEBUG] mod_dptools.c:1060
sofia/internal/1000 at 192.168.216.18 SET [RFC2822_DATE]=[Mon, 23 May 2011
18:13:09 -0500]
EXECUTE sofia/internal/1000 at 192.168.216.18 javascript(dtmf_test.js)
2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3034 AUDIO RTP
[sofia/internal/1000 at 192.168.216.18] 192.168.216.18 port 31886 ->
192.168.216.9 port 5062 codec: 8 ms: 20
2011-05-23 18:13:09.879013 [DEBUG] switch_rtp.c:1636 Starting timer
[soft] 160 bytes per 20ms
2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3296 Set 2833 dtmf send
payload to 101
2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3301 Set 2833 dtmf
receive payload to 101
2011-05-23 18:13:09.879013 [DEBUG] mod_sofia.c:681 Local SDP
sofia/internal/1000 at 192.168.216.18:
v=0
o=FreeSWITCH 1306160503 1306160504 IN IP4 192.168.216.18
s=FreeSWITCH
c=IN IP4 192.168.216.18
t=0 0
m=audio 31886 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
a=sendrecv

2011-05-23 18:13:09.879013 [DEBUG] switch_core_session.c:707 Send signal
sofia/internal/1000 at 192.168.216.18 [BREAK]
2011-05-23 18:13:09.879013 [DEBUG] switch_channel.c:2859
(sofia/internal/1000 at 192.168.216.18) Callstate Change RINGING -> ACTIVE
2011-05-23 18:13:09.879013 [NOTICE] mod_spidermonkey.c:2068 Channel
[sofia/internal/1000 at 192.168.216.18] has been answered
2011-05-23 18:13:09.879013 [DEBUG] dtmf_test.js:34 sayivrmenu:
menu=[mainmenu] validdigits=[0123#]
2011-05-23 18:13:09.879013 [DEBUG] sofia.c:4770 Channel
sofia/internal/1000 at 192.168.216.18 entering state [completed][200]
2011-05-23 18:13:09.879013 [DEBUG] switch_ivr_play_say.c:244 Handle
speak-text:[Press 0 for the Operator.] (en:en)
2011-05-23 18:13:09.879013 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]! -->cepstral is never installed. Why did Freeswitch
try to use cepstral?
2011-05-23 18:13:09.879013 [ERR] switch_ivr_play_say.c:2361 Invalid TTS
module!
2011-05-23 18:13:09.879013 [DEBUG] sofia.c:4770 Channel
sofia/internal/1000 at 192.168.216.18 entering state [ready][200]
2011-05-23 18:13:09.939014 [DEBUG] switch_rtp.c:3104 Correct ip/port
confirmed.
2011-05-23 18:13:15.979391 [DEBUG] switch_ivr_play_say.c:244 Handle
speak-text:[Press 0 for the Operator.] (en:en)
2011-05-23 18:13:15.979391 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!
2011-05-23 18:13:15.979391 [ERR] switch_ivr_play_say.c:2361 Invalid TTS
module!
2011-05-23 18:13:22.079772 [DEBUG] switch_ivr_play_say.c:244 Handle
speak-text:[Press 0 for the Operator.] (en:en)
2011-05-23 18:13:22.079772 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!
2011-05-23 18:13:22.079772 [ERR] switch_ivr_play_say.c:2361 Invalid TTS
module!

During installatio, I uncommented the following in modules.conf so
mod_flite is installed. mod_cepstral remains commented.
asr_tts/mod_unimrcp
asr_tts/mod_flite
asr_tts/mod_pocketsphinx

After installation, I uncommented mod_flite in
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml
    <!-- ASR /TTS -->
    <load module="mod_flite"/>
    <!-- <load module="mod_pocketsphinx"/> -->
    <!-- <load module="mod_cepstral"/> -->
    <!-- <load module="mod_tts_commandline"/> -->
    <!-- <load module="mod_rss"/> -->

Here's my dial-plan
  <extension name="test1">
        <condition field="destination_number" expression="^9911$">
                <action application="set" data="tts_engine=flite"/>
                <action application="set" data="tts_voice=kal"/>
                <action application="javascript" data="dtmf_test.js" />
        </condition>
  </extension>

dtmf_test.js is copied from
http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js to
/usr/local/freeswitch/scripts folder.
The phrase macro is also copied from that link to
/usr/local/freeswitch/conf/lang/en/demo folder.






From msc at freeswitch.org  Tue May 24 20:55:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 24 May 2011 09:55:10 -0700
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTimxzhdyq3Jp=wOGviF=JfoW=BPWRg@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTimxzhdyq3Jp=wOGviF=JfoW=BPWRg@mail.gmail.com>
Message-ID: <BANLkTi=967OF3e-diLDMeBuQ27q_vtaL+Q@mail.gmail.com>

If your DTMF detection is sketchy then I would record the calls and listen
to them. You need to make sure that the tones are coming through cleanly.
>From the symptoms you've described I have to wonder if the issue is with the
network or provider and not your FreeSWITCH box.

-MC

On Tue, May 24, 2011 at 3:29 AM, Sidharth Kshatriya <sid.kshatriya at gmail.com
> wrote:

> Thanks for your reply, Michael. I have tried doing a start dtmf but then
> I've noticed:
>
> 1.  My skype key presses get unreliable...
> 2. The dtmf detection is patchy with inband -- works sometimes and doesn't
> work sometimes :-)
>
> Any settings to keep in mind apart just adding a start dtmf ?
>
> Thanks,
>
> Sidharth
>
> On Mon, May 23, 2011 at 8:54 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Sidharth,
>>
>> A mobile phone will always send DTMFs in-band, so you need to be ready for
>> that scenario. I recommend you add this to your dialplan for inbound calls:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
>>
>> Test various scenarios and make sure they work - call from Skype, from
>> mobile phone, from a land line, etc. Let us know what happens.
>>
>> -MC
>>
>>  On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya <
>> sid.kshatriya at gmail.com> wrote:
>>
>>>  Dear Friends,
>>>
>>> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
>>> list). While I'm generally happy with their service there seems to be one
>>> fatal problem: My IVR does not recognize DTMF!
>>>
>>> I have set <param name="dtmf-type" value="rfc2833"/> in both
>>> sip_profile/internal.xml and sip_profile/external.xml
>>>
>>> The symptom of the problem is that making a call via skype will *always*make the IVR recognize the DTMF while using something like a mobile phone
>>> *almost always* won't!
>>>
>>> I've tried in-band detection too. I'm making international calls into my
>>> IVR and the reliability of the in-band detection is not so good, possibly
>>> because of the quality of the call.
>>>
>>> Can someone please help me?
>>>
>>> Thanks,
>>>
>>> Sidharth
>>>
>>> --
>>> Sidharth Kshatriya
>>> www.sidk.info
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 24 21:03:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 24 May 2011 10:03:58 -0700
Subject: [Freeswitch-users] How to troubleshoot "503 Maximum Calls In
	Progress"?
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F1220CB5293C@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F1220CB5293C@VMBX113.ihostexchange.net>
Message-ID: <BANLkTikuRXGQeA5NRceb7XunqeXgX5p-PQ@mail.gmail.com>

Have you been able to reproduce this issue? Be sure to get the SIP traffic
captured along with the debug log.
-MC

On Mon, May 23, 2011 at 3:49 PM, Chris Fowler <chris at fowler.cc> wrote:

> Hi,
>
>
>
> On our Production FreeSWITCH box:
>
>
>
> FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500)
>
>
>
> UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, 751
> microseconds
>
> 12519 session(s) since startup
>
> 2 session(s) 0/30
>
> 1000 session(s) max
>
> min idle cpu 0.00/100.00
>
>
>
> At 14:30:37 a reloadxml command was issued.
>
> At 14:30:43 the box started rejecting all calls with 503 error.
>
> At 14:38 it started working again.
>
>
>
>
>
> Looking at the code I see three conditions can trigger this behavior:
>
>
>
>               if (sess_count >= sess_max || !sofia_test_pflag(profile,
> PFLAG_RUNNING) || !switch_core_ready()) {
>
>                         nua_respond(nh, 503, "Maximum Calls In Progress",
> SIPTAG_RETRY_AFTER_STR("300"), TAG_END());
>
>
>
>                         switch_log_printf(SWITCH_CHANNEL_LOG,
> SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n");
>
>
>
>                         goto done;
>
>                }
>
>
>
> Logs show it wasn?t lack of available sessions.  How can
> sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and
> busy the system?
>
>
>
> Thoughts? Thx, Chris.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kerem.erciyes at gmail.com  Tue May 24 22:02:41 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Tue, 24 May 2011 21:02:41 +0300
Subject: [Freeswitch-users] DTMF problems
In-Reply-To: <4DDBB93B.2040604@tagnet.ru>
References: <4DDBA05C.6060204@tagnet.ru> <op.vvzjapbizwj8qf@solidsnake.local>
	<4DDBAE2B.6080208@tagnet.ru> <op.vvzmq7lyzwj8qf@solidsnake.local>
	<4DDBB93B.2040604@tagnet.ru>
Message-ID: <op.vvzy2rnezwj8qf@solidsnake.local>

No problem Boris. I never had much success with in-band DTMF but as M.  
Collins suggested in a separate thread there are some cases where you  
really need in-band detection (i.e. calls via GSM networks always send  
in-band DTMF), I suggest you try recording the calls and try to make sense  
of how much of the DTMF tone is being received, if there is stuttering due  
to echo cancellation or static on the line, etc...

On Tue, 24 May 2011 16:57:15 +0300, Boris Kovalenko <boris at tagnet.ru>  
wrote:

> Hello!
>
>      Thank You Kerem, I'll try. The main problem is that there is a
> little percent of wrong DTMF and a very nervous customer.
> Also what if not to use dtmf-relay on Cisco at all? What kind of
> problems may I have? I use G711alaw inside a network. Some transit calls
> (ie Internet -> PSTN) may use G729.
>
>> Yes Boris, try setting it lower incrementally and see if it helps. There
>> might be problems related to gain + echo on the line that confuses DTMF
>> detection, so check if the line has echo as well. Also sometimes I have
>> seen that the remote calling party has a POTS line with too much static  
>> it
>> can confuse DTMF detection. On other VoIP systems we solved this problem
>> by spying on the channel e.g.
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be
>> helpful in debugging the situation.
>>
>> On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko<boris at tagnet.ru>
>> wrote:
>>
<snip>
>> <snip>
<snip>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>




From yungwei at resolvity.com  Tue May 24 22:58:43 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 24 May 2011 14:58:43 -0400
Subject: [Freeswitch-users] Playing a wav file on a remote web server
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>

Hi,

I am currently using the following javascript snippet to play a wav file on a remote web server.
I'm wondering if there's another way to do it without involving a local file, just like mod_shout. 
Thanks.

 function on_dtmf(session, type, digits, arg)
 {
     dtmf_digits += digits.digit;
     console_log("dtmf_digits=" + dtmf_digits + "\n");
     return(true);
 }

 var ttsUrl = "http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin";
 var file = "/tmp/tts.wav";
 fetchURLFile(ttsUrl,file);
 session.streamFile(file, on_dtmf);


From kris at kriskinc.com  Tue May 24 23:01:55 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 24 May 2011 14:01:55 -0500
Subject: [Freeswitch-users] Setting gateway username dynamically
In-Reply-To: <da470c84f0fd55fce3ec812529064570@mail.gmail.com>
References: <da470c84f0fd55fce3ec812529064570@mail.gmail.com>
Message-ID: <BANLkTikknTRaP+PTs4EEbrU+ib+QDi9AmA@mail.gmail.com>

Don't use a gateway at all:

http://wiki.freeswitch.org/wiki/Variable_sip_auth_username

On Tue, May 24, 2011 at 1:17 AM, Liam Farr <liam at intersys.co.nz> wrote:
> Hi,
>
>
>
> I have a sip provider who requires me to set the username to be the same as
> the outbound caller id for the gateway (trunk) to them.
>
>
>
> Is there a way to do this dynamically for each call?
>
>
>
> e.g.
>
>
>
> <include>
>
> ??? <gateway name="SIP-Provider">
>
> ????? <param name="username" value="${outbound_caller_id_number}"/>
>
> ????? <param name="password" value="password"/>
>
> ????? <param name="realm" value="1.2.3.4"/>
>
> ????? <param name="proxy" value="1.2.3.4"/>
>
> ????? <param name="expire-seconds" value="800"/>
>
> ????? <param name="register" value="false"/>
>
> ????? <param name="retry-seconds" value="60"/>
>
> ????? <param name="context" value="public"/>
>
> ??? </gateway>
>
> </include>
>
>
>
>
>
> Cheers
>
>
>
> Liam
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 kris at kriskinc.com  Tue May 24 23:08:43 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 24 May 2011 14:08:43 -0500
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
Message-ID: <BANLkTi=VeCq=grE7T7iYQfYWrp9f8w+vSg@mail.gmail.com>

Michael,

  I'm not really sure what you mean by this...

  The DTMF method used between an endpoint and a SIP carrier is
largely determined by the SIP carrier, not the carrier or device on
the far end.

  On the PSTN DTMF is always inband.  If configured for out of band
DTMF it's up to the various media gateways in use by VoIP providers to
detect the inband DTMF, squelch it from the G711 input audio, and
create out of band events.

  Of course if you're configured for inband and using G711 it will be
inband all the way through.

P.S. - Cell phones either use AMR or EVRC these days and I would guess
they're using out of band DTMF too.  I've seen some documentation
claiming support for inband DTMF on either but that seems strange to
me...

On Mon, May 23, 2011 at 10:24 AM, Michael Collins <msc at freeswitch.org> wrote:
> Sidharth,
> A mobile phone will always send DTMFs in-band, so you need to be ready for
> that scenario. I recommend you add this to your dialplan for inbound calls:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> Test various scenarios and make sure they work - call from Skype, from
> mobile phone, from a land line, etc. Let us know what happens.
> -MC
> On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya
> <sid.kshatriya at gmail.com> wrote:
>>
>> Dear Friends,
>>
>> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
>> list). While I'm generally happy with their service there seems to be one
>> fatal problem: My IVR does not recognize DTMF!
>>
>> I have set <param name="dtmf-type" value="rfc2833"/> in both
>> sip_profile/internal.xml and sip_profile/external.xml
>>
>> The symptom of the problem is that making a call via skype will always
>> make the IVR recognize the DTMF while using something like a mobile phone
>> almost always won't!
>>
>> I've tried in-band detection too. I'm making international calls into my
>> IVR and the reliability of the in-band detection is not so good, possibly
>> because of the quality of the call.
>>
>> Can someone please help me?
>>
>> Thanks,
>>
>> Sidharth
>>
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Tue May 24 23:22:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 24 May 2011 12:22:15 -0700
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTi=VeCq=grE7T7iYQfYWrp9f8w+vSg@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTi=VeCq=grE7T7iYQfYWrp9f8w+vSg@mail.gmail.com>
Message-ID: <BANLkTi=GV_FdH-3d-wamOryLJXcXEM2smw@mail.gmail.com>

I guess what I mean is this: he needs to listen to what ACTUALLY comes down
the line. There's no substitute for the human ear. If the tones coming down
the line are not clean or otherwise are not recognizable then I would
suggest that looking for a solution inside of FreeSWITCH itself may not be
too helpful...

That being said, if the DTMFs are coming in cleanly then there's definitely
something hinky going on inside his system.

-MC

On Tue, May 24, 2011 at 12:08 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Michael,
>
>  I'm not really sure what you mean by this...
>
>  The DTMF method used between an endpoint and a SIP carrier is
> largely determined by the SIP carrier, not the carrier or device on
> the far end.
>
>  On the PSTN DTMF is always inband.  If configured for out of band
> DTMF it's up to the various media gateways in use by VoIP providers to
> detect the inband DTMF, squelch it from the G711 input audio, and
> create out of band events.
>
>  Of course if you're configured for inband and using G711 it will be
> inband all the way through.
>
> P.S. - Cell phones either use AMR or EVRC these days and I would guess
> they're using out of band DTMF too.  I've seen some documentation
> claiming support for inband DTMF on either but that seems strange to
> me...
>
> On Mon, May 23, 2011 at 10:24 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Sidharth,
> > A mobile phone will always send DTMFs in-band, so you need to be ready
> for
> > that scenario. I recommend you add this to your dialplan for inbound
> calls:
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> > Test various scenarios and make sure they work - call from Skype, from
> > mobile phone, from a land line, etc. Let us know what happens.
> > -MC
> > On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya
> > <sid.kshatriya at gmail.com> wrote:
> >>
> >> Dear Friends,
> >>
> >> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
> >> list). While I'm generally happy with their service there seems to be
> one
> >> fatal problem: My IVR does not recognize DTMF!
> >>
> >> I have set <param name="dtmf-type" value="rfc2833"/> in both
> >> sip_profile/internal.xml and sip_profile/external.xml
> >>
> >> The symptom of the problem is that making a call via skype will always
> >> make the IVR recognize the DTMF while using something like a mobile
> phone
> >> almost always won't!
> >>
> >> I've tried in-band detection too. I'm making international calls into my
> >> IVR and the reliability of the in-band detection is not so good,
> possibly
> >> because of the quality of the call.
> >>
> >> Can someone please help me?
> >>
> >> Thanks,
> >>
> >> Sidharth
> >>
> >> --
> >> Sidharth Kshatriya
> >> www.sidk.info
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 rhuddleston at gmail.com  Tue May 24 23:29:01 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 24 May 2011 15:29:01 -0400
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
	rfc2833 not working!
In-Reply-To: <BANLkTi=GV_FdH-3d-wamOryLJXcXEM2smw@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTi=VeCq=grE7T7iYQfYWrp9f8w+vSg@mail.gmail.com>
	<BANLkTi=GV_FdH-3d-wamOryLJXcXEM2smw@mail.gmail.com>
Message-ID: <064501cc1a48$d6ad9790$8408c6b0$@com>

And in case you are like me and can't tell the difference between a - and a
.

. --- .

 

Download a good DTMF decoder

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, May 24, 2011 3:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF Problems! In-band detection patchy,
rfc2833 not working!

 

I guess what I mean is this: he needs to listen to what ACTUALLY comes down
the line. There's no substitute for the human ear. If the tones coming down
the line are not clean or otherwise are not recognizable then I would
suggest that looking for a solution inside of FreeSWITCH itself may not be
too helpful...

 

That being said, if the DTMFs are coming in cleanly then there's definitely
something hinky going on inside his system.

 

-MC

 

On Tue, May 24, 2011 at 12:08 PM, Kristian Kielhofner <kris at kriskinc.com>
wrote:

Michael,

 I'm not really sure what you mean by this...

 The DTMF method used between an endpoint and a SIP carrier is
largely determined by the SIP carrier, not the carrier or device on
the far end.

 On the PSTN DTMF is always inband.  If configured for out of band
DTMF it's up to the various media gateways in use by VoIP providers to
detect the inband DTMF, squelch it from the G711 input audio, and
create out of band events.

 Of course if you're configured for inband and using G711 it will be
inband all the way through.

P.S. - Cell phones either use AMR or EVRC these days and I would guess
they're using out of band DTMF too.  I've seen some documentation
claiming support for inband DTMF on either but that seems strange to
me...


On Mon, May 23, 2011 at 10:24 AM, Michael Collins <msc at freeswitch.org>
wrote:
> Sidharth,
> A mobile phone will always send DTMFs in-band, so you need to be ready for
> that scenario. I recommend you add this to your dialplan for inbound
calls:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
> Test various scenarios and make sure they work - call from Skype, from
> mobile phone, from a land line, etc. Let us know what happens.
> -MC
> On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya
> <sid.kshatriya at gmail.com> wrote:
>>
>> Dear Friends,
>>
>> I'm using a voip carrier Voicenetwork.ca (recommended by someone on this
>> list). While I'm generally happy with their service there seems to be one
>> fatal problem: My IVR does not recognize DTMF!
>>
>> I have set <param name="dtmf-type" value="rfc2833"/> in both
>> sip_profile/internal.xml and sip_profile/external.xml
>>
>> The symptom of the problem is that making a call via skype will always
>> make the IVR recognize the DTMF while using something like a mobile phone
>> almost always won't!
>>
>> I've tried in-band detection too. I'm making international calls into my
>> IVR and the reliability of the in-band detection is not so good, possibly
>> because of the quality of the call.
>>
>> Can someone please help me?
>>
>> Thanks,
>>
>> Sidharth
>>
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Wed May 25 01:12:01 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 24 May 2011 23:12:01 +0200
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <7EB00560C8AFF348A7EA76EFD3DC5598332C0360@dtn1mbx01.gci.com>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
	<BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035F@dtn1mbx01.gci.com>
	<DF5E4E14-2338-4871-A73E-2291537069B9@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C0360@dtn1mbx01.gci.com>
Message-ID: <E8079B89-5EDC-4130-A331-DA923CBD292D@ipeva.fr>

Sean,

can you check the configuration of the SIP profile used by both legs ?
Perhaps you have late-negotiation enabled in one of them ?

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

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

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




Le 24/05/2011 ? 18:31, Sean Eichhorn a ?crit :

> Yep, no problem.
> Here it is.  Same endpoints.  The only thing that changed is the call direction.
>  
> Received Message :
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bK5Qtv3B04eS46g
>    From: "Test 2" <sip:xxxxxxxx75 at A.B.C.D>;tag=yy2XQKXSy7gtN
>    To: <sip:xxxxx02 at I.J.K.L>;tag=93BEF648-1A8B
>    Date: Tue, 24 May 2011 16:37:23 GMT
>    Call-ID: b7967af6-00c4-122f-19ba-000c29c18d38
>    CSeq: 12790979 INVITE
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
>    Allow-Events: telephone-event
>    Remote-Party-ID: <sip:xxxxx02 at I.J.K.L>;party=called;screen=no;privacy=off
>    Contact: <sip:xxxxx02 at I.J.K.L:5060>
>    Supported: replaces
>    Supported: sdp-anat
>    Server: Cisco-SIPGateway/IOS-12.x
>    Content-Type: application/sdp
>    Content-Disposition: session;handling=required
>    Content-Length: 334
>  
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 6942 3511 IN IP4 I.J.K.L
>    s=SIP Call
>    c=IN IP4 I.J.K.L
>    t=0 0
>    m=audio 17046 RTP/AVP 18 19 101 100
>    c=IN IP4 I.J.K.L
>    a=rtpmap:18 G729/8000
>    a=rtpmap:19 CN/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:100 X-NSE/8000
>    a=fmtp:100 192-194
>  
> Sent Message :
>  
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK9E31DBF
>    From: "Test 2" <sip:xxxxxxxx75 at A.B.C.D>;tag=65BCD9B0-115E
>    To: <sip:xxxxx02 at A.B.C.D>;tag=XN94Nrcp1yt7S
>    Call-ID: A4A3C9E0-855A11E0-8237C79A-B2A2FA42 at E.F.G.H
>    CSeq: 102 INVITE
>    Contact: <sip:xxxxx02 at A.B.C.D:5060;transport=udp>
>    User-Agent: GCI
>    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
>    Min-SE: 1800
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 333
>    Remote-Party-ID: "xxxxx02" <sip:xxxxx02 at A.B.C.D>;party=calling;privacy=off;screen=no
>  
>    v=0
>    o=FreeSWITCH 3912792884 3912792885 IN IP4 A.B.C.D
>    s=FreeSWITCH
>    c=IN IP4 A.B.C.D
>    t=0 0
>    m=audio 27728 RTP/AVP 18 100 19 101
>    c=IN IP4 A.B.C.D
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:100 X-NSE/8000
>    a=fmtp:100 192-194
>    a=rtpmap:19 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
> Sent: Monday, May 23, 2011 09:42
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] rtpmap line missing on answer
>  
> Sean,
>  
> can you show us the same packets, but for an incoming call (that works, if I understood you correctly) ?
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
>  
> Le 24/05/2011 ? 01:53, Sean Eichhorn a ?crit :
> 
> 
> Yeah, they?re negotiating the same codec on both sides.
> The problem is the way Cisco handles the negotiation.  Here?s an example of what I see in Freeswitch:
>  
> Received from CALLED :
>  
>   SIP/2.0 200 OK
>    Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bKXZZ18p8tD03cS
>    From: "pending" <sip:xxxxxxxx02 at A.B.C.D>;tag=NH1HrNe744HSB
>    To: <sip:sean at E.F.G.H:5060>;tag=622555CC-408
>    Date: Mon, 23 May 2011 23:50:56 GMT
>    Call-ID: 69258f8c-0038-122f-19ba-000c29c18d38
>    CSeq: 12760849 INVITE
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
>    Allow-Events: telephone-event
>    Contact: <sip:xxxxxxxx75 at E.F.G.H:5060>
>    Supported: replaces
>    Supported: sdp-anat
>    Server: Cisco-SIPGateway/IOS-12.x
>    Content-Type: application/sdp
>    Content-Disposition: session;handling=required
>    Content-Length: 311
>  
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 4080 7999 IN IP4 E.F.G.H
>    s=SIP Call
>    c=IN IP4 E.F.G.H
>    t=0 0
>    m=audio 17122 RTP/AVP 0 19 101 100
>    c=IN IP4 E.F.G.H
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:19 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:100 X-NSE/8000
>    a=fmtp:100 192-194
>  
> Sent to CALLING :
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP I.J.K.L:5060;branch=z9hG4bK4C2B6C64
>    From: <sip:xxxxxxxx02@ I.J.K.L >;tag=9020153C-21B3
>    To: <sip:xxxxxxxx75 at A.B.C.D>;tag=m87rptX37UU6F
>    Call-ID: AFADEF2E-84CC11E0-B1D4F23A-27804614@ I.J.K.L
>    CSeq: 101 INVITE
>    Contact: <sip:xxxxxxxx75@ A.B.C.D:5060;transport=udp>
>    User-Agent: GCI
>    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
>    Min-SE: 1800
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 261
>    Remote-Party-ID: "Outbound Call" <sip:sean@ A.B.C.D >;party=calling;privacy=off;screen=no
>  
>    v=0
>    o=FreeSWITCH 1995227603 1995227604 IN IP4 A.B.C.D
>    s=FreeSWITCH
>    c=IN IP4 A.B.C.D
>    t=0 0
>    m=audio 24636 RTP/AVP 0 19 101
>    c=IN IP4 A.B.C.D
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:19 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>  
> This is not the case when the call direction is reversed.  Like I said, the only difference appears to be that one direction is responding with a 183 while this one is responding with a 200.
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From seichhorn at gci.com  Wed May 25 01:33:28 2011
From: seichhorn at gci.com (Sean Eichhorn)
Date: Tue, 24 May 2011 13:33:28 -0800
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <E8079B89-5EDC-4130-A331-DA923CBD292D@ipeva.fr>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
	<BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035F@dtn1mbx01.gci.com>
	<DF5E4E14-2338-4871-A73E-2291537069B9@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C0360@dtn1mbx01.gci.com>
	<E8079B89-5EDC-4130-A331-DA923CBD292D@ipeva.fr>
Message-ID: <7EB00560C8AFF348A7EA76EFD3DC5598332C0367@dtn1mbx01.gci.com>

Actually, I just found another interesting piece of the puzzle.
On the outbound call (that works), the gateway responds with a 183/SDP message.
The inbound call (that doesn?t work) returns a 180 Ringing/NO SDP message.
Both legs are using the same profile, so I don?t think that?s an issue.  For some reason though, if a 183/SDP is sent, Freeswitch forwards the rtpmap attribute.  If no 183 is detected, it deletes the attribute.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Tuesday, May 24, 2011 01:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] rtpmap line missing on answer

Sean,

can you check the configuration of the SIP profile used by both legs ?
Perhaps you have late-negotiation enabled in one of them ?

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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


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From bobc at devassert.com  Wed May 25 02:51:16 2011
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 25 May 2011 10:51:16 +1200
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTimxzhdyq3Jp=wOGviF=JfoW=BPWRg@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTimxzhdyq3Jp=wOGviF=JfoW=BPWRg@mail.gmail.com>
Message-ID: <BANLkTi=eini48X7tKDpbq2jdh0BR6RCt_w@mail.gmail.com>

Just my two cents, but we had similar problems with mobile dtmf, and
found the best combination was to set INFO rather than RFC2833 and to
always use start_dtmf .

That also allowed the soft phone clients to operate properly as well,
and if the client is configurable it is best to tell the soft phone
what type of dtmf to use other wise we got the same symtoms as you
described, namely multiple key presses when you only press it once

Regards

Bob

On Tue, May 24, 2011 at 10:29 PM, Sidharth Kshatriya
<sid.kshatriya at gmail.com> wrote:
> Thanks for your reply, Michael. I have tried doing a start dtmf but then
> I've noticed:
> 1. ?My skype key presses get unreliable...
> 2. The dtmf detection is patchy with inband -- works sometimes and doesn't
> work sometimes :-)
>
> Any settings to keep in mind apart just adding a start dtmf ?
> Thanks,
> Sidharth
>


From curriegrad2004 at gmail.com  Wed May 25 03:30:24 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 24 May 2011 16:30:24 -0700
Subject: [Freeswitch-users] Scientific Linux 6 (RHEL6) and Tickless Kernel
	(and possibly Fedora releases)
Message-ID: <BANLkTikXxRmpJ-SuVLuH_dyp8V+_nGRksA@mail.gmail.com>

I was recently going through the configuration options on compiling a
custom kernel for Scientific Linux 6 (aka RHEL 6 Clone) and I happened
to find the kernel is being compiled to run tickless by default. Red
Hat's documentation also confirms that the Kernel is indeed tickless
by default:

http://docs.redhat.com/docs/en-US/Red_Hat_Enterprise_Linux/6/html/Power_Management_Guide/ASPM.html

Not knowing what a tickless meant, I went on and googled and found my
answer to what a tickless kernel is. Then I decided to search up on
tickless kernels with FreeSwitch and found out that 2 years ago
somebody brought up the tickless kernel issue on this mailing list
before and bkw did recommend in the past that users should disable the
tickless feature of the kernel. Fast forwarding today is it still the
case that when a user runs FreeSwitch, they should disable the
tickless feature of the kernel? If so, I'd probably be adding the grub
option of disabling the tickless feature with the kernel and adding
this tidbit of information to the wiki.


From liam at intersys.co.nz  Wed May 25 04:47:16 2011
From: liam at intersys.co.nz (Liam Farr)
Date: Wed, 25 May 2011 12:47:16 +1200
Subject: [Freeswitch-users] Setting gateway username dynamically
In-Reply-To: <BANLkTikknTRaP+PTs4EEbrU+ib+QDi9AmA@mail.gmail.com>
References: <da470c84f0fd55fce3ec812529064570@mail.gmail.com>
	<BANLkTikknTRaP+PTs4EEbrU+ib+QDi9AmA@mail.gmail.com>
Message-ID: <BANLkTimXYVTVmEq4dVm1sRTRDHxuy8joWg@mail.gmail.com>

Thanks,

Ended up using   <action application="bridge"
data="{sip_contact_user=${outbound_caller_id_number}}sofia/external/${
destination_number}@1.2.3.4:5060" /> in my outbound dialplan.

On Wed, May 25, 2011 at 7:01 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Don't use a gateway at all:
>
> http://wiki.freeswitch.org/wiki/Variable_sip_auth_username
>
> On Tue, May 24, 2011 at 1:17 AM, Liam Farr <liam at intersys.co.nz> wrote:
> > Hi,
> >
> >
> >
> > I have a sip provider who requires me to set the username to be the same
> as
> > the outbound caller id for the gateway (trunk) to them.
> >
> >
> >
> > Is there a way to do this dynamically for each call?
> >
> >
> >
> > e.g.
> >
> >
> >
> > <include>
> >
> >     <gateway name="SIP-Provider">
> >
> >       <param name="username" value="${outbound_caller_id_number}"/>
> >
> >       <param name="password" value="password"/>
> >
> >       <param name="realm" value="1.2.3.4"/>
> >
> >       <param name="proxy" value="1.2.3.4"/>
> >
> >       <param name="expire-seconds" value="800"/>
> >
> >       <param name="register" value="false"/>
> >
> >       <param name="retry-seconds" value="60"/>
> >
> >       <param name="context" value="public"/>
> >
> >     </gateway>
> >
> > </include>
> >
> >
> >
> >
> >
> > Cheers
> >
> >
> >
> > Liam
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>



-- 


Kind Regards


Liam Farr

Intersys Ltd
Email: liam at intersys.co.nz
Mobile: +64-22-6107884
Skype: nz_liam
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From liam at intersys.co.nz  Wed May 25 04:56:38 2011
From: liam at intersys.co.nz (Liam Farr)
Date: Wed, 25 May 2011 12:56:38 +1200
Subject: [Freeswitch-users] Outbound SIP invites without codec list
Message-ID: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>

Hi,

My outbound SIP invites are going out without a codec list?
Could someone point me in the right direction on how to set / insert this?


send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:
   ------------------------------------------------------------------------
   INVITE sip:0800000000 at 1.2.3.4:5060 SIP/2.0
   Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr
   Max-Forwards: 69
   From: "Liam - Test" <sip:12345678 at 9.8.7.6>;tag=ryFQea59e6Upe
   To: <sip:0800000000 at 1.2.3.4:5060>
   Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43
   CSeq: 12805602 INVITE
   Contact: <sip:12345678 at 9.8.7.6:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 189
   X-FS-Support: update_display
   Remote-Party-ID: "Liam - Test" <sip:12345678 at 9.8.7.6
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6
   s=FreeSWITCH
   c=IN IP4 9.8.7.6
   t=0 0
   m=audio 18646 RTP/AVP 0 8 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   ------------------------------------------------------------------------


Thanks

Liam
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From frisch.alan at gmail.com  Wed May 25 07:17:14 2011
From: frisch.alan at gmail.com (Alan Frisch)
Date: Tue, 24 May 2011 23:17:14 -0400
Subject: [Freeswitch-users] Do ignore_display_updates=true only work on an
	Outgoing/B-Leg?
Message-ID: <BANLkTi=vBf0KDWo0eTWTOTZ6by+YuaohSQ@mail.gmail.com>

Scenario is two FS boxes (FS1 and FS2) and OpenSIPs.

FS1 sends Invite with "X-FS-Support: update_display" to FS2 via OpenSIPs
FS2 configured with "global_setvar ignore_display_updates=yes"

Even with ignore_display_updates enabled on FS2, FS2 responds with
"X-FS-Display-Name" and "X-FS-Display-Numbers".

FS1 is a box not under my administration, so is there a way to disable
this response on FS2?  I could easily strip the X-FS-Support header in
OpenSIPs to prevent this, but seems like a kludgy way of getting
things done.

Is there something I am overlooking?


From boris at tagnet.ru  Wed May 25 07:22:51 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 25 May 2011 09:22:51 +0600
Subject: [Freeswitch-users] DTMF problems
In-Reply-To: <op.vvzy2rnezwj8qf@solidsnake.local>
References: <4DDBA05C.6060204@tagnet.ru>
	<op.vvzjapbizwj8qf@solidsnake.local>	<4DDBAE2B.6080208@tagnet.ru>
	<op.vvzmq7lyzwj8qf@solidsnake.local>	<4DDBB93B.2040604@tagnet.ru>
	<op.vvzy2rnezwj8qf@solidsnake.local>
Message-ID: <4DDC760B.6080208@tagnet.ru>

Hello!

Hmm... we are talking about calls coming from PRI so (AFAIK) DTMF is 
always in-band, isn't? And Cisco is responsible to detect it and send 
out-of-band. IMHO bad call quality (e.g. call from GSM) is a good 
explanation.
> No problem Boris. I never had much success with in-band DTMF but as M.
> Collins suggested in a separate thread there are some cases where you
> really need in-band detection (i.e. calls via GSM networks always send
> in-band DTMF), I suggest you try recording the calls and try to make sense
> of how much of the DTMF tone is being received, if there is stuttering due
> to echo cancellation or static on the line, etc...
>
> On Tue, 24 May 2011 16:57:15 +0300, Boris Kovalenko<boris at tagnet.ru>
> wrote:
>
>> Hello!
>>
>>       Thank You Kerem, I'll try. The main problem is that there is a
>> little percent of wrong DTMF and a very nervous customer.
>> Also what if not to use dtmf-relay on Cisco at all? What kind of
>> problems may I have? I use G711alaw inside a network. Some transit calls
>> (ie Internet ->  PSTN) may use G729.
>>
>>> Yes Boris, try setting it lower incrementally and see if it helps. There
>>> might be problems related to gain + echo on the line that confuses DTMF
>>> detection, so check if the line has echo as well. Also sometimes I have
>>> seen that the remote calling party has a POTS line with too much static
>>> it
>>> can confuse DTMF detection. On other VoIP systems we solved this problem
>>> by spying on the channel e.g.
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be
>>> helpful in debugging the situation.
>>>
>>> On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko<boris at tagnet.ru>
>>> wrote:
>>>
> <snip>
>>> <snip>
> <snip>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sid.kshatriya at gmail.com  Wed May 25 09:04:40 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 25 May 2011 10:34:40 +0530
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTi=eini48X7tKDpbq2jdh0BR6RCt_w@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTimxzhdyq3Jp=wOGviF=JfoW=BPWRg@mail.gmail.com>
	<BANLkTi=eini48X7tKDpbq2jdh0BR6RCt_w@mail.gmail.com>
Message-ID: <BANLkTinY5AA96ruq5bz7Rd5gh_iH_QZTrA@mail.gmail.com>

Thanks for your tips Bob.

So I have done the following
1. Used start_dtmf before my lua IVR script starts
2. Asked my VOIP carrier to send me INFO events

Now I need to know what I need to do on my internal and external SIP
profiles in terms of config (if anything, from the default config).
Currently when I get a call on an external line I simply bridge it to my
internal IVR number.

Thanks,

Sidharth

On Wed, May 25, 2011 at 4:21 AM, Bob Coleman <bobc at devassert.com> wrote:

> Just my two cents, but we had similar problems with mobile dtmf, and
> found the best combination was to set INFO rather than RFC2833 and to
> always use start_dtmf .
>
> That also allowed the soft phone clients to operate properly as well,
> and if the client is configurable it is best to tell the soft phone
> what type of dtmf to use other wise we got the same symtoms as you
> described, namely multiple key presses when you only press it once
>
> Regards
>
> Bob
>
> On Tue, May 24, 2011 at 10:29 PM, Sidharth Kshatriya
> <sid.kshatriya at gmail.com> wrote:
> > Thanks for your reply, Michael. I have tried doing a start dtmf but then
> > I've noticed:
> > 1.  My skype key presses get unreliable...
> > 2. The dtmf detection is patchy with inband -- works sometimes and
> doesn't
> > work sometimes :-)
> >
> > Any settings to keep in mind apart just adding a start dtmf ?
> > Thanks,
> > Sidharth
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sidharth Kshatriya
www.sidk.info
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From sid.kshatriya at gmail.com  Wed May 25 09:08:40 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 25 May 2011 10:38:40 +0530
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTi=GV_FdH-3d-wamOryLJXcXEM2smw@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTi=VeCq=grE7T7iYQfYWrp9f8w+vSg@mail.gmail.com>
	<BANLkTi=GV_FdH-3d-wamOryLJXcXEM2smw@mail.gmail.com>
Message-ID: <BANLkTikZUgytz0x7hfHGa=WgX4_G0d6W4A@mail.gmail.com>

The tech support guys from my VOIP carrier had the following to say:

"Have you told freeswitch to ANSWER the call prior to the the IVR coming
into play as by default it would answer EARLY MEDIA thus allowing only 1 way
audio and no DTMF."

What does this mean? I answer the call within the lua script.

On Wed, May 25, 2011 at 12:52 AM, Michael Collins <msc at freeswitch.org>wrote:

> I guess what I mean is this: he needs to listen to what ACTUALLY comes down
> the line. There's no substitute for the human ear. If the tones coming down
> the line are not clean or otherwise are not recognizable then I would
> suggest that looking for a solution inside of FreeSWITCH itself may not be
> too helpful...
>
> That being said, if the DTMFs are coming in cleanly then there's definitely
> something hinky going on inside his system.
>
> -MC
>
>
> On Tue, May 24, 2011 at 12:08 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> Michael,
>>
>>  I'm not really sure what you mean by this...
>>
>>  The DTMF method used between an endpoint and a SIP carrier is
>> largely determined by the SIP carrier, not the carrier or device on
>> the far end.
>>
>>  On the PSTN DTMF is always inband.  If configured for out of band
>> DTMF it's up to the various media gateways in use by VoIP providers to
>> detect the inband DTMF, squelch it from the G711 input audio, and
>> create out of band events.
>>
>>  Of course if you're configured for inband and using G711 it will be
>> inband all the way through.
>>
>> P.S. - Cell phones either use AMR or EVRC these days and I would guess
>> they're using out of band DTMF too.  I've seen some documentation
>> claiming support for inband DTMF on either but that seems strange to
>> me...
>>
>> On Mon, May 23, 2011 at 10:24 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > Sidharth,
>> > A mobile phone will always send DTMFs in-band, so you need to be ready
>> for
>> > that scenario. I recommend you add this to your dialplan for inbound
>> calls:
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
>> > Test various scenarios and make sure they work - call from Skype, from
>> > mobile phone, from a land line, etc. Let us know what happens.
>> > -MC
>> > On Sat, May 21, 2011 at 1:11 AM, Sidharth Kshatriya
>> > <sid.kshatriya at gmail.com> wrote:
>> >>
>> >> Dear Friends,
>> >>
>> >> I'm using a voip carrier Voicenetwork.ca (recommended by someone on
>> this
>> >> list). While I'm generally happy with their service there seems to be
>> one
>> >> fatal problem: My IVR does not recognize DTMF!
>> >>
>> >> I have set <param name="dtmf-type" value="rfc2833"/> in both
>> >> sip_profile/internal.xml and sip_profile/external.xml
>> >>
>> >> The symptom of the problem is that making a call via skype will always
>> >> make the IVR recognize the DTMF while using something like a mobile
>> phone
>> >> almost always won't!
>> >>
>> >> I've tried in-band detection too. I'm making international calls into
>> my
>> >> IVR and the reliability of the in-band detection is not so good,
>> possibly
>> >> because of the quality of the call.
>> >>
>> >> Can someone please help me?
>> >>
>> >> Thanks,
>> >>
>> >> Sidharth
>> >>
>> >> --
>> >> Sidharth Kshatriya
>> >> www.sidk.info
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From ayhkor at gmail.com  Wed May 25 09:12:57 2011
From: ayhkor at gmail.com (deniro)
Date: Wed, 25 May 2011 01:12:57 -0400
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
	<BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
	<BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>
Message-ID: <BANLkTin1V6TinJWsOVorkwAMTXc88G_mcg@mail.gmail.com>

Thanks for the replies
I made some research and run series of tests.
I am using  perl+FCGI (fastcgi)  and my http server is  working fine as
tested with multiple perl scripts.
 I think the problem is handling the CGI within the cdr.pl file(obtained
from freeswitch wiki)

 so, how can I change below script to use with FCGI instead of CGI::Simple

use XML::Simple; # Get from CPAN
use CGI::Simple; # Get from CPAN
use Data::Dumper;
# dump into a place for further review
open(FILEOUT,">",'/tmp/fs-xml-cdr2.log');
print FILEOUT "Test successful..\n";
cloe(FILEOUT);
# dump into a place for further review
open(FILEOUT,">>",'/tmp/fs-xml-cdr.log');
# $cgi object has handy methods
my $cgi = new CGI::Simple;


Thx
deniro




On Mon, May 23, 2011 at 3:50 AM, Steven Ayre <steveayre at gmail.com> wrote:

>  > 2-- when dialing into conference, getting following from fs_cli
>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>> > [http://www.xxx.com/PERL/cdr.pl]
>>
>> There isn't a lot of info to work on here, but I'd look at your web
>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>> means you have a bad gateway).
>>
>
> Check your web server logs - there may be some useful message. It sounds
> like the cgi-bin/perl handler hasn't been configured correctly, the script
> isn't executable, or it's not returning some headers. Try visiting the URL
> in your webbrowser (you
> won't be submitting a CDR but you can at least check the script runs
> without an (unexpected) error.
>
>
>
>
>
> On 23 May 2011 07:38, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
>> > several questions;
>> > 1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
>> > trying to figure out if my problem is with cdr or with curl
>>
>> No, they are not functionally related... easy to get mixed up because
>> they both use HTTP and XML. One does not depend on the other. However,
>> you *can* configure mod_xml_cdr (or any other module) with
>> mod_xml_curl.
>>
>>
>>
>> > 2-- when dialing into conference, getting following from fs_cli
>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>> > [http://www.xxx.com/PERL/cdr.pl]
>>
>> There isn't a lot of info to work on here, but I'd look at your web
>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>> means you have a bad gateway).
>>
>>
>>
>> > 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
>> >  <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>
>>
>> Looks good to me, but check it with the wiki:
>> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration
>>
>>
>>
>> > 4-- if I need mod_xml_curl (question 1)
>> > what should this line be like in xml_curl.conf.xml?
>> > <param name="gateway-url" value="http://www.xxx.com:80<http://www.xxx.com/>
>> "
>> > bindings="dialplan"/>
>>
>> This one is a little tricker to get right.  Again, the wiki is might
>> be the best place to reference:
>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring
>>
>>
>>
>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From e.kornev at dcn.ru  Wed May 25 09:49:36 2011
From: e.kornev at dcn.ru (=?UTF-8?B?ItCa0L7RgNC90LXQsiDQlS7QoS4i?=)
Date: Wed, 25 May 2011 11:49:36 +0600
Subject: [Freeswitch-users] Outbound SIP invites without codec list
In-Reply-To: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>
References: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>
Message-ID: <4DDC9870.7010404@dcn.ru>

m=audio 18646 RTP/AVP 0 8 101 13
                                                ^ ^ - these are codecs


> Hi,
>
> My outbound SIP invites are going out without a codec list? 
> Could someone point me in the right direction on how to set / insert this?
>
>
> send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:
>   
>  ------------------------------------------------------------------------
>    INVITE sip:0800000000 at 1.2.3.4:5060 
> <http://sip:0800000000 at 1.2.3.4:5060> SIP/2.0
>    Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr
>    Max-Forwards: 69
>    From: "Liam - Test" <sip:12345678 at 9.8.7.6 
> <mailto:sip%3A12345678 at 9.8.7.6>>;tag=ryFQea59e6Upe
>    To: <sip:0800000000 at 1.2.3.4:5060 <http://sip:0800000000 at 1.2.3.4:5060>>
>    Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43
>    CSeq: 12805602 INVITE
>    Contact: <sip:12345678 at 9.8.7.6:5060 <http://sip:12345678 at 9.8.7.6:5060>>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 189
>    X-FS-Support: update_display
>    Remote-Party-ID: "Liam - Test" <sip:12345678 at 9.8.7.6 
> <mailto:sip%3A12345678 at 9.8.7.6>>;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6
>    s=FreeSWITCH
>    c=IN IP4 9.8.7.6
>    t=0 0
>    m=audio 18646 RTP/AVP 0 8 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>   
>  ------------------------------------------------------------------------
>
>
> Thanks
>
> Liam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 



? ?????????, ?????? ?.?.
???????????? ?????? ????
??? "???"
(343) 378-3100 + 170



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From steveayre at gmail.com  Wed May 25 10:21:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 25 May 2011 07:21:47 +0100
Subject: [Freeswitch-users] Outbound SIP invites without codec list
In-Reply-To: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>
References: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>
Message-ID: <BANLkTin08DDhiOnvZAcc5ZAVqKCUvV7Xpw@mail.gmail.com>

"m=audio 18646 RTP/AVP 0 8 101 13"

That is the codec list.
  0 = PCMU (G711 u-law)
  8 = PCMA (G711 a-law)
  101 = DTMF RFC2833
  13 = Comfort Noise

Numbers 96-127 are dynamic. Anything else is static and defined by IANA (
http://www.iana.org/assignments/rtp-parameters).

A dynamic payload type needs a rtpmap line to map the number to the codec
name. That is not required for static payload types, because they have a
preset rtpmap.

FreeSWITCH doesn't add the a=rtpmap lines for static types by default. The
advantage of doing so is it makes your packets smaller - less bandwidth
usage, and less likely to get a fragmented packet.

Some (very few) devices are broken though and require the rtpmap lines for
all codecs. These have broken SIP implementations because the RFC states the
rtpmap is optional for static types. There is a compatibility option if
you're using one of those and really do require it, but changes are you
don't: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp

-Steve



On 25 May 2011 01:56, Liam Farr <liam at intersys.co.nz> wrote:

> Hi,
>
> My outbound SIP invites are going out without a codec list?
> Could someone point me in the right direction on how to set / insert this?
>
>
> send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:
>    ------------------------------------------------------------------------
>    INVITE sip:0800000000 at 1.2.3.4:5060 SIP/2.0
>    Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr
>    Max-Forwards: 69
>    From: "Liam - Test" <sip:12345678 at 9.8.7.6>;tag=ryFQea59e6Upe
>    To: <sip:0800000000 at 1.2.3.4:5060>
>    Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43
>    CSeq: 12805602 INVITE
>    Contact: <sip:12345678 at 9.8.7.6:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 189
>    X-FS-Support: update_display
>    Remote-Party-ID: "Liam - Test" <sip:12345678 at 9.8.7.6
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6
>    s=FreeSWITCH
>    c=IN IP4 9.8.7.6
>    t=0 0
>    m=audio 18646 RTP/AVP 0 8 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    ------------------------------------------------------------------------
>
>
> Thanks
>
>  Liam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From liam at intersys.co.nz  Wed May 25 11:28:34 2011
From: liam at intersys.co.nz (Liam Farr)
Date: Wed, 25 May 2011 19:28:34 +1200
Subject: [Freeswitch-users] Outbound SIP invites without codec list
In-Reply-To: <BANLkTin08DDhiOnvZAcc5ZAVqKCUvV7Xpw@mail.gmail.com>
References: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>
	<BANLkTin08DDhiOnvZAcc5ZAVqKCUvV7Xpw@mail.gmail.com>
Message-ID: <d30cd50e8d90ac3f2ff1841ca15367b0@mail.gmail.com>

Hi,



I set <action application="export" data="verbose_sdp=true" /> in my dial
plan and this fixed it.



My provider is rather strict about going through a testing / certification
process for any ?new? hardware that they connect to their network, and
wanted to see the full codec list in the SIP invite, (even though as you say
it?s not required under the RFC?s).





Thanks



Liam



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
*Sent:* Wednesday, 25 May 2011 6:22 p.m.
*To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] Outbound SIP invites without codec list



"m=audio 18646 RTP/AVP 0 8 101 13"

That is the codec list.
  0 = PCMU (G711 u-law)
  8 = PCMA (G711 a-law)
  101 = DTMF RFC2833
  13 = Comfort Noise

Numbers 96-127 are dynamic. Anything else is static and defined by IANA (
http://www.iana.org/assignments/rtp-parameters).

A dynamic payload type needs a rtpmap line to map the number to the codec
name. That is not required for static payload types, because they have a
preset rtpmap.

FreeSWITCH doesn't add the a=rtpmap lines for static types by default. The
advantage of doing so is it makes your packets smaller - less bandwidth
usage, and less likely to get a fragmented packet.

Some (very few) devices are broken though and require the rtpmap lines for
all codecs. These have broken SIP implementations because the RFC states the
rtpmap is optional for static types. There is a compatibility option if
you're using one of those and really do require it, but changes are you
don't: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp

-Steve


On 25 May 2011 01:56, Liam Farr <liam at intersys.co.nz> wrote:

Hi,



My outbound SIP invites are going out without a codec list?
Could someone point me in the right direction on how to set / insert this?





send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:

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

   INVITE sip:0800000000 at 1.2.3.4:5060 SIP/2.0

   Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr

   Max-Forwards: 69

   From: "Liam - Test" <sip:12345678 at 9.8.7.6>;tag=ryFQea59e6Upe

   To: <sip:0800000000 at 1.2.3.4:5060>

   Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43

   CSeq: 12805602 INVITE

   Contact: <sip:12345678 at 9.8.7.6:5060>

   User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, hold, refer

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 189

   X-FS-Support: update_display

   Remote-Party-ID: "Liam - Test" <sip:12345678 at 9.8.7.6
>;party=calling;screen=yes;privacy=off



   v=0

   o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6

   s=FreeSWITCH

   c=IN IP4 9.8.7.6

   t=0 0

   m=audio 18646 RTP/AVP 0 8 101 13

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

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





Thanks



Liam


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Wed May 25 12:32:04 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 25 May 2011 08:32:04 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
	<BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>
Message-ID: <BANLkTimX6S3rSZF8jALpbk0XF1+sQ0daYg@mail.gmail.com>

Good Morning List As Promised.

I Have tested with G711 the problem remains and appears not be affected by
the codec change.

Here i have my debug log :
http://pastebin.freeswitch.org/16369

Just a short explanation :

the call XXXX1010 is Leg A and is currently valet_park()

call XXXX371 is Leg B and is currently on park()

Strangely FS says LEG A is out of order but i can hear MOH and sometimes
with this exact software it connects fine.

Hope you can help me out i will keep testing during the day.

Regards
A/T


2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com>

> Ok Anthony.
>
> I will provide you with data regarding the use of G711 and full debug logs.
> Will keep an eye on the logs.
>
> Regards
> A/T
>
>
> 2011/5/24 Anthony Minessale <anthony.minessale at gmail.com>
>
>> The +OK only means the attempt to bridge was successful, if something
>> else goes wrong after that, you will not know because it happens
>> later.
>>
>> As suggested, look at the cause of the hangup on the failed bridge.
>>
>>
>> On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
>> <eagle.antonio at gmail.com> wrote:
>> > Hello List , Good Morning.
>> >
>> > In my fight to get ESL & Python & Freeswitch all to behave properly i
>> > noticed a possible BUG ( need the veterans to confirm).
>> >
>> > Scenario :
>> >
>> > Originate & ValetPark () + MOH OR  Just Park() , Both show the same
>> problem.
>> > Codec G729
>> >
>> > I Make another call:
>> > originate & park()
>> > codec G729
>> >
>> > Now Python ESL Inbound Or FS Console :
>> >
>> > uuid_bridge uuid1 uuid2
>> >
>> > +OK uuid
>> >
>> > Now one of the two things happen :
>> >
>> > 1) One the call gets connected hurray :) , Audio Perfect  , etc.
>> >
>> > 2) The call gets dropped :(
>> >
>> > In both cases uuid_bridge reports +OK even in the case the call is
>> dropped.
>> > Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>> > SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
>> > could this be related to the use of G729 ?
>> >
>> > I have lots of available licenses.
>> >
>> > Regards
>> > Ant?nio Teixeira
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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 peter.olsson at visionutveckling.se  Wed May 25 12:52:59 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 25 May 2011 10:52:59 +0200
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTimX6S3rSZF8jALpbk0XF1+sQ0daYg@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
	<BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>,
	<BANLkTimX6S3rSZF8jALpbk0XF1+sQ0daYg@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F9@cooper>

You need to provide the entire debug trace from this, nor just only after calling uuid_bridge.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Antonio Teixeira [eagle.antonio at gmail.com]
Skickat: den 25 maj 2011 10:32
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK

Good Morning List As Promised.

I Have tested with G711 the problem remains and appears not be affected by the codec change.

Here i have my debug log :
http://pastebin.freeswitch.org/16369

Just a short explanation :

the call XXXX1010 is Leg A and is currently valet_park()

call XXXX371 is Leg B and is currently on park()

Strangely FS says LEG A is out of order but i can hear MOH and sometimes with this exact software it connects fine.

Hope you can help me out i will keep testing during the day.

Regards
A/T


2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>>
Ok Anthony.

I will provide you with data regarding the use of G711 and full debug logs.
Will keep an eye on the logs.

Regards
A/T


2011/5/24 Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
The +OK only means the attempt to bridge was successful, if something
else goes wrong after that, you will not know because it happens
later.

As suggested, look at the cause of the hangup on the failed bridge.


On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
<eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>> wrote:
> Hello List , Good Morning.
>
> In my fight to get ESL & Python & Freeswitch all to behave properly i
> noticed a possible BUG ( need the veterans to confirm).
>
> Scenario :
>
> Originate & ValetPark () + MOH OR  Just Park() , Both show the same problem.
> Codec G729
>
> I Make another call:
> originate & park()
> codec G729
>
> Now Python ESL Inbound Or FS Console :
>
> uuid_bridge uuid1 uuid2
>
> +OK uuid
>
> Now one of the two things happen :
>
> 1) One the call gets connected hurray :) , Audio Perfect  , etc.
>
> 2) The call gets dropped :(
>
> In both cases uuid_bridge reports +OK even in the case the call is dropped.
> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
> could this be related to the use of G729 ?
>
> I have lots of available licenses.
>
> Regards
> Ant?nio Teixeira
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

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

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

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

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


!DSPAM:4ddcbfed32761527616399!


From bobc at devassert.com  Wed May 25 13:12:33 2011
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 25 May 2011 21:12:33 +1200
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTinY5AA96ruq5bz7Rd5gh_iH_QZTrA@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTimxzhdyq3Jp=wOGviF=JfoW=BPWRg@mail.gmail.com>
	<BANLkTi=eini48X7tKDpbq2jdh0BR6RCt_w@mail.gmail.com>
	<BANLkTinY5AA96ruq5bz7Rd5gh_iH_QZTrA@mail.gmail.com>
Message-ID: <BANLkTins+GfZgK2NNYn+OFg5Vz0WtcKUPA@mail.gmail.com>

In your SIP profile eg internal.xml change the RFC2833 to info.

eg <param name="dtmf-type" value="info"/>

Then just call start_dtmf at the start of the call, like you said and
it definitely helped my cause

Bob

On Wed, May 25, 2011 at 5:04 PM, Sidharth Kshatriya
<sid.kshatriya at gmail.com> wrote:
> Thanks for your tips Bob.
> So I have done the following
> 1. Used start_dtmf before my lua IVR script starts
> 2. Asked my VOIP carrier to send me INFO events
> Now I need to know what I need to do on my internal and external SIP
> profiles in terms of config (if anything, from the default config).
> Currently when I get a call on an external line I simply bridge it to my
> internal IVR number.
> Thanks,
> Sidharth
>


From bobc at devassert.com  Wed May 25 13:21:55 2011
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 25 May 2011 21:21:55 +1200
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTikZUgytz0x7hfHGa=WgX4_G0d6W4A@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTi=VeCq=grE7T7iYQfYWrp9f8w+vSg@mail.gmail.com>
	<BANLkTi=GV_FdH-3d-wamOryLJXcXEM2smw@mail.gmail.com>
	<BANLkTikZUgytz0x7hfHGa=WgX4_G0d6W4A@mail.gmail.com>
Message-ID: <BANLkTikb2B=B5uRaF=W-DHGgsoiDe-dt_g@mail.gmail.com>

I always set the following in the dialplan to let the user know to
wait, ie it keeps ringing

<action application="set" data="instant_ringback=true"/>
<action application="ring_ready" />

I am not sure I understand what the support guy says, but the above
lets me get ready to answer the call without the user waiting in
silence.

Bob

On Wed, May 25, 2011 at 5:08 PM, Sidharth Kshatriya
<sid.kshatriya at gmail.com> wrote:
> The tech support guys from my VOIP carrier had the following to say:
> "Have you told freeswitch to ANSWER the call prior to the the IVR coming
> into play as by default it would answer EARLY MEDIA thus allowing only 1 way
> audio and no DTMF."
> What does this mean? I answer the call within the lua script.


From anton.vazir at gmail.com  Wed May 25 13:26:46 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 25 May 2011 14:26:46 +0500
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F9@cooper>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
	<BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>
	<BANLkTimX6S3rSZF8jALpbk0XF1+sQ0daYg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F9@cooper>
Message-ID: <BANLkTinFdKhyy2HkRLB8Dohtc5CL0vhbiA@mail.gmail.com>

Anthony, I do use the same scheme, and did not experienced the
problem. But I do use park everywhere instead of ValetPark - can it be
the reason?

2011/5/25 Peter Olsson <peter.olsson at visionutveckling.se>:
> You need to provide the entire debug trace from this, nor just only after calling uuid_bridge.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Antonio Teixeira [eagle.antonio at gmail.com]
> Skickat: den 25 maj 2011 10:32
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
>
> Good Morning List As Promised.
>
> I Have tested with G711 the problem remains and appears not be affected by the codec change.
>
> Here i have my debug log :
> http://pastebin.freeswitch.org/16369
>
> Just a short explanation :
>
> the call XXXX1010 is Leg A and is currently valet_park()
>
> call XXXX371 is Leg B and is currently on park()
>
> Strangely FS says LEG A is out of order but i can hear MOH and sometimes with this exact software it connects fine.
>
> Hope you can help me out i will keep testing during the day.
>
> Regards
> A/T
>
>
> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>>
> Ok Anthony.
>
> I will provide you with data regarding the use of G711 and full debug logs.
> Will keep an eye on the logs.
>
> Regards
> A/T
>
>
> 2011/5/24 Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
> The +OK only means the attempt to bridge was successful, if something
> else goes wrong after that, you will not know because it happens
> later.
>
> As suggested, look at the cause of the hangup on the failed bridge.
>
>
> On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
> <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>> wrote:
>> Hello List , Good Morning.
>>
>> In my fight to get ESL & Python & Freeswitch all to behave properly i
>> noticed a possible BUG ( need the veterans to confirm).
>>
>> Scenario :
>>
>> Originate & ValetPark () + MOH OR ?Just Park() , Both show the same problem.
>> Codec G729
>>
>> I Make another call:
>> originate & park()
>> codec G729
>>
>> Now Python ESL Inbound Or FS Console :
>>
>> uuid_bridge uuid1 uuid2
>>
>> +OK uuid
>>
>> Now one of the two things happen :
>>
>> 1) One the call gets connected hurray :) , Audio Perfect ?, etc.
>>
>> 2) The call gets dropped :(
>>
>> In both cases uuid_bridge reports +OK even in the case the call is dropped.
>> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>> SYN/ASYNC Problems) ?I still get sometimes ( not always) a dropped call
>> could this be related to the use of G729 ?
>>
>> I have lots of available licenses.
>>
>> Regards
>> Ant?nio Teixeira
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> !DSPAM:4ddcbfed32761527616399!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From rmorin at blie-ent.com  Wed May 25 15:18:43 2011
From: rmorin at blie-ent.com (Rob Morin)
Date: Wed, 25 May 2011 07:18:43 -0400
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <04b201cc1a21$e8825400$b986fc00$@blie-ent.com>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>	<59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>
	<04b201cc1a21$e8825400$b986fc00$@blie-ent.com>
Message-ID: <060401cc1acd$82adae20$88090a60$@blie-ent.com>

Is the fact that I was previously running the snapshot from 25 March
adequate to meet this test?

 

From: Rob Morin [mailto:rmorin at blie-ent.com] 
Sent: Tuesday, May 24, 2011 10:50 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

I can, if someone can tell me where to get the snapshot.  I looked at
files.freeswitch.org and didn?t see one.

 

Prior to updating to the 5-17 snapshot, I was having the same problems with
the 3-25 snapshot that I was using.

 

Thank you,

Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Tuesday, May 24, 2011 1:47 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

following Yehavi's comment, could you downgrade your GIT version to a
previous version, like one month old ?

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 

 

 

Le 24/05/2011 ? 02:59, Rob Morin a ?crit :

 

David,

 

It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and
Cisco/Linksys SPA 2102. Same story in both cases.

 

I had my carrier help me troubleshoot and we can see the SIP request from
the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but
the response from the A leg isn?t forwarded back to the B leg. I can produce
traces that show this.

 

When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t
respond or forward it. So the connection dies.  I have every reason to
believe that I?d be able to send the fax if FS would respond with a 488
indicating T.38 is not acceptable. But instead, all I get is silence. Again,
I can produce traces that demonstrate this.

 

There?s another similar email  trail out there,
http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821
994.html, but I?m not certain that they?re related.

 

Thank you for your help!
Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Monday, May 23, 2011 6:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

perhaps you should not consider T38 is 100% interoperable.

You may tell us what ATA is that, because some of them are nice piece of
junk.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 





 

Le 23/05/2011 ? 20:51, Rob Morin a ?crit :





I?m having problems faxing if the B leg offers T.38.  Several scenarios, but
always the same result.

 

First, the architecture

 

FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another
carrier (PSTN) and the fax machine)

 

On my ATA, T.38 is enabled (Auto Detect).

 

If the destination supports T.38, the Carrier will offer it to Freeswitch.
If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
faxes go through.

 

I?ve set Freeswitch as:

      <action application="set" data="fax_enable_t38=true"/>

      <action application="export" data="t38_passthru=true"/> 

When that?s the case, and the B leg offers T.38, Freeswitch passes through
the offer and the ATA responds, 3 times. But the response never gets sent to
the B leg, so it terminates the call after about 30 seconds.

 

When I set Freeswitch as:

      <action application="set" data="fax_enable_t38=false"/>

The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t
pass the offer through and the B leg terminates the call after about 30
seconds.

 

I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
2011-05-17 22-51-47 -0500) ).

 

I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
?passthrough? has had the most success, mostly when the other ends aren?t
T.38 capable and it stays in G711u.

 

Is there something else I need to do to enable T.38 in passthrough mode? 

 

Thank you,
Rob Morin

 

PS ? I can provide tcpdumps of this, or whatever else is necessary.

 

 

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

 

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

 

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From adam.kelloway at newpace.ca  Wed May 25 17:19:52 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Wed, 25 May 2011 10:19:52 -0300
Subject: [Freeswitch-users] transfer to extension without pattern matching
Message-ID: <4DDD01F8.5030506@newpace.ca>

Hi there,

The wiki page describing transfer 
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer) states 
the following:

"Immediately transfer the calling channel to a new context. If there 
happens to be an xml extension named <destination_number> then control 
is "warped" directly to that extension. Otherwise it goes through the 
entire context checking for a match."

Is this a correct statement? It seems that even when I try to transfer 
control directly to an extension, the entire context is still checked 
for a match.

I am using the following:

..
<application="transfer" data="test XML default"/>
..

<extension name="test">
<condition field="destination_number" expression="^test$">
     ...
</condition>
</extension>

Is there a way to disable the pattern matching/dial plan processing?

Thanks,

Adam


From avi at avimarcus.net  Wed May 25 17:27:04 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 25 May 2011 16:27:04 +0300
Subject: [Freeswitch-users] transfer to extension without pattern
	matching
In-Reply-To: <4DDD01F8.5030506@newpace.ca>
References: <4DDD01F8.5030506@newpace.ca>
Message-ID: <BANLkTimUVmzC+CtnGCMDJNx4rR0w-a8rzA@mail.gmail.com>

That sounds like it's describing
http://wiki.freeswitch.org/wiki/Variable_auto_hunt
 <http://wiki.freeswitch.org/wiki/Variable_auto_hunt>You need to turn that
on. I'm not aware that even transfer does that automatically, so afaik, it
is indeed wrong - but is a feature that FreeSWITCH possesses if you turn it
on.
-Avi

On Wed, May 25, 2011 at 4:19 PM, Adam Kelloway <adam.kelloway at newpace.ca>wrote:

> Hi there,
>
> The wiki page describing transfer
> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer) states
> the following:
>
> "Immediately transfer the calling channel to a new context. If there
> happens to be an xml extension named <destination_number> then control
> is "warped" directly to that extension. Otherwise it goes through the
> entire context checking for a match."
>
> Is this a correct statement? It seems that even when I try to transfer
> control directly to an extension, the entire context is still checked
> for a match.
>
> I am using the following:
>
> ..
> <application="transfer" data="test XML default"/>
> ..
>
> <extension name="test">
> <condition field="destination_number" expression="^test$">
>     ...
> </condition>
> </extension>
>
> Is there a way to disable the pattern matching/dial plan processing?
>
> Thanks,
>
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Wed May 25 19:10:32 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 25 May 2011 17:10:32 +0200
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <060401cc1acd$82adae20$88090a60$@blie-ent.com>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>	<59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>
	<04b201cc1a21$e8825400$b986fc00$@blie-ent.com>
	<060401cc1acd$82adae20$88090a60$@blie-ent.com>
Message-ID: <E92FE87C-23E1-4DEE-9318-3A7D1035B9EE@ipeva.fr>

Rob,

On my test box, I run a GIT from 05-02, and I don't have any issue.
But I just did the test, and the call flow is different from yours.

When I send a fax with my endpoint (Zoiper Softphone), I got the remote fax tone, and then my Zoiper sends the T38-REINVITE, not the remote gateway.
So I guess there are 2 possible call flows, because I remember in some circumstances, the remote sends the RE-INVITE.

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

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

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




Le 25/05/2011 ? 13:18, Rob Morin a ?crit :

> Is the fact that I was previously running the snapshot from 25 March adequate to meet this test?
>  
> From: Rob Morin [mailto:rmorin at blie-ent.com] 
> Sent: Tuesday, May 24, 2011 10:50 AM
> To: 'FreeSWITCH Users Help'
> Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>  
> I can, if someone can tell me where to get the snapshot.  I looked at files.freeswitch.org and didn?t see one.
>  
> Prior to updating to the 5-17 snapshot, I was having the same problems with the 3-25 snapshot that I was using.
>  
> Thank you,
> Rob
>  
> From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
> Sent: Tuesday, May 24, 2011 1:47 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>  
> Rob,
>  
> following Yehavi's comment, could you downgrade your GIT version to a previous version, like one month old ?
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
>  
> 
>  
> Le 24/05/2011 ? 02:59, Rob Morin a ?crit :
>  
> 
> David,
>  
> It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and Cisco/Linksys SPA 2102. Same story in both cases.
>  
> I had my carrier help me troubleshoot and we can see the SIP request from the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but the response from the A leg isn?t forwarded back to the B leg. I can produce traces that show this.
>  
> When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t respond or forward it. So the connection dies.  I have every reason to believe that I?d be able to send the fax if FS would respond with a 488 indicating T.38 is not acceptable. But instead, all I get is silence. Again, I can produce traces that demonstrate this.
>  
> There?s another similar email  trail out there, http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821994.html, but I?m not certain that they?re related.
>  
> Thank you for your help!
> Rob
>  
> From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
> Sent: Monday, May 23, 2011 6:11 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>  
> Rob,
>  
> perhaps you should not consider T38 is 100% interoperable.
> You may tell us what ATA is that, because some of them are nice piece of junk.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
> 
>  
> Le 23/05/2011 ? 20:51, Rob Morin a ?crit :
> 
> 
> 
> I?m having problems faxing if the B leg offers T.38.  Several scenarios, but always the same result.
>  
> First, the architecture
>  
> FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another carrier (PSTN) and the fax machine)
>  
> On my ATA, T.38 is enabled (Auto Detect).
>  
> If the destination supports T.38, the Carrier will offer it to Freeswitch. If the Destination doesn?t support T.38, the B leg doesn?t offer it and the faxes go through.
>  
> I?ve set Freeswitch as:
>       <action application="set" data="fax_enable_t38=true"/>
>       <action application="export" data="t38_passthru=true"/> 
> When that?s the case, and the B leg offers T.38, Freeswitch passes through the offer and the ATA responds, 3 times. But the response never gets sent to the B leg, so it terminates the call after about 30 seconds.
>  
> When I set Freeswitch as:
>       <action application="set" data="fax_enable_t38=false"/>
> The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t pass the offer through and the B leg terminates the call after about 30 seconds.
>  
> I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d 2011-05-17 22-51-47 -0500) ).
>  
> I?ve had no success attempting to configure FS as a gateway. Frankly, T.38 ?passthrough? has had the most success, mostly when the other ends aren?t T.38 capable and it stays in G711u.
>  
> Is there something else I need to do to enable T.38 in passthrough mode? 
>  
> Thank you,
> Rob Morin
>  
> PS ? I can provide tcpdumps of this, or whatever else is necessary.
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From robert.hadley at teotech.com  Wed May 25 20:33:14 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 25 May 2011 09:33:14 -0700
Subject: [Freeswitch-users] transfer to extension without pattern
 matching
In-Reply-To: <BANLkTimUVmzC+CtnGCMDJNx4rR0w-a8rzA@mail.gmail.com>
References: <4DDD01F8.5030506@newpace.ca>
	<BANLkTimUVmzC+CtnGCMDJNx4rR0w-a8rzA@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BD8C592B2@VA3DIAXVS351.RED001.local>


Also check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension.  You would want to use it with the auto-hunt feature to skip extra pattern matching.
-Robert

From: Avi Marcus [mailto:avi at avimarcus.net]
Sent: Wednesday, May 25, 2011 6:27 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] transfer to extension without pattern matching

That sounds like it's describing http://wiki.freeswitch.org/wiki/Variable_auto_hunt
You need to turn that on. I'm not aware that even transfer does that automatically, so afaik, it is indeed wrong - but is a feature that FreeSWITCH possesses if you turn it on.
-Avi

On Wed, May 25, 2011 at 4:19 PM, Adam Kelloway <adam.kelloway at newpace.ca<mailto:adam.kelloway at newpace.ca>> wrote:
Hi there,

The wiki page describing transfer
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer) states
the following:

"Immediately transfer the calling channel to a new context. If there
happens to be an xml extension named <destination_number> then control
is "warped" directly to that extension. Otherwise it goes through the
entire context checking for a match."

Is this a correct statement? It seems that even when I try to transfer
control directly to an extension, the entire context is still checked
for a match.

I am using the following:

..
<application="transfer" data="test XML default"/>
..

<extension name="test">
<condition field="destination_number" expression="^test$">
    ...
</condition>
</extension>

Is there a way to disable the pattern matching/dial plan processing?

Thanks,

Adam

_______________________________________________
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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 msc at freeswitch.org  Wed May 25 20:38:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 May 2011 09:38:38 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTikSN0hNbvjhYM9QFMte_Ah6s1RAGg@mail.gmail.com>

Hi folks!

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_25

We have a few news items to discuss and if there's time I will share with
you the technique I demonstrated for using bind_digit_action in conferences.

Talk to you soon,
Michael
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From benkokakao at gmail.com  Wed May 25 20:52:04 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 25 May 2011 18:52:04 +0200
Subject: [Freeswitch-users] xml_cdr: end_epoch is 0,
	billsec are negative - how does this happen?
Message-ID: <BANLkTikiEL3H=wkXELgZOBBS5bM60hf5CA@mail.gmail.com>

Hello!

My xml-cdrs are saved with an end_epoch of 0/end_stamp 1970-01-01 and
negative duration, billsec etc.(e.g. -1306336317) -
start_epoch/start_stamp values are correct. What's causing this
behaviour?
I'm running FreeSWITCH on a OpenVZ-Container(2.6.32-5-openvz-amd64) -
could there be a relation to the cdr-issue?

Regards
Christian


From rmorin at blie-ent.com  Wed May 25 20:56:29 2011
From: rmorin at blie-ent.com (Rob Morin)
Date: Wed, 25 May 2011 12:56:29 -0400
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <E92FE87C-23E1-4DEE-9318-3A7D1035B9EE@ipeva.fr>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>	<59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>	<04b201cc1a21$e8825400$b986fc00$@blie-ent.com>	<060401cc1acd$82adae20$88090a60$@blie-ent.com>
	<E92FE87C-23E1-4DEE-9318-3A7D1035B9EE@ipeva.fr>
Message-ID: <067601cc1afc$b210ada0$163208e0$@blie-ent.com>

Thank you.

 

So, I believe that there are two potential problems, fixing either of which
will allow me to send faxes.

 

1.       When T.38 is enabled in passthrough mode and the B leg offers T.38,
Freeswitch needs to forward the invite and the responses. Currently it isn?t
forwarding the response back to the B leg.

2.       When T.38 is not enabled and the B leg offers T.38, Freeswitch
needs to respond with a 488 indicating T.38 is not acceptable.  Currently it
doesn?t respond at all.

 

How would I go about making or requesting these fixes?

Thank you,
Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Wednesday, May 25, 2011 11:11 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

On my test box, I run a GIT from 05-02, and I don't have any issue.

But I just did the test, and the call flow is different from yours.

 

When I send a fax with my endpoint (Zoiper Softphone), I got the remote fax
tone, and then my Zoiper sends the T38-REINVITE, not the remote gateway.

So I guess there are 2 possible call flows, because I remember in some
circumstances, the remote sends the RE-INVITE.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 





 

Le 25/05/2011 ? 13:18, Rob Morin a ?crit :





Is the fact that I was previously running the snapshot from 25 March
adequate to meet this test?

 

From: Rob Morin [mailto:rmorin at blie-ent.com] 
Sent: Tuesday, May 24, 2011 10:50 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

I can, if someone can tell me where to get the snapshot.  I looked at
files.freeswitch.org and didn?t see one.

 

Prior to updating to the 5-17 snapshot, I was having the same problems with
the 3-25 snapshot that I was using.

 

Thank you,

Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Tuesday, May 24, 2011 1:47 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

following Yehavi's comment, could you downgrade your GIT version to a
previous version, like one month old ?

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 

 

 

Le 24/05/2011 ? 02:59, Rob Morin a ?crit :

 

David,

 

It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and
Cisco/Linksys SPA 2102. Same story in both cases.

 

I had my carrier help me troubleshoot and we can see the SIP request from
the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but
the response from the A leg isn?t forwarded back to the B leg. I can produce
traces that show this.

 

When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t
respond or forward it. So the connection dies.  I have every reason to
believe that I?d be able to send the fax if FS would respond with a 488
indicating T.38 is not acceptable. But instead, all I get is silence. Again,
I can produce traces that demonstrate this.

 

There?s another similar email  trail out there,
http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821
994.html, but I?m not certain that they?re related.

 

Thank you for your help!
Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Monday, May 23, 2011 6:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

perhaps you should not consider T38 is 100% interoperable.

You may tell us what ATA is that, because some of them are nice piece of
junk.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 






 

Le 23/05/2011 ? 20:51, Rob Morin a ?crit :






I?m having problems faxing if the B leg offers T.38.  Several scenarios, but
always the same result.

 

First, the architecture

 

FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another
carrier (PSTN) and the fax machine)

 

On my ATA, T.38 is enabled (Auto Detect).

 

If the destination supports T.38, the Carrier will offer it to Freeswitch.
If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
faxes go through.

 

I?ve set Freeswitch as:

      <action application="set" data="fax_enable_t38=true"/>

      <action application="export" data="t38_passthru=true"/> 

When that?s the case, and the B leg offers T.38, Freeswitch passes through
the offer and the ATA responds, 3 times. But the response never gets sent to
the B leg, so it terminates the call after about 30 seconds.

 

When I set Freeswitch as:

      <action application="set" data="fax_enable_t38=false"/>

The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t
pass the offer through and the B leg terminates the call after about 30
seconds.

 

I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
2011-05-17 22-51-47 -0500) ).

 

I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
?passthrough? has had the most success, mostly when the other ends aren?t
T.38 capable and it stays in G711u.

 

Is there something else I need to do to enable T.38 in passthrough mode? 

 

Thank you,
Rob Morin

 

PS ? I can provide tcpdumps of this, or whatever else is necessary.

 

 

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

 

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

 

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

 

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From steveayre at gmail.com  Wed May 25 20:59:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 25 May 2011 17:59:09 +0100
Subject: [Freeswitch-users] xml_cdr: end_epoch is 0,
 billsec are negative - how does this happen?
In-Reply-To: <BANLkTikiEL3H=wkXELgZOBBS5bM60hf5CA@mail.gmail.com>
References: <BANLkTikiEL3H=wkXELgZOBBS5bM60hf5CA@mail.gmail.com>
Message-ID: <BANLkTi=xkB-9EXSSFGG-V3V3k7Qte7hXBQ@mail.gmail.com>

What version of FreeSWITCH are you running?

There was briefly a version with bad timestamps a few weeks/months ago.

-Steve



On 25 May 2011 17:52, Christian Benke <benkokakao at gmail.com> wrote:

> Hello!
>
> My xml-cdrs are saved with an end_epoch of 0/end_stamp 1970-01-01 and
> negative duration, billsec etc.(e.g. -1306336317) -
> start_epoch/start_stamp values are correct. What's causing this
> behaviour?
> I'm running FreeSWITCH on a OpenVZ-Container(2.6.32-5-openvz-amd64) -
> could there be a relation to the cdr-issue?
>
> Regards
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From benkokakao at gmail.com  Wed May 25 21:06:38 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 25 May 2011 19:06:38 +0200
Subject: [Freeswitch-users] xml_cdr: end_epoch is 0,
 billsec are negative - how does this happen?
In-Reply-To: <BANLkTi=xkB-9EXSSFGG-V3V3k7Qte7hXBQ@mail.gmail.com>
References: <BANLkTikiEL3H=wkXELgZOBBS5bM60hf5CA@mail.gmail.com>
	<BANLkTi=xkB-9EXSSFGG-V3V3k7Qte7hXBQ@mail.gmail.com>
Message-ID: <BANLkTincLEUuBrsL2xQKC_SA5rjOxNZwGg@mail.gmail.com>

> What version of FreeSWITCH are you running?
>
> There was briefly a version with bad timestamps a few weeks/months ago.

FreeSWITCH Version 1.0.head (git-5510618 2011-04-29 08-57-00 -0500)

Thanks for the quick reply, time flies by, didn't realize i didn't
update for so long :-) Will report back when i'm finished...


From spencer at 5ninesolutions.com  Wed May 25 08:43:33 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 24 May 2011 23:43:33 -0500
Subject: [Freeswitch-users] Outbound SIP invites without codec list
In-Reply-To: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>
References: <BANLkTi=0GrHMRUOgM6b4n3xtC2YmRL2Ndw@mail.gmail.com>
Message-ID: <49E2DEFD-AF3E-4B7A-8315-83886DEA458B@5ninesolutions.com>

There is actually a codec list but those are static RTP payload types.
You are offering PCMU, PCMA, RFC2833 DTMF, and Comfort Noise 

See:
http://www.iana.org/assignments/rtp-parameters

If you need the full SDP for even known static types set verbose_sdp=true in vars.xml.
http://wiki.freeswitch.org/wiki/Variable_verbose_sdp

Spencer


On May 24, 2011, at 7:56 PM, Liam Farr wrote:

> Hi,
> 
> My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this?
> 
> 
> send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:
>    ------------------------------------------------------------------------
>    INVITE sip:0800000000 at 1.2.3.4:5060 SIP/2.0
>    Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr
>    Max-Forwards: 69
>    From: "Liam - Test" <sip:12345678 at 9.8.7.6>;tag=ryFQea59e6Upe
>    To: <sip:0800000000 at 1.2.3.4:5060>
>    Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43
>    CSeq: 12805602 INVITE
>    Contact: <sip:12345678 at 9.8.7.6:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 189
>    X-FS-Support: update_display
>    Remote-Party-ID: "Liam - Test" <sip:12345678 at 9.8.7.6>;party=calling;screen=yes;privacy=off
> 
>    v=0
>    o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6
>    s=FreeSWITCH
>    c=IN IP4 9.8.7.6
>    t=0 0
>    m=audio 18646 RTP/AVP 0 8 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    ------------------------------------------------------------------------
> 
> 
> Thanks
> 
> Liam
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bsevier at bandwidth.com  Wed May 25 20:58:01 2011
From: bsevier at bandwidth.com (Blake Sevier)
Date: Wed, 25 May 2011 10:58:01 -0600
Subject: [Freeswitch-users] FS not recognizing 200 OK after far-end call is
	answered
Message-ID: <341cf7e4fac39ca6e0b3e4b29166d3f8@mail.gmail.com>

Hi all,



I?m new, so please bear with me.  My setup is fairly basic, and the routing
all seems to work just fine:



FS receives call from an OpenSIPS proxy, it answers the call, records it and
plays back a short prompt before invoking the ?bridge? function to place an
outbound call via the same OpenSIPS proxy to an external SIP endpoint (in
testing, just an Asterisk instance that answers the call).  This all works
just fine, except FS doesn?t seem to recognize the 200 OK it receives back
for leg b after the Asterisk instances answers the call.  It apparently does
recognize/accept the 100 Trying, as FS doesn?t send additional INVITEs.  I?m
not seeing any issues with the SIP messaging received by FS ? it?s
apparently formatted correctly.

To note, I?ve also tested calls direct to the Asterisk instance (bypassing
OpenSIPS proxy), and get similar results.  My guess is that I?ve got
something incorrectly configured within my dialplan??  However, not sure how
to debug any further to figure out why the 200 Oks are not being
recognized.  Eventually, after 60s, FS ends the call with a NO_ANSWER.

Below is a link to a full call trace from FS ? from start of call to when I
see multiple 200 OKs from Asterisk.

http://pastebin.com/bmEtiW9X



Any help would be greatly appreciated.  If any further details are needed,
please let me know



Thank you!!!



Blake
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From lautram.mathieu at gmail.com  Wed May 25 16:05:07 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Wed, 25 May 2011 14:05:07 +0200
Subject: [Freeswitch-users] Errors while sending faxes
Message-ID: <BANLkTim9y6HAzBstbnZZYVeM0=kK39Oqkg@mail.gmail.com>

Hi all,

After sending faxes (it works very well know, thanks to Anthony Minessale) ,
sometimes I've got errors like that:

- Disconnected after permitted retries
- No response after sending a page
- Received a DCN from remote after sending a page
- Received bad response to DCS ot TCF
- The call dropped prematurely
- Timed out waiting for initial communication

Those faxes are about 25% of all faxes that I send. I'm using the last git
version.
Where this problem comes from? Is there a way to fix this?
I know that sometimes the remote end could be broken or could be an old fax
machine, but it seems strange that I have so many errors like that.

Thank you for your help :-)

-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From tang.du at hotmail.com  Wed May 25 07:38:51 2011
From: tang.du at hotmail.com (tangdu)
Date: Tue, 24 May 2011 20:38:51 -0700 (PDT)
Subject: [Freeswitch-users] load mod_gsmopen error
Message-ID: <1306294731594-6401282.post@n2.nabble.com>

I useing FS 1.0.7 ?Connected to nokia 3208C cellphone with usb?I can
communicate with cellphone by minicom?When I load mod_gsmopen , following
errors message.
Have you compiled mod_gsmopen correctly? Perhaps the  AT Command of 3208c
dose not match ?I need to modify gsmopen.conf.xml.
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1364 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1364 ][none      ][-1,-1,-1] globals.debug=0
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1366 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1366 ][none      ][-1,-1,-1] globals.debug=8
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1372 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1372 ][none      ][-1,-1,-1] globals.dialplan=XML
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1378 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1378 ][none      ][-1,-1,-1] globals.context=context_1
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1369 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1369 ][none      ][-1,-1,-1] globals.hold_music=
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1375 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1375 ][none      ][-1,-1,-1] globals.destination=9999
2011-05-25 10:25:33.694973 [WARNING] mod_gsmopen.cpp:1860 rev ?????(nil)|37    
][WARNINGA  1860 ][interface1][-1, 0, 0] STARTING interface_id=1
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1861 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1861 ][interface1][-1, 0, 0] id=1
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1862 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1862 ][interface1][-1, 0, 0] name=interface1
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1863 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1863 ][interface1][-1, 0, 0] hold-music=
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1864 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1864 ][interface1][-1, 0, 0] context=context_1
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1865 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1865 ][interface1][-1, 0, 0] dialplan=XML
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1866 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1866 ][interface1][-1, 0, 0] destination=5000
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1867 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1867 ][interface1][-1, 0, 0]
controldevice_name=/dev/ttyACM0
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1869 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1869 ][interface1][-1, 0, 0] alsacname=plughw:1
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1870 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1870 ][interface1][-1, 0, 0] alsapname=plughw:1
2011-05-25 10:25:33.694973 [DEBUG] mod_gsmopen.cpp:1885 rev ?????(nil)|37    
][DEBUG_GSMOPEN  1885 ][interface1][-1, 0, 0] gsmopen_serial_sync_period=300
2011-05-25 10:25:34.711673 [DEBUG] gsmopen_protocol.cpp:485 rev
?????(nil)|37     ][DEBUG_GSMOPEN  485  ][interface1][-1, 0, 0] Syncing
Serial, fd=68, protocol=2
2011-05-25 10:25:34.731728 [DEBUG] gsmopen_protocol.cpp:2581 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2581 ][interface1][-1, 0, 0]
ucs2_in=006300690061006F0020003100320033002000620065006C00E80020043D043E0432043E044104420438002005DC05E7002005E805D005EA0020FE8EFEE0FEA0FEE4FECBFE9300204EBA5927
2011-05-25 10:25:34.731728 [DEBUG] gsmopen_protocol.cpp:2602 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2602 ][interface1][-1, 0, 0] 1 ciao in=,
inleft=76, out=, outleft=1000, converted=, utf8_out=
2011-05-25 10:25:34.731728 [DEBUG] gsmopen_protocol.cpp:2614 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2614 ][interface1][-1, 0, 0] iconv_res=0, 
in=, inleft=0, out=, outleft=933, converted=, utf8_out=ciao 123 bel? ???????
?? ??? ????????? ???
2011-05-25 10:25:34.731728 [DEBUG] gsmopen_protocol.cpp:632 rev
?????(nil)|37     ][DEBUG_GSMOPEN  632  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:35.231548 [DEBUG] gsmopen_protocol.cpp:689 rev
?????(nil)|37     ][DEBUG_GSMOPEN  689  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:35.731444 [DEBUG] gsmopen_protocol.cpp:2362 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2362 ][interface1][-1, 0, 0] sending: AT
2011-05-25 10:25:35.731444 [DEBUG] gsmopen_protocol.cpp:2264 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2264 ][interface1][-1, 0, 0] sent data...
(A)
2011-05-25 10:25:35.731444 [DEBUG] gsmopen_protocol.cpp:2264 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2264 ][interface1][-1, 0, 0] sent data...
(T)
2011-05-25 10:25:35.731444 [DEBUG] gsmopen_protocol.cpp:2290 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2290 ][interface1][-1, 0, 0] sent
(carriage return)
2011-05-25 10:25:38.230865 [ERR] gsmopen_protocol.cpp:696 rev ?????(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-25 10:25:38.230865 [ERR] mod_gsmopen.cpp:1907 rev ?????(nil)|37    
][ERRORA  1907 ][interface1][-1, 0, 0] gsmopen_serial_config failed, let's
try again
2011-05-25 10:25:38.342857 [DEBUG] gsmopen_protocol.cpp:2581 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2581 ][interface1][-1, 0, 0]
ucs2_in=006300690061006F0020003100320033002000620065006C00E80020043D043E0432043E044104420438002005DC05E7002005E805D005EA0020FE8EFEE0FEA0FEE4FECBFE9300204EBA5927
2011-05-25 10:25:38.342857 [DEBUG] gsmopen_protocol.cpp:2602 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2602 ][interface1][-1, 0, 0] 1 ciao in=,
inleft=76, out=, outleft=1000, converted=, utf8_out=
2011-05-25 10:25:38.342857 [DEBUG] gsmopen_protocol.cpp:2614 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2614 ][interface1][-1, 0, 0] iconv_res=0, 
in=, inleft=0, out=, outleft=933, converted=, utf8_out=ciao 123 bel? ???????
?? ??? ????????? ???
2011-05-25 10:25:38.342857 [DEBUG] gsmopen_protocol.cpp:632 rev
?????(nil)|37     ][DEBUG_GSMOPEN  632  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:38.842711 [DEBUG] gsmopen_protocol.cpp:689 rev
?????(nil)|37     ][DEBUG_GSMOPEN  689  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:39.342567 [DEBUG] gsmopen_protocol.cpp:2362 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2362 ][interface1][-1, 0, 0] sending: AT
2011-05-25 10:25:39.342567 [DEBUG] gsmopen_protocol.cpp:2264 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2264 ][interface1][-1, 0, 0] sent data...
(A)
2011-05-25 10:25:39.342567 [DEBUG] gsmopen_protocol.cpp:2264 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2264 ][interface1][-1, 0, 0] sent data...
(T)
2011-05-25 10:25:39.342567 [DEBUG] gsmopen_protocol.cpp:2290 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2290 ][interface1][-1, 0, 0] sent
(carriage return)
2011-05-25 10:25:41.842030 [ERR] gsmopen_protocol.cpp:696 rev ?????(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-25 10:25:41.842030 [ERR] mod_gsmopen.cpp:1913 rev ?????(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 1: gsmopen_serial_config failed,
let's try again
2011-05-25 10:25:41.962079 [DEBUG] gsmopen_protocol.cpp:2581 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2581 ][interface1][-1, 0, 0]
ucs2_in=006300690061006F0020003100320033002000620065006C00E80020043D043E0432043E044104420438002005DC05E7002005E805D005EA0020FE8EFEE0FEA0FEE4FECBFE9300204EBA5927
2011-05-25 10:25:41.962079 [DEBUG] gsmopen_protocol.cpp:2602 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2602 ][interface1][-1, 0, 0] 1 ciao in=,
inleft=76, out=, outleft=1000, converted=, utf8_out=
2011-05-25 10:25:41.962079 [DEBUG] gsmopen_protocol.cpp:2614 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2614 ][interface1][-1, 0, 0] iconv_res=0, 
in=, inleft=0, out=, outleft=933, converted=, utf8_out=ciao 123 bel? ???????
?? ??? ????????? ???
2011-05-25 10:25:41.962079 [DEBUG] gsmopen_protocol.cpp:632 rev
?????(nil)|37     ][DEBUG_GSMOPEN  632  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:42.461918 [DEBUG] gsmopen_protocol.cpp:689 rev
?????(nil)|37     ][DEBUG_GSMOPEN  689  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:42.945855 [DEBUG] gsmopen_protocol.cpp:2362 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2362 ][interface1][-1, 0, 0] sending: AT
2011-05-25 10:25:42.945855 [DEBUG] gsmopen_protocol.cpp:2264 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2264 ][interface1][-1, 0, 0] sent data...
(A)
2011-05-25 10:25:42.945855 [DEBUG] gsmopen_protocol.cpp:2246 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2246 ][interface1][-1, 0, 0] Error
sending (T): -1 (???????????
2011-05-25 10:25:43.052861 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 0 -1 (???????????
2011-05-25 10:25:43.158728 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 1 -1 (???????????
2011-05-25 10:25:43.263940 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 2 -1 (???????????
2011-05-25 10:25:43.346794 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 3 -1 (???????????
2011-05-25 10:25:43.446754 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 4 -1 (???????????
2011-05-25 10:25:43.546738 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 5 -1 (???????????
2011-05-25 10:25:43.646742 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 6 -1 (???????????
2011-05-25 10:25:43.746669 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 7 -1 (???????????
2011-05-25 10:25:43.846762 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 8 -1 (???????????
2011-05-25 10:25:43.946646 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (T): 9 -1 (???????????
2011-05-25 10:25:44.046635 [ERR] gsmopen_protocol.cpp:2259 rev ?????(nil)|37    
][ERRORA  2259 ][interface1][-1, 0, 0] Error RE-sending (T): 10 -1
(????????????????????????)
2011-05-25 10:25:44.046635 [ERR] gsmopen_protocol.cpp:2364 rev ?????(nil)|37    
][ERRORA  2364 ][interface1][-1, 0, 0] Error sending data...
(????????????????????????) 
2011-05-25 10:25:44.046635 [ERR] gsmopen_protocol.cpp:696 rev ?????(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-25 10:25:44.046635 [ERR] mod_gsmopen.cpp:1913 rev ?????(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 2: gsmopen_serial_config failed,
let's try again
2011-05-25 10:25:44.146615 [DEBUG] gsmopen_protocol.cpp:2581 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2581 ][interface1][-1, 0, 0]
ucs2_in=006300690061006F0020003100320033002000620065006C00E80020043D043E0432043E044104420438002005DC05E7002005E805D005EA0020FE8EFEE0FEA0FEE4FECBFE9300204EBA5927
2011-05-25 10:25:44.146615 [DEBUG] gsmopen_protocol.cpp:2602 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2602 ][interface1][-1, 0, 0] 1 ciao in=,
inleft=76, out=, outleft=1000, converted=, utf8_out=
2011-05-25 10:25:44.146615 [DEBUG] gsmopen_protocol.cpp:2614 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2614 ][interface1][-1, 0, 0] iconv_res=0, 
in=, inleft=0, out=, outleft=933, converted=, utf8_out=ciao 123 bel? ???????
?? ??? ????????? ???
2011-05-25 10:25:44.146615 [DEBUG] gsmopen_protocol.cpp:632 rev
?????(nil)|37     ][DEBUG_GSMOPEN  632  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:44.659482 [DEBUG] gsmopen_protocol.cpp:689 rev
?????(nil)|37     ][DEBUG_GSMOPEN  689  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:45.159364 [DEBUG] gsmopen_protocol.cpp:2362 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2362 ][interface1][-1, 0, 0] sending: AT
2011-05-25 10:25:45.159364 [DEBUG] gsmopen_protocol.cpp:2246 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2246 ][interface1][-1, 0, 0] Error
sending (A): -1 (???????????
2011-05-25 10:25:45.259338 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 0 -1 (???????????
2011-05-25 10:25:45.359318 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 1 -1 (???????????
2011-05-25 10:25:45.459297 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 2 -1 (???????????
2011-05-25 10:25:45.559284 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 3 -1 (???????????
2011-05-25 10:25:45.659269 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 4 -1 (???????????
2011-05-25 10:25:45.759244 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 5 -1 (???????????
2011-05-25 10:25:45.859225 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 6 -1 (???????????
2011-05-25 10:25:45.959211 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 7 -1 (???????????
2011-05-25 10:25:46.059224 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 8 -1 (???????????
2011-05-25 10:25:46.159231 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 9 -1 (???????????
2011-05-25 10:25:46.259170 [ERR] gsmopen_protocol.cpp:2259 rev ?????(nil)|37    
][ERRORA  2259 ][interface1][-1, 0, 0] Error RE-sending (A): 10 -1
(????????????????????????)
2011-05-25 10:25:46.259170 [ERR] gsmopen_protocol.cpp:2364 rev ?????(nil)|37    
][ERRORA  2364 ][interface1][-1, 0, 0] Error sending data...
(????????????????????????) 
2011-05-25 10:25:46.259170 [ERR] gsmopen_protocol.cpp:696 rev ?????(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-25 10:25:46.259170 [ERR] mod_gsmopen.cpp:1913 rev ?????(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 3: gsmopen_serial_config failed,
let's try again
2011-05-25 10:25:46.349267 [DEBUG] gsmopen_protocol.cpp:2581 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2581 ][interface1][-1, 0, 0]
ucs2_in=006300690061006F0020003100320033002000620065006C00E80020043D043E0432043E044104420438002005DC05E7002005E805D005EA0020FE8EFEE0FEA0FEE4FECBFE9300204EBA5927
2011-05-25 10:25:46.349267 [DEBUG] gsmopen_protocol.cpp:2602 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2602 ][interface1][-1, 0, 0] 1 ciao in=,
inleft=76, out=, outleft=1000, converted=, utf8_out=
2011-05-25 10:25:46.349267 [DEBUG] gsmopen_protocol.cpp:2614 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2614 ][interface1][-1, 0, 0] iconv_res=0, 
in=, inleft=0, out=, outleft=933, converted=, utf8_out=ciao 123 bel? ???????
?? ??? ????????? ???
2011-05-25 10:25:46.349267 [DEBUG] gsmopen_protocol.cpp:632 rev
?????(nil)|37     ][DEBUG_GSMOPEN  632  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:46.849077 [DEBUG] gsmopen_protocol.cpp:689 rev
?????(nil)|37     ][DEBUG_GSMOPEN  689  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:47.348955 [DEBUG] gsmopen_protocol.cpp:2362 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2362 ][interface1][-1, 0, 0] sending: AT
2011-05-25 10:25:47.348955 [DEBUG] gsmopen_protocol.cpp:2246 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2246 ][interface1][-1, 0, 0] Error
sending (A): -1 (???????????
2011-05-25 10:25:47.448923 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 0 -1 (???????????
2011-05-25 10:25:47.548908 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 1 -1 (???????????
2011-05-25 10:25:47.648879 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 2 -1 (???????????
2011-05-25 10:25:47.748874 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 3 -1 (???????????
2011-05-25 10:25:47.848821 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 4 -1 (???????????
2011-05-25 10:25:47.948790 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 5 -1 (???????????
2011-05-25 10:25:48.048751 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 6 -1 (???????????
2011-05-25 10:25:48.148729 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 7 -1 (???????????
2011-05-25 10:25:48.248690 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 8 -1 (???????????
2011-05-25 10:25:48.348666 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 9 -1 (???????????
2011-05-25 10:25:48.448626 [ERR] gsmopen_protocol.cpp:2259 rev ?????(nil)|37    
][ERRORA  2259 ][interface1][-1, 0, 0] Error RE-sending (A): 10 -1
(????????????????????????)
2011-05-25 10:25:48.448626 [ERR] gsmopen_protocol.cpp:2364 rev ?????(nil)|37    
][ERRORA  2364 ][interface1][-1, 0, 0] Error sending data...
(????????????????????????) 
2011-05-25 10:25:48.448626 [ERR] gsmopen_protocol.cpp:696 rev ?????(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-25 10:25:48.448626 [ERR] mod_gsmopen.cpp:1913 rev ?????(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 4: gsmopen_serial_config failed,
let's try again
2011-05-25 10:25:48.557620 [DEBUG] gsmopen_protocol.cpp:2581 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2581 ][interface1][-1, 0, 0]
ucs2_in=006300690061006F0020003100320033002000620065006C00E80020043D043E0432043E044104420438002005DC05E7002005E805D005EA0020FE8EFEE0FEA0FEE4FECBFE9300204EBA5927
2011-05-25 10:25:48.557620 [DEBUG] gsmopen_protocol.cpp:2602 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2602 ][interface1][-1, 0, 0] 1 ciao in=,
inleft=76, out=, outleft=1000, converted=, utf8_out=
2011-05-25 10:25:48.557620 [DEBUG] gsmopen_protocol.cpp:2614 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2614 ][interface1][-1, 0, 0] iconv_res=0, 
in=, inleft=0, out=, outleft=933, converted=, utf8_out=ciao 123 bel? ???????
?? ??? ????????? ???
2011-05-25 10:25:48.557620 [DEBUG] gsmopen_protocol.cpp:632 rev
?????(nil)|37     ][DEBUG_GSMOPEN  632  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:49.057476 [DEBUG] gsmopen_protocol.cpp:689 rev
?????(nil)|37     ][DEBUG_GSMOPEN  689  ][interface1][-1, 0, 0] sleeping for
500000 usec
2011-05-25 10:25:49.557325 [DEBUG] gsmopen_protocol.cpp:2362 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2362 ][interface1][-1, 0, 0] sending: AT
2011-05-25 10:25:49.557325 [DEBUG] gsmopen_protocol.cpp:2246 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2246 ][interface1][-1, 0, 0] Error
sending (A): -1 (???????????
2011-05-25 10:25:49.657297 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 0 -1 (???????????
2011-05-25 10:25:49.757270 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 1 -1 (???????????
2011-05-25 10:25:49.857348 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 2 -1 (???????????
2011-05-25 10:25:49.957216 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 3 -1 (???????????
2011-05-25 10:25:50.057183 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 4 -1 (???????????
2011-05-25 10:25:50.157168 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 5 -1 (???????????
2011-05-25 10:25:50.257146 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 6 -1 (???????????
2011-05-25 10:25:50.357131 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 7 -1 (???????????
2011-05-25 10:25:50.457108 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 8 -1 (???????????
2011-05-25 10:25:50.557100 [DEBUG] gsmopen_protocol.cpp:2254 rev
?????(nil)|37     ][DEBUG_GSMOPEN  2254 ][interface1][-1, 0, 0] Error
RE-sending (A): 9 -1 (???????????
2011-05-25 10:25:50.658078 [ERR] gsmopen_protocol.cpp:2259 rev ?????(nil)|37    
][ERRORA  2259 ][interface1][-1, 0, 0] Error RE-sending (A): 10 -1
(????????????????????????)
2011-05-25 10:25:50.658078 [ERR] gsmopen_protocol.cpp:2364 rev ?????(nil)|37    
][ERRORA  2364 ][interface1][-1, 0, 0] Error sending data...
(????????????????????????) 
2011-05-25 10:25:50.658078 [ERR] gsmopen_protocol.cpp:696 rev ?????(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-25 10:25:50.658078 [ERR] mod_gsmopen.cpp:1913 rev ?????(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 5: gsmopen_serial_config failed,
let's try again
2011-05-25 10:25:50.658078 [ERR] mod_gsmopen.cpp:1917 rev ?????(nil)|37    
][ERRORA  1917 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED
2011-05-25 10:25:50.658078 [ERR] mod_gsmopen.cpp:3146 rev ?????(nil)|37    
][ERRORA  3146 ][interface1][-1, 0, 0] ALARM on interface interface1: 
2011-05-25 10:25:50.658078 [NOTICE] switch_loadable_module.c:145 Adding
Endpoint 'gsmopen'
2011-05-25 10:25:50.658078 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsm'
2011-05-25 10:25:50.658078 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen'
2011-05-25 10:25:50.658078 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen_boost_audio'
2011-05-25 10:25:50.658078 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen_dump'
2011-05-25 10:25:50.658078 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen_sendsms'
2011-05-25 10:25:50.658078 [NOTICE] switch_loadable_module.c:379 Adding Chat
interface 'SMS'


Regards
Tang du

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/load-mod-gsmopen-error-tp6401282p6401282.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From adam.kelloway at newpace.ca  Wed May 25 21:36:12 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Wed, 25 May 2011 14:36:12 -0300
Subject: [Freeswitch-users] transfer to extension without pattern
 matching
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BD8C592B2@VA3DIAXVS351.RED001.local>
References: <4DDD01F8.5030506@newpace.ca>	<BANLkTimUVmzC+CtnGCMDJNx4rR0w-a8rzA@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BD8C592B2@VA3DIAXVS351.RED001.local>
Message-ID: <4DDD3E0C.7050903@newpace.ca>

That's great, thanks folks!

On 3:59 PM, Robert Hadley wrote:
>
> Also check out 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension.  
> You would want to use it with the auto-hunt feature to skip extra 
> pattern matching.
>
> -Robert
>
> *From:*Avi Marcus [mailto:avi at avimarcus.net]
> *Sent:* Wednesday, May 25, 2011 6:27 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] transfer to extension without 
> pattern matching
>
> That sounds like it's describing 
> http://wiki.freeswitch.org/wiki/Variable_auto_hunt
>
> You need to turn that on. I'm not aware that even transfer does that 
> automatically, so afaik, it is indeed wrong - but is a feature that 
> FreeSWITCH possesses if you turn it on.
> -Avi
>
> On Wed, May 25, 2011 at 4:19 PM, Adam Kelloway 
> <adam.kelloway at newpace.ca <mailto:adam.kelloway at newpace.ca>> wrote:
>
> Hi there,
>
> The wiki page describing transfer
> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer) states
> the following:
>
> "Immediately transfer the calling channel to a new context. If there
> happens to be an xml extension named <destination_number> then control
> is "warped" directly to that extension. Otherwise it goes through the
> entire context checking for a match."
>
> Is this a correct statement? It seems that even when I try to transfer
> control directly to an extension, the entire context is still checked
> for a match.
>
> I am using the following:
>
> ..
> <application="transfer" data="test XML default"/>
> ..
>
> <extension name="test">
> <condition field="destination_number" expression="^test$">
>     ...
> </condition>
> </extension>
>
> Is there a way to disable the pattern matching/dial plan processing?
>
> Thanks,
>
> Adam
>
> _______________________________________________
> 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
>


-- 
Adam Kelloway
Software Engineer
NewPace Technology Development Inc.
adam.kelloway at newpace.ca
+1 902-406-8375 x1031
www.newpace.ca

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From Spencer at 5ninesolutions.com  Wed May 25 22:48:02 2011
From: Spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 25 May 2011 13:48:02 -0500
Subject: [Freeswitch-users] CNAM Providers
Message-ID: <4AC25B32-5EF4-4D68-A391-A03B1F3EBCAC@5ninesolutions.com>

Hello all,
We are considering adding US CNAM lookups on our inbound DIDs.  Does anyone have any experience with any of the providers listed on the wiki or any others for that matter. (i.e. accuracy, uptime, query response time, etc..)?

Thanks,
Spencer





From sos at sokhapkin.dyndns.org  Wed May 25 23:06:20 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 25 May 2011 15:06:20 -0400
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <4AC25B32-5EF4-4D68-A391-A03B1F3EBCAC@5ninesolutions.com>
References: <4AC25B32-5EF4-4D68-A391-A03B1F3EBCAC@5ninesolutions.com>
Message-ID: <201105251506.20730.sos@sokhapkin.dyndns.org>

http://www.callwithus.com/API#cnam

On Wednesday 25 May 2011, Spencer Thomason wrote:
> Hello all,
> We are considering adding US CNAM lookups on our inbound DIDs.  Does anyone
> have any experience with any of the providers listed on the wiki or any
> others for that matter. (i.e. accuracy, uptime, query response time,
> etc..)?
> 
> 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 bsevier at bandwidth.com  Wed May 25 23:07:39 2011
From: bsevier at bandwidth.com (Blake Sevier)
Date: Wed, 25 May 2011 13:07:39 -0600
Subject: [Freeswitch-users] FS not recognizing 200 OK after far-end call
	is answered
In-Reply-To: 341cf7e4fac39ca6e0b3e4b29166d3f8@mail.gmail.com
References: 341cf7e4fac39ca6e0b3e4b29166d3f8@mail.gmail.com
Message-ID: <7c5250ca8ee69193c4ff72aa04ba4bec@mail.gmail.com>

Please ignore ? it appears it?s due to the ?Session-Expires: -1? from
Asterisk ? that?s causing FS to drop the 200 OK.



Thanks,



Blake



*From:* Blake Sevier [mailto:bsevier at bandwidth.com]
*Sent:* Wednesday, May 25, 2011 10:58 AM
*To:* 'freeswitch-users at lists.freeswitch.org'
*Subject:* FS not recognizing 200 OK after far-end call is answered



Hi all,



I?m new, so please bear with me.  My setup is fairly basic, and the routing
all seems to work just fine:



FS receives call from an OpenSIPS proxy, it answers the call, records it and
plays back a short prompt before invoking the ?bridge? function to place an
outbound call via the same OpenSIPS proxy to an external SIP endpoint (in
testing, just an Asterisk instance that answers the call).  This all works
just fine, except FS doesn?t seem to recognize the 200 OK it receives back
for leg b after the Asterisk instances answers the call.  It apparently does
recognize/accept the 100 Trying, as FS doesn?t send additional INVITEs.  I?m
not seeing any issues with the SIP messaging received by FS ? it?s
apparently formatted correctly.

To note, I?ve also tested calls direct to the Asterisk instance (bypassing
OpenSIPS proxy), and get similar results.  My guess is that I?ve got
something incorrectly configured within my dialplan??  However, not sure how
to debug any further to figure out why the 200 Oks are not being
recognized.  Eventually, after 60s, FS ends the call with a NO_ANSWER.

Below is a link to a full call trace from FS ? from start of call to when I
see multiple 200 OKs from Asterisk.

http://pastebin.com/bmEtiW9X



Any help would be greatly appreciated.  If any further details are needed,
please let me know



Thank you!!!



Blake
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From yungwei at resolvity.com  Wed May 25 23:25:12 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 25 May 2011 15:25:12 -0400
Subject: [Freeswitch-users] Playing a wav file on a remote web server
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC4E9@VMBX102.ihostexchange.net>

Any suggestions?

-----Original Message-----
Subject: [Freeswitch-users] Playing a wav file on a remote web server

Hi,

I am currently using the following javascript snippet to play a wav file on a remote web server.
I'm wondering if there's another way to do it without involving a local file, just like mod_shout. 
Thanks.

 function on_dtmf(session, type, digits, arg)
 {
     dtmf_digits += digits.digit;
     console_log("dtmf_digits=" + dtmf_digits + "\n");
     return(true);
 }

 var ttsUrl = "http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin";
 var file = "/tmp/tts.wav";
 fetchURLFile(ttsUrl,file);
 session.streamFile(file, on_dtmf);

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


From steveayre at gmail.com  Wed May 25 23:39:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 25 May 2011 20:39:54 +0100
Subject: [Freeswitch-users] Playing a wav file on a remote web server
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>
Message-ID: <BANLkTinQrzhrUJz6dbSZ0UQ06SaVOWG8Gg@mail.gmail.com>

Have you tried:
session.streamFile("
http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin
")




On 24 May 2011 19:58, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I am currently using the following javascript snippet to play a wav file on
> a remote web server.
> I'm wondering if there's another way to do it without involving a local
> file, just like mod_shout.
> Thanks.
>
>  function on_dtmf(session, type, digits, arg)
>  {
>     dtmf_digits += digits.digit;
>     console_log("dtmf_digits=" + dtmf_digits + "\n");
>     return(true);
>  }
>
>  var ttsUrl = "
> http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin
> ";
>  var file = "/tmp/tts.wav";
>  fetchURLFile(ttsUrl,file);
>  session.streamFile(file, on_dtmf);
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yudha2008 at gmail.com  Thu May 26 00:27:32 2011
From: yudha2008 at gmail.com (Baskar)
Date: Wed, 25 May 2011 16:27:32 -0400
Subject: [Freeswitch-users] Inbound dialplan
Message-ID: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>

Hi All,,

In my inbound dial plan xml file  i have set two conditions enabled

Condition1:

In inbound dial plan callers are given an option to key in the extension
number and reach the appropriate extension (Example: 1001 or 1002 or 1003
etc).

Condition2:

The second condition routes call to a default extension in scenarios where
the caller does not specify any extension number (Example: Default extension
is 1007).


Both the above conditions are working fine.

Now I need to set up another condition where after keying in the extension
number the call gets transferred to the appropriate extension and if the
extension is busy(Example: say extension 1003 is busy) it should be hunted
to a default extension (example: 1007). How can we set up this condition in
dial plan?

Can any one guide me.
-- 
Thanks with Regards,

N.Baskar
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From p2.freeswitch-list at coleinternet.com  Wed May 25 22:55:55 2011
From: p2.freeswitch-list at coleinternet.com (Jay)
Date: Wed, 25 May 2011 11:55:55 -0700
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <4AC25B32-5EF4-4D68-A391-A03B1F3EBCAC@5ninesolutions.com>
References: <4AC25B32-5EF4-4D68-A391-A03B1F3EBCAC@5ninesolutions.com>
Message-ID: <4DDD50BB.1090901@coleinternet.com>

On 5/25/11 11:48 AM, Spencer Thomason wrote:
> Hello all,
> We are considering adding US CNAM lookups on our inbound DIDs.  Does anyone have any experience with any of the providers listed on the wiki or any others for that matter. (i.e. accuracy, uptime, query response time, etc..)?
whitepages.com is working ok...   they don't have everything.. but they're free.

mod_cidlookup

flowroute has (on request) an add-on to their DID offering  (Type: Standard CNAM) for an extra $2 /month per DID.


From brad at tech21.com  Wed May 25 23:29:59 2011
From: brad at tech21.com (Brad Mina)
Date: Wed, 25 May 2011 12:29:59 -0700
Subject: [Freeswitch-users] Playing a wav file on a remote web server
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC4E9@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950AC3AC4E9@VMBX102.ihostexchange.net>
Message-ID: <8697D8C7-6FCA-4723-8424-11BA5580101E@tech21.com>

Is there anything preventing you from using mod_shout? You could use mod_shout very easily from a number of languages. 



On May 25, 2011, at 12:25 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Any suggestions?
> 
> -----Original Message-----
> Subject: [Freeswitch-users] Playing a wav file on a remote web server
> 
> Hi,
> 
> I am currently using the following javascript snippet to play a wav file on a remote web server.
> I'm wondering if there's another way to do it without involving a local file, just like mod_shout. 
> Thanks.
> 
> function on_dtmf(session, type, digits, arg)
> {
>     dtmf_digits += digits.digit;
>     console_log("dtmf_digits=" + dtmf_digits + "\n");
>     return(true);
> }
> 
> var ttsUrl = "http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin";
> var file = "/tmp/tts.wav";
> fetchURLFile(ttsUrl,file);
> session.streamFile(file, on_dtmf);
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From yungwei at resolvity.com  Thu May 26 00:42:01 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 25 May 2011 16:42:01 -0400
Subject: [Freeswitch-users] Playing a wav file on a remote web server
In-Reply-To: <BANLkTinQrzhrUJz6dbSZ0UQ06SaVOWG8Gg@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>
	<BANLkTinQrzhrUJz6dbSZ0UQ06SaVOWG8Gg@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC538@VMBX102.ihostexchange.net>

Yes, and I got the following error.
[ERR] switch_core_file.c:122 Invalid file format [http] for [127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin]!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, May 25, 2011 2:40 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Playing a wav file on a remote web server

Have you tried:
session.streamFile("http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin")



On 24 May 2011 19:58, Yungwei Chen <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:
Hi,

I am currently using the following javascript snippet to play a wav file on a remote web server.
I'm wondering if there's another way to do it without involving a local file, just like mod_shout.
Thanks.

 function on_dtmf(session, type, digits, arg)
 {
    dtmf_digits += digits.digit;
    console_log("dtmf_digits=" + dtmf_digits + "\n");
    return(true);
 }

 var ttsUrl = "http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin";
 var file = "/tmp/tts.wav";
 fetchURLFile(ttsUrl,file);
 session.streamFile(file, on_dtmf);

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

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From msc at freeswitch.org  Thu May 26 00:48:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 May 2011 13:48:11 -0700
Subject: [Freeswitch-users] DTMF Problems! In-band detection patchy,
 rfc2833 not working!
In-Reply-To: <BANLkTikZUgytz0x7hfHGa=WgX4_G0d6W4A@mail.gmail.com>
References: <BANLkTi=A3azwFX+wH_DAAFRpzoaO96tmgQ@mail.gmail.com>
	<BANLkTimEocC1+WvpYNL9nsC4eS7AeVLkvw@mail.gmail.com>
	<BANLkTi=VeCq=grE7T7iYQfYWrp9f8w+vSg@mail.gmail.com>
	<BANLkTi=GV_FdH-3d-wamOryLJXcXEM2smw@mail.gmail.com>
	<BANLkTikZUgytz0x7hfHGa=WgX4_G0d6W4A@mail.gmail.com>
Message-ID: <BANLkTimr7JggVoaTtmx9Nt1SzYk0wLHKDw@mail.gmail.com>

On Tue, May 24, 2011 at 10:08 PM, Sidharth Kshatriya <
sid.kshatriya at gmail.com> wrote:

> The tech support guys from my VOIP carrier had the following to say:
>
> "Have you told freeswitch to ANSWER the call prior to the the IVR coming
> into play as by default it would answer EARLY MEDIA thus allowing only 1 way
> audio and no DTMF."
>
> What does this mean? I answer the call within the lua script.
>

If you have something like "session:answer()" in your script then you're
good. You can also call the "answer" app prior to sending the call to the
Lua script. If you do a SIP trace and you see FS sending back a 200 OK then
you are telling the VoIP provider that you've answered the call. If you see
a 180 or 183 w/SDP but no 200 OK then that would indicate you've only
established early media and not actually answered the call. It sounds like
the carrier does not send audio in early media, which basically makes sense.
Look at the early media article on our wiki if you need to learn more about
it...

-MC
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From yungwei at resolvity.com  Thu May 26 01:16:49 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 25 May 2011 17:16:49 -0400
Subject: [Freeswitch-users] Playing a wav file on a remote web server
In-Reply-To: <8697D8C7-6FCA-4723-8424-11BA5580101E@tech21.com>
References: <33095823FD21DF429B481B5163264B7950AC3AC1D8@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950AC3AC4E9@VMBX102.ihostexchange.net>
	<8697D8C7-6FCA-4723-8424-11BA5580101E@tech21.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC564@VMBX102.ihostexchange.net>

No. I just want to explore more options.
It would be nice to support playing remote wav files.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brad Mina
Sent: Wednesday, May 25, 2011 2:30 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Playing a wav file on a remote web server

Is there anything preventing you from using mod_shout? You could use mod_shout very easily from a number of languages. 



On May 25, 2011, at 12:25 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Any suggestions?
> 
> -----Original Message-----
> Subject: [Freeswitch-users] Playing a wav file on a remote web server
> 
> Hi,
> 
> I am currently using the following javascript snippet to play a wav file on a remote web server.
> I'm wondering if there's another way to do it without involving a local file, just like mod_shout. 
> Thanks.
> 
> function on_dtmf(session, type, digits, arg)
> {
>     dtmf_digits += digits.digit;
>     console_log("dtmf_digits=" + dtmf_digits + "\n");
>     return(true);
> }
> 
> var ttsUrl = "http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20pin";
> var file = "/tmp/tts.wav";
> fetchURLFile(ttsUrl,file);
> session.streamFile(file, on_dtmf);
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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> http://www.freeswitch.org
> 
> _______________________________________________
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> http://www.freeswitch.org

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From msc at freeswitch.org  Thu May 26 01:25:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 May 2011 14:25:45 -0700
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
Message-ID: <BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>

If I understand correctly you're wanting to handle a bridge scenario where
the target extension is busy, etc. Most likely you just need to set
continue_on_fail=true so that your dialplan continues in the case of the
target extension not picking up.

-MC

On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:

> Hi All,,
>
> In my inbound dial plan xml file  i have set two conditions enabled
>
> Condition1:
>
> In inbound dial plan callers are given an option to key in the extension
> number and reach the appropriate extension (Example: 1001 or 1002 or 1003
> etc).
>
> Condition2:
>
> The second condition routes call to a default extension in scenarios where
> the caller does not specify any extension number (Example: Default extension
> is 1007).
>
>
> Both the above conditions are working fine.
>
> Now I need to set up another condition where after keying in the extension
> number the call gets transferred to the appropriate extension and if the
> extension is busy(Example: say extension 1003 is busy) it should be hunted
> to a default extension (example: 1007). How can we set up this condition in
> dial plan?
>
> Can any one guide me.
> --
> Thanks with Regards,
>
> N.Baskar
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu May 26 01:33:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 May 2011 14:33:07 -0700
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTin1V6TinJWsOVorkwAMTXc88G_mcg@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
	<BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
	<BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>
	<BANLkTin1V6TinJWsOVorkwAMTXc88G_mcg@mail.gmail.com>
Message-ID: <BANLkTi=SPWEzBs0vAS=T_2wErdawdHo=FA@mail.gmail.com>

http://search.cpan.org/~markstos/CGI.pm-3.54/lib/CGI/Fast.pm

That module looks like it is built on top of FCGI and gives you a CGI-like
interface.

-MC

On Tue, May 24, 2011 at 10:12 PM, deniro <ayhkor at gmail.com> wrote:

>
> Thanks for the replies
> I made some research and run series of tests.
> I am using  perl+FCGI (fastcgi)  and my http server is  working fine as
> tested with multiple perl scripts.
>  I think the problem is handling the CGI within the cdr.pl file(obtained
> from freeswitch wiki)
>
>  so, how can I change below script to use with FCGI instead of CGI::Simple
>
> use XML::Simple; # Get from CPAN
> use CGI::Simple; # Get from CPAN
> use Data::Dumper;
> # dump into a place for further review
> open(FILEOUT,">",'/tmp/fs-xml-cdr2.log');
> print FILEOUT "Test successful..\n";
> cloe(FILEOUT);
> # dump into a place for further review
> open(FILEOUT,">>",'/tmp/fs-xml-cdr.log');
> # $cgi object has handy methods
> my $cgi = new CGI::Simple;
>
>
> Thx
> deniro
>
>
>
>
> On Mon, May 23, 2011 at 3:50 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>>  > 2-- when dialing into conference, getting following from fs_cli
>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>
>>> There isn't a lot of info to work on here, but I'd look at your web
>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>> means you have a bad gateway).
>>>
>>
>> Check your web server logs - there may be some useful message. It sounds
>> like the cgi-bin/perl handler hasn't been configured correctly, the script
>> isn't executable, or it's not returning some headers. Try visiting the URL
>> in your webbrowser (you
>> won't be submitting a CDR but you can at least check the script runs
>> without an (unexpected) error.
>>
>>
>>
>>
>>
>> On 23 May 2011 07:38, Gabriel Gunderson <gabe at gundy.org> wrote:
>>
>>> On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
>>> > several questions;
>>> > 1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
>>> > trying to figure out if my problem is with cdr or with curl
>>>
>>> No, they are not functionally related... easy to get mixed up because
>>> they both use HTTP and XML. One does not depend on the other. However,
>>> you *can* configure mod_xml_cdr (or any other module) with
>>> mod_xml_curl.
>>>
>>>
>>>
>>> > 2-- when dialing into conference, getting following from fs_cli
>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>
>>> There isn't a lot of info to work on here, but I'd look at your web
>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>> means you have a bad gateway).
>>>
>>>
>>>
>>> > 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
>>> >  <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>
>>>
>>> Looks good to me, but check it with the wiki:
>>> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration
>>>
>>>
>>>
>>> > 4-- if I need mod_xml_curl (question 1)
>>> > what should this line be like in xml_curl.conf.xml?
>>> > <param name="gateway-url" value="http://www.xxx.com:80<http://www.xxx.com/>
>>> "
>>> > bindings="dialplan"/>
>>>
>>> This one is a little tricker to get right.  Again, the wiki is might
>>> be the best place to reference:
>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring
>>>
>>>
>>>
>>> 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
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing 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 msc at freeswitch.org  Thu May 26 01:34:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 May 2011 14:34:49 -0700
Subject: [Freeswitch-users] rtpmap line missing on answer
In-Reply-To: <7EB00560C8AFF348A7EA76EFD3DC5598332C0367@dtn1mbx01.gci.com>
References: <7EB00560C8AFF348A7EA76EFD3DC5598332C0358@dtn1mbx01.gci.com>
	<E1AEB5C4-B1CD-49EC-891E-D832BBD23D6A@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035A@dtn1mbx01.gci.com>
	<BANLkTiky3gUeZz=7R43BnnYYEXFxx_JBUw@mail.gmail.com>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C035F@dtn1mbx01.gci.com>
	<DF5E4E14-2338-4871-A73E-2291537069B9@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C0360@dtn1mbx01.gci.com>
	<E8079B89-5EDC-4130-A331-DA923CBD292D@ipeva.fr>
	<7EB00560C8AFF348A7EA76EFD3DC5598332C0367@dtn1mbx01.gci.com>
Message-ID: <BANLkTikNYC125snAH0eNWNDJPkJMgk-0+w@mail.gmail.com>

FWIW, you can use "pre_answer" to send 183 w/SDP from FS; "ring_ready" will
send 180. See the early media article on the wiki for more information.
-MC

On Tue, May 24, 2011 at 2:33 PM, Sean Eichhorn <seichhorn at gci.com> wrote:

>  Actually, I just found another interesting piece of the puzzle.
>
> On the outbound call (that works), the gateway responds with a 183/SDP
> message.
>
> The inbound call (that doesn?t work) returns a 180 Ringing/NO SDP message.
>
> Both legs are using the same profile, so I don?t think that?s an issue.
> For some reason though, if a 183/SDP is sent, Freeswitch forwards the rtpmap
> attribute.  If no 183 is detected, it deletes the attribute.
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David
> Ponzone
> *Sent:* Tuesday, May 24, 2011 01:12
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] rtpmap line missing on answer
>
>
>
> Sean,
>
>
>
> can you check the configuration of the SIP profile used by both legs ?
>
> Perhaps you have late-negotiation enabled in one of them ?
>
>
>
> David Ponzone  Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel:      01 74 03 18 97
>
> gsm:   06 66 98 76 34
>
>
>
> Service Client IPeva
>
> tel:      0811 46 26 26
>
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
>
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
>
> * *
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yudha2008 at gmail.com  Thu May 26 02:27:35 2011
From: yudha2008 at gmail.com (Baskar)
Date: Wed, 25 May 2011 18:27:35 -0400
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
Message-ID: <BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>

Hi Michael,

Thanks for your quick reply. Below is the code for the dial plan inbound
ivr. Can you please specify where i should be inserting the
continue_on_fail=true line in the code.

Default.xml

    <extension name="inbound">
      <condition field="destination_number" expression="^(XXXXXXXXXX)$" />
      <condition wday="2-6" hour="8-17">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="accountcode=Inbound"/>
        <action application="answer"/>
        <action application="start_dtmf" />
        <action application="ivr" data="ivr_inbound"/>
        <action application="record_session"
data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
        <action application="transfer" data="1018 XML default"/>
      </condition>
    </extension>

ivr_inbound.xml

<menu name="ivr_inbound"

 greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
     invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
     exit-sound="voicemail/vm-goodbye.wav"
     confirm-macro=""
     confirm-key=""
     tts-engine="flite"
     tts-voice="rms"
     confirm-attempts="3"
     timeout="10000"
     inter-digit-timeout="2000"
     max-failures="3"
     max-timeouts="1"
     digit-len="4">
   <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
param="transfer $1 XML default"/>
   <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
default"/>
</menu>

*--
Thanks with Regards,

N.Baskar

*
On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org> wrote:

> If I understand correctly you're wanting to handle a bridge scenario where
> the target extension is busy, etc. Most likely you just need to set
> continue_on_fail=true so that your dialplan continues in the case of the
> target extension not picking up.
>
> -MC
>
> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
>
>> Hi All,,
>>
>> In my inbound dial plan xml file  i have set two conditions enabled
>>
>> Condition1:
>>
>> In inbound dial plan callers are given an option to key in the extension
>> number and reach the appropriate extension (Example: 1001 or 1002 or 1003
>> etc).
>>
>> Condition2:
>>
>> The second condition routes call to a default extension in scenarios where
>> the caller does not specify any extension number (Example: Default extension
>> is 1007).
>>
>>
>> Both the above conditions are working fine.
>>
>> Now I need to set up another condition where after keying in the extension
>> number the call gets transferred to the appropriate extension and if the
>> extension is busy(Example: say extension 1003 is busy) it should be hunted
>> to a default extension (example: 1007). How can we set up this condition in
>> dial plan?
>>
>> Can any one guide me.
>> --
>> Thanks with Regards,
>>
>> N.Baskar
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From adminjew at gmail.com  Thu May 26 02:40:12 2011
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 25 May 2011 18:40:12 -0400
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <201105251506.20730.sos@sokhapkin.dyndns.org>
References: <4AC25B32-5EF4-4D68-A391-A03B1F3EBCAC@5ninesolutions.com>
	<201105251506.20730.sos@sokhapkin.dyndns.org>
Message-ID: <BANLkTikosE0xgty5gK-+J6TNYw-xhGJ9BA@mail.gmail.com>

callwithus is nice but they are not that up to date on all listings.


Yitzchok


On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> http://www.callwithus.com/API#cnam
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From krice at freeswitch.org  Thu May 26 03:03:49 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 25 May 2011 18:03:49 -0500
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <BANLkTikosE0xgty5gK-+J6TNYw-xhGJ9BA@mail.gmail.com>
Message-ID: <CA02F505.18B5B%krice@freeswitch.org>

You have to keep in mind that even with SS7 interconnects with companies
like callwithus they may or may not have access to the full national cnam
infrastructure

CNAM is hosted by the LEC that owns the DIDs and you have to have an
interconnect agreement with them to get to their CNAM...  Simply having an
SS7 connection is not enuff

K


On 5/25/11 5:40 PM, "Yitzchok" <adminjew at gmail.com> wrote:

> callwithus is nice but they are not that up to date on all listings.
> 
> 
> Yitzchok
> 
> 
> On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> wrote:
>> http://www.callwithus.com/API#cnam
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From adminjew at gmail.com  Thu May 26 03:33:50 2011
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 25 May 2011 19:33:50 -0400
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <CA02F505.18B5B%krice@freeswitch.org>
References: <BANLkTikosE0xgty5gK-+J6TNYw-xhGJ9BA@mail.gmail.com>
	<CA02F505.18B5B%krice@freeswitch.org>
Message-ID: <BANLkTi=s12cXHNd7zMJ9ya4j7_jAPeDh=w@mail.gmail.com>

Is there any CNAM provider that have access to all or most (C/I)LEC's?


Yitzchok


On Wed, May 25, 2011 at 7:03 PM, Ken Rice <krice at freeswitch.org> wrote:

>  You have to keep in mind that even with SS7 interconnects with companies
> like callwithus they may or may not have access to the full national cnam
> infrastructure
>
> CNAM is hosted by the LEC that owns the DIDs and you have to have an
> interconnect agreement with them to get to their CNAM...  Simply having an
> SS7 connection is not enuff
>
> K
>
>
>
> On 5/25/11 5:40 PM, "Yitzchok" <adminjew at gmail.com> wrote:
>
> callwithus is nice but they are not that up to date on all listings.
>
>
> Yitzchok
>
>
> On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> wrote:
>
> http://www.callwithus.com/API#cnam
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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From krice at freeswitch.org  Thu May 26 04:24:51 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 25 May 2011 19:24:51 -0500
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <BANLkTi=s12cXHNd7zMJ9ya4j7_jAPeDh=w@mail.gmail.com>
Message-ID: <CA030803.18B5F%krice@freeswitch.org>

You could go with someone like http://tnsi.com/ but its not cheap


On 5/25/11 6:33 PM, "Yitzchok" <adminjew at gmail.com> wrote:

> Is there any CNAM provider that have access to all or most (C/I)LEC's?
> 
> 
> Yitzchok
> 
> 
> On Wed, May 25, 2011 at 7:03 PM, Ken Rice <krice at freeswitch.org> wrote:
>> You have to keep in mind that even with SS7 interconnects with companies like
>> callwithus they may or may not have access to the full national cnam
>> infrastructure
>> 
>> CNAM is hosted by the LEC that owns the DIDs and you have to have an
>> interconnect agreement with them to get to their CNAM... ?Simply having an
>> SS7 connection is not enuff
>> 
>> K
>> 
>> 
>> 
>> On 5/25/11 5:40 PM, "Yitzchok" <adminjew at gmail.com
>> <http://adminjew at gmail.com> > wrote:
>> 
>>> callwithus is nice but they are not that up to date on all listings.
>>> 
>>> 
>>> Yitzchok
>>> 
>>> 
>>> On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
>>> <http://sos at sokhapkin.dyndns.org> > wrote:
>>>> http://www.callwithus.com/API#cnam
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frisch.alan at gmail.com  Thu May 26 05:04:29 2011
From: frisch.alan at gmail.com (Alan Frisch)
Date: Wed, 25 May 2011 21:04:29 -0400
Subject: [Freeswitch-users] Freeswitch DBH - Possible to create connection
	pool to other DBs?
Message-ID: <BANLkTimFS0HJNOD2p6+T+ziRUrkPvDKCUw@mail.gmail.com>

Been playing with the DBH functionality in LUA.  The connection
pooling works great if you need to access the core DSN... but if you
need to access an external DB, it seems that FS will open a single
connection for the maximum number of concurrent calls accessing the
LUA script.

In * one can limit the connection pool to any DB in res_odbc.conf...
but is there a way of doing this in FS?

After a learning curve and lots of time, tthis is the one hangup in FS
that is preventing me from trashing * altogether.  Help me put * in
the dustbin for good!


From yungwei at resolvity.com  Thu May 26 07:39:30 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 25 May 2011 23:39:30 -0400
Subject: [Freeswitch-users] Making an outbound call to a cell phone and
 letting the callee interact with a JavaScript program
Message-ID: <33095823FD21DF429B481B5163264B7950AC4FB5EC@VMBX102.ihostexchange.net>

Hi,

I am wondering which application allows me to do the following:

1. make an outbound call from a SIP phone to a cell phone.
2. When the call is answered, run a JavaScript application (so the caller can interact with the JavaScript program.)

Thanks. 

From ayhkor at gmail.com  Thu May 26 09:22:22 2011
From: ayhkor at gmail.com (deniro)
Date: Thu, 26 May 2011 01:22:22 -0400
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTi=SPWEzBs0vAS=T_2wErdawdHo=FA@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
	<BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
	<BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>
	<BANLkTin1V6TinJWsOVorkwAMTXc88G_mcg@mail.gmail.com>
	<BANLkTi=SPWEzBs0vAS=T_2wErdawdHo=FA@mail.gmail.com>
Message-ID: <BANLkTimXTTJ5L2RhqCs6CQ9T_Wqk4DS2Rg@mail.gmail.com>

Thanks a lot. The link was a good help.
we tested with sample code in the link and no more 502 errors with the
sample code in the link.

We tried to make changes in our cdr.pl perl (using mod_xml_cdr) program
accordingly
we made  good progres in generatng  xml file, however we are still getting
502 errors from cdr.pl.
Probably something else in our cdr.pl  giving us that, we are still trying
to figure out  what  piece is that.

In the mean time,  in generated  xml  we see a line like this
   * <^[7]\d{4}\$>77555</^[7]\d{4}\$>*
and looks like this coming from
*$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
"conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav", "",
'^[7]\d{4}\$');*
and because of above xml line I am unable to parse xml.
(above tested with changes and changes reflect on xml line)

why do you think
above $session->playAndGetDigits   is producing xml  line like
"<^[7]\d{4}\$>77555</^[7]\d{4}\$>"

any ideas appreciated.
why and how to fix it.

thx again
deniro--










On Wed, May 25, 2011 at 5:33 PM, Michael Collins <msc at freeswitch.org> wrote:

> http://search.cpan.org/~markstos/CGI.pm-3.54/lib/CGI/Fast.pm
>
> That module looks like it is built on top of FCGI and gives you a CGI-like
> interface.
>
> -MC
>
>
> On Tue, May 24, 2011 at 10:12 PM, deniro <ayhkor at gmail.com> wrote:
>
>>
>> Thanks for the replies
>> I made some research and run series of tests.
>> I am using  perl+FCGI (fastcgi)  and my http server is  working fine as
>> tested with multiple perl scripts.
>>  I think the problem is handling the CGI within the cdr.pl file(obtained
>> from freeswitch wiki)
>>
>>  so, how can I change below script to use with FCGI instead of
>> CGI::Simple
>>
>> use XML::Simple; # Get from CPAN
>> use CGI::Simple; # Get from CPAN
>> use Data::Dumper;
>> # dump into a place for further review
>> open(FILEOUT,">",'/tmp/fs-xml-cdr2.log');
>> print FILEOUT "Test successful..\n";
>> cloe(FILEOUT);
>> # dump into a place for further review
>> open(FILEOUT,">>",'/tmp/fs-xml-cdr.log');
>> # $cgi object has handy methods
>> my $cgi = new CGI::Simple;
>>
>>
>> Thx
>> deniro
>>
>>
>>
>>
>> On Mon, May 23, 2011 at 3:50 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>>  > 2-- when dialing into conference, getting following from fs_cli
>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>
>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>>> means you have a bad gateway).
>>>>
>>>
>>> Check your web server logs - there may be some useful message. It sounds
>>> like the cgi-bin/perl handler hasn't been configured correctly, the script
>>> isn't executable, or it's not returning some headers. Try visiting the URL
>>> in your webbrowser (you
>>> won't be submitting a CDR but you can at least check the script runs
>>> without an (unexpected) error.
>>>
>>>
>>>
>>>
>>>
>>> On 23 May 2011 07:38, Gabriel Gunderson <gabe at gundy.org> wrote:
>>>
>>>> On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
>>>> > several questions;
>>>> > 1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
>>>> > trying to figure out if my problem is with cdr or with curl
>>>>
>>>> No, they are not functionally related... easy to get mixed up because
>>>> they both use HTTP and XML. One does not depend on the other. However,
>>>> you *can* configure mod_xml_cdr (or any other module) with
>>>> mod_xml_curl.
>>>>
>>>>
>>>>
>>>> > 2-- when dialing into conference, getting following from fs_cli
>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>
>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>>> means you have a bad gateway).
>>>>
>>>>
>>>>
>>>> > 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
>>>> >  <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>
>>>>
>>>> Looks good to me, but check it with the wiki:
>>>> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration
>>>>
>>>>
>>>>
>>>> > 4-- if I need mod_xml_curl (question 1)
>>>> > what should this line be like in xml_curl.conf.xml?
>>>> > <param name="gateway-url" value="http://www.xxx.com:80<http://www.xxx.com/>
>>>> "
>>>> > bindings="dialplan"/>
>>>>
>>>> This one is a little tricker to get right.  Again, the wiki is might
>>>> be the best place to reference:
>>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring
>>>>
>>>>
>>>>
>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Thu May 26 09:49:48 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 26 May 2011 07:49:48 +0200
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
Message-ID: <741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>

Baskar,

If you want this behavior only for calls only coming through the IVR, I think you will have to use bridge rather than transfer.
it would look like this:

   <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/" param="execute_extension set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>

or something like it :)

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

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

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




Le 26/05/2011 ? 00:27, Baskar a ?crit :

> Hi Michael,
> 
> Thanks for your quick reply. Below is the code for the dial plan inbound ivr. Can you please specify where i should be inserting the continue_on_fail=true line in the code.
> 
> Default.xml
> 
>     <extension name="inbound">
>       <condition field="destination_number" expression="^(XXXXXXXXXX)$" />
>       <condition wday="2-6" hour="8-17">
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="accountcode=Inbound"/>
>         <action application="answer"/>
>         <action application="start_dtmf" />
>         <action application="ivr" data="ivr_inbound"/>
>         <action application="record_session" data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>         <action application="transfer" data="1018 XML default"/>
>       </condition>
>     </extension>
> 
> ivr_inbound.xml
> 
> <menu name="ivr_inbound"
>      greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>      exit-sound="voicemail/vm-goodbye.wav"
>      confirm-macro=""
>      confirm-key=""
>      tts-engine="flite"
>      tts-voice="rms"
>      confirm-attempts="3"
>      timeout="10000"
>      inter-digit-timeout="2000"
>      max-failures="3"
>      max-timeouts="1"
>      digit-len="4">
>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/" param="transfer $1 XML default"/>
>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML default"/>
> </menu>
> 
> -- 
> Thanks with Regards,
> 
> N.Baskar 
> 
> 
> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org> wrote:
> If I understand correctly you're wanting to handle a bridge scenario where the target extension is busy, etc. Most likely you just need to set continue_on_fail=true so that your dialplan continues in the case of the target extension not picking up.
> 
> -MC
> 
> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
> Hi All,,
>  
> In my inbound dial plan xml file  i have set two conditions enabled 
> 
> Condition1: 
> 
> In inbound dial plan callers are given an option to key in the extension number and reach the appropriate extension (Example: 1001 or 1002 or 1003 etc). 
> 
> Condition2:
> 
> The second condition routes call to a default extension in scenarios where the caller does not specify any extension number (Example: Default extension is 1007).
> 
>  
> Both the above conditions are working fine.
> 
> Now I need to set up another condition where after keying in the extension number the call gets transferred to the appropriate extension and if the extension is busy(Example: say extension 1003 is busy) it should be hunted to a default extension (example: 1007). How can we set up this condition in dial plan?
> 
> Can any one guide me.
> -- 
> Thanks with Regards,
> 
> N.Baskar 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lautram.mathieu at gmail.com  Thu May 26 09:54:14 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Thu, 26 May 2011 07:54:14 +0200
Subject: [Freeswitch-users] Errors while sending faxes
Message-ID: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>

Hi all,

After sending faxes (it works very well know, thanks to Anthony Minessale) ,
sometimes I've got errors like that:

- Disconnected after permitted retries
- No response after sending a page
- Received a DCN from remote after sending a page
- Received bad response to DCS ot TCF
- The call dropped prematurely
- Timed out waiting for initial communication

Those faxes are about 25% of all faxes that I send. I'm using the last git
version.
Where this problem comes from? Is there a way to fix this?
I know that sometimes the remote end could be broken or could be an old fax
machine, but it seems strange that I have so many errors like that.

Thank you for your help :-)

-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From infos at madovsky.org  Thu May 26 10:04:13 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 26 May 2011 02:04:13 -0400
Subject: [Freeswitch-users] Errors while sending faxes
References: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>
Message-ID: <ED055084A034476C877440F041F8D4E1@e1705>

Please don't send several same email
  ----- Original Message ----- 
  From: Mathieu Lautram 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 26, 2011 1:54 AM
  Subject: [Freeswitch-users] Errors while sending faxes


  Hi all,


  After sending faxes (it works very well know, thanks to Anthony Minessale) , sometimes I've got errors like that:


  - Disconnected after permitted retries
  - No response after sending a page
  - Received a DCN from remote after sending a page
  - Received bad response to DCS ot TCF
  - The call dropped prematurely
  - Timed out waiting for initial communication


  Those faxes are about 25% of all faxes that I send. I'm using the last git version.
  Where this problem comes from? Is there a way to fix this?
  I know that sometimes the remote end could be broken or could be an old fax machine, but it seems strange that I have so many errors like that.


  Thank you for your help :-)

  -- 
  Mathieu LAUTRAM
  Application developer

  BJT Partners - FRANCE
  +33 1 79 75 99 60
  +33 6 61 59 07 25



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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From steveayre at gmail.com  Thu May 26 10:25:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 26 May 2011 07:25:52 +0100
Subject: [Freeswitch-users] Freeswitch DBH - Possible to create
 connection pool to other DBs?
In-Reply-To: <BANLkTimFS0HJNOD2p6+T+ziRUrkPvDKCUw@mail.gmail.com>
References: <BANLkTimFS0HJNOD2p6+T+ziRUrkPvDKCUw@mail.gmail.com>
Message-ID: <BANLkTinOV_ZW8Ow2p4JUbCT8i8ck1DY3tg@mail.gmail.com>

Not sure what you mean here...

When a script runs it'll reuse a connection from the pool if there is one.
If there isn't it'll open a new one.

If you only every have one copy of your script running at a time you'll only
get a single connection. But if you get 2 copies running at the same time
you'll get 2, 3 copies 3 etc.


-Steve


On 26 May 2011 02:04, Alan Frisch <frisch.alan at gmail.com> wrote:

> Been playing with the DBH functionality in LUA.  The connection
> pooling works great if you need to access the core DSN... but if you
> need to access an external DB, it seems that FS will open a single
> connection for the maximum number of concurrent calls accessing the
> LUA script.
>
> In * one can limit the connection pool to any DB in res_odbc.conf...
> but is there a way of doing this in FS?
>
> After a learning curve and lots of time, tthis is the one hangup in FS
> that is preventing me from trashing * altogether.  Help me put * in
> the dustbin for good!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu May 26 10:34:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 26 May 2011 07:34:52 +0100
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTimXTTJ5L2RhqCs6CQ9T_Wqk4DS2Rg@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
	<BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
	<BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>
	<BANLkTin1V6TinJWsOVorkwAMTXc88G_mcg@mail.gmail.com>
	<BANLkTi=SPWEzBs0vAS=T_2wErdawdHo=FA@mail.gmail.com>
	<BANLkTimXTTJ5L2RhqCs6CQ9T_Wqk4DS2Rg@mail.gmail.com>
Message-ID: <BANLkTinuScKyL+sSJ7vnt0czPqcaerKmCw@mail.gmail.com>

*$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
"conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav", "",
'^[7]\d{4}\$');

*It looks like this usage is wrong. You're giving 9 parameters but the Wiki
seems to say it's only 8. It looks like it's using the 9th as a channel
variable name to store the result in.

Try this:
*$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
"conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav",
'^[7]\d{4}\$');*

-Steve



On 26 May 2011 06:22, deniro <ayhkor at gmail.com> wrote:

> Thanks a lot. The link was a good help.
> we tested with sample code in the link and no more 502 errors with the
> sample code in the link.
>
> We tried to make changes in our cdr.pl perl (using mod_xml_cdr) program
> accordingly
> we made  good progres in generatng  xml file, however we are still getting
> 502 errors from cdr.pl.
> Probably something else in our cdr.pl  giving us that, we are still trying
> to figure out  what  piece is that.
>
> In the mean time,  in generated  xml  we see a line like this
>    * <^[7]\d{4}\$>77555</^[7]\d{4}\$>
> *
> and looks like this coming from
> *$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
> "conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav", "",
> '^[7]\d{4}\$');*
> and because of above xml line I am unable to parse xml.
> (above tested with changes and changes reflect on xml line)
>
> why do you think
> above $session->playAndGetDigits   is producing xml  line like
> "<^[7]\d{4}\$>77555</^[7]\d{4}\$>"
>
> any ideas appreciated.
> why and how to fix it.
>
> thx again
> deniro--
>
>
>
>
>
>
>
>
>
>
> On Wed, May 25, 2011 at 5:33 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> http://search.cpan.org/~markstos/CGI.pm-3.54/lib/CGI/Fast.pm
>>
>> That module looks like it is built on top of FCGI and gives you a CGI-like
>> interface.
>>
>> -MC
>>
>>
>> On Tue, May 24, 2011 at 10:12 PM, deniro <ayhkor at gmail.com> wrote:
>>
>>>
>>> Thanks for the replies
>>> I made some research and run series of tests.
>>> I am using  perl+FCGI (fastcgi)  and my http server is  working fine as
>>> tested with multiple perl scripts.
>>>  I think the problem is handling the CGI within the cdr.pl file(obtained
>>> from freeswitch wiki)
>>>
>>>  so, how can I change below script to use with FCGI instead of
>>> CGI::Simple
>>>
>>> use XML::Simple; # Get from CPAN
>>> use CGI::Simple; # Get from CPAN
>>> use Data::Dumper;
>>> # dump into a place for further review
>>> open(FILEOUT,">",'/tmp/fs-xml-cdr2.log');
>>> print FILEOUT "Test successful..\n";
>>> cloe(FILEOUT);
>>> # dump into a place for further review
>>> open(FILEOUT,">>",'/tmp/fs-xml-cdr.log');
>>> # $cgi object has handy methods
>>> my $cgi = new CGI::Simple;
>>>
>>>
>>> Thx
>>> deniro
>>>
>>>
>>>
>>>
>>> On Mon, May 23, 2011 at 3:50 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>>  > 2-- when dialing into conference, getting following from fs_cli
>>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>>
>>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>>>> means you have a bad gateway).
>>>>>
>>>>
>>>> Check your web server logs - there may be some useful message. It sounds
>>>> like the cgi-bin/perl handler hasn't been configured correctly, the script
>>>> isn't executable, or it's not returning some headers. Try visiting the URL
>>>> in your webbrowser (you
>>>> won't be submitting a CDR but you can at least check the script runs
>>>> without an (unexpected) error.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On 23 May 2011 07:38, Gabriel Gunderson <gabe at gundy.org> wrote:
>>>>
>>>>> On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
>>>>> > several questions;
>>>>> > 1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
>>>>> > trying to figure out if my problem is with cdr or with curl
>>>>>
>>>>> No, they are not functionally related... easy to get mixed up because
>>>>> they both use HTTP and XML. One does not depend on the other. However,
>>>>> you *can* configure mod_xml_cdr (or any other module) with
>>>>> mod_xml_curl.
>>>>>
>>>>>
>>>>>
>>>>> > 2-- when dialing into conference, getting following from fs_cli
>>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>>
>>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>>>> means you have a bad gateway).
>>>>>
>>>>>
>>>>>
>>>>> > 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
>>>>> >  <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>
>>>>>
>>>>> Looks good to me, but check it with the wiki:
>>>>> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration
>>>>>
>>>>>
>>>>>
>>>>> > 4-- if I need mod_xml_curl (question 1)
>>>>> > what should this line be like in xml_curl.conf.xml?
>>>>> > <param name="gateway-url" value="http://www.xxx.com:80<http://www.xxx.com/>
>>>>> "
>>>>> > bindings="dialplan"/>
>>>>>
>>>>> This one is a little tricker to get right.  Again, the wiki is might
>>>>> be the best place to reference:
>>>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring
>>>>>
>>>>>
>>>>>
>>>>> 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
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lautram.mathieu at gmail.com  Thu May 26 11:49:11 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Thu, 26 May 2011 09:49:11 +0200
Subject: [Freeswitch-users] Errors while sending faxes
In-Reply-To: <ED055084A034476C877440F041F8D4E1@e1705>
References: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>
	<ED055084A034476C877440F041F8D4E1@e1705>
Message-ID: <BANLkTimUOY1W2=nT97Pv93Nv-rYvO8f8tA@mail.gmail.com>

Yes, I'm sorry but I didn't see my post after 2 hours so I thought that I
didn't post it well.

2011/5/26 Madovsky <infos at madovsky.org>

>  Please don't send several same email
>
> ----- Original Message -----
> *From:* Mathieu Lautram <lautram.mathieu at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, May 26, 2011 1:54 AM
> *Subject:* [Freeswitch-users] Errors while sending faxes
>
> Hi all,
>
> After sending faxes (it works very well know, thanks to Anthony Minessale)
> , sometimes I've got errors like that:
>
> - Disconnected after permitted retries
> - No response after sending a page
> - Received a DCN from remote after sending a page
> - Received bad response to DCS ot TCF
> - The call dropped prematurely
> - Timed out waiting for initial communication
>
> Those faxes are about 25% of all faxes that I send. I'm using the last git
> version.
> Where this problem comes from? Is there a way to fix this?
> I know that sometimes the remote end could be broken or could be an old fax
> machine, but it seems strange that I have so many errors like that.
>
> Thank you for your help :-)
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From david.ponzone at ipeva.fr  Thu May 26 12:17:47 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 26 May 2011 10:17:47 +0200
Subject: [Freeswitch-users] Errors while sending faxes
In-Reply-To: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>
References: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>
Message-ID: <121674B8-550C-416D-ACF1-7FD868944C77@ipeva.fr>

Mathieu,

you should perhaps start by describing your configuration, the endpoints involved, where FS is involved (T38-relay or T38-gateway or pure TDM), etc...

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

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

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




Le 26/05/2011 ? 07:54, Mathieu Lautram a ?crit :

> Hi all,
> 
> After sending faxes (it works very well know, thanks to Anthony Minessale) , sometimes I've got errors like that:
> 
> - Disconnected after permitted retries
> - No response after sending a page
> - Received a DCN from remote after sending a page
> - Received bad response to DCS ot TCF
> - The call dropped prematurely
> - Timed out waiting for initial communication
> 
> Those faxes are about 25% of all faxes that I send. I'm using the last git version.
> Where this problem comes from? Is there a way to fix this?
> I know that sometimes the remote end could be broken or could be an old fax machine, but it seems strange that I have so many errors like that.
> 
> Thank you for your help :-)
> 
> -- 
> Mathieu LAUTRAM
> Application developer
> 
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From u2nsam at gmail.com  Thu May 26 12:41:21 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 26 May 2011 14:11:21 +0530
Subject: [Freeswitch-users] matching word
Message-ID: <BANLkTinfe5rvkz_qQDDPT617pjcLsTswtw@mail.gmail.com>

Friends,

How can i match words in expressions,

is it by expression="^(\y\w+\y)$"  or could be something else.

Regards
Sam
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From lzwierko at gmail.com  Thu May 26 13:26:04 2011
From: lzwierko at gmail.com (=?ISO-8859-2?Q?=A3ukasz_Zwierko?=)
Date: Thu, 26 May 2011 11:26:04 +0200
Subject: [Freeswitch-users] Codec selection for incoming call
Message-ID: <BANLkTinFPNQb_jm88C_hFa5a653uXVwHNg@mail.gmail.com>

Hi guys,

Is is possible to manually select which codec will be used on the A-leg
(incoming call) when using outbound event socket to control the call?
My case is quite basic I guess: I want to answer the call and pass it to an
application which will record raw data (so I'm not bridging the call
anywhere).
It is important for me to be able to make a per-call selection of codec.
I've been digging through the wiki the whole morning, but I've only found
out how to force a codec on the outgoing call...

thanks for any advice,

Lukasz
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From tang.du at hotmail.com  Thu May 26 13:32:13 2011
From: tang.du at hotmail.com (tangdu)
Date: Thu, 26 May 2011 02:32:13 -0700 (PDT)
Subject: [Freeswitch-users] Error load mod_gsmopen
Message-ID: <1306402333534-6406292.post@n2.nabble.com>

Hi all,
I have followed this wiki page to intall gsmopen module 
http://wiki.freeswitch.org/wiki/GSMopen#Linux
Installed following prerequisites packages
# yum -y install alsa-lib-devel alsa-utils
I downloaded gsmlib-1.10-12ubuntu1 and patched, compiled it no error. 
#cd
/#/usr/local/src/freeswitch-1.0.7/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1
#make clean #make install 
Nokia-3208c connected with USB. Use minicom,I can dialout from GSM.
When I load mod_gsmopen, following error message. What's wrong?

freeswitch at localhost.localdomain> load mod_gsmopen
2011-05-26 15:27:45.465196 [INFO] mod_enum.c:808 ENUM Reloaded
2011-05-26 15:27:45.465196 [INFO] switch_time.c:954 Timezone reloaded 530
definitions
2011-05-26 15:27:45.503006 [WARNING] mod_gsmopen.cpp:1860 rev [(nil)|37    
][WARNINGA  1860 ][interface1][-1, 0, 0] STARTING interface_id=1
2011-05-26 15:27:51.507876 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-26 15:27:51.507876 [ERR] mod_gsmopen.cpp:1907 rev [(nil)|37    
][ERRORA  1907 ][interface1][-1, 0, 0] gsmopen_serial_config failed, let's
try again
2011-05-26 15:27:55.629412 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-26 15:27:55.629412 [ERR] mod_gsmopen.cpp:1913 rev [(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 1: gsmopen_serial_config failed,
let's try again
2011-05-26 15:27:57.854725 [ERR] gsmopen_protocol.cpp:2285 rev [(nil)|37    
][ERRORA  2285 ][interface1][-1, 0, 0] Error RE-sending (carriage return):
10 -1 (Invalid or incomplete multibyte or wide character)
2011-05-26 15:27:57.854725 [ERR] gsmopen_protocol.cpp:2364 rev [(nil)|37    
][ERRORA  2364 ][interface1][-1, 0, 0] Error sending data... (Invalid or
incomplete multibyte or wide character) 
2011-05-26 15:27:57.854725 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-26 15:27:57.854725 [ERR] mod_gsmopen.cpp:1913 rev [(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 2: gsmopen_serial_config failed,
let's try again
2011-05-26 15:28:00.621621 [ERR] gsmopen_protocol.cpp:2259 rev [(nil)|37    
][ERRORA  2259 ][interface1][-1, 0, 0] Error RE-sending (A): 10 -1 (Invalid
or incomplete multibyte or wide character)
2011-05-26 15:28:00.621621 [ERR] gsmopen_protocol.cpp:2364 rev [(nil)|37    
][ERRORA  2364 ][interface1][-1, 0, 0] Error sending data... (Invalid or
incomplete multibyte or wide character) 
2011-05-26 15:28:00.621621 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37    
][ERRORA  696  ][interface1][-1, 0, 0] no response to AT
2011-05-26 15:28:00.621621 [ERR] mod_gsmopen.cpp:1913 rev [(nil)|37    
][ERRORA  1913 ][interface1][-1, 0, 0] 3: gsmopen_serial_config failed,
let's try again
........................
freeswitch at localhost.localdomain> 2011-05-26 15:28:05.063925 [ERR]
mod_gsmopen.cpp:1917 rev [(nil)|37     ][ERRORA  1917 ][interface1][-1, 0,
0] STARTING interface_id=1 FAILED
2011-05-26 15:28:05.063925 [ERR] mod_gsmopen.cpp:3146 rev [(nil)|37    
][ERRORA  3146 ][interface1][-1, 0, 0] ALARM on interface interface1: 
2011-05-26 15:28:05.063925 [CONSOLE] switch_loadable_module.c:946
Successfully Loaded [mod_gsmopen]
2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:145 Adding
Endpoint 'gsmopen'
2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsm'
2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen'
2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen_boost_audio'
2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen_dump'
2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'gsmopen_sendsms'
2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:379 Adding Chat
interface 'SMS'

Thank you
Tang du

--
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From benkokakao at gmail.com  Thu May 26 13:36:21 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 26 May 2011 11:36:21 +0200
Subject: [Freeswitch-users] xml_cdr: end_epoch is 0,
 billsec are negative - how does this happen?
In-Reply-To: <BANLkTincLEUuBrsL2xQKC_SA5rjOxNZwGg@mail.gmail.com>
References: <BANLkTikiEL3H=wkXELgZOBBS5bM60hf5CA@mail.gmail.com>
	<BANLkTi=xkB-9EXSSFGG-V3V3k7Qte7hXBQ@mail.gmail.com>
	<BANLkTincLEUuBrsL2xQKC_SA5rjOxNZwGg@mail.gmail.com>
Message-ID: <BANLkTimQrMcL=kLjtdZ1TbbonAMkj6BtDQ@mail.gmail.com>

On 25 May 2011 19:06, Christian Benke <benkokakao at gmail.com> wrote:
>> What version of FreeSWITCH are you running?
>>
>> There was briefly a version with bad timestamps a few weeks/months ago.
>
> FreeSWITCH Version 1.0.head (git-5510618 2011-04-29 08-57-00 -0500)

The update fixed the issue, cheers!


From william at xofap.com  Thu May 26 14:06:43 2011
From: william at xofap.com (William Alianto)
Date: Thu, 26 May 2011 17:06:43 +0700
Subject: [Freeswitch-users] Bridging Gateway from another server
Message-ID: <4DDE2633.2030002@xofap.com>

I'm trying to do an outbound scenario as following :

Client --> FS1 --> FS2 --> SBC

It maybe more simple if I add the SBC as gateway to FS1, but it only 
accept IP from FS2 due to IP restriction from provider. Is there any 
possible solution for this scenario?


From Nabble at slickdeals.endjunk.com  Thu May 26 14:26:51 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 26 May 2011 03:26:51 -0700 (PDT)
Subject: [Freeswitch-users] matching word
In-Reply-To: <BANLkTinfe5rvkz_qQDDPT617pjcLsTswtw@mail.gmail.com>
References: <BANLkTinfe5rvkz_qQDDPT617pjcLsTswtw@mail.gmail.com>
Message-ID: <1306405611055-6406420.post@n2.nabble.com>


samir wrote:
> 
> Friends,
> 
> How can i match words in expressions,
> 
> is it by expression="^(\y\w+\y)$"  or could be something else.
> 
> Regards
> Sam
I understand ^(\w)$ will match any word, but not with \y. Perhaps, if you
tell us exactly what you want to accomplish, someone will chime in for a
better solution.

-----
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/matching-word-tp6406170p6406420.html
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From david.ponzone at ipeva.fr  Thu May 26 14:59:59 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 26 May 2011 12:59:59 +0200
Subject: [Freeswitch-users] Question about P-Asserted-Identity
Message-ID: <83CC0F6E-8DD4-4285-A4CA-85C2ABAE8905@ipeva.fr>

All,

I am using a GIT version of beginning of May.
I receive a P-Asserted-Identity from a provider, but I can't find the corresponding variable in the XML CDR.
Is there a known bug that was fixed ?

While I send this email, I am upgrading, so I'll let you know if that was it.

Thanks

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

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

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




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From vetali100 at gmail.com  Thu May 26 16:10:32 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Thu, 26 May 2011 15:10:32 +0300
Subject: [Freeswitch-users] Codec selection for incoming call
In-Reply-To: <BANLkTinFPNQb_jm88C_hFa5a653uXVwHNg@mail.gmail.com>
References: <BANLkTinFPNQb_jm88C_hFa5a653uXVwHNg@mail.gmail.com>
Message-ID: <BANLkTimGUh7r3TC=m-bmsU1WxEHNmenLOA@mail.gmail.com>

Probably late negotiation is what you need:

http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requires_param.29

Vitalie

2011/5/26 ?ukasz Zwierko <lzwierko at gmail.com>

> Hi guys,
>
> Is is possible to manually select which codec will be used on the A-leg
> (incoming call) when using outbound event socket to control the call?
> My case is quite basic I guess: I want to answer the call and pass it to an
> application which will record raw data (so I'm not bridging the call
> anywhere).
> It is important for me to be able to make a per-call selection of codec.
> I've been digging through the wiki the whole morning, but I've only found
> out how to force a codec on the outgoing call...
>
> thanks for any advice,
>
> Lukasz
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anton.vazir at gmail.com  Thu May 26 18:44:04 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 26 May 2011 19:44:04 +0500
Subject: [Freeswitch-users] ESL event filter work scheme
Message-ID: <BANLkTimwWSXNzQqYJu7PfhkmaZeOitYAxQ@mail.gmail.com>

Trying to find out if the given is intended behavior or not:
I am noticed that I do not receive the BACKGROUND_JOB event if filter
for the given event is added after filter on Unique-ID filter.

e.g. if I do

filter Unique-ID XXXXX
filter Unique-ID YYYY
filter Event-Name BACKGROUND_JOB

I do not receive the BACKGROUND_JOB event at all, only events
consisting filtered UUIDs

but if I do:

filter Event-Name BACKGROUND_JOB
filter Unique-ID XXXXX
filter Unique-ID YYYY

I receive it, and other events with my UUID too.

Seems a little odd, since described that it's a filter in
Is the given intended behavior or a bug?


From frisch.alan at gmail.com  Thu May 26 19:23:13 2011
From: frisch.alan at gmail.com (Alan Frisch)
Date: Thu, 26 May 2011 11:23:13 -0400
Subject: [Freeswitch-users] Freeswitch DBH - Possible to create
 connection pool to other DBs?
In-Reply-To: <BANLkTinOV_ZW8Ow2p4JUbCT8i8ck1DY3tg@mail.gmail.com>
References: <BANLkTimFS0HJNOD2p6+T+ziRUrkPvDKCUw@mail.gmail.com>
	<BANLkTinOV_ZW8Ow2p4JUbCT8i8ck1DY3tg@mail.gmail.com>
Message-ID: <BANLkTi=rwJhjR=-p58shS3Ef2eabgpgAPg@mail.gmail.com>

Steve,

Yeah I think the key might be the dbh:release() function.  Haven't
tested it yet, but rewrote my script so that it releases the DB
handles once the call is in progress.  Since the CDR/billing is
handledby XML CDR no need to keep a handle on the connection.

Will have to test it later and see how many connections get spawned.

AF.

On Thu, May 26, 2011 at 2:25 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Not sure what you mean here...
>
> When a script runs it'll reuse a connection from the pool if there is one.
> If there isn't it'll open a new one.
>
> If you only every have one copy of your script running at a time you'll only
> get a single connection. But if you get 2 copies running at the same time
> you'll get 2, 3 copies 3 etc.
>
>
> -Steve
>
>
> On 26 May 2011 02:04, Alan Frisch <frisch.alan at gmail.com> wrote:
>>
>> Been playing with the DBH functionality in LUA. ?The connection
>> pooling works great if you need to access the core DSN... but if you
>> need to access an external DB, it seems that FS will open a single
>> connection for the maximum number of concurrent calls accessing the
>> LUA script.
>>
>> In * one can limit the connection pool to any DB in res_odbc.conf...
>> but is there a way of doing this in FS?
>>
>> After a learning curve and lots of time, tthis is the one hangup in FS
>> that is preventing me from trashing * altogether. ?Help me put * in
>> the dustbin for good!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu May 26 19:27:17 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 26 May 2011 10:27:17 -0500
Subject: [Freeswitch-users] Duplicate Pastebin Post
Message-ID: <9916A1DD-56D4-4A53-AD97-D985319B80BF@5ninesolutions.com>

Hello,
I accidentally made a duplicate pastebin post.. :-/ Could someone with admin access please remove http://pastebin.freeswitch.org/16379


Thanks,
Spencer




From adam.kelloway at newpace.ca  Thu May 26 19:36:08 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Thu, 26 May 2011 12:36:08 -0300
Subject: [Freeswitch-users] event inconsistency
Message-ID: <4DDE7368.8060904@newpace.ca>

Hi there,

Has anyone ever had problems with events not being sent out over socket 
connections reliably? I am noticing that sometimes event notifications 
are sent, and sometimes they are not. For example, I have subscribed to 
CHANNEL_HANGUP, but depending on the timing of when I hangup the call 
(or when the dial plan hangs up), the event is not received. I confirmed 
with tcpdump that it is not even sent over the socket.

Has anyone else seen his? What might be the problem?

Thanks


From peter.olsson at visionutveckling.se  Thu May 26 19:43:32 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 26 May 2011 17:43:32 +0200
Subject: [Freeswitch-users] event inconsistency
In-Reply-To: <4DDE7368.8060904@newpace.ca>
References: <4DDE7368.8060904@newpace.ca>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6301@cooper>

Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.

Besides that I've never seen any problems.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Adam Kelloway [adam.kelloway at newpace.ca]
Skickat: den 26 maj 2011 17:36
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] event inconsistency

Hi there,

Has anyone ever had problems with events not being sent out over socket
connections reliably? I am noticing that sometimes event notifications
are sent, and sometimes they are not. For example, I have subscribed to
CHANNEL_HANGUP, but depending on the timing of when I hangup the call
(or when the dial plan hangs up), the event is not received. I confirmed
with tcpdump that it is not even sent over the socket.

Has anyone else seen his? What might be the problem?

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

!DSPAM:4dde73da32761960016355!



From yungwei at resolvity.com  Thu May 26 19:51:54 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Thu, 26 May 2011 11:51:54 -0400
Subject: [Freeswitch-users] using $${var} and ${var} in javascript
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC695@VMBX102.ihostexchange.net>

Hi,

I am wondering if there's a way to use $${base_dir}, ${strftime(%Y-%m-%d-%H-%M-%S), and ${destination_number} in a javascript program. 
For example, new_session.execute("record_session", "$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav");
Thanks.


From steveayre at gmail.com  Thu May 26 20:01:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 26 May 2011 17:01:14 +0100
Subject: [Freeswitch-users] using $${var} and ${var} in javascript
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC695@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC695@VMBX102.ihostexchange.net>
Message-ID: <BANLkTikBHB+9-2QnoxGS8sEFF+sZ0C57=g@mail.gmail.com>

Not sure about ${{var}}, but you can get ${var} channel variables with
session:getVariable

-Steve


On 26 May 2011 16:51, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I am wondering if there's a way to use $${base_dir},
> ${strftime(%Y-%m-%d-%H-%M-%S), and ${destination_number} in a javascript
> program.
> For example, new_session.execute("record_session",
> "$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav");
> 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 spencer at 5ninesolutions.com  Thu May 26 20:04:26 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 26 May 2011 11:04:26 -0500
Subject: [Freeswitch-users] Codec Negotiation Help
Message-ID: <42E549A5-8F9F-4C57-B358-755A02EDDDDD@5ninesolutions.com>

Hello all,
I have a problem regarding the codec negotiation on an outbound call.  My setup is like this:

Polycom IP 650 (1-n) -NAT-> FS  --> Our Signaling Proxy --> ITSP Proxy ---> ITSP Cisco GW

I'd like to use different codecs for different call paths (in order of pref), g729 in passthru only:
IP-650         ->  IP-650   G722, PCMU, G729
Inbound      ->  IP-650   PCMU
IP-650         ->  Outbound PCMU,G729

I have two sofia profiles, internal, public IPv4:5060 and external, public:IPv4:5080.

The phones use the internal profile and the external profile only communicates with our signaling proxy (no media proxy).
On the internal one:
CODECS IN        	G722,PCMU,G729,GSM
CODECS OUT       	G722,PCMU,G729,GSM
NOMEDIA          	false
LATE-NEG         	true

External:
CODECS IN        	PCMU,G729
CODECS OUT       	PCMU,G729
NOMEDIA          	false
LATE-NEG         	true

I have inbound-codec-negotiation set to greedy on both profiles and on outbound calls set absolute_codec_string=PCMU,G729 to prevent transcoding.  Note that mod_g729 is enabled for passthru only.  

The problem I have is this:
We use the dynamic routing module in OpenSIPS to select an outbound provider/GW, all support PCMU and G729.  On one of the routes, the Cisco IOS GW on this route has G729, PCMU configured as its codec pref.  

I have included a ladder diagram to better illustrate the problem but in a nutshell, the polycom negotiates PCMU with FS, FS asks for both PCMU and G729, the cisco GW sends G729 and FS sends a 488 because it can't transcode.  I would like to keep G729 in the outbound prefs because some routes might not support PCMU.  Should I set one of the profiles to generous, and if so which one?

When someone makes an outbound call the following happens (ladder diagram):
http://pastebin.freeswitch.org/16380


Sorry for the novella, :-)

Thanks!
Spencer



From djbinter at gmail.com  Thu May 26 20:47:28 2011
From: djbinter at gmail.com (DJB International)
Date: Thu, 26 May 2011 09:47:28 -0700
Subject: [Freeswitch-users] Codec Negotiation Help
In-Reply-To: <42E549A5-8F9F-4C57-B358-755A02EDDDDD@5ninesolutions.com>
References: <42E549A5-8F9F-4C57-B358-755A02EDDDDD@5ninesolutions.com>
Message-ID: <BANLkTi=wjx5JuFVBbzHcw-aSE7SW4AdrKA@mail.gmail.com>

Your profile has late negotion enabled.  I believe you can set
inherit_codec=true, so that it will force A leg to use the same codec as B
leg offered.


On Thu, May 26, 2011 at 9:04 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> I have a problem regarding the codec negotiation on an outbound call.  My
> setup is like this:
>
> Polycom IP 650 (1-n) -NAT-> FS  --> Our Signaling Proxy --> ITSP Proxy --->
> ITSP Cisco GW
>
> I'd like to use different codecs for different call paths (in order of
> pref), g729 in passthru only:
> IP-650         ->  IP-650   G722, PCMU, G729
> Inbound      ->  IP-650   PCMU
> IP-650         ->  Outbound PCMU,G729
>
> I have two sofia profiles, internal, public IPv4:5060 and external,
> public:IPv4:5080.
>
> The phones use the internal profile and the external profile only
> communicates with our signaling proxy (no media proxy).
> On the internal one:
> CODECS IN               G722,PCMU,G729,GSM
> CODECS OUT              G722,PCMU,G729,GSM
> NOMEDIA                 false
> LATE-NEG                true
>
> External:
> CODECS IN               PCMU,G729
> CODECS OUT              PCMU,G729
> NOMEDIA                 false
> LATE-NEG                true
>
> I have inbound-codec-negotiation set to greedy on both profiles and on
> outbound calls set absolute_codec_string=PCMU,G729 to prevent transcoding.
>  Note that mod_g729 is enabled for passthru only.
>
> The problem I have is this:
> We use the dynamic routing module in OpenSIPS to select an outbound
> provider/GW, all support PCMU and G729.  On one of the routes, the Cisco IOS
> GW on this route has G729, PCMU configured as its codec pref.
>
> I have included a ladder diagram to better illustrate the problem but in a
> nutshell, the polycom negotiates PCMU with FS, FS asks for both PCMU and
> G729, the cisco GW sends G729 and FS sends a 488 because it can't transcode.
>  I would like to keep G729 in the outbound prefs because some routes might
> not support PCMU.  Should I set one of the profiles to generous, and if so
> which one?
>
> When someone makes an outbound call the following happens (ladder diagram):
> http://pastebin.freeswitch.org/16380
>
>
> Sorry for the novella, :-)
>
> 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 u2nsam at gmail.com  Thu May 26 21:02:19 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 26 May 2011 22:32:19 +0530
Subject: [Freeswitch-users] matching word
In-Reply-To: <1306405611055-6406420.post@n2.nabble.com>
References: <BANLkTinfe5rvkz_qQDDPT617pjcLsTswtw@mail.gmail.com>
	<1306405611055-6406420.post@n2.nabble.com>
Message-ID: <BANLkTimBXuivtsVJSrX0Ck_STsQvZD2ozA@mail.gmail.com>

Thanks for that,

I want to use that when someone with username sam at gmail.com will be routed
to a voicemail dialplan, with domain partitioning.

here still i am not able to achiveve it with domain.

<domain name="gmail.com">

    <extension name="public_extensions">
      <condition field="destination_number" expression="^(\w)$">
          <action application="info"/>
        <action application="voicemail" data="default $${domain_name} $1"/>
      </condition>
    </extension>

</domain>

here above gmail.com is examplary .

when there is an invite coming from different server to FS with
sam at gmail.com and it is routed directly on voicemail context to route via
public.xml

here first the call is not recognized to the context gmail.com but when i
remove that it goes to voicemail directly properly and stores the voicemail.

but the voicemail is not stored under domain name gmail.com which should
store it ideally under domain name as folder, which it is just storing it
under user sam folder.

This is what I want to achieve.

Regards
Sam




On Thu, May 26, 2011 at 3:56 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> samir wrote:
> >
> > Friends,
> >
> > How can i match words in expressions,
> >
> > is it by expression="^(\y\w+\y)$"  or could be something else.
> >
> > Regards
> > Sam
> I understand ^(\w)$ will match any word, but not with \y. Perhaps, if you
> tell us exactly what you want to accomplish, someone will chime in for a
> better solution.
>
> -----
> 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/matching-word-tp6406170p6406420.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 spencer at 5ninesolutions.com  Thu May 26 21:56:14 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 26 May 2011 12:56:14 -0500
Subject: [Freeswitch-users] Codec Negotiation Help
In-Reply-To: <BANLkTi=wjx5JuFVBbzHcw-aSE7SW4AdrKA@mail.gmail.com>
References: <42E549A5-8F9F-4C57-B358-755A02EDDDDD@5ninesolutions.com>
	<BANLkTi=wjx5JuFVBbzHcw-aSE7SW4AdrKA@mail.gmail.com>
Message-ID: <BE3CBF8A-CA92-482D-A994-C9FC0FBA1FBB@5ninesolutions.com>

Thanks for you help, I'll try that.  So I'm a little confused with the negotiation process with two profiles.  Lets say for  example that I have a device that only supports GSM on the internal profile and a call comes in on the external profile as ULAW, G729.  If I set inherit_codec=true would Freeswitch then transcode?  How does the disable-transcoding option work in regards to two profiles?  I.e. Only one profile has transcoding disabled but a call traverses both of them.. (2 legs, one bridge).  Which profile would the transcoding need to be enabled (or not disabled rather)?

Thanks,
Spencer


On May 26, 2011, at 11:47 AM, DJB International wrote:

> Your profile has late negotion enabled.  I believe you can set inherit_codec=true, so that it will force A leg to use the same codec as B leg offered.
> 
> 
> On Thu, May 26, 2011 at 9:04 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello all,
> I have a problem regarding the codec negotiation on an outbound call.  My setup is like this:
> 
> Polycom IP 650 (1-n) -NAT-> FS  --> Our Signaling Proxy --> ITSP Proxy ---> ITSP Cisco GW
> 
> I'd like to use different codecs for different call paths (in order of pref), g729 in passthru only:
> IP-650         ->  IP-650   G722, PCMU, G729
> Inbound      ->  IP-650   PCMU
> IP-650         ->  Outbound PCMU,G729
> 
> I have two sofia profiles, internal, public IPv4:5060 and external, public:IPv4:5080.
> 
> The phones use the internal profile and the external profile only communicates with our signaling proxy (no media proxy).
> On the internal one:
> CODECS IN               G722,PCMU,G729,GSM
> CODECS OUT              G722,PCMU,G729,GSM
> NOMEDIA                 false
> LATE-NEG                true
> 
> External:
> CODECS IN               PCMU,G729
> CODECS OUT              PCMU,G729
> NOMEDIA                 false
> LATE-NEG                true
> 
> I have inbound-codec-negotiation set to greedy on both profiles and on outbound calls set absolute_codec_string=PCMU,G729 to prevent transcoding.  Note that mod_g729 is enabled for passthru only.
> 
> The problem I have is this:
> We use the dynamic routing module in OpenSIPS to select an outbound provider/GW, all support PCMU and G729.  On one of the routes, the Cisco IOS GW on this route has G729, PCMU configured as its codec pref.
> 
> I have included a ladder diagram to better illustrate the problem but in a nutshell, the polycom negotiates PCMU with FS, FS asks for both PCMU and G729, the cisco GW sends G729 and FS sends a 488 because it can't transcode.  I would like to keep G729 in the outbound prefs because some routes might not support PCMU.  Should I set one of the profiles to generous, and if so which one?
> 
> When someone makes an outbound call the following happens (ladder diagram):
> http://pastebin.freeswitch.org/16380
> 
> 
> Sorry for the novella, :-)
> 
> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 26 22:01:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 26 May 2011 13:01:06 -0500
Subject: [Freeswitch-users] using $${var} and ${var} in javascript
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC695@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC695@VMBX102.ihostexchange.net>
Message-ID: <BANLkTimABai73K1vbbxRXtxh5LpdfCjFgA@mail.gmail.com>

no $$ are available at runtime its special for xml pre-processor
but globals can be accessed at runtime with ${var}

using execute_extension as your example should expand vars.


On Thu, May 26, 2011 at 10:51 AM, Yungwei Chen <yungwei at resolvity.com> wrote:
> Hi,
>
> I am wondering if there's a way to use $${base_dir}, ${strftime(%Y-%m-%d-%H-%M-%S), and ${destination_number} in a javascript program.
> For example, new_session.execute("record_session", "$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav");
> Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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 mitch.capper at gmail.com  Thu May 26 22:04:05 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 26 May 2011 11:04:05 -0700
Subject: [Freeswitch-users] Duplicate Pastebin Post
In-Reply-To: <9916A1DD-56D4-4A53-AD97-D985319B80BF@5ninesolutions.com>
References: <9916A1DD-56D4-4A53-AD97-D985319B80BF@5ninesolutions.com>
Message-ID: <BANLkTin+6HsaF-UEJoVUQbhu9tRe9A6cOg@mail.gmail.com>

Don't worry about it pb is just scratchspace so it doesn't matter.

~Mitch

On Thu, May 26, 2011 at 8:27 AM, Spencer Thomason
<spencer at 5ninesolutions.com> wrote:
> Hello,
> I accidentally made a duplicate pastebin post.. :-/ Could someone with admin access please remove http://pastebin.freeswitch.org/16379
>
>
> 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 david.ponzone at ipeva.fr  Thu May 26 22:24:49 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 26 May 2011 20:24:49 +0200
Subject: [Freeswitch-users] Question about P-Asserted-Identity
In-Reply-To: <83CC0F6E-8DD4-4285-A4CA-85C2ABAE8905@ipeva.fr>
References: <83CC0F6E-8DD4-4285-A4CA-85C2ABAE8905@ipeva.fr>
Message-ID: <6FE1AF40-7902-41B9-A5F5-DA4F3BDC5EAD@ipeva.fr>

I upgraded to latest GIT and it's the same.

The P-Asserted-Identity I receive from the provider is of the form:

<sip:12356789>
without any @domain part.

Could it be the reason it is not imported in the call variables ?

Thank you

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

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

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




Le 26/05/2011 ? 12:59, David Ponzone a ?crit :

> All,
> 
> I am using a GIT version of beginning of May.
> I receive a P-Asserted-Identity from a provider, but I can't find the corresponding variable in the XML CDR.
> Is there a known bug that was fixed ?
> 
> While I send this email, I am upgrading, so I'll let you know if that was it.
> 
> Thanks
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 

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From sherwood.mcgowan at gmail.com  Thu May 26 05:21:57 2011
From: sherwood.mcgowan at gmail.com (Sherwood McGowan)
Date: Wed, 25 May 2011 20:21:57 -0500
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <CA030803.18B5F%krice@freeswitch.org>
References: <CA030803.18B5F%krice@freeswitch.org>
Message-ID: <23FCA551-BEE6-4588-B447-1B47B8BC9DD3@gmail.com>

AccuData is one I've used with great success in the past. 

No matter who you use, however, keep in mind that you can keep your costs down by keeping a local cache in a database. Just check the local cache to see if the number matches a record, an if it's not older than your configured TTL, use the record :-)

Sent from my iPhone

On May 25, 2011, at 7:24 PM, Ken Rice <krice at freeswitch.org> wrote:

> You could go with someone like http://tnsi.com/ but its not cheap
> 
> 
> On 5/25/11 6:33 PM, "Yitzchok" <adminjew at gmail.com> wrote:
> 
> Is there any CNAM provider that have access to all or most (C/I)LEC's?
> 
> 
> Yitzchok
> 
> 
> On Wed, May 25, 2011 at 7:03 PM, Ken Rice <krice at freeswitch.org> wrote:
> You have to keep in mind that even with SS7 interconnects with companies like callwithus they may or may not have access to the full national cnam infrastructure
> 
> CNAM is hosted by the LEC that owns the DIDs and you have to have an interconnect agreement with them to get to their CNAM...  Simply having an SS7 connection is not enuff
> 
> K
> 
> 
> 
> On 5/25/11 5:40 PM, "Yitzchok" <adminjew at gmail.com <http://adminjew at gmail.com> > wrote:
> 
> callwithus is nice but they are not that up to date on all listings.
> 
> 
> Yitzchok
> 
> 
> On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org <http://sos at sokhapkin.dyndns.org> > wrote:
> http://www.callwithus.com/API#cnam
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <http://FreeSWITCH-users at lists.freeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://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 yungwei at resolvity.com  Thu May 26 22:45:29 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Thu, 26 May 2011 14:45:29 -0400
Subject: [Freeswitch-users] Transferring to a cell phone
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>

Hi,

I installed freeswitch from a recent snapshot on CentOS 5.
I configured a gateway for my SIP provider in /usr/loca/freeswitch/conf/sip_profiles/external/my.xml, and I can make an outbound call to my cell phone using the following javascript snippet.
   var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");

Now I am trying to make a call from a land-line to a javascript application in freeswich, which plays a recording and then transfers the call to a cell phone.
But transferring the call to a cell phone doesn't work for some reason. 

Here's how I transfer the call to a cell phone in javascript:
   session.execute("transfer", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");

According to freeswitch.log, the problem seems to be that freeswitch cannot find a dialplan that matches the destiantion number, {ignore_early_media=true}sofia/gateway/sip_provider/1231231234.
What am I missing here? Thanks.



From msc at freeswitch.org  Thu May 26 22:49:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 May 2011 11:49:48 -0700
Subject: [Freeswitch-users] matching word
In-Reply-To: <BANLkTimBXuivtsVJSrX0Ck_STsQvZD2ozA@mail.gmail.com>
References: <BANLkTinfe5rvkz_qQDDPT617pjcLsTswtw@mail.gmail.com>
	<1306405611055-6406420.post@n2.nabble.com>
	<BANLkTimBXuivtsVJSrX0Ck_STsQvZD2ozA@mail.gmail.com>
Message-ID: <BANLkTinai5FLezOrHGSv2kAU7wbUKNVU3Q@mail.gmail.com>

Your regex is only going to match a single alphanumeric character. Are you
trying to match an actual email address or what? Once you know what you hope
to match it will be easier to know which regex to use. For the record a
semi-proper email grabbing regex is like this:

^([a-zA-Z0-9._%+-]+@[a-zA-Z0-9.-]+\.[a-zA-Z]{2,6})$

Matching emails is an interesting exercise. If you're not all into "making
it perfect" and just want to capture "foo at bar" then do something like this:

^([^@]+ at .*)$

-MC

On Thu, May 26, 2011 at 10:02 AM, Sam <u2nsam at gmail.com> wrote:

> Thanks for that,
>
> I want to use that when someone with username sam at gmail.com will be routed
> to a voicemail dialplan, with domain partitioning.
>
> here still i am not able to achiveve it with domain.
>
> <domain name="gmail.com">
>
>     <extension name="public_extensions">
>       <condition field="destination_number" expression="^(\w)$">
>           <action application="info"/>
>         <action application="voicemail" data="default $${domain_name} $1"/>
>       </condition>
>     </extension>
>
> </domain>
>
> here above gmail.com is examplary .
>
> when there is an invite coming from different server to FS with
> sam at gmail.com and it is routed directly on voicemail context to route via
> public.xml
>
> here first the call is not recognized to the context gmail.com but when i
> remove that it goes to voicemail directly properly and stores the voicemail.
>
> but the voicemail is not stored under domain name gmail.com which should
> store it ideally under domain name as folder, which it is just storing it
> under user sam folder.
>
> This is what I want to achieve.
>
> Regards
> Sam
>
>
>
>
>
> On Thu, May 26, 2011 at 3:56 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:
>
>>
>> samir wrote:
>> >
>> > Friends,
>> >
>> > How can i match words in expressions,
>> >
>> > is it by expression="^(\y\w+\y)$"  or could be something else.
>> >
>> > Regards
>> > Sam
>> I understand ^(\w)$ will match any word, but not with \y. Perhaps, if you
>> tell us exactly what you want to accomplish, someone will chime in for a
>> better solution.
>>
>> -----
>> 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/matching-word-tp6406170p6406420.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 adam.kelloway at newpace.ca  Thu May 26 22:51:56 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Thu, 26 May 2011 15:51:56 -0300
Subject: [Freeswitch-users] event inconsistency
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6301@cooper>
References: <4DDE7368.8060904@newpace.ca>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6301@cooper>
Message-ID: <4DDEA14C.4070000@newpace.ca>

I got the latest git yesterday from scratch just to make sure. Is there 
any way I can see debug output of the send operations?

I didn't notice this behaviour when I was running the 1.0.6 tar, it is 
only since I started using git that I saw it.

On 3:59 PM, Peter Olsson wrote:
> Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.
>
> Besides that I've never seen any problems.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Adam Kelloway [adam.kelloway at newpace.ca]
> Skickat: den 26 maj 2011 17:36
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] event inconsistency
>
> Hi there,
>
> Has anyone ever had problems with events not being sent out over socket
> connections reliably? I am noticing that sometimes event notifications
> are sent, and sometimes they are not. For example, I have subscribed to
> CHANNEL_HANGUP, but depending on the timing of when I hangup the call
> (or when the dial plan hangs up), the event is not received. I confirmed
> with tcpdump that it is not even sent over the socket.
>
> Has anyone else seen his? What might be the problem?
>
> 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
>
> !DSPAM:4dde73da32761960016355!
>
>
>


-- 
Adam Kelloway
Software Engineer
NewPace Technology Development Inc.
adam.kelloway at newpace.ca
+1 902-406-8375 x1031
www.newpace.ca



From tculjaga at gmail.com  Thu May 26 23:42:23 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 26 May 2011 21:42:23 +0200
Subject: [Freeswitch-users] 300 message without Diversion header
Message-ID: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>

hello,

i got a strange issue ... I'm using FS as a redirect server (route server)
and it does work fine ... except if the original INVITE contains a Diversion
header, the same header is lost in the responding 300 or 302 message.
Is this by design or its a bug ?


this is a part of the code in sofia where u construct redirect response
message...

                if (argc > 1) {
                    nua_respond(tech_pvt->nh, SIP_300_MULTIPLE_CHOICES,
SIPTAG_CONTACT_STR(dest),
                                TAG_IF(!zstr(extra_headers),
SIPTAG_HEADER_STR(extra_headers)), TAG_END());
                } else {
                    nua_respond(tech_pvt->nh, SIP_302_MOVED_TEMPORARILY,
SIPTAG_CONTACT_STR(dest),
                                TAG_IF(!zstr(extra_headers),
SIPTAG_HEADER_STR(extra_headers)), TAG_END());
                }


so, why the original diversion header is missing ? ... m'I missing something
?


also, should the 3000/302 message always contain a diversion field saying
the call is diverted ?


Thanks for your answer,
Tihomir.
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From anthony.minessale at gmail.com  Thu May 26 23:44:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 26 May 2011 14:44:32 -0500
Subject: [Freeswitch-users] event inconsistency
In-Reply-To: <4DDEA14C.4070000@newpace.ca>
References: <4DDE7368.8060904@newpace.ca>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6301@cooper>
	<4DDEA14C.4070000@newpace.ca>
Message-ID: <BANLkTinPK1zSv4NJSxEgN8oTfi2+OeJNrw@mail.gmail.com>

how are you consuming events? are you using ESL?

I recommend using fs_cli as a control to rule out faulty client logic.

start fs_cli  and issue the events command from there

/events channel_hangup
/log 0

then watch for the events there.

P.S. you did not mention your platform or any other pertinent details
about your setup.



On Thu, May 26, 2011 at 1:51 PM, Adam Kelloway <adam.kelloway at newpace.ca> wrote:
> I got the latest git yesterday from scratch just to make sure. Is there
> any way I can see debug output of the send operations?
>
> I didn't notice this behaviour when I was running the 1.0.6 tar, it is
> only since I started using git that I saw it.
>
> On 3:59 PM, Peter Olsson wrote:
>> Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.
>>
>> Besides that I've never seen any problems.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Adam Kelloway [adam.kelloway at newpace.ca]
>> Skickat: den 26 maj 2011 17:36
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] event inconsistency
>>
>> Hi there,
>>
>> Has anyone ever had problems with events not being sent out over socket
>> connections reliably? I am noticing that sometimes event notifications
>> are sent, and sometimes they are not. For example, I have subscribed to
>> CHANNEL_HANGUP, but depending on the timing of when I hangup the call
>> (or when the dial plan hangs up), the event is not received. I confirmed
>> with tcpdump that it is not even sent over the socket.
>>
>> Has anyone else seen his? What might be the problem?
>>
>> 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
>>
>> !DSPAM:4dde73da32761960016355!
>>
>>
>>
>
>
> --
> Adam Kelloway
> Software Engineer
> NewPace Technology Development Inc.
> adam.kelloway at newpace.ca
> +1 902-406-8375 x1031
> www.newpace.ca
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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 krice at freeswitch.org  Thu May 26 23:53:19 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 26 May 2011 14:53:19 -0500
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <23FCA551-BEE6-4588-B447-1B47B8BC9DD3@gmail.com>
Message-ID: <CA0419DF.18BB6%krice@freeswitch.org>

Keep in mind you need to look at your CNAM agreement... MOST CNAM
interconnect agreements SPECIFICALLY BAR caching and by doing so you may be
opening yourself to litigation from a 1000lb gorilla in the market place


On 5/25/11 8:21 PM, "Sherwood McGowan" <sherwood.mcgowan at gmail.com> wrote:

> AccuData is one I've used with great success in the past.
> 
> No matter who you use, however, keep in mind that you can keep your costs down
> by keeping a local cache in a database. Just check the local cache to see if
> the number matches a record, an if it's not older than your configured TTL,
> use the record :-)
> 
> Sent from my iPhone
> 
> On May 25, 2011, at 7:24 PM, Ken Rice <krice at freeswitch.org> wrote:
> 
>> You could go with someone like http://tnsi.com/ <http://tnsi.com/>  but its
>> not cheap
>> 
>> 
>> On 5/25/11 6:33 PM, "Yitzchok" <adminjew at gmail.com
>> <mailto:adminjew at gmail.com> > wrote:
>> 
>>> Is there any CNAM provider that have access to all or most (C/I)LEC's?
>>> 
>>> 
>>> Yitzchok
>>> 
>>> 
>>> On Wed, May 25, 2011 at 7:03 PM, Ken Rice <krice at freeswitch.org
>>> <mailto:krice at freeswitch.org> > wrote:
>>>> You have to keep in mind that even with SS7 interconnects with companies
>>>> like callwithus they may or may not have access to the full national cnam
>>>> infrastructure
>>>> 
>>>> CNAM is hosted by the LEC that owns the DIDs and you have to have an
>>>> interconnect agreement with them to get to their CNAM...  Simply having an
>>>> SS7 connection is not enuff
>>>> 
>>>> K
>>>> 
>>>> 
>>>> 
>>>> On 5/25/11 5:40 PM, "Yitzchok" <adminjew at gmail.com
>>>> <mailto:adminjew at gmail.com>  <http://adminjew at gmail.com
>>>> <http://adminjew at gmail.com> > > wrote:
>>>> 
>>>>> callwithus is nice but they are not that up to date on all listings.
>>>>> 
>>>>> 
>>>>> Yitzchok
>>>>> 
>>>>> 
>>>>> On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
>>>>> <mailto:sos at sokhapkin.dyndns.org>  <http://sos at sokhapkin.dyndns.org
>>>>> <http://sos at sokhapkin.dyndns.org> > > wrote:
>>>>>> http://www.callwithus.com/API#cnam <http://www.callwithus.com/API#cnam>
>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org
>>>>> <http://FreeSWITCH-users at lists.freeswitch.org> >
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> <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
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org <http://www.freeswitch.org>
>>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://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 brian at freeswitch.org  Thu May 26 23:54:52 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 26 May 2011 14:54:52 -0500
Subject: [Freeswitch-users] 300 message without Diversion header
In-Reply-To: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>
References: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>
Message-ID: <BC87F03A-02BC-4B24-89AE-202A080F0538@freeswitch.org>

Check your variables. via uuid_dump/info

/b

On May 26, 2011, at 2:42 PM, Tihomir Culjaga wrote:

> hello,
> 
> i got a strange issue ... I'm using FS as a redirect server (route server)
> and it does work fine ... except if the original INVITE contains a Diversion
> header, the same header is lost in the responding 300 or 302 message.
> Is this by design or its a bug ?
> 
> 
> this is a part of the code in sofia where u construct redirect response
> message...
> 
>                if (argc > 1) {
>                    nua_respond(tech_pvt->nh, SIP_300_MULTIPLE_CHOICES,
> SIPTAG_CONTACT_STR(dest),
>                                TAG_IF(!zstr(extra_headers),
> SIPTAG_HEADER_STR(extra_headers)), TAG_END());
>                } else {
>                    nua_respond(tech_pvt->nh, SIP_302_MOVED_TEMPORARILY,
> SIPTAG_CONTACT_STR(dest),
>                                TAG_IF(!zstr(extra_headers),
> SIPTAG_HEADER_STR(extra_headers)), TAG_END());
>                }
> 
> 
> so, why the original diversion header is missing ? ... m'I missing something
> ?
> 
> 
> also, should the 3000/302 message always contain a diversion field saying
> the call is diverted ?
> 
> 
> Thanks for your answer,
> Tihomir.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri May 27 00:08:13 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Thu, 26 May 2011 16:08:13 -0400
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <CA0419DF.18BB6%krice@freeswitch.org>
References: <23FCA551-BEE6-4588-B447-1B47B8BC9DD3@gmail.com>
	<CA0419DF.18BB6%krice@freeswitch.org>
Message-ID: <114601cc1be0$a54664e0$efd32ea0$@com>

I got to be honest - I can't see that caching is really going to be any type
of savior. Unless you have customers that are constantly calling the same
people (residential?)

 

Now for a fun topic - how about E911.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Thursday, May 26, 2011 3:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] CNAM Providers

 

Keep in mind you need to look at your CNAM agreement... MOST CNAM
interconnect agreements SPECIFICALLY BAR caching and by doing so you may be
opening yourself to litigation from a 1000lb gorilla in the market place


On 5/25/11 8:21 PM, "Sherwood McGowan" <sherwood.mcgowan at gmail.com> wrote:

AccuData is one I've used with great success in the past. 

No matter who you use, however, keep in mind that you can keep your costs
down by keeping a local cache in a database. Just check the local cache to
see if the number matches a record, an if it's not older than your
configured TTL, use the record :-)

Sent from my iPhone

On May 25, 2011, at 7:24 PM, Ken Rice <krice at freeswitch.org> wrote:

You could go with someone like http://tnsi.com/ <http://tnsi.com/>  but its
not cheap


On 5/25/11 6:33 PM, "Yitzchok" <adminjew at gmail.com
<mailto:adminjew at gmail.com> > wrote:

Is there any CNAM provider that have access to all or most (C/I)LEC's?


Yitzchok


On Wed, May 25, 2011 at 7:03 PM, Ken Rice <krice at freeswitch.org
<mailto:krice at freeswitch.org> > wrote:

You have to keep in mind that even with SS7 interconnects with companies
like callwithus they may or may not have access to the full national cnam
infrastructure

CNAM is hosted by the LEC that owns the DIDs and you have to have an
interconnect agreement with them to get to their CNAM...  Simply having an
SS7 connection is not enuff

K



On 5/25/11 5:40 PM, "Yitzchok" <adminjew at gmail.com
<mailto:adminjew at gmail.com>  <http://adminjew at gmail.com
<http://adminjew at gmail.com> > > wrote:

callwithus is nice but they are not that up to date on all listings.


Yitzchok


On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
<mailto:sos at sokhapkin.dyndns.org>  <http://sos at sokhapkin.dyndns.org
<http://sos at sokhapkin.dyndns.org> > > wrote:

http://www.callwithus.com/API#cnam <http://www.callwithus.com/API#cnam> 

 

  _____  

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

 

  _____  

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

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

 

  _____  

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.ponzone at ipeva.fr  Fri May 27 00:14:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 26 May 2011 22:14:28 +0200
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
Message-ID: <0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>

You have to use "bridge" instead of "transfer".
"Transfer" is to transfer the call to another part of the dialplan, with the same or another dialed number.

The Wiki is your friend :)

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

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

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




Le 26/05/2011 ? 20:45, Yungwei Chen a ?crit :

> Hi,
> 
> I installed freeswitch from a recent snapshot on CentOS 5.
> I configured a gateway for my SIP provider in /usr/loca/freeswitch/conf/sip_profiles/external/my.xml, and I can make an outbound call to my cell phone using the following javascript snippet.
>   var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
> 
> Now I am trying to make a call from a land-line to a javascript application in freeswich, which plays a recording and then transfers the call to a cell phone.
> But transferring the call to a cell phone doesn't work for some reason. 
> 
> Here's how I transfer the call to a cell phone in javascript:
>   session.execute("transfer", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
> 
> According to freeswitch.log, the problem seems to be that freeswitch cannot find a dialplan that matches the destiantion number, {ignore_early_media=true}sofia/gateway/sip_provider/1231231234.
> What am I missing here? Thanks.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Fri May 27 01:05:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 May 2011 14:05:23 -0700
Subject: [Freeswitch-users] Codec Negotiation Help
In-Reply-To: <BE3CBF8A-CA92-482D-A994-C9FC0FBA1FBB@5ninesolutions.com>
References: <42E549A5-8F9F-4C57-B358-755A02EDDDDD@5ninesolutions.com>
	<BANLkTi=wjx5JuFVBbzHcw-aSE7SW4AdrKA@mail.gmail.com>
	<BE3CBF8A-CA92-482D-A994-C9FC0FBA1FBB@5ninesolutions.com>
Message-ID: <BANLkTin-Z6gjD_QqxVWPJ229NoX4JU_YZQ@mail.gmail.com>

I believe in the scenario you describe that the call would fail since the
incoming call cannot inherit the only codec supported by the b leg device
(GSM).

-MC

On Thu, May 26, 2011 at 10:56 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Thanks for you help, I'll try that.  So I'm a little confused with the
> negotiation process with two profiles.  Lets say for  example that I have a
> device that only supports GSM on the internal profile and a call comes in on
> the external profile as ULAW, G729.  If I set inherit_codec=true would
> Freeswitch then transcode?  How does the disable-transcoding option work in
> regards to two profiles?  I.e. Only one profile has transcoding disabled but
> a call traverses both of them.. (2 legs, one bridge).  Which profile would
> the transcoding need to be enabled (or not disabled rather)?
>
> Thanks,
> Spencer
>
>
> On May 26, 2011, at 11:47 AM, DJB International wrote:
>
> Your profile has late negotion enabled.  I believe you can set
> inherit_codec=true, so that it will force A leg to use the same codec as B
> leg offered.
>
>
> On Thu, May 26, 2011 at 9:04 AM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> Hello all,
>> I have a problem regarding the codec negotiation on an outbound call.  My
>> setup is like this:
>>
>> Polycom IP 650 (1-n) -NAT-> FS  --> Our Signaling Proxy --> ITSP Proxy
>> ---> ITSP Cisco GW
>>
>> I'd like to use different codecs for different call paths (in order of
>> pref), g729 in passthru only:
>> IP-650         ->  IP-650   G722, PCMU, G729
>> Inbound      ->  IP-650   PCMU
>> IP-650         ->  Outbound PCMU,G729
>>
>> I have two sofia profiles, internal, public IPv4:5060 and external,
>> public:IPv4:5080.
>>
>> The phones use the internal profile and the external profile only
>> communicates with our signaling proxy (no media proxy).
>> On the internal one:
>> CODECS IN               G722,PCMU,G729,GSM
>> CODECS OUT              G722,PCMU,G729,GSM
>> NOMEDIA                 false
>> LATE-NEG                true
>>
>> External:
>> CODECS IN               PCMU,G729
>> CODECS OUT              PCMU,G729
>> NOMEDIA                 false
>> LATE-NEG                true
>>
>> I have inbound-codec-negotiation set to greedy on both profiles and on
>> outbound calls set absolute_codec_string=PCMU,G729 to prevent transcoding.
>>  Note that mod_g729 is enabled for passthru only.
>>
>> The problem I have is this:
>> We use the dynamic routing module in OpenSIPS to select an outbound
>> provider/GW, all support PCMU and G729.  On one of the routes, the Cisco IOS
>> GW on this route has G729, PCMU configured as its codec pref.
>>
>> I have included a ladder diagram to better illustrate the problem but in a
>> nutshell, the polycom negotiates PCMU with FS, FS asks for both PCMU and
>> G729, the cisco GW sends G729 and FS sends a 488 because it can't transcode.
>>  I would like to keep G729 in the outbound prefs because some routes might
>> not support PCMU.  Should I set one of the profiles to generous, and if so
>> which one?
>>
>> When someone makes an outbound call the following happens (ladder
>> diagram):
>> http://pastebin.freeswitch.org/16380
>>
>>
>> Sorry for the novella, :-)
>>
>> 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
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri May 27 01:20:30 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 26 May 2011 16:20:30 -0500
Subject: [Freeswitch-users] Codec Negotiation Help
In-Reply-To: <BANLkTin-Z6gjD_QqxVWPJ229NoX4JU_YZQ@mail.gmail.com>
References: <42E549A5-8F9F-4C57-B358-755A02EDDDDD@5ninesolutions.com>
	<BANLkTi=wjx5JuFVBbzHcw-aSE7SW4AdrKA@mail.gmail.com>
	<BE3CBF8A-CA92-482D-A994-C9FC0FBA1FBB@5ninesolutions.com>
	<BANLkTin-Z6gjD_QqxVWPJ229NoX4JU_YZQ@mail.gmail.com>
Message-ID: <90BEEBF8-041B-4AC1-B8BE-33A5391EFB33@5ninesolutions.com>

That's what I was thinking.  I don't actually have any devices like this but a few of our phones do support iLBC and we have be toying with the idea of using that on a few links that are slower.  Is there any option to setting inherit_codec=true that would "fall back" to transcoding if need be?

Spencer


On May 26, 2011, at 4:05 PM, Michael Collins wrote:

> I believe in the scenario you describe that the call would fail since the incoming call cannot inherit the only codec supported by the b leg device (GSM).
> 
> -MC
> 
> On Thu, May 26, 2011 at 10:56 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Thanks for you help, I'll try that.  So I'm a little confused with the negotiation process with two profiles.  Lets say for  example that I have a device that only supports GSM on the internal profile and a call comes in on the external profile as ULAW, G729.  If I set  would Freeswitch then transcode?  How does the disable-transcoding option work in regards to two profiles?  I.e. Only one profile has transcoding disabled but a call traverses both of them.. (2 legs, one bridge).  Which profile would the transcoding need to be enabled (or not disabled rather)?
> 
> Thanks,
> Spencer
> 
> 
> On May 26, 2011, at 11:47 AM, DJB International wrote:
> 
>> Your profile has late negotion enabled.  I believe you can set inherit_codec=true, so that it will force A leg to use the same codec as B leg offered.
>> 
>> 
>> On Thu, May 26, 2011 at 9:04 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> Hello all,
>> I have a problem regarding the codec negotiation on an outbound call.  My setup is like this:
>> 
>> Polycom IP 650 (1-n) -NAT-> FS  --> Our Signaling Proxy --> ITSP Proxy ---> ITSP Cisco GW
>> 
>> I'd like to use different codecs for different call paths (in order of pref), g729 in passthru only:
>> IP-650         ->  IP-650   G722, PCMU, G729
>> Inbound      ->  IP-650   PCMU
>> IP-650         ->  Outbound PCMU,G729
>> 
>> I have two sofia profiles, internal, public IPv4:5060 and external, public:IPv4:5080.
>> 
>> The phones use the internal profile and the external profile only communicates with our signaling proxy (no media proxy).
>> On the internal one:
>> CODECS IN               G722,PCMU,G729,GSM
>> CODECS OUT              G722,PCMU,G729,GSM
>> NOMEDIA                 false
>> LATE-NEG                true
>> 
>> External:
>> CODECS IN               PCMU,G729
>> CODECS OUT              PCMU,G729
>> NOMEDIA                 false
>> LATE-NEG                true
>> 
>> I have inbound-codec-negotiation set to greedy on both profiles and on outbound calls set absolute_codec_string=PCMU,G729 to prevent transcoding.  Note that mod_g729 is enabled for passthru only.
>> 
>> The problem I have is this:
>> We use the dynamic routing module in OpenSIPS to select an outbound provider/GW, all support PCMU and G729.  On one of the routes, the Cisco IOS GW on this route has G729, PCMU configured as its codec pref.
>> 
>> I have included a ladder diagram to better illustrate the problem but in a nutshell, the polycom negotiates PCMU with FS, FS asks for both PCMU and G729, the cisco GW sends G729 and FS sends a 488 because it can't transcode.  I would like to keep G729 in the outbound prefs because some routes might not support PCMU.  Should I set one of the profiles to generous, and if so which one?
>> 
>> When someone makes an outbound call the following happens (ladder diagram):
>> http://pastebin.freeswitch.org/16380
>> 
>> 
>> Sorry for the novella, :-)
>> 
>> 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
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gmaruzz at gmail.com  Fri May 27 01:24:34 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 26 May 2011 23:24:34 +0200
Subject: [Freeswitch-users] load mod_gsmopen error
In-Reply-To: <1306294731594-6401282.post@n2.nabble.com>
References: <1306294731594-6401282.post@n2.nabble.com>
Message-ID: <BANLkTi=xtUmPZOceK=4bT_-NmRV93FR=JA@mail.gmail.com>

On Wed, May 25, 2011 at 5:38 AM, tangdu <tang.du at hotmail.com> wrote:
> I useing FS 1.0.7 ?Connected to nokia 3208C cellphone with usb?I can
> communicate with cellphone by minicom?When I load mod_gsmopen , following
> errors message.
> Have you compiled mod_gsmopen correctly? Perhaps the ?AT Command of 3208c
> dose not match ?I need to modify gsmopen.conf.xml.

Yep, compilation is ok.
But it seems that the AT commands of 3208 are giving problems...



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From yudha2008 at gmail.com  Fri May 27 01:34:48 2011
From: yudha2008 at gmail.com (Baskar)
Date: Thu, 26 May 2011 17:34:48 -0400
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
Message-ID: <BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>

Hi David,

I tried the dial plan suggested by you several times but the moment i key in
the extension it take my input but exits the ivr_inbound.xml and goes to
default.xml. It does not bridge me to the extension i key in but instead
comes out of the xml and connects me to the extension in the default.xml.
Below is the output log of the tests that i have made.


Output Log:

2011-05-26 17:24:59.065884 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 5:840
2011-05-26 17:24:59.341669 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:720
2011-05-26 17:24:59.745697 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960
2011-05-26 17:25:00.041720 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 7:1120
2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:376 digits '5107'
2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:470 action regex [5107]
[/^([1-6][0-9][0-9][0-9])$/] [2]
2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:488 IVR action on menu
'ivr_inbound' matched '5107' param
'execute_extensionset:continue_on_fail=true,bridge
sofia/internal/5107%XX.XX.XX.XX,bridge sofia/internal/5998%XX.XX.XX.XX'
2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:492
switch_ivr_menu_execute todo=[2]
2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:580 exit-sound
'voicemail/vm-goodbye.wav'

--
Thanks with Regards,
N.Baskar

On Thu, May 26, 2011 at 1:49 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Baskar,
>
> If you want this behavior only for calls only coming through the IVR, I
> think you will have to use bridge rather than transfer.
> it would look like this:
>
>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
> param="execute_extension
> set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>
>
> or something like it :)
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 26/05/2011 ? 00:27, Baskar a ?crit :
>
> Hi Michael,
>
> Thanks for your quick reply. Below is the code for the dial plan inbound
> ivr. Can you please specify where i should be inserting the
> continue_on_fail=true line in the code.
>
> Default.xml
>
>     <extension name="inbound">
>       <condition field="destination_number" expression="^(XXXXXXXXXX)$" />
>       <condition wday="2-6" hour="8-17">
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="accountcode=Inbound"/>
>         <action application="answer"/>
>         <action application="start_dtmf" />
>         <action application="ivr" data="ivr_inbound"/>
>         <action application="record_session"
> data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>         <action application="transfer" data="1018 XML default"/>
>       </condition>
>     </extension>
>
> ivr_inbound.xml
>
> <menu name="ivr_inbound"
>
>  greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>      exit-sound="voicemail/vm-goodbye.wav"
>      confirm-macro=""
>      confirm-key=""
>      tts-engine="flite"
>      tts-voice="rms"
>      confirm-attempts="3"
>      timeout="10000"
>      inter-digit-timeout="2000"
>      max-failures="3"
>      max-timeouts="1"
>      digit-len="4">
>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
> param="transfer $1 XML default"/>
>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
> default"/>
> </menu>
>
> *--
> Thanks with Regards,
>
> N.Baskar
>
> *
> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> If I understand correctly you're wanting to handle a bridge scenario where
>> the target extension is busy, etc. Most likely you just need to set
>> continue_on_fail=true so that your dialplan continues in the case of the
>> target extension not picking up.
>>
>> -MC
>>
>> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
>>
>>> Hi All,,
>>>
>>> In my inbound dial plan xml file  i have set two conditions enabled
>>>
>>> Condition1:
>>>
>>> In inbound dial plan callers are given an option to key in the extension
>>> number and reach the appropriate extension (Example: 1001 or 1002 or 1003
>>> etc).
>>>
>>> Condition2:
>>>
>>> The second condition routes call to a default extension in scenarios
>>> where the caller does not specify any extension number (Example: Default
>>> extension is 1007).
>>>
>>>
>>> Both the above conditions are working fine.
>>>
>>> Now I need to set up another condition where after keying in the
>>> extension number the call gets transferred to the appropriate extension and
>>> if the extension is busy(Example: say extension 1003 is busy) it should be
>>> hunted to a default extension (example: 1007). How can we set up this
>>> condition in dial plan?
>>>
>>> Can any one guide me.
>>> --
>>> Thanks with Regards,
>>>
>>> N.Baskar
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-
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From msc at freeswitch.org  Fri May 27 01:43:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 May 2011 14:43:24 -0700
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
Message-ID: <BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>

That's because you are telling it to go to the default context:
 <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
param="transfer $1 XML default"/>
Change that "default" to whatever context you want the destination number to
be in...

-MC

On Thu, May 26, 2011 at 2:34 PM, Baskar <yudha2008 at gmail.com> wrote:

> Hi David,
>
> I tried the dial plan suggested by you several times but the moment i key
> in the extension it take my input but exits the ivr_inbound.xml and goes to
> default.xml. It does not bridge me to the extension i key in but instead
> comes out of the xml and connects me to the extension in the default.xml.
> Below is the output log of the tests that i have made.
>
>
> Output Log:
>
> 2011-05-26 17:24:59.065884 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 5:840
> 2011-05-26 17:24:59.341669 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:720
> 2011-05-26 17:24:59.745697 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960
> 2011-05-26 17:25:00.041720 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 7:1120
> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:376 digits '5107'
> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:470 action regex
> [5107] [/^([1-6][0-9][0-9][0-9])$/] [2]
> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:488 IVR action on menu
> 'ivr_inbound' matched '5107' param
> 'execute_extensionset:continue_on_fail=true,bridge
> sofia/internal/5107%XX.XX.XX.XX,bridge sofia/internal/5998%XX.XX.XX.XX'
> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:492
> switch_ivr_menu_execute todo=[2]
> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:580 exit-sound
> 'voicemail/vm-goodbye.wav'
>
> --
>
> Thanks with Regards,
> N.Baskar
>
> On Thu, May 26, 2011 at 1:49 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Baskar,
>>
>> If you want this behavior only for calls only coming through the IVR, I
>> think you will have to use bridge rather than transfer.
>> it would look like this:
>>
>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>> param="execute_extension
>> set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>
>>
>> or something like it :)
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 26/05/2011 ? 00:27, Baskar a ?crit :
>>
>> Hi Michael,
>>
>> Thanks for your quick reply. Below is the code for the dial plan inbound
>> ivr. Can you please specify where i should be inserting the
>> continue_on_fail=true line in the code.
>>
>> Default.xml
>>
>>     <extension name="inbound">
>>       <condition field="destination_number" expression="^(XXXXXXXXXX)$" />
>>        <condition wday="2-6" hour="8-17">
>>         <action application="set" data="hangup_after_bridge=true"/>
>>         <action application="set" data="accountcode=Inbound"/>
>>         <action application="answer"/>
>>         <action application="start_dtmf" />
>>         <action application="ivr" data="ivr_inbound"/>
>>         <action application="record_session"
>> data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>>         <action application="transfer" data="1018 XML default"/>
>>       </condition>
>>     </extension>
>>
>> ivr_inbound.xml
>>
>> <menu name="ivr_inbound"
>>
>>  greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>      exit-sound="voicemail/vm-goodbye.wav"
>>      confirm-macro=""
>>      confirm-key=""
>>      tts-engine="flite"
>>      tts-voice="rms"
>>      confirm-attempts="3"
>>      timeout="10000"
>>      inter-digit-timeout="2000"
>>      max-failures="3"
>>      max-timeouts="1"
>>      digit-len="4">
>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>> param="transfer $1 XML default"/>
>>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
>> default"/>
>> </menu>
>>
>> *--
>> Thanks with Regards,
>>
>> N.Baskar
>>
>> *
>> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> If I understand correctly you're wanting to handle a bridge scenario
>>> where the target extension is busy, etc. Most likely you just need to set
>>> continue_on_fail=true so that your dialplan continues in the case of the
>>> target extension not picking up.
>>>
>>> -MC
>>>
>>> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>
>>>> Hi All,,
>>>>
>>>> In my inbound dial plan xml file  i have set two conditions enabled
>>>>
>>>> Condition1:
>>>>
>>>> In inbound dial plan callers are given an option to key in the extension
>>>> number and reach the appropriate extension (Example: 1001 or 1002 or 1003
>>>> etc).
>>>>
>>>> Condition2:
>>>>
>>>> The second condition routes call to a default extension in scenarios
>>>> where the caller does not specify any extension number (Example: Default
>>>> extension is 1007).
>>>>
>>>>
>>>> Both the above conditions are working fine.
>>>>
>>>> Now I need to set up another condition where after keying in the
>>>> extension number the call gets transferred to the appropriate extension and
>>>> if the extension is busy(Example: say extension 1003 is busy) it should be
>>>> hunted to a default extension (example: 1007). How can we set up this
>>>> condition in dial plan?
>>>>
>>>> Can any one guide me.
>>>> --
>>>> Thanks with Regards,
>>>>
>>>> N.Baskar
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> -
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yudha2008 at gmail.com  Fri May 27 02:02:09 2011
From: yudha2008 at gmail.com (Baskar)
Date: Thu, 26 May 2011 18:02:09 -0400
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
Message-ID: <BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>

HI Mike,

I am looking at a condition where if the extension that i enter is busy then
it should automatically route to the extension i specify in the ivr_inbound
xml.

-- 
Thanks with Regards,
N.Baskar

On Thu, May 26, 2011 at 5:43 PM, Michael Collins <msc at freeswitch.org> wrote:

> That's because you are telling it to go to the default context:
>     <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
> param="transfer $1 XML default"/>
> Change that "default" to whatever context you want the destination number
> to be in...
>
> -MC
>
> On Thu, May 26, 2011 at 2:34 PM, Baskar <yudha2008 at gmail.com> wrote:
>
>> Hi David,
>>
>> I tried the dial plan suggested by you several times but the moment i key
>> in the extension it take my input but exits the ivr_inbound.xml and goes to
>> default.xml. It does not bridge me to the extension i key in but instead
>> comes out of the xml and connects me to the extension in the default.xml.
>> Below is the output log of the tests that i have made.
>>
>>
>> Output Log:
>>
>> 2011-05-26 17:24:59.065884 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 5:840
>> 2011-05-26 17:24:59.341669 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:720
>> 2011-05-26 17:24:59.745697 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960
>> 2011-05-26 17:25:00.041720 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 7:1120
>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:376 digits '5107'
>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:470 action regex
>> [5107] [/^([1-6][0-9][0-9][0-9])$/] [2]
>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:488 IVR action on
>> menu 'ivr_inbound' matched '5107' param
>> 'execute_extensionset:continue_on_fail=true,bridge
>> sofia/internal/5107%XX.XX.XX.XX,bridge sofia/internal/5998%XX.XX.XX.XX'
>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:492
>> switch_ivr_menu_execute todo=[2]
>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:580 exit-sound
>> 'voicemail/vm-goodbye.wav'
>>
>> --
>>
>> Thanks with Regards,
>> N.Baskar
>>
>> On Thu, May 26, 2011 at 1:49 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> Baskar,
>>>
>>> If you want this behavior only for calls only coming through the IVR, I
>>> think you will have to use bridge rather than transfer.
>>> it would look like this:
>>>
>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>> param="execute_extension
>>> set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>
>>>
>>> or something like it :)
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 26/05/2011 ? 00:27, Baskar a ?crit :
>>>
>>> Hi Michael,
>>>
>>> Thanks for your quick reply. Below is the code for the dial plan inbound
>>> ivr. Can you please specify where i should be inserting the
>>> continue_on_fail=true line in the code.
>>>
>>> Default.xml
>>>
>>>     <extension name="inbound">
>>>       <condition field="destination_number" expression="^(XXXXXXXXXX)$"
>>> />
>>>        <condition wday="2-6" hour="8-17">
>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>         <action application="set" data="accountcode=Inbound"/>
>>>         <action application="answer"/>
>>>         <action application="start_dtmf" />
>>>         <action application="ivr" data="ivr_inbound"/>
>>>         <action application="record_session"
>>> data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>>>         <action application="transfer" data="1018 XML default"/>
>>>       </condition>
>>>     </extension>
>>>
>>> ivr_inbound.xml
>>>
>>> <menu name="ivr_inbound"
>>>
>>>  greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>>>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>      exit-sound="voicemail/vm-goodbye.wav"
>>>      confirm-macro=""
>>>      confirm-key=""
>>>      tts-engine="flite"
>>>      tts-voice="rms"
>>>      confirm-attempts="3"
>>>      timeout="10000"
>>>      inter-digit-timeout="2000"
>>>      max-failures="3"
>>>      max-timeouts="1"
>>>      digit-len="4">
>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>> param="transfer $1 XML default"/>
>>>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
>>> default"/>
>>> </menu>
>>>
>>> *--
>>> Thanks with Regards,
>>>
>>> N.Baskar
>>>
>>> *
>>> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> If I understand correctly you're wanting to handle a bridge scenario
>>>> where the target extension is busy, etc. Most likely you just need to set
>>>> continue_on_fail=true so that your dialplan continues in the case of the
>>>> target extension not picking up.
>>>>
>>>> -MC
>>>>
>>>> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>>
>>>>> Hi All,,
>>>>>
>>>>> In my inbound dial plan xml file  i have set two conditions enabled
>>>>>
>>>>> Condition1:
>>>>>
>>>>> In inbound dial plan callers are given an option to key in the
>>>>> extension number and reach the appropriate extension (Example: 1001 or 1002
>>>>> or 1003 etc).
>>>>>
>>>>> Condition2:
>>>>>
>>>>> The second condition routes call to a default extension in scenarios
>>>>> where the caller does not specify any extension number (Example: Default
>>>>> extension is 1007).
>>>>>
>>>>>
>>>>> Both the above conditions are working fine.
>>>>>
>>>>> Now I need to set up another condition where after keying in the
>>>>> extension number the call gets transferred to the appropriate extension and
>>>>> if the extension is busy(Example: say extension 1003 is busy) it should be
>>>>> hunted to a default extension (example: 1007). How can we set up this
>>>>> condition in dial plan?
>>>>>
>>>>> Can any one guide me.
>>>>> --
>>>>> Thanks with Regards,
>>>>>
>>>>> N.Baskar
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> -
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mitch.capper at gmail.com  Fri May 27 02:05:23 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 26 May 2011 15:05:23 -0700
Subject: [Freeswitch-users] CNAM Providers
In-Reply-To: <114601cc1be0$a54664e0$efd32ea0$@com>
References: <23FCA551-BEE6-4588-B447-1B47B8BC9DD3@gmail.com>
	<CA0419DF.18BB6%krice@freeswitch.org>
	<114601cc1be0$a54664e0$efd32ea0$@com>
Message-ID: <BANLkTim-vA_n4ix0N3FYMMq1d7rSH-xJsg@mail.gmail.com>

Flowroute has E911 I believe.
~Mitch

On Thu, May 26, 2011 at 1:08 PM, Robert Huddleston
<rhuddleston at gmail.com> wrote:
> I got to be honest ? I can?t see that caching is really going to be any type
> of savior. Unless you have customers that are constantly calling the same
> people (residential?)
>
>
>
> Now for a fun topic ? how about E911.
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Thursday, May 26, 2011 3:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] CNAM Providers
>
>
>
> Keep in mind you need to look at your CNAM agreement... MOST CNAM
> interconnect agreements SPECIFICALLY BAR caching and by doing so you may be
> opening yourself to litigation from a 1000lb gorilla in the market place
>
>
> On 5/25/11 8:21 PM, "Sherwood McGowan" <sherwood.mcgowan at gmail.com> wrote:
>
> AccuData is one I've used with great success in the past.
>
> No matter who you use, however, keep in mind that you can keep your costs
> down by keeping a local cache in a database. Just check the local cache to
> see if the number matches a record, an if it's not older than your
> configured TTL, use the record :-)
>
> Sent from my iPhone
>
> On May 25, 2011, at 7:24 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> You could go with someone like http://tnsi.com/ <http://tnsi.com/> ?but its
> not cheap
>
>
> On 5/25/11 6:33 PM, "Yitzchok" <adminjew at gmail.com
> <mailto:adminjew at gmail.com> > wrote:
>
> Is there any CNAM provider that have access to all or most (C/I)LEC's?
>
>
> Yitzchok
>
>
> On Wed, May 25, 2011 at 7:03 PM, Ken Rice <krice at freeswitch.org
> <mailto:krice at freeswitch.org> > wrote:
>
> You have to keep in mind that even with SS7 interconnects with companies
> like callwithus they may or may not have access to the full national cnam
> infrastructure
>
> CNAM is hosted by the LEC that owns the DIDs and you have to have an
> interconnect agreement with them to get to their CNAM... ?Simply having an
> SS7 connection is not enuff
>
> K
>
>
>
> On 5/25/11 5:40 PM, "Yitzchok" <adminjew at gmail.com
> <mailto:adminjew at gmail.com> ?<http://adminjew at gmail.com
> <http://adminjew at gmail.com> > > wrote:
>
> callwithus is nice but they are not that up to date on all listings.
>
>
> Yitzchok
>
>
> On Wed, May 25, 2011 at 3:06 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> <mailto:sos at sokhapkin.dyndns.org> ?<http://sos at sokhapkin.dyndns.org
> <http://sos at sokhapkin.dyndns.org> > > wrote:
>
> http://www.callwithus.com/API#cnam <http://www.callwithus.com/API#cnam>
>
>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> ?<http://FreeSWITCH-users at lists.freeswitch.org
> <http://FreeSWITCH-users at lists.freeswitch.org> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> <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
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org>
>
>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lists at kempgen.net  Fri May 27 02:08:30 2011
From: lists at kempgen.net (Philipp Kempgen)
Date: Fri, 27 May 2011 00:08:30 +0200
Subject: [Freeswitch-users] Question about P-Asserted-Identity
In-Reply-To: <6FE1AF40-7902-41B9-A5F5-DA4F3BDC5EAD@ipeva.fr>
References: <83CC0F6E-8DD4-4285-A4CA-85C2ABAE8905@ipeva.fr>
	<6FE1AF40-7902-41B9-A5F5-DA4F3BDC5EAD@ipeva.fr>
Message-ID: <4DDECF5E.1040904@kempgen.net>

David Ponzone wrote:
>
> The P-Asserted-Identity I receive from the provider is of the form:
>
> <sip:12356789>
> without any @domain part.

Actually that's a SIP-URI with a hostname ("12356789") but
without a userinfo part. (see ABNF rules in RFC 3261)

It's a valid SIP-URI in theory but without userinfo so it doesn't
make too much sense as a caller ID in a PAI header.

Tell your provider to fix it.


    Philipp

-- 
http://twitter.com/kempgen


From msc at freeswitch.org  Fri May 27 02:33:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 May 2011 15:33:08 -0700
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
Message-ID: <BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>

Pastebin the debug output of a call to a busy phone.
-MC

On Thu, May 26, 2011 at 3:02 PM, Baskar <yudha2008 at gmail.com> wrote:

> HI Mike,
>
> I am looking at a condition where if the extension that i enter is busy
> then it should automatically route to the extension i specify in the
> ivr_inbound xml.
>
> --
>
> Thanks with Regards,
> N.Baskar
>
> On Thu, May 26, 2011 at 5:43 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> That's because you are telling it to go to the default context:
>>     <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>> param="transfer $1 XML default"/>
>> Change that "default" to whatever context you want the destination number
>> to be in...
>>
>> -MC
>>
>> On Thu, May 26, 2011 at 2:34 PM, Baskar <yudha2008 at gmail.com> wrote:
>>
>>> Hi David,
>>>
>>> I tried the dial plan suggested by you several times but the moment i key
>>> in the extension it take my input but exits the ivr_inbound.xml and goes to
>>> default.xml. It does not bridge me to the extension i key in but instead
>>> comes out of the xml and connects me to the extension in the default.xml.
>>> Below is the output log of the tests that i have made.
>>>
>>>
>>> Output Log:
>>>
>>> 2011-05-26 17:24:59.065884 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 5:840
>>> 2011-05-26 17:24:59.341669 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:720
>>> 2011-05-26 17:24:59.745697 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960
>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 7:1120
>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:376 digits '5107'
>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:470 action regex
>>> [5107] [/^([1-6][0-9][0-9][0-9])$/] [2]
>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:488 IVR action on
>>> menu 'ivr_inbound' matched '5107' param
>>> 'execute_extensionset:continue_on_fail=true,bridge
>>> sofia/internal/5107%XX.XX.XX.XX,bridge sofia/internal/5998%XX.XX.XX.XX'
>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:492
>>> switch_ivr_menu_execute todo=[2]
>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:580 exit-sound
>>> 'voicemail/vm-goodbye.wav'
>>>
>>> --
>>>
>>> Thanks with Regards,
>>> N.Baskar
>>>
>>> On Thu, May 26, 2011 at 1:49 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>
>>>> Baskar,
>>>>
>>>> If you want this behavior only for calls only coming through the IVR, I
>>>> think you will have to use bridge rather than transfer.
>>>> it would look like this:
>>>>
>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>> param="execute_extension
>>>> set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>
>>>>
>>>> or something like it :)
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 26/05/2011 ? 00:27, Baskar a ?crit :
>>>>
>>>> Hi Michael,
>>>>
>>>> Thanks for your quick reply. Below is the code for the dial plan inbound
>>>> ivr. Can you please specify where i should be inserting the
>>>> continue_on_fail=true line in the code.
>>>>
>>>> Default.xml
>>>>
>>>>     <extension name="inbound">
>>>>       <condition field="destination_number" expression="^(XXXXXXXXXX)$"
>>>> />
>>>>        <condition wday="2-6" hour="8-17">
>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>         <action application="set" data="accountcode=Inbound"/>
>>>>         <action application="answer"/>
>>>>         <action application="start_dtmf" />
>>>>         <action application="ivr" data="ivr_inbound"/>
>>>>         <action application="record_session"
>>>> data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>>>>         <action application="transfer" data="1018 XML default"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>> ivr_inbound.xml
>>>>
>>>> <menu name="ivr_inbound"
>>>>
>>>>  greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>>>>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>>      exit-sound="voicemail/vm-goodbye.wav"
>>>>      confirm-macro=""
>>>>      confirm-key=""
>>>>      tts-engine="flite"
>>>>      tts-voice="rms"
>>>>      confirm-attempts="3"
>>>>      timeout="10000"
>>>>      inter-digit-timeout="2000"
>>>>      max-failures="3"
>>>>      max-timeouts="1"
>>>>      digit-len="4">
>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>> param="transfer $1 XML default"/>
>>>>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
>>>> default"/>
>>>> </menu>
>>>>
>>>> *--
>>>> Thanks with Regards,
>>>>
>>>> N.Baskar
>>>>
>>>> *
>>>> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> If I understand correctly you're wanting to handle a bridge scenario
>>>>> where the target extension is busy, etc. Most likely you just need to set
>>>>> continue_on_fail=true so that your dialplan continues in the case of the
>>>>> target extension not picking up.
>>>>>
>>>>> -MC
>>>>>
>>>>> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>>>
>>>>>> Hi All,,
>>>>>>
>>>>>> In my inbound dial plan xml file  i have set two conditions enabled
>>>>>>
>>>>>> Condition1:
>>>>>>
>>>>>> In inbound dial plan callers are given an option to key in the
>>>>>> extension number and reach the appropriate extension (Example: 1001 or 1002
>>>>>> or 1003 etc).
>>>>>>
>>>>>> Condition2:
>>>>>>
>>>>>> The second condition routes call to a default extension in scenarios
>>>>>> where the caller does not specify any extension number (Example: Default
>>>>>> extension is 1007).
>>>>>>
>>>>>>
>>>>>> Both the above conditions are working fine.
>>>>>>
>>>>>> Now I need to set up another condition where after keying in the
>>>>>> extension number the call gets transferred to the appropriate extension and
>>>>>> if the extension is busy(Example: say extension 1003 is busy) it should be
>>>>>> hunted to a default extension (example: 1007). How can we set up this
>>>>>> condition in dial plan?
>>>>>>
>>>>>> Can any one guide me.
>>>>>> --
>>>>>> Thanks with Regards,
>>>>>>
>>>>>> N.Baskar
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>>  _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> -
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From lists at kempgen.net  Fri May 27 02:38:11 2011
From: lists at kempgen.net (Philipp Kempgen)
Date: Fri, 27 May 2011 00:38:11 +0200
Subject: [Freeswitch-users] 300 message without Diversion header
In-Reply-To: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>
References: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>
Message-ID: <4DDED653.20505@kempgen.net>

Tihomir Culjaga wrote:

> should the 3000/302 message always contain a diversion field saying
> the call is diverted ?

http://tools.ietf.org/html/rfc5806#section-5.3
says it "SHOULD".

The Diversion header could be encrypted (/removed?) for privacy
reasons.
http://tools.ietf.org/html/rfc5806#section-7


    Philipp

-- 
http://twitter.com/kempgen


From lautram.mathieu at gmail.com  Fri May 27 02:43:04 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Fri, 27 May 2011 00:43:04 +0200
Subject: [Freeswitch-users] Errors while sending faxes
In-Reply-To: <121674B8-550C-416D-ACF1-7FD868944C77@ipeva.fr>
References: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>
	<121674B8-550C-416D-ACF1-7FD868944C77@ipeva.fr>
Message-ID: <BANLkTikCcUN2SWwOavh7QKnaYpMvGmGBug@mail.gmail.com>

David,

I am running freeswitch on CentOS. Faxes are sent by SFR which provides a
T38-gateway. What I don't know is the endpoints involved. Indeed, I send
faxes to our clients so I don't know what type of fax machine they have.

I know I give you only few informations but what I'd like to know is despite
those responses, are the faxes correctly sent?

Thank you for your help :-)


2011/5/26 David Ponzone <david.ponzone at ipeva.fr>

> Mathieu,
>
> you should perhaps start by describing your configuration, the endpoints
> involved, where FS is involved (T38-relay or T38-gateway or pure TDM),
> etc...
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 26/05/2011 ? 07:54, Mathieu Lautram a ?crit :
>
> Hi all,
>
> After sending faxes (it works very well know, thanks to Anthony Minessale)
> , sometimes I've got errors like that:
>
> - Disconnected after permitted retries
> - No response after sending a page
> - Received a DCN from remote after sending a page
> - Received bad response to DCS ot TCF
> - The call dropped prematurely
> - Timed out waiting for initial communication
>
> Those faxes are about 25% of all faxes that I send. I'm using the last git
> version.
> Where this problem comes from? Is there a way to fix this?
> I know that sometimes the remote end could be broken or could be an old fax
> machine, but it seems strange that I have so many errors like that.
>
> Thank you for your help :-)
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From yudha2008 at gmail.com  Fri May 27 02:45:19 2011
From: yudha2008 at gmail.com (Baskar)
Date: Thu, 26 May 2011 18:45:19 -0400
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
	<BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
Message-ID: <BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>

HI Mike,

I have pasted my log in path http://pastebin.freeswitch.org/16388

<http://pastebin.freeswitch.org/16388>--
Thanks with Regards,
N.Baskar

On Thu, May 26, 2011 at 6:33 PM, Michael Collins <msc at freeswitch.org> wrote:

> Pastebin the debug output of a call to a busy phone.
> -MC
>
>
> On Thu, May 26, 2011 at 3:02 PM, Baskar <yudha2008 at gmail.com> wrote:
>
>> HI Mike,
>>
>> I am looking at a condition where if the extension that i enter is busy
>> then it should automatically route to the extension i specify in the
>> ivr_inbound xml.
>>
>> --
>>
>> Thanks with Regards,
>> N.Baskar
>>
>> On Thu, May 26, 2011 at 5:43 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> That's because you are telling it to go to the default context:
>>>     <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>> param="transfer $1 XML default"/>
>>> Change that "default" to whatever context you want the destination number
>>> to be in...
>>>
>>> -MC
>>>
>>> On Thu, May 26, 2011 at 2:34 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>
>>>> Hi David,
>>>>
>>>> I tried the dial plan suggested by you several times but the moment i
>>>> key in the extension it take my input but exits the ivr_inbound.xml and goes
>>>> to default.xml. It does not bridge me to the extension i key in but instead
>>>> comes out of the xml and connects me to the extension in the default.xml.
>>>> Below is the output log of the tests that i have made.
>>>>
>>>>
>>>> Output Log:
>>>>
>>>> 2011-05-26 17:24:59.065884 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 5:840
>>>> 2011-05-26 17:24:59.341669 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:720
>>>> 2011-05-26 17:24:59.745697 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960
>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>> 7:1120
>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:376 digits '5107'
>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:470 action regex
>>>> [5107] [/^([1-6][0-9][0-9][0-9])$/] [2]
>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:488 IVR action on
>>>> menu 'ivr_inbound' matched '5107' param
>>>> 'execute_extensionset:continue_on_fail=true,bridge
>>>> sofia/internal/5107%XX.XX.XX.XX,bridge sofia/internal/5998%XX.XX.XX.XX'
>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:492
>>>> switch_ivr_menu_execute todo=[2]
>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:580 exit-sound
>>>> 'voicemail/vm-goodbye.wav'
>>>>
>>>> --
>>>>
>>>> Thanks with Regards,
>>>> N.Baskar
>>>>
>>>> On Thu, May 26, 2011 at 1:49 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>>
>>>>> Baskar,
>>>>>
>>>>> If you want this behavior only for calls only coming through the IVR, I
>>>>> think you will have to use bridge rather than transfer.
>>>>> it would look like this:
>>>>>
>>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>>> param="execute_extension
>>>>> set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>
>>>>>
>>>>> or something like it :)
>>>>>
>>>>>  David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>>> l'exp?diteur.*
>>>>> *
>>>>> *
>>>>>
>>>>>
>>>>>
>>>>> Le 26/05/2011 ? 00:27, Baskar a ?crit :
>>>>>
>>>>> Hi Michael,
>>>>>
>>>>> Thanks for your quick reply. Below is the code for the dial plan
>>>>> inbound ivr. Can you please specify where i should be inserting the
>>>>> continue_on_fail=true line in the code.
>>>>>
>>>>> Default.xml
>>>>>
>>>>>     <extension name="inbound">
>>>>>       <condition field="destination_number" expression="^(XXXXXXXXXX)$"
>>>>> />
>>>>>        <condition wday="2-6" hour="8-17">
>>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>>         <action application="set" data="accountcode=Inbound"/>
>>>>>         <action application="answer"/>
>>>>>         <action application="start_dtmf" />
>>>>>         <action application="ivr" data="ivr_inbound"/>
>>>>>         <action application="record_session"
>>>>> data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>>>>>         <action application="transfer" data="1018 XML default"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>> ivr_inbound.xml
>>>>>
>>>>> <menu name="ivr_inbound"
>>>>>
>>>>>  greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>>>>>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>>>      exit-sound="voicemail/vm-goodbye.wav"
>>>>>      confirm-macro=""
>>>>>      confirm-key=""
>>>>>      tts-engine="flite"
>>>>>      tts-voice="rms"
>>>>>      confirm-attempts="3"
>>>>>      timeout="10000"
>>>>>      inter-digit-timeout="2000"
>>>>>      max-failures="3"
>>>>>      max-timeouts="1"
>>>>>      digit-len="4">
>>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>>> param="transfer $1 XML default"/>
>>>>>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
>>>>> default"/>
>>>>> </menu>
>>>>>
>>>>> *--
>>>>> Thanks with Regards,
>>>>>
>>>>> N.Baskar
>>>>>
>>>>> *
>>>>> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> If I understand correctly you're wanting to handle a bridge scenario
>>>>>> where the target extension is busy, etc. Most likely you just need to set
>>>>>> continue_on_fail=true so that your dialplan continues in the case of the
>>>>>> target extension not picking up.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>>>>
>>>>>>> Hi All,,
>>>>>>>
>>>>>>> In my inbound dial plan xml file  i have set two conditions enabled
>>>>>>>
>>>>>>> Condition1:
>>>>>>>
>>>>>>> In inbound dial plan callers are given an option to key in the
>>>>>>> extension number and reach the appropriate extension (Example: 1001 or 1002
>>>>>>> or 1003 etc).
>>>>>>>
>>>>>>> Condition2:
>>>>>>>
>>>>>>> The second condition routes call to a default extension in scenarios
>>>>>>> where the caller does not specify any extension number (Example: Default
>>>>>>> extension is 1007).
>>>>>>>
>>>>>>>
>>>>>>> Both the above conditions are working fine.
>>>>>>>
>>>>>>> Now I need to set up another condition where after keying in the
>>>>>>> extension number the call gets transferred to the appropriate extension and
>>>>>>> if the extension is busy(Example: say extension 1003 is busy) it should be
>>>>>>> hunted to a default extension (example: 1007). How can we set up this
>>>>>>> condition in dial plan?
>>>>>>>
>>>>>>> Can any one guide me.
>>>>>>> --
>>>>>>> Thanks with Regards,
>>>>>>>
>>>>>>> N.Baskar
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>>  _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> -
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://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 yungwei at resolvity.com  Fri May 27 03:41:24 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Thu, 26 May 2011 19:41:24 -0400
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>

Thanks for your reply. Using bridge fixed the problem. But I cannot hear anything both ways. Any idea?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Thursday, May 26, 2011 3:14 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

You have to use "bridge" instead of "transfer".
"Transfer" is to transfer the call to another part of the dialplan, with the same or another dialed number.

The Wiki is your friend :)

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 26/05/2011 ? 20:45, Yungwei Chen a ?crit :


Hi,

I installed freeswitch from a recent snapshot on CentOS 5.
I configured a gateway for my SIP provider in /usr/loca/freeswitch/conf/sip_profiles/external/my.xml, and I can make an outbound call to my cell phone using the following javascript snippet.
  var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");

Now I am trying to make a call from a land-line to a javascript application in freeswich, which plays a recording and then transfers the call to a cell phone.
But transferring the call to a cell phone doesn't work for some reason.

Here's how I transfer the call to a cell phone in javascript:
  session.execute("transfer", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");

According to freeswitch.log, the problem seems to be that freeswitch cannot find a dialplan that matches the destiantion number, {ignore_early_media=true}sofia/gateway/sip_provider/1231231234.
What am I missing here? Thanks.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From ayhkor at gmail.com  Fri May 27 03:50:24 2011
From: ayhkor at gmail.com (deniro)
Date: Thu, 26 May 2011 19:50:24 -0400
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTinuScKyL+sSJ7vnt0czPqcaerKmCw@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
	<BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
	<BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>
	<BANLkTin1V6TinJWsOVorkwAMTXc88G_mcg@mail.gmail.com>
	<BANLkTi=SPWEzBs0vAS=T_2wErdawdHo=FA@mail.gmail.com>
	<BANLkTimXTTJ5L2RhqCs6CQ9T_Wqk4DS2Rg@mail.gmail.com>
	<BANLkTinuScKyL+sSJ7vnt0czPqcaerKmCw@mail.gmail.com>
Message-ID: <BANLkTimiusAYub8dTw-UnZRCHxbOJR-o0Q@mail.gmail.com>

tried it and looks like  that XML line is  gone.
but now I am unable to join to conference by making suggested change.
played around it, no luck to join conference....
it doesnt get DTMF as I check from fs_cli
   *switch_cpp.cpp:1177 Got dtmf:  -- ..*
* *
I also checked usage and it  seems with 8 parameters.
I can't remember how I got it with 9 parameters before.
Anyway, It works(joins conference) with 9 parametes only.
I am using freeswitch 1.0.6 with ubuntu 10.04
can that usage be different from version to version?
thx


On Thu, May 26, 2011 at 2:34 AM, Steven Ayre <steveayre at gmail.com> wrote:

> *$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
> "conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav", "",
> '^[7]\d{4}\$');
>
> *
> It looks like this usage is wrong. You're giving 9 parameters but the Wiki
> seems to say it's only 8. It looks like it's using the 9th as a channel
> variable name to store the result in.
>
> Try this:
>
> *$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
> "conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav",
> '^[7]\d{4}\$');*
>
> -Steve
>
>
>
>
> On 26 May 2011 06:22, deniro <ayhkor at gmail.com> wrote:
>
>> Thanks a lot. The link was a good help.
>> we tested with sample code in the link and no more 502 errors with the
>> sample code in the link.
>>
>> We tried to make changes in our cdr.pl perl (using mod_xml_cdr) program
>> accordingly
>> we made  good progres in generatng  xml file, however we are still getting
>> 502 errors from cdr.pl.
>> Probably something else in our cdr.pl  giving us that, we are still
>> trying to figure out  what  piece is that.
>>
>> In the mean time,  in generated  xml  we see a line like this
>>    * <^[7]\d{4}\$>77555</^[7]\d{4}\$>
>> *
>> and looks like this coming from
>> *$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
>> "conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav", "",
>> '^[7]\d{4}\$');*
>> and because of above xml line I am unable to parse xml.
>> (above tested with changes and changes reflect on xml line)
>>
>> why do you think
>> above $session->playAndGetDigits   is producing xml  line like
>> "<^[7]\d{4}\$>77555</^[7]\d{4}\$>"
>>
>> any ideas appreciated.
>> why and how to fix it.
>>
>> thx again
>> deniro--
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, May 25, 2011 at 5:33 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> http://search.cpan.org/~markstos/CGI.pm-3.54/lib/CGI/Fast.pm
>>>
>>> That module looks like it is built on top of FCGI and gives you a
>>> CGI-like interface.
>>>
>>> -MC
>>>
>>>
>>> On Tue, May 24, 2011 at 10:12 PM, deniro <ayhkor at gmail.com> wrote:
>>>
>>>>
>>>> Thanks for the replies
>>>> I made some research and run series of tests.
>>>> I am using  perl+FCGI (fastcgi)  and my http server is  working fine as
>>>> tested with multiple perl scripts.
>>>>  I think the problem is handling the CGI within the cdr.plfile(obtained from freeswitch wiki)
>>>>
>>>>  so, how can I change below script to use with FCGI instead of
>>>> CGI::Simple
>>>>
>>>> use XML::Simple; # Get from CPAN
>>>> use CGI::Simple; # Get from CPAN
>>>> use Data::Dumper;
>>>> # dump into a place for further review
>>>> open(FILEOUT,">",'/tmp/fs-xml-cdr2.log');
>>>> print FILEOUT "Test successful..\n";
>>>> cloe(FILEOUT);
>>>> # dump into a place for further review
>>>> open(FILEOUT,">>",'/tmp/fs-xml-cdr.log');
>>>> # $cgi object has handy methods
>>>> my $cgi = new CGI::Simple;
>>>>
>>>>
>>>> Thx
>>>> deniro
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, May 23, 2011 at 3:50 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>>  > 2-- when dialing into conference, getting following from fs_cli
>>>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>>>
>>>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>>>>> means you have a bad gateway).
>>>>>>
>>>>>
>>>>> Check your web server logs - there may be some useful message. It
>>>>> sounds like the cgi-bin/perl handler hasn't been configured correctly, the
>>>>> script isn't executable, or it's not returning some headers. Try visiting
>>>>> the URL in your webbrowser (you
>>>>> won't be submitting a CDR but you can at least check the script runs
>>>>> without an (unexpected) error.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 23 May 2011 07:38, Gabriel Gunderson <gabe at gundy.org> wrote:
>>>>>
>>>>>> On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
>>>>>> > several questions;
>>>>>> > 1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
>>>>>> > trying to figure out if my problem is with cdr or with curl
>>>>>>
>>>>>> No, they are not functionally related... easy to get mixed up because
>>>>>> they both use HTTP and XML. One does not depend on the other. However,
>>>>>> you *can* configure mod_xml_cdr (or any other module) with
>>>>>> mod_xml_curl.
>>>>>>
>>>>>>
>>>>>>
>>>>>> > 2-- when dialing into conference, getting following from fs_cli
>>>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>>>
>>>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>>>> setup and not FS for the mix-up (particularly true as "HTTP Error 502"
>>>>>> means you have a bad gateway).
>>>>>>
>>>>>>
>>>>>>
>>>>>> > 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml
>>>>>> >  <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>
>>>>>>
>>>>>> Looks good to me, but check it with the wiki:
>>>>>> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration
>>>>>>
>>>>>>
>>>>>>
>>>>>> > 4-- if I need mod_xml_curl (question 1)
>>>>>> > what should this line be like in xml_curl.conf.xml?
>>>>>> > <param name="gateway-url" value="http://www.xxx.com:80<http://www.xxx.com/>
>>>>>> "
>>>>>> > bindings="dialplan"/>
>>>>>>
>>>>>> This one is a little tricker to get right.  Again, the wiki is might
>>>>>> be the best place to reference:
>>>>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring
>>>>>>
>>>>>>
>>>>>>
>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri May 27 04:35:59 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 26 May 2011 19:35:59 -0500
Subject: [Freeswitch-users] PRI Signaling Converters
Message-ID: <DEEBF8BA-93F3-4EC0-AFB1-A5FD6F4340BB@5ninesolutions.com>

Hello all,
Does anyone have any experience with any PRI -> SIP signaling converters?  Basically what I'm trying to do is migrate a Dialogic PBX that has a PRI card to a SIP trunk and eventually to a Freeswitch based PBX.  And since this will be a transitional piece of equipment cost does become a factor.  

Thanks,
Spencer



From krice at freeswitch.org  Fri May 27 04:45:12 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 26 May 2011 19:45:12 -0500
Subject: [Freeswitch-users] PRI Signaling Converters
In-Reply-To: <DEEBF8BA-93F3-4EC0-AFB1-A5FD6F4340BB@5ninesolutions.com>
Message-ID: <CA045E48.18BE8%krice@freeswitch.org>

You can drop a Sangoma PRI card straight into the FreeSWITCH box

That's basically all any PRI-> SIP converter really is anyway

Contact me offlist if you would like a quote on one

K


On 5/26/11 7:35 PM, "Spencer Thomason" <spencer at 5ninesolutions.com> wrote:

> Hello all,
> Does anyone have any experience with any PRI -> SIP signaling converters?
> Basically what I'm trying to do is migrate a Dialogic PBX that has a PRI card
> to a SIP trunk and eventually to a Freeswitch based PBX.  And since this will
> be a transitional piece of equipment cost does become a factor.
> 
> 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 cyril.zlachevsky at gmail.com  Fri May 27 03:03:13 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Fri, 27 May 2011 02:03:13 +0300
Subject: [Freeswitch-users] FreeSWITCH as SIP softswitch
Message-ID: <4DDEDC31.40601@gmail.com>

Hello,
I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server.
But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help.

FreeSWITCH server (IP 88.198.XXX.XXX) have to receive SIP calls from the one IP address (X-Lite 
softphone with dynamic IP from subnet 92.112.0.0/16) and forward this calls to the hardware SIP 
phone (Asotel, IP 195.225.XXX.XXX). SIP mode on hardware softphone (Asotel) is peer-to-peer.

Direct calls from X-Lite to Asotel complete with success - I always hear incoming ring when dial 
7777. But through FreeSWITCH calls always fail.

I created four configuration files - all other configs I left unchanged:
1) conf/sip-profiles/internal/X-Lite.xml:
<include>
   <gateway name="X-Lite">
     <param name="username" value="inboundtest"/>
     <param name="password" value="test"/>
     <param name="register" value="true"/>
   </gateway>
</include>

2) conf/sip-profiles/external/Asotel.xml:
<include>
   <gateway name="Asotel">
     <param name="realm" value="195.225.XXX.XXX"/>
     <param name="username" value="outboundtest"/>
     <param name="password" value="test"/>
     <param name="register" value="true"/>
   </gateway>
</include>

3) conf/dialplan/public/test.xml:
<include>
   <extension name="test">
       <condition field="destination_number" expression="^7777$">
         <action application="bridge" data="sofia/gateway/Asotel/$1"/>
       </condition>
     </extension>
</include>

4) conf/directory/default/inboundtest.xml:
<include>
   <user id="inboundtest" cidr="92.112.0.0/16">
     <params>
       <param name="from-domain" value="88.198.XXX.XXX"/>
       <param name="password" value="test"/>
     </params>
   </user>
</include>


This debug from Asotel ip-phone:
---begin---
Incoming CallLeg at callleg created 0x57d334 Incoming CallLeg at MsgReceived 0x57d334  *** $1 was 
being Invited ***
 >>> All call occupied. <<<
No slot availabe for this call...
FindIPCall...All Slot is Busy
RvSipCallLegReject(486), hCallLeg: 57d334
--> Message Sent (Message type: 1) (call-leg 57d334)
SIP/2.0 486 Busy Here
From: "inboundtest"<sip:outboundtest at 195.225.XXX.XXX;transport=udp>;tag=vgpp5vSBgcX6p
To: <sip:$1 at 195.225.XXX.XXX>;tag=c3e19fb6-13c4-4ddecb5a-1f577-5f2a
Call-ID: 232b19fe-0285-122f-b5b5-1b5bdf4f9807
CSeq: 12887277 INVITE
Via: SIP/2.0/UDP 88.198.XXX.XXX:5080;rport=5080;branch=z9hG4bKv0XUcNtvK3c2K
Supported: replaces
User-Agent: FXS_GW (1asipfxs.109)
Content-Length: 0
---end---

In freeswitch log I can see this:
---begin---
[NOTICE] switch_channel.c:816 New Channel sofia/internal inboundtest at 88.198.XXX.XXX 
[bf8e8081-eaf1-453e-a643-ee03df36ba0f]
[INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->7777 in context public
[NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [4beaba1f-c9c6-4ed7-94c5-efec453e895a]
[NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [USER_BUSY]
[INFO] mod_dptools.c:2685 Originate Failed.  Cause: USER_BUSY
[NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest at 88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY]
[NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest at 88.198.XXX.XXX) Ended
[NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest at 88.198.XXX.XXX [CS_DESTROY]
[NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended
[NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY]
---end---

In my last tests I called to the voip-provider test number and got UNALLOCATED_NUMBER disconnect cause:
---begin---
[NOTICE] switch_channel.c:816 New Channel sofia/internal/inboundtest at 88.198.XXX.XXX 
[ee9e4e33-0676-4c9c-9952-ff97c4d8db18]
[INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->555 in context public
[NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [b8278ac0-8e90-4d48-bdbf-8d0f608ec35a]
[NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER]
[INFO] mod_dptools.c:2685 Originate Failed.  Cause: UNALLOCATED_NUMBER
[NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest at 88.198.XXX.XXX [CS_EXECUTE] 
[UNALLOCATED_NUMBER]
[NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest at 88.198.XXX.XXX) Ended
[NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest at 88.198.XXX.XXX [CS_DESTROY]
[NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended
[NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY]
---end---


What am I missing here?

Thanks for your help.


From msc at freeswitch.org  Fri May 27 05:28:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 May 2011 18:28:30 -0700
Subject: [Freeswitch-users] trying to get mod_xml_cdr working
In-Reply-To: <BANLkTimiusAYub8dTw-UnZRCHxbOJR-o0Q@mail.gmail.com>
References: <BANLkTikEtt+Eu3p4mCBc53XxRi=e=+e14Q@mail.gmail.com>
	<BANLkTinEccOrZm86rPy05GyqdyvRpziT7g@mail.gmail.com>
	<BANLkTik-rdgQBMbrxvmzbJswL_sCfzg0Kg@mail.gmail.com>
	<BANLkTin1V6TinJWsOVorkwAMTXc88G_mcg@mail.gmail.com>
	<BANLkTi=SPWEzBs0vAS=T_2wErdawdHo=FA@mail.gmail.com>
	<BANLkTimXTTJ5L2RhqCs6CQ9T_Wqk4DS2Rg@mail.gmail.com>
	<BANLkTinuScKyL+sSJ7vnt0czPqcaerKmCw@mail.gmail.com>
	<BANLkTimiusAYub8dTw-UnZRCHxbOJR-o0Q@mail.gmail.com>
Message-ID: <BANLkTi=Xhm+eY9XR2YD4wg_RkGftxuCnQA@mail.gmail.com>

1.0.6 is ancient. Get on the latest git - it is more stable and has more
features and fewer bugs than 1.0.6.
-MC

On Thu, May 26, 2011 at 4:50 PM, deniro <ayhkor at gmail.com> wrote:

> tried it and looks like  that XML line is  gone.
> but now I am unable to join to conference by making suggested change.
> played around it, no luck to join conference....
> it doesnt get DTMF as I check from fs_cli
>    *switch_cpp.cpp:1177 Got dtmf:  -- ..*
> * *
> I also checked usage and it  seems with 8 parameters.
> I can't remember how I got it with 9 parameters before.
> Anyway, It works(joins conference) with 9 parametes only.
> I am using freeswitch 1.0.6 with ubuntu 10.04
> can that usage be different from version to version?
> thx
>
>
> On Thu, May 26, 2011 at 2:34 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> *$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
>> "conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav", "",
>> '^[7]\d{4}\$');
>>
>> *
>> It looks like this usage is wrong. You're giving 9 parameters but the Wiki
>> seems to say it's only 8. It looks like it's using the 9th as a channel
>> variable name to store the result in.
>>
>> Try this:
>>
>> *$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
>> "conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav",
>> '^[7]\d{4}\$');*
>>
>> -Steve
>>
>>
>>
>>
>> On 26 May 2011 06:22, deniro <ayhkor at gmail.com> wrote:
>>
>>> Thanks a lot. The link was a good help.
>>> we tested with sample code in the link and no more 502 errors with the
>>> sample code in the link.
>>>
>>> We tried to make changes in our cdr.pl perl (using mod_xml_cdr) program
>>> accordingly
>>> we made  good progres in generatng  xml file, however we are still
>>> getting 502 errors from cdr.pl.
>>> Probably something else in our cdr.pl  giving us that, we are still
>>> trying to figure out  what  piece is that.
>>>
>>> In the mean time,  in generated  xml  we see a line like this
>>>    * <^[7]\d{4}\$>77555</^[7]\d{4}\$>
>>> *
>>> and looks like this coming from
>>> *$PIN_temp=$session->playAndGetDigits(1, 5, 1, 8000, "#",
>>> "conference/conf-pin.wav", "ivr/ivr-that_was_an_invalid_entry.wav", "",
>>> '^[7]\d{4}\$');*
>>> and because of above xml line I am unable to parse xml.
>>> (above tested with changes and changes reflect on xml line)
>>>
>>> why do you think
>>> above $session->playAndGetDigits   is producing xml  line like
>>> "<^[7]\d{4}\$>77555</^[7]\d{4}\$>"
>>>
>>> any ideas appreciated.
>>> why and how to fix it.
>>>
>>> thx again
>>> deniro--
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, May 25, 2011 at 5:33 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> http://search.cpan.org/~markstos/CGI.pm-3.54/lib/CGI/Fast.pm
>>>>
>>>> That module looks like it is built on top of FCGI and gives you a
>>>> CGI-like interface.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Tue, May 24, 2011 at 10:12 PM, deniro <ayhkor at gmail.com> wrote:
>>>>
>>>>>
>>>>> Thanks for the replies
>>>>> I made some research and run series of tests.
>>>>> I am using  perl+FCGI (fastcgi)  and my http server is  working fine as
>>>>> tested with multiple perl scripts.
>>>>>  I think the problem is handling the CGI within the cdr.plfile(obtained from freeswitch wiki)
>>>>>
>>>>>  so, how can I change below script to use with FCGI instead of
>>>>> CGI::Simple
>>>>>
>>>>> use XML::Simple; # Get from CPAN
>>>>> use CGI::Simple; # Get from CPAN
>>>>> use Data::Dumper;
>>>>> # dump into a place for further review
>>>>> open(FILEOUT,">",'/tmp/fs-xml-cdr2.log');
>>>>> print FILEOUT "Test successful..\n";
>>>>> cloe(FILEOUT);
>>>>> # dump into a place for further review
>>>>> open(FILEOUT,">>",'/tmp/fs-xml-cdr.log');
>>>>> # $cgi object has handy methods
>>>>> my $cgi = new CGI::Simple;
>>>>>
>>>>>
>>>>> Thx
>>>>> deniro
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Mon, May 23, 2011 at 3:50 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>
>>>>>>  > 2-- when dialing into conference, getting following from fs_cli
>>>>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>>>>
>>>>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>>>>> setup and not FS for the mix-up (particularly true as "HTTP Error
>>>>>>> 502"
>>>>>>> means you have a bad gateway).
>>>>>>>
>>>>>>
>>>>>> Check your web server logs - there may be some useful message. It
>>>>>> sounds like the cgi-bin/perl handler hasn't been configured correctly, the
>>>>>> script isn't executable, or it's not returning some headers. Try visiting
>>>>>> the URL in your webbrowser (you
>>>>>> won't be submitting a CDR but you can at least check the script runs
>>>>>> without an (unexpected) error.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 23 May 2011 07:38, Gabriel Gunderson <gabe at gundy.org> wrote:
>>>>>>
>>>>>>> On Sun, May 22, 2011 at 6:27 PM, deniro <ayhkor at gmail.com> wrote:
>>>>>>> > several questions;
>>>>>>> > 1-- do I really need to  load mod_xml_curl for mod_xml_cdr?
>>>>>>> > trying to figure out if my problem is with cdr or with curl
>>>>>>>
>>>>>>> No, they are not functionally related... easy to get mixed up because
>>>>>>> they both use HTTP and XML. One does not depend on the other.
>>>>>>> However,
>>>>>>> you *can* configure mod_xml_cdr (or any other module) with
>>>>>>> mod_xml_curl.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> > 2-- when dialing into conference, getting following from fs_cli
>>>>>>> > [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server
>>>>>>> > [http://www.xxx.com/PERL/cdr.pl]
>>>>>>>
>>>>>>> There isn't a lot of info to work on here, but I'd look at your web
>>>>>>> setup and not FS for the mix-up (particularly true as "HTTP Error
>>>>>>> 502"
>>>>>>> means you have a bad gateway).
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> > 3-- I am not quiete sure if this line is correct in
>>>>>>> xml_cdr.conf.xml
>>>>>>> >  <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/>
>>>>>>>
>>>>>>> Looks good to me, but check it with the wiki:
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_xml_cdr#Sample_Configuration
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> > 4-- if I need mod_xml_curl (question 1)
>>>>>>> > what should this line be like in xml_curl.conf.xml?
>>>>>>> > <param name="gateway-url" value="http://www.xxx.com:80<http://www.xxx.com/>
>>>>>>> "
>>>>>>> > bindings="dialplan"/>
>>>>>>>
>>>>>>> This one is a little tricker to get right.  Again, the wiki is might
>>>>>>> be the best place to reference:
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#Configuring
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing 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 msc at freeswitch.org  Fri May 27 05:33:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 May 2011 18:33:11 -0700
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
	<BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
	<BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>
Message-ID: <BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>

Okay, walk us through what you did to get here. You dialed in from the
outside, got to the IVR, and then dialed 5107? Please confirm. Also, I see
that you have two endpoints in your bridge line. Are you deliberately trying
to dial two separate endpoints at the same time?

-MC

On Thu, May 26, 2011 at 3:45 PM, Baskar <yudha2008 at gmail.com> wrote:

> HI Mike,
>
> I have pasted my log in path http://pastebin.freeswitch.org/16388
>
> <http://pastebin.freeswitch.org/16388>--
> Thanks with Regards,
> N.Baskar
>
> On Thu, May 26, 2011 at 6:33 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Pastebin the debug output of a call to a busy phone.
>> -MC
>>
>>
>> On Thu, May 26, 2011 at 3:02 PM, Baskar <yudha2008 at gmail.com> wrote:
>>
>>> HI Mike,
>>>
>>> I am looking at a condition where if the extension that i enter is busy
>>> then it should automatically route to the extension i specify in the
>>> ivr_inbound xml.
>>>
>>> --
>>>
>>> Thanks with Regards,
>>> N.Baskar
>>>
>>> On Thu, May 26, 2011 at 5:43 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> That's because you are telling it to go to the default context:
>>>>     <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>> param="transfer $1 XML default"/>
>>>> Change that "default" to whatever context you want the destination
>>>> number to be in...
>>>>
>>>> -MC
>>>>
>>>> On Thu, May 26, 2011 at 2:34 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>>
>>>>> Hi David,
>>>>>
>>>>> I tried the dial plan suggested by you several times but the moment i
>>>>> key in the extension it take my input but exits the ivr_inbound.xml and goes
>>>>> to default.xml. It does not bridge me to the extension i key in but instead
>>>>> comes out of the xml and connects me to the extension in the default.xml.
>>>>> Below is the output log of the tests that i have made.
>>>>>
>>>>>
>>>>> Output Log:
>>>>>
>>>>> 2011-05-26 17:24:59.065884 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>> 5:840
>>>>> 2011-05-26 17:24:59.341669 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>> 1:720
>>>>> 2011-05-26 17:24:59.745697 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>> 0:960
>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>> 7:1120
>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:376 digits '5107'
>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:470 action regex
>>>>> [5107] [/^([1-6][0-9][0-9][0-9])$/] [2]
>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:488 IVR action on
>>>>> menu 'ivr_inbound' matched '5107' param
>>>>> 'execute_extensionset:continue_on_fail=true,bridge
>>>>> sofia/internal/5107%XX.XX.XX.XX,bridge sofia/internal/5998%XX.XX.XX.XX'
>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:492
>>>>> switch_ivr_menu_execute todo=[2]
>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:580 exit-sound
>>>>> 'voicemail/vm-goodbye.wav'
>>>>>
>>>>> --
>>>>>
>>>>> Thanks with Regards,
>>>>> N.Baskar
>>>>>
>>>>> On Thu, May 26, 2011 at 1:49 AM, David Ponzone <david.ponzone at ipeva.fr
>>>>> > wrote:
>>>>>
>>>>>> Baskar,
>>>>>>
>>>>>> If you want this behavior only for calls only coming through the
>>>>>> IVR, I think you will have to use bridge rather than transfer.
>>>>>> it would look like this:
>>>>>>
>>>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>>>> param="execute_extension
>>>>>> set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>
>>>>>>
>>>>>> or something like it :)
>>>>>>
>>>>>>  David Ponzone  Direction Technique
>>>>>> email: david.ponzone at ipeva.fr
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>>
>>>>>> Service Client IPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>
>>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et
>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>>>>> susceptible d'alt?ration. **IPeva** d?cline toute responsabilit? au
>>>>>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
>>>>>> pas destinataire de ce message, merci de le d?truire imm?diatement et
>>>>>> d'avertir l'exp?diteur.*
>>>>>> *
>>>>>> *
>>>>>>
>>>>>>
>>>>>>
>>>>>> Le 26/05/2011 ? 00:27, Baskar a ?crit :
>>>>>>
>>>>>> Hi Michael,
>>>>>>
>>>>>> Thanks for your quick reply. Below is the code for the dial plan
>>>>>> inbound ivr. Can you please specify where i should be inserting the
>>>>>> continue_on_fail=true line in the code.
>>>>>>
>>>>>> Default.xml
>>>>>>
>>>>>>     <extension name="inbound">
>>>>>>       <condition field="destination_number"
>>>>>> expression="^(XXXXXXXXXX)$" />
>>>>>>        <condition wday="2-6" hour="8-17">
>>>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>>>         <action application="set" data="accountcode=Inbound"/>
>>>>>>         <action application="answer"/>
>>>>>>         <action application="start_dtmf" />
>>>>>>         <action application="ivr" data="ivr_inbound"/>
>>>>>>         <action application="record_session"
>>>>>> data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>>>>>>         <action application="transfer" data="1018 XML default"/>
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>
>>>>>> ivr_inbound.xml
>>>>>>
>>>>>> <menu name="ivr_inbound"
>>>>>>
>>>>>>  greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>>>>>>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>>>>      exit-sound="voicemail/vm-goodbye.wav"
>>>>>>      confirm-macro=""
>>>>>>      confirm-key=""
>>>>>>      tts-engine="flite"
>>>>>>      tts-voice="rms"
>>>>>>      confirm-attempts="3"
>>>>>>      timeout="10000"
>>>>>>      inter-digit-timeout="2000"
>>>>>>      max-failures="3"
>>>>>>      max-timeouts="1"
>>>>>>      digit-len="4">
>>>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>>>> param="transfer $1 XML default"/>
>>>>>>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
>>>>>> default"/>
>>>>>> </menu>
>>>>>>
>>>>>> *--
>>>>>> Thanks with Regards,
>>>>>>
>>>>>> N.Baskar
>>>>>>
>>>>>> *
>>>>>> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>> If I understand correctly you're wanting to handle a bridge scenario
>>>>>>> where the target extension is busy, etc. Most likely you just need to set
>>>>>>> continue_on_fail=true so that your dialplan continues in the case of the
>>>>>>> target extension not picking up.
>>>>>>>
>>>>>>> -MC
>>>>>>>
>>>>>>> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi All,,
>>>>>>>>
>>>>>>>> In my inbound dial plan xml file  i have set two conditions enabled
>>>>>>>>
>>>>>>>> Condition1:
>>>>>>>>
>>>>>>>> In inbound dial plan callers are given an option to key in the
>>>>>>>> extension number and reach the appropriate extension (Example: 1001 or 1002
>>>>>>>> or 1003 etc).
>>>>>>>>
>>>>>>>> Condition2:
>>>>>>>>
>>>>>>>> The second condition routes call to a default extension in scenarios
>>>>>>>> where the caller does not specify any extension number (Example: Default
>>>>>>>> extension is 1007).
>>>>>>>>
>>>>>>>>
>>>>>>>> Both the above conditions are working fine.
>>>>>>>>
>>>>>>>> Now I need to set up another condition where after keying in the
>>>>>>>> extension number the call gets transferred to the appropriate extension and
>>>>>>>> if the extension is busy(Example: say extension 1003 is busy) it should be
>>>>>>>> hunted to a default extension (example: 1007). How can we set up this
>>>>>>>> condition in dial plan?
>>>>>>>>
>>>>>>>> Can any one guide me.
>>>>>>>> --
>>>>>>>> Thanks with Regards,
>>>>>>>>
>>>>>>>> N.Baskar
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>  _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> -
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri May 27 05:35:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 May 2011 18:35:59 -0700
Subject: [Freeswitch-users] Errors while sending faxes
In-Reply-To: <BANLkTikCcUN2SWwOavh7QKnaYpMvGmGBug@mail.gmail.com>
References: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>
	<121674B8-550C-416D-ACF1-7FD868944C77@ipeva.fr>
	<BANLkTikCcUN2SWwOavh7QKnaYpMvGmGBug@mail.gmail.com>
Message-ID: <BANLkTinM-PLmx9dVqd6U=873ej4s-TBPBA@mail.gmail.com>

Console debug logs on pastebin would help. Use pastebin.freeswitch.org and
use "FreeSWITCH Log" as the syntax highlighting.
-MC

On Thu, May 26, 2011 at 3:43 PM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> David,
>
> I am running freeswitch on CentOS. Faxes are sent by SFR which provides a
> T38-gateway. What I don't know is the endpoints involved. Indeed, I send
> faxes to our clients so I don't know what type of fax machine they have.
>
> I know I give you only few informations but what I'd like to know is
> despite those responses, are the faxes correctly sent?
>
> Thank you for your help :-)
>
>
> 2011/5/26 David Ponzone <david.ponzone at ipeva.fr>
>
>> Mathieu,
>>
>> you should perhaps start by describing your configuration, the endpoints
>> involved, where FS is involved (T38-relay or T38-gateway or pure TDM),
>> etc...
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 26/05/2011 ? 07:54, Mathieu Lautram a ?crit :
>>
>> Hi all,
>>
>> After sending faxes (it works very well know, thanks to Anthony Minessale)
>> , sometimes I've got errors like that:
>>
>> - Disconnected after permitted retries
>> - No response after sending a page
>> - Received a DCN from remote after sending a page
>> - Received bad response to DCS ot TCF
>> - The call dropped prematurely
>> - Timed out waiting for initial communication
>>
>> Those faxes are about 25% of all faxes that I send. I'm using the last git
>> version.
>> Where this problem comes from? Is there a way to fix this?
>> I know that sometimes the remote end could be broken or could be an old
>> fax machine, but it seems strange that I have so many errors like that.
>>
>> Thank you for your help :-)
>>
>> --
>> Mathieu LAUTRAM
>> Application developer
>>
>> BJT Partners - FRANCE
>> +33 1 79 75 99 60
>> +33 6 61 59 07 25
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sipmaillist at gmail.com  Fri May 27 05:46:41 2011
From: sipmaillist at gmail.com (Jakson Kalsson)
Date: Fri, 27 May 2011 09:46:41 +0800
Subject: [Freeswitch-users] XCAP Server
Message-ID: <BANLkTi=fTWDFsbOQ4Z1ynwTDcHP1+nhZGw@mail.gmail.com>

Dear all, does free switch support XCAP?

Thanks

-- 
havesoftware, Inc.
http://www.havesoftware.com


Jakson Kalsson
Senior Programmer
jakkalsoon at havesoftware.com
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From liam at intersys.co.nz  Fri May 27 08:43:44 2011
From: liam at intersys.co.nz (Liam Farr)
Date: Fri, 27 May 2011 16:43:44 +1200
Subject: [Freeswitch-users] Privacy: id / Inbound
Message-ID: <b40b1dbda07415ec494f074af3204d8e@mail.gmail.com>

Hi,



When I receive an inbound call with Privacy: id set in the SIP invite
freeswitch isn?t stripping off the caller I?d when passing the call onto an
internal extension?



The inbound invite looks like this;



recv 1091 bytes from udp/[23.51.15.101]:5060 at 04:14:28.246255:

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

   INVITE sip:92568888 at 202.55.98.6 SIP/2.0

   Max-Forwards: 69

   Supported: 100rel

   To: <sip:92568888 at 202.55.98.6>

   From: <sip:225553389 at 23.51.15.101>;tag=3515458594-499967

   P-Asserted-Identity: <sip:225553389 at 23.51.15.101>

   Privacy: id

   Call-ID: 23213457-3515458594-499959 at MSX-101.squiggly.net

   CSeq: 1 INVITE

   Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK

   Via: SIP/2.0/UDP 23.51.15.101:5060
;branch=z9hG4bKf91a7732fbc823dc2c3252d3f7ec83fc

   Contact: <sip:225553389 at 23.51.15.101:5060;tgrp=2Degrees_sst03>

   Expires: 180

   Call-Info:
<sip:23.51.15.101>;method="NOTIFY;Event=telephone-event;Duration=1000"

   Allow-Events: telephone-event

   Content-Type: application/sdp

   Content-Length: 369



   v=0

   o=MSX-101 8787 8322 IN IP4 23.51.15.101

   s=sip call

   c=IN IP4 23.51.15.103

   t=0 0

   m=audio 29708 RTP/AVP 8 0 18 4 100 101

   a=rtpmap:8 PCMA/8000

   a=rtpmap:0 PCMU/8000

   a=rtpmap:18 G729/8000

   a=fmtp:18 annexb=yes

   a=rtpmap:4 G723/8000

   a=fmtp:4 bitrate=6.3;annexa=yes

   a=rtpmap:100 X-NSE/8000

   a=fmtp:100 192-194

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16





With the internal invite looking like this;



send 1083 bytes to udp/[202.55.98.122]:5080 at 04:14:28.260497:

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

   INVITE sip:555 at 202.55.98.122:5080 SIP/2.0

   Via: SIP/2.0/UDP 202.55.98.7;rport;branch=z9hG4bK20Bv5jcFpj00c

   Max-Forwards: 67

   From: "225553389" <sip:225553389 at 202.55.98.7>;tag=8eK40BpUtXSUB

   To: <sip:555 at 202.55.98.122:5080>

   Call-ID: a7bce49f-02ba-122f-a19a-8903425aa839

   CSeq: 12898770 INVITE

   Contact: <sip:mod_sofia at 202.55.98.7:5060>

   User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported

   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

   Privacy: id

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 203

   X-FS-Support: update_display

   P-Asserted-Identity: "225553389" <sip:225553389 at 202.55.98.7>



   v=0

   o=FreeSWITCH 1306450788 1306450789 IN IP4 202.55.98.7

   s=FreeSWITCH

   c=IN IP4 202.55.98.7

   t=0 0

   m=audio 18880 RTP/AVP 8 9 0 101 13

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=ptime:20





As you can see the caller id is passed onto the internal extension, where I
really need to strip it off / replace it with restricted / anonymous. Is
there a way to catch this as a variable in the inbound dial plan?









Thanks



Liam
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From liam at intersys.co.nz  Fri May 27 09:40:51 2011
From: liam at intersys.co.nz (Liam Farr)
Date: Fri, 27 May 2011 17:40:51 +1200
Subject: [Freeswitch-users] Privacy: id / Inbound
In-Reply-To: <b40b1dbda07415ec494f074af3204d8e@mail.gmail.com>
References: <b40b1dbda07415ec494f074af3204d8e@mail.gmail.com>
Message-ID: <BANLkTik1nnOwhOLfAPGKAyCJJj7WbsEXFQ@mail.gmail.com>

I have now resolved this issue with;

   <condition field="privacy_hide_number" expression="true" continue="true"
break="never">
      <action application="set" data="sip_h_Privacy=id"/>
      <action application="set" data="effective_caller_id_number=Unknown"/>
      <action application="set" data="effective_caller_id_name=Unknown"/>
   </condition>


On Fri, May 27, 2011 at 4:43 PM, Liam Farr <liam at intersys.co.nz> wrote:

> Hi,
>
>
>
> When I receive an inbound call with Privacy: id set in the SIP invite
> freeswitch isn?t stripping off the caller I?d when passing the call onto an
> internal extension?
>
>
>
> The inbound invite looks like this;
>
>
>
> recv 1091 bytes from udp/[23.51.15.101]:5060 at 04:14:28.246255:
>
>    ------------------------------------------------------------------------
>
>    INVITE sip:92568888 at 202.55.98.6 SIP/2.0
>
>    Max-Forwards: 69
>
>    Supported: 100rel
>
>    To: <sip:92568888 at 202.55.98.6>
>
>    From: <sip:225553389 at 23.51.15.101>;tag=3515458594-499967
>
>    P-Asserted-Identity: <sip:225553389 at 23.51.15.101>
>
>    Privacy: id
>
>    Call-ID: 23213457-3515458594-499959 at MSX-101.squiggly.net
>
>    CSeq: 1 INVITE
>
>    Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK
>
>    Via: SIP/2.0/UDP 23.51.15.101:5060
> ;branch=z9hG4bKf91a7732fbc823dc2c3252d3f7ec83fc
>
>    Contact: <sip:225553389 at 23.51.15.101:5060;tgrp=2Degrees_sst03>
>
>    Expires: 180
>
>    Call-Info:
> <sip:23.51.15.101>;method="NOTIFY;Event=telephone-event;Duration=1000"
>
>    Allow-Events: telephone-event
>
>    Content-Type: application/sdp
>
>    Content-Length: 369
>
>
>
>    v=0
>
>    o=MSX-101 8787 8322 IN IP4 23.51.15.101
>
>    s=sip call
>
>    c=IN IP4 23.51.15.103
>
>    t=0 0
>
>    m=audio 29708 RTP/AVP 8 0 18 4 100 101
>
>    a=rtpmap:8 PCMA/8000
>
>    a=rtpmap:0 PCMU/8000
>
>    a=rtpmap:18 G729/8000
>
>    a=fmtp:18 annexb=yes
>
>    a=rtpmap:4 G723/8000
>
>    a=fmtp:4 bitrate=6.3;annexa=yes
>
>    a=rtpmap:100 X-NSE/8000
>
>    a=fmtp:100 192-194
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>
>
>
>
> With the internal invite looking like this;
>
>
>
> send 1083 bytes to udp/[202.55.98.122]:5080 at 04:14:28.260497:
>
>    ------------------------------------------------------------------------
>
>    INVITE sip:555 at 202.55.98.122:5080 SIP/2.0
>
>    Via: SIP/2.0/UDP 202.55.98.7;rport;branch=z9hG4bK20Bv5jcFpj00c
>
>    Max-Forwards: 67
>
>    From: "225553389" <sip:225553389 at 202.55.98.7>;tag=8eK40BpUtXSUB
>
>    To: <sip:555 at 202.55.98.122:5080>
>
>    Call-ID: a7bce49f-02ba-122f-a19a-8903425aa839
>
>    CSeq: 12898770 INVITE
>
>    Contact: <sip:mod_sofia at 202.55.98.7:5060>
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported
>
>    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
>
>    Privacy: id
>
>    Content-Type: application/sdp
>
>    Content-Disposition: session
>
>    Content-Length: 203
>
>    X-FS-Support: update_display
>
>    P-Asserted-Identity: "225553389" <sip:225553389 at 202.55.98.7>
>
>
>
>    v=0
>
>    o=FreeSWITCH 1306450788 1306450789 IN IP4 202.55.98.7
>
>    s=FreeSWITCH
>
>    c=IN IP4 202.55.98.7
>
>    t=0 0
>
>    m=audio 18880 RTP/AVP 8 9 0 101 13
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>    a=ptime:20
>
>
>
>
>
> As you can see the caller id is passed onto the internal extension, where I
> really need to strip it off / replace it with restricted / anonymous. Is
> there a way to catch this as a variable in the inbound dial plan?
>
>
>
>
>
>
>
>
>
> Thanks
>
>
>
> Liam
>
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From david.ponzone at ipeva.fr  Fri May 27 09:51:13 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 27 May 2011 07:51:13 +0200
Subject: [Freeswitch-users] Question about P-Asserted-Identity
In-Reply-To: <4DDECF5E.1040904@kempgen.net>
References: <83CC0F6E-8DD4-4285-A4CA-85C2ABAE8905@ipeva.fr>
	<6FE1AF40-7902-41B9-A5F5-DA4F3BDC5EAD@ipeva.fr>
	<4DDECF5E.1040904@kempgen.net>
Message-ID: <F6884262-600D-4B58-946B-570207648893@ipeva.fr>

Philipp,

thanks, that's what I was thinking, and you're totally right about the missing userinfo part!
I gonna tell them.

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

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

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




Le 27/05/2011 ? 00:08, Philipp Kempgen a ?crit :

> David Ponzone wrote:
>> 
>> The P-Asserted-Identity I receive from the provider is of the form:
>> 
>> <sip:12356789>
>> without any @domain part.
> 
> Actually that's a SIP-URI with a hostname ("12356789") but
> without a userinfo part. (see ABNF rules in RFC 3261)
> 
> It's a valid SIP-URI in theory but without userinfo so it doesn't
> make too much sense as a caller ID in a PAI header.
> 
> Tell your provider to fix it.
> 
> 
>    Philipp
> 
> -- 
> http://twitter.com/kempgen
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Fri May 27 09:57:48 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 27 May 2011 07:57:48 +0200
Subject: [Freeswitch-users] Errors while sending faxes
In-Reply-To: <BANLkTikCcUN2SWwOavh7QKnaYpMvGmGBug@mail.gmail.com>
References: <BANLkTinFGzdJK2wLyNkfwEGqmKm-UxDEsA@mail.gmail.com>
	<121674B8-550C-416D-ACF1-7FD868944C77@ipeva.fr>
	<BANLkTikCcUN2SWwOavh7QKnaYpMvGmGBug@mail.gmail.com>
Message-ID: <9DA5CEB2-65D9-4DAD-877C-01430A0CC161@ipeva.fr>

Well, 2 comments:
-you dont know the CPE endpoints involved ??? the endpoint is the SIP ATA which talks T38. I suppose you manage that
-SFR ? Ok, good luck with that :) I was not even aware they proposed that. I know they don't on their wholesale SIP trunk product, so that must be a direct SIP trunk (direct to customer).

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

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

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




Le 27/05/2011 ? 00:43, Mathieu Lautram a ?crit :

> David,
> 
> I am running freeswitch on CentOS. Faxes are sent by SFR which provides a T38-gateway. What I don't know is the endpoints involved. Indeed, I send faxes to our clients so I don't know what type of fax machine they have.
> 
> I know I give you only few informations but what I'd like to know is despite those responses, are the faxes correctly sent?
> 
> Thank you for your help :-)
> 
> 
> 2011/5/26 David Ponzone <david.ponzone at ipeva.fr>
> Mathieu,
> 
> you should perhaps start by describing your configuration, the endpoints involved, where FS is involved (T38-relay or T38-gateway or pure TDM), etc...
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 26/05/2011 ? 07:54, Mathieu Lautram a ?crit :
> 
>> Hi all,
>> 
>> After sending faxes (it works very well know, thanks to Anthony Minessale) , sometimes I've got errors like that:
>> 
>> - Disconnected after permitted retries
>> - No response after sending a page
>> - Received a DCN from remote after sending a page
>> - Received bad response to DCS ot TCF
>> - The call dropped prematurely
>> - Timed out waiting for initial communication
>> 
>> Those faxes are about 25% of all faxes that I send. I'm using the last git version.
>> Where this problem comes from? Is there a way to fix this?
>> I know that sometimes the remote end could be broken or could be an old fax machine, but it seems strange that I have so many errors like that.
>> 
>> Thank you for your help :-)
>> 
>> -- 
>> Mathieu LAUTRAM
>> Application developer
>> 
>> BJT Partners - FRANCE
>> +33 1 79 75 99 60
>> +33 6 61 59 07 25
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Mathieu LAUTRAM
> Application developer
> 
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From zetruger at gmail.com  Fri May 27 10:30:13 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Fri, 27 May 2011 10:30:13 +0400
Subject: [Freeswitch-users] FreeSWITCH as SIP softswitch
In-Reply-To: <4DDEDC31.40601@gmail.com>
References: <4DDEDC31.40601@gmail.com>
Message-ID: <BANLkTi=tYbGQZid0SnjTtPFom-NqmK=D4g@mail.gmail.com>

Use directory for SIP clients (hardphone, softphone) instead gateways.
Use gateway just for SIP providers (SIP trunks).

Simple test (run from FS console):
originate sofia/internal/outboundtest at x.x.x.x:p
&bridge(sofia/internal/inboundtest at x.x.x.x:p)

2011/5/27 Cyril Zlachevsky <cyril.zlachevsky at gmail.com>:
> Hello,
> I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server.
> But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help.
>
> FreeSWITCH server (IP 88.198.XXX.XXX) have to receive SIP calls from the one IP address (X-Lite
> softphone with dynamic IP from subnet 92.112.0.0/16) and forward this calls to the hardware SIP
> phone (Asotel, IP 195.225.XXX.XXX). SIP mode on hardware softphone (Asotel) is peer-to-peer.
>
> Direct calls from X-Lite to Asotel complete with success - I always hear incoming ring when dial
> 7777. But through FreeSWITCH calls always fail.
>
> I created four configuration files - all other configs I left unchanged:
> 1) conf/sip-profiles/internal/X-Lite.xml:
> <include>
> ? <gateway name="X-Lite">
> ? ? <param name="username" value="inboundtest"/>
> ? ? <param name="password" value="test"/>
> ? ? <param name="register" value="true"/>
> ? </gateway>
> </include>
>
> 2) conf/sip-profiles/external/Asotel.xml:
> <include>
> ? <gateway name="Asotel">
> ? ? <param name="realm" value="195.225.XXX.XXX"/>
> ? ? <param name="username" value="outboundtest"/>
> ? ? <param name="password" value="test"/>
> ? ? <param name="register" value="true"/>
> ? </gateway>
> </include>
>
> 3) conf/dialplan/public/test.xml:
> <include>
> ? <extension name="test">
> ? ? ? <condition field="destination_number" expression="^7777$">
> ? ? ? ? <action application="bridge" data="sofia/gateway/Asotel/$1"/>
> ? ? ? </condition>
> ? ? </extension>
> </include>
>
> 4) conf/directory/default/inboundtest.xml:
> <include>
> ? <user id="inboundtest" cidr="92.112.0.0/16">
> ? ? <params>
> ? ? ? <param name="from-domain" value="88.198.XXX.XXX"/>
> ? ? ? <param name="password" value="test"/>
> ? ? </params>
> ? </user>
> </include>
>
>
> This debug from Asotel ip-phone:
> ---begin---
> Incoming CallLeg at callleg created 0x57d334 Incoming CallLeg at MsgReceived 0x57d334 ?*** $1 was
> being Invited ***
> ?>>> All call occupied. <<<
> No slot availabe for this call...
> FindIPCall...All Slot is Busy
> RvSipCallLegReject(486), hCallLeg: 57d334
> --> Message Sent (Message type: 1) (call-leg 57d334)
> SIP/2.0 486 Busy Here
> From: "inboundtest"<sip:outboundtest at 195.225.XXX.XXX;transport=udp>;tag=vgpp5vSBgcX6p
> To: <sip:$1 at 195.225.XXX.XXX>;tag=c3e19fb6-13c4-4ddecb5a-1f577-5f2a
> Call-ID: 232b19fe-0285-122f-b5b5-1b5bdf4f9807
> CSeq: 12887277 INVITE
> Via: SIP/2.0/UDP 88.198.XXX.XXX:5080;rport=5080;branch=z9hG4bKv0XUcNtvK3c2K
> Supported: replaces
> User-Agent: FXS_GW (1asipfxs.109)
> Content-Length: 0
> ---end---
>
> In freeswitch log I can see this:
> ---begin---
> [NOTICE] switch_channel.c:816 New Channel sofia/internal inboundtest at 88.198.XXX.XXX
> [bf8e8081-eaf1-453e-a643-ee03df36ba0f]
> [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->7777 in context public
> [NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [4beaba1f-c9c6-4ed7-94c5-efec453e895a]
> [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [USER_BUSY]
> [INFO] mod_dptools.c:2685 Originate Failed. ?Cause: USER_BUSY
> [NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest at 88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY]
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest at 88.198.XXX.XXX) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest at 88.198.XXX.XXX [CS_DESTROY]
> [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY]
> ---end---
>
> In my last tests I called to the voip-provider test number and got UNALLOCATED_NUMBER disconnect cause:
> ---begin---
> [NOTICE] switch_channel.c:816 New Channel sofia/internal/inboundtest at 88.198.XXX.XXX
> [ee9e4e33-0676-4c9c-9952-ff97c4d8db18]
> [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->555 in context public
> [NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [b8278ac0-8e90-4d48-bdbf-8d0f608ec35a]
> [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER]
> [INFO] mod_dptools.c:2685 Originate Failed. ?Cause: UNALLOCATED_NUMBER
> [NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest at 88.198.XXX.XXX [CS_EXECUTE]
> [UNALLOCATED_NUMBER]
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest at 88.198.XXX.XXX) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest at 88.198.XXX.XXX [CS_DESTROY]
> [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY]
> ---end---
>
>
> What am I missing here?
>
> Thanks for your help.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From u2nsam at gmail.com  Fri May 27 10:41:45 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 27 May 2011 12:11:45 +0530
Subject: [Freeswitch-users] matching word
In-Reply-To: <BANLkTinai5FLezOrHGSv2kAU7wbUKNVU3Q@mail.gmail.com>
References: <BANLkTinfe5rvkz_qQDDPT617pjcLsTswtw@mail.gmail.com>
	<1306405611055-6406420.post@n2.nabble.com>
	<BANLkTimBXuivtsVJSrX0Ck_STsQvZD2ozA@mail.gmail.com>
	<BANLkTinai5FLezOrHGSv2kAU7wbUKNVU3Q@mail.gmail.com>
Message-ID: <BANLkTi=KtPFKMSPjd0D-iLpPT4yXsOxD=g@mail.gmail.com>

Is my dialplan correct ?

<extension name="public_extensions">
      <condition field="destination_number" expression="^(sam)$">
      <action application="info"/>
    <action application="voicemail" data="default $${domain}
${sip_to_host}"/>
      </condition>
    </extension>


I want to match three any alphanumeric 3 digit/letters

Also i wan to store the voicemail into the folders where the herachy would
be      :::          Domain --> Username --> Voicemail file

Regards
Sam




On Fri, May 27, 2011 at 12:19 AM, Michael Collins <msc at freeswitch.org>wrote:

> Your regex is only going to match a single alphanumeric character. Are you
> trying to match an actual email address or what? Once you know what you hope
> to match it will be easier to know which regex to use. For the record a
> semi-proper email grabbing regex is like this:
>
> ^([a-zA-Z0-9._%+-]+@[a-zA-Z0-9.-]+\.[a-zA-Z]{2,6})$
>
> Matching emails is an interesting exercise. If you're not all into "making
> it perfect" and just want to capture "foo at bar" then do something like
> this:
>
> ^([^@]+ at .*)$
>
> -MC
>
>
> On Thu, May 26, 2011 at 10:02 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Thanks for that,
>>
>> I want to use that when someone with username sam at gmail.com will be
>> routed to a voicemail dialplan, with domain partitioning.
>>
>> here still i am not able to achiveve it with domain.
>>
>> <domain name="gmail.com">
>>
>>     <extension name="public_extensions">
>>       <condition field="destination_number" expression="^(\w)$">
>>           <action application="info"/>
>>         <action application="voicemail" data="default $${domain_name}
>> $1"/>
>>       </condition>
>>     </extension>
>>
>> </domain>
>>
>> here above gmail.com is examplary .
>>
>> when there is an invite coming from different server to FS with
>> sam at gmail.com and it is routed directly on voicemail context to route via
>> public.xml
>>
>> here first the call is not recognized to the context gmail.com but when i
>> remove that it goes to voicemail directly properly and stores the voicemail.
>>
>> but the voicemail is not stored under domain name gmail.com which should
>> store it ideally under domain name as folder, which it is just storing it
>> under user sam folder.
>>
>> This is what I want to achieve.
>>
>> Regards
>> Sam
>>
>>
>>
>>
>>
>> On Thu, May 26, 2011 at 3:56 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:
>>
>>>
>>> samir wrote:
>>> >
>>> > Friends,
>>> >
>>> > How can i match words in expressions,
>>> >
>>> > is it by expression="^(\y\w+\y)$"  or could be something else.
>>> >
>>> > Regards
>>> > Sam
>>> I understand ^(\w)$ will match any word, but not with \y. Perhaps, if you
>>> tell us exactly what you want to accomplish, someone will chime in for a
>>> better solution.
>>>
>>> -----
>>> 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/matching-word-tp6406170p6406420.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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Fri May 27 12:00:21 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 27 May 2011 10:00:21 +0200
Subject: [Freeswitch-users] 300 message without Diversion header
In-Reply-To: <4DDED653.20505@kempgen.net>
References: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>
	<4DDED653.20505@kempgen.net>
Message-ID: <BANLkTinAYUoFyZSba9_cNwfjp2ubZKtapg@mail.gmail.com>

On Fri, May 27, 2011 at 12:38 AM, Philipp Kempgen <lists at kempgen.net> wrote:

> Tihomir Culjaga wrote:
>
> > should the 3000/302 message always contain a diversion field saying
> > the call is diverted ?
>
> http://tools.ietf.org/html/rfc5806#section-5.3
> says it "SHOULD".
>

everything in RFC is "SHOULD" or "MAY" ... this is not ETSI.


>
> The Diversion header could be encrypted (/removed?) for privacy
> reasons.
> http://tools.ietf.org/html/rfc5806#section-7
>
>
>
there is a Flag for privacy on Diversion header and its not supposed to be
removed in this case :=)

   diversion-privacy = "privacy" "=" ( "full" | "name" |
                       "uri" | "off" | token | quoted-string )






>    Philipp
>
> --
> http://twitter.com/kempgen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Fri May 27 12:17:22 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 27 May 2011 10:17:22 +0200
Subject: [Freeswitch-users] 300 message without Diversion header
In-Reply-To: <BC87F03A-02BC-4B24-89AE-202A080F0538@freeswitch.org>
References: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>
	<BC87F03A-02BC-4B24-89AE-202A080F0538@freeswitch.org>
Message-ID: <BANLkTi=bubgQXH54VviNjSzp1-QeVF6z1Q@mail.gmail.com>

On Thu, May 26, 2011 at 9:54 PM, Brian West <brian at freeswitch.org> wrote:

> Check your variables. via uuid_dump/info
>



here they are :=)


EXECUTE sofia/external/00681038515000402 at 127.0.0.1:5061 info()
2011-05-26 10:05:29.230485 [INFO] mod_dptools.c:1203 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-Call-State: [RINGING]
Channel-State-Number: [4]
Channel-Name: [sofia/external/00681038515000402 at 127.0.0.1:5061]
Unique-ID: [17ebc5e5-87cf-4fc9-a5b7-c69c0e682d41]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMA]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMA]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [00681038515000402]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [00681038515000402]
Caller-Caller-ID-Number: [00681038515000402]
Caller-Network-Addr: [10.4.62.88]
Caller-ANI: [00681038515000402]
Caller-Destination-Number: [30003016094191500]
Caller-Unique-ID: [17ebc5e5-87cf-4fc9-a5b7-c69c0e682d41]
Caller-Source: [mod_sofia]
Caller-Context: [public]
Caller-Channel-Name: [sofia/external/00681038515000402 at 127.0.0.1:5061]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1306407929129882]
Caller-Channel-Created-Time: [1306407929129882]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_direction: [inbound]
variable_uuid: [17ebc5e5-87cf-4fc9-a5b7-c69c0e682d41]
variable_sip_local_network_addr: [10.4.62.88]
variable_sip_network_ip: [10.4.62.88]
variable_sip_network_port: [5061]
variable_sip_received_ip: [10.4.62.88]
variable_sip_received_port: [5061]
variable_sip_via_protocol: [udp]
variable_sip_from_user: [00681038515000402]
variable_sip_from_port: [5061]
variable_sip_from_uri: [00681038515000402 at 127.0.0.1:5061]
variable_sip_from_host: [127.0.0.1]
variable_sip_from_user_stripped: [00681038515000402]
variable_sip_from_tag: [3]
variable_sofia_profile_name: [external]
variable_sip_full_via: [SIP/2.0/UDP 127.0.0.1:5061
;branch=z9hG4bK-354-3-0;received=10.4.62.88]
variable_sip_from_display: [00681038515000402]
variable_sip_full_from: [00681038515000402 <
sip:00681038515000402 at 127.0.0.1:5061>;tag=3]
variable_sip_to_display: [30003016094191500]
variable_sip_full_to: [30003016094191500 <
sip:30003016094191500 at 10.4.62.88:5060>]
variable_sip_req_user: [30003016094191500]
variable_sip_req_port: [5060]
variable_sip_req_uri: [30003016094191500 at 10.4.62.88:5060]
variable_sip_req_host: [10.4.62.88]
variable_sip_to_user: [30003016094191500]
variable_sip_to_port: [5060]
variable_sip_to_uri: [30003016094191500 at 10.4.62.88:5060]
variable_sip_to_host: [10.4.62.88]
variable_sip_contact_user: [00681038515000402]
variable_sip_contact_uri: [00681038515000402 at 127.0.0.1]
variable_sip_contact_host: [127.0.0.1]
variable_channel_name: [sofia/external/00681038515000402 at 127.0.0.1:5061]
variable_sip_call_id: [3-354 at 127.0.0.1]
variable_sip_via_host: [127.0.0.1]
variable_sip_via_port: [5061]
variable_max_forwards: [70]
variable_sip_h_Diversion: [<sip:030230003038516608363 at 10.4.13.11
;user=phone>;reason=deflection;counter=1]
variable_switch_r_sdp: [v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMA/8000
]
variable_remote_media_ip: [127.0.0.1]
variable_remote_media_port: [6000]
variable_sip_audio_recv_pt: [0]
variable_sip_use_codec_name: [PCMA]
variable_sip_use_codec_rate: [8000]
variable_sip_use_codec_ptime: [20]
variable_read_codec: [PCMA]
variable_read_rate: [8000]
variable_write_codec: [PCMA]
variable_write_rate: [8000]
variable_dtmf_type: [none]
variable_endpoint_disposition: [RECEIVED]
variable_intf: [false]
variable_aPfx: [006810]
variable_divertTo: [385030230003038516608363]
variable_aNum: [38515000402]
variable_IP_ADDR: [10.4.62.88:5060]
variable_bPfx: [300030]
variable_bNum: [16094191500]
variable_caller_id_number: [006810385030230003038516608363]
variable_my_sid: [[006810 38515000402 -> 300030 16094191500 :
gid-68ab-5969]]
variable_red_contact: [<sip:01002016094191500 at 10.4.62.88:5060
;user=phone>;q=0.99,<sip:04901016094191500 at 10.4.62.88:5060
;user=phone>;q=0.98,<sip:03861016094191500 at 10.4.62.88:5060
;user=phone>;q=0.97,<sip:00001016094191500 at 10.4.62.88:5060
;user=phone>;q=0.96]

variable_authResult: [0]
variable_componentStatus: [0:0]
variable_current_application: [info]




Invite:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.62.88:5060 SIP/2.0
   Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-354-3-0
   Max-Forwards: 70
   Contact: <sip:00681038515000402 at 127.0.0.1>
   Diversion: <sip:030230003038516608363 at 10.4.13.11
;user=phone>;reason=deflection;counter=1
   From: 00681038515000402 <sip:00681038515000402 at 127.0.0.1:5061>;tag=3
   To: 30003016094191500 <sip:30003016094191500 at 10.4.62.88:5060>
   Call-ID: 3-354 at 127.0.0.1
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   129

   v=0
   o=user1 53655765 2353687637 IN IP4 127.0.0.1
   s=-
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMA/8000
   ------------------------------------------------------------------------



300:
   ------------------------------------------------------------------------
   SIP/2.0 300 Multiple Choices
   Via: SIP/2.0/UDP 127.0.0.1:5061
;branch=z9hG4bK-354-3-0;received=10.4.62.88
   From: 00681038515000402 <sip:00681038515000402 at 127.0.0.1:5061>;tag=3
   To: 30003016094191500 <sip:30003016094191500 at 10.4.62.88:5060
>;tag=N0pScX59jUjHm
   Call-ID: 3-354 at 127.0.0.1
   CSeq: 1 INVITE
   Contact: <sip:01002016094191500 at 10.4.62.88:5060;user=phone>;q=0.99
   Contact: <sip:04901016094191500 at 10.4.62.88:5060;user=phone>;q=0.98
   Contact: <sip:03861016094191500 at 10.4.62.88:5060;user=phone>;q=0.97
   Contact: <sip:00001016094191500 at 10.4.62.88:5060;user=phone>;q=0.96
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Length: 0

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





>
> /b
>
> On May 26, 2011, at 2:42 PM, Tihomir Culjaga wrote:
>
> > hello,
> >
> > i got a strange issue ... I'm using FS as a redirect server (route
> server)
> > and it does work fine ... except if the original INVITE contains a
> Diversion
> > header, the same header is lost in the responding 300 or 302 message.
> > Is this by design or its a bug ?
> >
> >
> > this is a part of the code in sofia where u construct redirect response
> > message...
> >
> >                if (argc > 1) {
> >                    nua_respond(tech_pvt->nh, SIP_300_MULTIPLE_CHOICES,
> > SIPTAG_CONTACT_STR(dest),
> >                                TAG_IF(!zstr(extra_headers),
> > SIPTAG_HEADER_STR(extra_headers)), TAG_END());
> >                } else {
> >                    nua_respond(tech_pvt->nh, SIP_302_MOVED_TEMPORARILY,
> > SIPTAG_CONTACT_STR(dest),
> >                                TAG_IF(!zstr(extra_headers),
> > SIPTAG_HEADER_STR(extra_headers)), TAG_END());
> >                }
> >
> >
> > so, why the original diversion header is missing ? ... m'I missing
> something
> > ?
> >
> >
> > also, should the 3000/302 message always contain a diversion field saying
> > the call is diverted ?
> >
> >
> > Thanks for your answer,
> > Tihomir.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 27 12:37:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 27 May 2011 09:37:50 +0100
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
Message-ID: <BANLkTim0-D3ut6J-Lf8dJ_AweWhw6Njcbg@mail.gmail.com>

Is there any NAT or firewall between the endpoints? That's usually the
reason for media going missing.

-Steve


On 27 May 2011 00:41, Yungwei Chen <yungwei at resolvity.com> wrote:

> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David
> Ponzone
> *Sent:* Thursday, May 26, 2011 3:14 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Transferring to a cell phone
>
>
>
> You have to use "bridge" instead of "transfer".
>
> "Transfer" is to transfer the call to another part of the dialplan, with
> the same or another dialed number.
>
>
>
> The Wiki is your friend :)
>
>
>
> David Ponzone  Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel:      01 74 03 18 97
>
> gsm:   06 66 98 76 34
>
>
>
> Service Client IPeva
>
> tel:      0811 46 26 26
>
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
>
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
>
> * *
>
>
>
>
>
> Le 26/05/2011 ? 20:45, Yungwei Chen a ?crit :
>
>
>
> Hi,
>
> I installed freeswitch from a recent snapshot on CentOS 5.
> I configured a gateway for my SIP provider in
> /usr/loca/freeswitch/conf/sip_profiles/external/my.xml, and I can make an
> outbound call to my cell phone using the following javascript snippet.
>   var new_session = new
> Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>
> Now I am trying to make a call from a land-line to a javascript application
> in freeswich, which plays a recording and then transfers the call to a cell
> phone.
> But transferring the call to a cell phone doesn't work for some reason.
>
> Here's how I transfer the call to a cell phone in javascript:
>   session.execute("transfer",
> "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>
> According to freeswitch.log, the problem seems to be that freeswitch cannot
> find a dialplan that matches the destiantion number,
> {ignore_early_media=true}sofia/gateway/sip_provider/1231231234.
> What am I missing here? 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 steveayre at gmail.com  Fri May 27 12:39:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 27 May 2011 09:39:36 +0100
Subject: [Freeswitch-users] FreeSWITCH as SIP softswitch
In-Reply-To: <BANLkTi=tYbGQZid0SnjTtPFom-NqmK=D4g@mail.gmail.com>
References: <4DDEDC31.40601@gmail.com>
	<BANLkTi=tYbGQZid0SnjTtPFom-NqmK=D4g@mail.gmail.com>
Message-ID: <BANLkTima=+vACXHSu6yeMcRyjXYVTgeKVw@mail.gmail.com>

User directory = Inbound registration
Gateway = Outbound registration

-Steve


On 27 May 2011 07:30, ???? ???????? <zetruger at gmail.com> wrote:

> Use directory for SIP clients (hardphone, softphone) instead gateways.
> Use gateway just for SIP providers (SIP trunks).
>
> Simple test (run from FS console):
> originate sofia/internal/outboundtest at x.x.x.x:p
> &bridge(sofia/internal/inboundtest at x.x.x.x:p)
>
> 2011/5/27 Cyril Zlachevsky <cyril.zlachevsky at gmail.com>:
> > Hello,
> > I have success with compiling and installing latest (1.0.7) FreeSWITCH on
> my server.
> > But all my today attempts to configure FreeSWITCH as softswitch failed
> and I really need help.
> >
> > FreeSWITCH server (IP 88.198.XXX.XXX) have to receive SIP calls from the
> one IP address (X-Lite
> > softphone with dynamic IP from subnet 92.112.0.0/16) and forward this
> calls to the hardware SIP
> > phone (Asotel, IP 195.225.XXX.XXX). SIP mode on hardware softphone
> (Asotel) is peer-to-peer.
> >
> > Direct calls from X-Lite to Asotel complete with success - I always hear
> incoming ring when dial
> > 7777. But through FreeSWITCH calls always fail.
> >
> > I created four configuration files - all other configs I left unchanged:
> > 1) conf/sip-profiles/internal/X-Lite.xml:
> > <include>
> >   <gateway name="X-Lite">
> >     <param name="username" value="inboundtest"/>
> >     <param name="password" value="test"/>
> >     <param name="register" value="true"/>
> >   </gateway>
> > </include>
> >
> > 2) conf/sip-profiles/external/Asotel.xml:
> > <include>
> >   <gateway name="Asotel">
> >     <param name="realm" value="195.225.XXX.XXX"/>
> >     <param name="username" value="outboundtest"/>
> >     <param name="password" value="test"/>
> >     <param name="register" value="true"/>
> >   </gateway>
> > </include>
> >
> > 3) conf/dialplan/public/test.xml:
> > <include>
> >   <extension name="test">
> >       <condition field="destination_number" expression="^7777$">
> >         <action application="bridge" data="sofia/gateway/Asotel/$1"/>
> >       </condition>
> >     </extension>
> > </include>
> >
> > 4) conf/directory/default/inboundtest.xml:
> > <include>
> >   <user id="inboundtest" cidr="92.112.0.0/16">
> >     <params>
> >       <param name="from-domain" value="88.198.XXX.XXX"/>
> >       <param name="password" value="test"/>
> >     </params>
> >   </user>
> > </include>
> >
> >
> > This debug from Asotel ip-phone:
> > ---begin---
> > Incoming CallLeg at callleg created 0x57d334 Incoming CallLeg at
> MsgReceived 0x57d334  *** $1 was
> > being Invited ***
> >  >>> All call occupied. <<<
> > No slot availabe for this call...
> > FindIPCall...All Slot is Busy
> > RvSipCallLegReject(486), hCallLeg: 57d334
> > --> Message Sent (Message type: 1) (call-leg 57d334)
> > SIP/2.0 486 Busy Here
> > From: "inboundtest"<sip:outboundtest at 195.225.XXX.XXX
> ;transport=udp>;tag=vgpp5vSBgcX6p
> > To: <sip:$1 at 195.225.XXX.XXX>;tag=c3e19fb6-13c4-4ddecb5a-1f577-5f2a
> > Call-ID: 232b19fe-0285-122f-b5b5-1b5bdf4f9807
> > CSeq: 12887277 INVITE
> > Via: SIP/2.0/UDP
> 88.198.XXX.XXX:5080;rport=5080;branch=z9hG4bKv0XUcNtvK3c2K
> > Supported: replaces
> > User-Agent: FXS_GW (1asipfxs.109)
> > Content-Length: 0
> > ---end---
> >
> > In freeswitch log I can see this:
> > ---begin---
> > [NOTICE] switch_channel.c:816 New Channel sofia/internal
> inboundtest at 88.198.XXX.XXX
> > [bf8e8081-eaf1-453e-a643-ee03df36ba0f]
> > [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->7777
> in context public
> > [NOTICE] switch_channel.c:816 New Channel sofia/external/$1
> [4beaba1f-c9c6-4ed7-94c5-efec453e895a]
> > [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA]
> [USER_BUSY]
> > [INFO] mod_dptools.c:2685 Originate Failed.  Cause: USER_BUSY
> > [NOTICE] mod_dptools.c:2799 Hangup
> sofia/internal/inboundtest at 88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY]
> > [NOTICE] switch_core_session.c:1304 Session 1
> (sofia/internal/inboundtest at 88.198.XXX.XXX) Ended
> > [NOTICE] switch_core_session.c:1306 Close Channel
> sofia/internal/inboundtest at 88.198.XXX.XXX [CS_DESTROY]
> > [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended
> > [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1
> [CS_DESTROY]
> > ---end---
> >
> > In my last tests I called to the voip-provider test number and got
> UNALLOCATED_NUMBER disconnect cause:
> > ---begin---
> > [NOTICE] switch_channel.c:816 New Channel
> sofia/internal/inboundtest at 88.198.XXX.XXX
> > [ee9e4e33-0676-4c9c-9952-ff97c4d8db18]
> > [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->555
> in context public
> > [NOTICE] switch_channel.c:816 New Channel sofia/external/$1
> [b8278ac0-8e90-4d48-bdbf-8d0f608ec35a]
> > [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA]
> [UNALLOCATED_NUMBER]
> > [INFO] mod_dptools.c:2685 Originate Failed.  Cause: UNALLOCATED_NUMBER
> > [NOTICE] mod_dptools.c:2799 Hangup
> sofia/internal/inboundtest at 88.198.XXX.XXX [CS_EXECUTE]
> > [UNALLOCATED_NUMBER]
> > [NOTICE] switch_core_session.c:1304 Session 1
> (sofia/internal/inboundtest at 88.198.XXX.XXX) Ended
> > [NOTICE] switch_core_session.c:1306 Close Channel
> sofia/internal/inboundtest at 88.198.XXX.XXX [CS_DESTROY]
> > [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended
> > [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1
> [CS_DESTROY]
> > ---end---
> >
> >
> > What am I missing here?
> >
> > Thanks for your help.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Fri May 27 12:55:40 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 27 May 2011 10:55:40 +0200
Subject: [Freeswitch-users] Bridging Gateway from another server
In-Reply-To: <4DDE2633.2030002@xofap.com>
References: <4DDE2633.2030002@xofap.com>
Message-ID: <BANLkTinGsRnpx_g6V3-Lk1BZT2XwayvRXg@mail.gmail.com>

on FS1 machine define FS2 as gateway and bridge the call to it...



<extension name="To_FS2">
  <condition field="destination_number" expression="(.*)">
      <action application="bridge" data="sofia/gateway/FS2/$1"/>
   </condition>
</extension>



or without creating any gateways just bridge to FS2 IP address...

<extension name="To_FS2">
  <condition field="destination_number" expression="(.*)">
      <action application="bridge" data="sofia/profilename/$1 at FS2_IP"/>
   </condition>
</extension>





On Thu, May 26, 2011 at 12:06 PM, William Alianto <william at xofap.com> wrote:

> I'm trying to do an outbound scenario as following :
>
> Client --> FS1 --> FS2 --> SBC
>
> It maybe more simple if I add the SBC as gateway to FS1, but it only
> accept IP from FS2 due to IP restriction from provider. Is there any
> possible solution for this scenario?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From william at xofap.com  Fri May 27 13:33:46 2011
From: william at xofap.com (William Alianto)
Date: Fri, 27 May 2011 16:33:46 +0700
Subject: [Freeswitch-users] Bridging Gateway from another server
In-Reply-To: <BANLkTinGsRnpx_g6V3-Lk1BZT2XwayvRXg@mail.gmail.com>
References: <4DDE2633.2030002@xofap.com>
	<BANLkTinGsRnpx_g6V3-Lk1BZT2XwayvRXg@mail.gmail.com>
Message-ID: <4DDF6FFA.9060108@xofap.com>

I tried both scenario, but it return as "Incompatible Destination" 
instead. Do I missed some step?

On 2:59, Tihomir Culjaga wrote:
> on FS1 machine define FS2 as gateway and bridge the call to it...
>
>
>
> <extension name="To_FS2">
> <condition field="destination_number" expression="(.*)">
> <action application="bridge" data="sofia/gateway/FS2/$1"/>
> </condition>
> </extension>
>
>
>
> or without creating any gateways just bridge to FS2 IP address...
>
> <extension name="To_FS2">
> <condition field="destination_number" expression="(.*)">
> <action application="bridge" data="sofia/profilename/$1 at FS2_IP"/>
> </condition>
> </extension>
>
>
>
>
>
> On Thu, May 26, 2011 at 12:06 PM, William Alianto <william at xofap.com 
> <mailto:william at xofap.com>> wrote:
>
>     I'm trying to do an outbound scenario as following :
>
>     Client --> FS1 --> FS2 --> SBC
>
>     It maybe more simple if I add the SBC as gateway to FS1, but it only
>     accept IP from FS2 due to IP restriction from provider. Is there any
>     possible solution for this scenario?
>
>     _______________________________________________
>     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,

William

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From lzwierko at gmail.com  Fri May 27 14:33:56 2011
From: lzwierko at gmail.com (=?ISO-8859-2?Q?=A3ukasz_Zwierko?=)
Date: Fri, 27 May 2011 12:33:56 +0200
Subject: [Freeswitch-users] Codec selection for incoming call
In-Reply-To: <BANLkTimGUh7r3TC=m-bmsU1WxEHNmenLOA@mail.gmail.com>
References: <BANLkTinFPNQb_jm88C_hFa5a653uXVwHNg@mail.gmail.com>
	<BANLkTimGUh7r3TC=m-bmsU1WxEHNmenLOA@mail.gmail.com>
Message-ID: <BANLkTikfkODs1G_qrGW6rs=2BzSyBw0_Jg@mail.gmail.com>

thanks man, worked like a charm, I was setting the 'codec_string' only but
it also requires

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

to work,

thanks,
Lukasz


2011/5/26 Vitalie Colosov <vetali100 at gmail.com>

> Probably late negotiation is what you need:
>
> http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requires_param.29
>
> Vitalie
>
>  2011/5/26 ?ukasz Zwierko <lzwierko at gmail.com>
>
>> Hi guys,
>>
>> Is is possible to manually select which codec will be used on the A-leg
>> (incoming call) when using outbound event socket to control the call?
>> My case is quite basic I guess: I want to answer the call and pass it to
>> an application which will record raw data (so I'm not bridging the call
>> anywhere).
>> It is important for me to be able to make a per-call selection of codec.
>> I've been digging through the wiki the whole morning, but I've only found
>> out how to force a codec on the outgoing call...
>>
>> thanks for any advice,
>>
>> Lukasz
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Fri May 27 14:37:29 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 27 May 2011 12:37:29 +0200
Subject: [Freeswitch-users] Bridging Gateway from another server
In-Reply-To: <4DDF6FFA.9060108@xofap.com>
References: <4DDE2633.2030002@xofap.com>
	<BANLkTinGsRnpx_g6V3-Lk1BZT2XwayvRXg@mail.gmail.com>
	<4DDF6FFA.9060108@xofap.com>
Message-ID: <BANLkTi=XnmTVO-DWqAdcCMY-gWCvrOZCOw@mail.gmail.com>

yap,

can u provide a siptrace (sofia profile <your profile> siptrace on) along
with debug logs on both FS ?

put them on pastebin.

T.

On Fri, May 27, 2011 at 11:33 AM, William Alianto <william at xofap.com> wrote:

>  I tried both scenario, but it return as "Incompatible Destination"
> instead. Do I missed some step?
>
>
> On 2:59, Tihomir Culjaga wrote:
>
> on FS1 machine define FS2 as gateway and bridge the call to it...
>
>
>
>  <extension name="To_FS2">
>   <condition field="destination_number" expression="(.*)">
>       <action application="bridge" data="sofia/gateway/FS2/$1"/>
>    </condition>
> </extension>
>
>
>
>  or without creating any gateways just bridge to FS2 IP address...
>
>  <extension name="To_FS2">
>   <condition field="destination_number" expression="(.*)">
>       <action application="bridge" data="sofia/profilename/$1 at FS2_IP"/>
>    </condition>
> </extension>
>
>
>
>
>
> On Thu, May 26, 2011 at 12:06 PM, William Alianto <william at xofap.com>wrote:
>
>> I'm trying to do an outbound scenario as following :
>>
>> Client --> FS1 --> FS2 --> SBC
>>
>> It maybe more simple if I add the SBC as gateway to FS1, but it only
>> accept IP from FS2 due to IP restriction from provider. Is there any
>> possible solution for this scenario?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cyril.zlachevsky at gmail.com  Fri May 27 14:46:37 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Fri, 27 May 2011 13:46:37 +0300
Subject: [Freeswitch-users] FreeSWITCH as SIP softswitch
Message-ID: <BANLkTi=xCoT_QzhZAphd=gt-+0Hy4nd7gA@mail.gmail.com>

Can't understand you Answer, Ivan!
You mean I have to replace
<action application="bridge" data="sofia/gateway/Asotel/$1"/>
to
<action application="bridge" data="sofia/directory/Asotel/$1"/>
in conf/dialplan/public/test.xml file?


2011/5/27  <freeswitch-users-request at lists.freeswitch.org>:
> From:?"???? ????????" <zetruger at gmail.com>
> Use directory for SIP clients (hardphone, softphone) instead gateways.
> Use gateway just for SIP providers (SIP trunks).
>
> Simple test (run from FS console):
> originate sofia/internal/outboundtest at x.x.x.x:p
> &bridge(sofia/internal/inboundtest at x.x.x.x:p)
>
> 2011/5/27 Cyril Zlachevsky <cyril.zlachevsky at gmail.com>:
>> Hello,
>> I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server.
>> But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help.


From zetruger at gmail.com  Fri May 27 15:31:14 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Fri, 27 May 2011 15:31:14 +0400
Subject: [Freeswitch-users] FreeSWITCH as SIP softswitch
In-Reply-To: <BANLkTi=xCoT_QzhZAphd=gt-+0Hy4nd7gA@mail.gmail.com>
References: <BANLkTi=xCoT_QzhZAphd=gt-+0Hy4nd7gA@mail.gmail.com>
Message-ID: <BANLkTim2iageGDngBpCj9LZ+oy1Hb0e0Tw@mail.gmail.com>

<action application="bridge"
data="sofia/<SIP_PROFILE_NAME>/inboundtest at x.x.x.x:p"/>
You must reconfigure FreeSWITCH. Create user directory xml files.

27 ??? 2011??. 14:46 ???????????? Cyril Zlachevsky
<cyril.zlachevsky at gmail.com> ???????:
> Can't understand you Answer, Ivan!
> You mean I have to replace
> <action application="bridge" data="sofia/gateway/Asotel/$1"/>
> to
> <action application="bridge" data="sofia/directory/Asotel/$1"/>
> in conf/dialplan/public/test.xml file?
>
>
> 2011/5/27 ?<freeswitch-users-request at lists.freeswitch.org>:
>> From:?"???? ????????" <zetruger at gmail.com>
>> Use directory for SIP clients (hardphone, softphone) instead gateways.
>> Use gateway just for SIP providers (SIP trunks).
>>
>> Simple test (run from FS console):
>> originate sofia/internal/outboundtest at x.x.x.x:p
>> &bridge(sofia/internal/inboundtest at x.x.x.x:p)
>>
>> 2011/5/27 Cyril Zlachevsky <cyril.zlachevsky at gmail.com>:
>>> Hello,
>>> I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server.
>>> But all my today attempts to configure FreeSWITCH as softswitch failed and I really need 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
>


From steveayre at gmail.com  Fri May 27 15:52:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 27 May 2011 12:52:27 +0100
Subject: [Freeswitch-users] Bridging Gateway from another server
In-Reply-To: <4DDF6FFA.9060108@xofap.com>
References: <4DDE2633.2030002@xofap.com>
	<BANLkTinGsRnpx_g6V3-Lk1BZT2XwayvRXg@mail.gmail.com>
	<4DDF6FFA.9060108@xofap.com>
Message-ID: <BANLkTikrR9O3sh23x9RC5D6FcpfJvwE4JQ@mail.gmail.com>

That means a codec negotiation problem.

Remember you'll need to configure your dialplan on FS2 so the calls from FS1
go to the gateway.

-Steve


On 27 May 2011 10:33, William Alianto <william at xofap.com> wrote:

>  I tried both scenario, but it return as "Incompatible Destination"
> instead. Do I missed some step?
>
>
> On 2:59, Tihomir Culjaga wrote:
>
> on FS1 machine define FS2 as gateway and bridge the call to it...
>
>
>
>  <extension name="To_FS2">
>   <condition field="destination_number" expression="(.*)">
>       <action application="bridge" data="sofia/gateway/FS2/$1"/>
>    </condition>
> </extension>
>
>
>
>  or without creating any gateways just bridge to FS2 IP address...
>
>  <extension name="To_FS2">
>   <condition field="destination_number" expression="(.*)">
>       <action application="bridge" data="sofia/profilename/$1 at FS2_IP"/>
>    </condition>
> </extension>
>
>
>
>
>
> On Thu, May 26, 2011 at 12:06 PM, William Alianto <william at xofap.com>wrote:
>
>> I'm trying to do an outbound scenario as following :
>>
>> Client --> FS1 --> FS2 --> SBC
>>
>> It maybe more simple if I add the SBC as gateway to FS1, but it only
>> accept IP from FS2 due to IP restriction from provider. Is there any
>> possible solution for this scenario?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 at slickdeals.endjunk.com  Fri May 27 16:22:07 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 27 May 2011 05:22:07 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH as SIP softswitch
In-Reply-To: <BANLkTi=xCoT_QzhZAphd=gt-+0Hy4nd7gA@mail.gmail.com>
References: <4DDEDC31.40601@gmail.com>
	<BANLkTi=xCoT_QzhZAphd=gt-+0Hy4nd7gA@mail.gmail.com>
Message-ID: <1306498927596-6410870.post@n2.nabble.com>


Cyril Zlachevsky wrote:
> 
> Can't understand you Answer, Ivan!
There are several pre-configured extensions under the conf/directory/default
directory on your FS machine. Its default password is set on conf/vars.xml
file. You can configure your X-Lite softphone and/or any ATA device to
register to your FS using one of the extensions.

-----
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-as-SIP-softswitch-tp6409421p6410870.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Fri May 27 16:28:19 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 27 May 2011 05:28:19 -0700 (PDT)
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
Message-ID: <1306499299750-6410887.post@n2.nabble.com>


Yungwei Chen wrote:
> 
> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
After switching to using bridge function, does this also happen when you
make an outbound call to your cell phone using your javascript?

-----
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/Transferring-to-a-cell-phone-tp6408246p6410887.html
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From adam.kelloway at newpace.ca  Fri May 27 17:38:37 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Fri, 27 May 2011 10:38:37 -0300
Subject: [Freeswitch-users] event inconsistency
In-Reply-To: <BANLkTinPK1zSv4NJSxEgN8oTfi2+OeJNrw@mail.gmail.com>
References: <4DDE7368.8060904@newpace.ca>	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6301@cooper>	<4DDEA14C.4070000@newpace.ca>
	<BANLkTinPK1zSv4NJSxEgN8oTfi2+OeJNrw@mail.gmail.com>
Message-ID: <4DDFA95D.9020200@newpace.ca>

Thank you for that suggestion,

I am running the latest git FS (from two days ago) with Ubuntu 10.10. My 
event-listening app uses ESL (the org.freeswitch.esl java library) to 
listen for the events. I see the same behaviour when I run the app on 
the local host or on a separate machine. Currently, I am doing the 
latter. Running tcpdump on the FS host shows that all traffic to the 
port that my app is listening on is reliably received.

Listening for the event from fs_cli shows that event does get fired 
consistently, but tcpdump does not show it as being sent over the 
socket. Again, the behaviour is not consistent. Sometimes I do see the 
event sent out (both in tcpdump and in my app) and sometimes I don't. 
There doesn't appear to be any differences in the FS logs.

Is the event notification to the fs_cli done in the same way as an esl 
notification?

Thanks,

Adam

On 3:59 PM, Anthony Minessale wrote:
> how are you consuming events? are you using ESL?
>
> I recommend using fs_cli as a control to rule out faulty client logic.
>
> start fs_cli  and issue the events command from there
>
> /events channel_hangup
> /log 0
>
> then watch for the events there.
>
> P.S. you did not mention your platform or any other pertinent details
> about your setup.
>
>
>
> On Thu, May 26, 2011 at 1:51 PM, Adam Kelloway<adam.kelloway at newpace.ca>  wrote:
>> I got the latest git yesterday from scratch just to make sure. Is there
>> any way I can see debug output of the send operations?
>>
>> I didn't notice this behaviour when I was running the 1.0.6 tar, it is
>> only since I started using git that I saw it.
>>
>> On 3:59 PM, Peter Olsson wrote:
>>> Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.
>>>
>>> Besides that I've never seen any problems.
>>>
>>> /Peter
>>> ________________________________________
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Adam Kelloway [adam.kelloway at newpace.ca]
>>> Skickat: den 26 maj 2011 17:36
>>> Till: FreeSWITCH Users Help
>>> ?mne: [Freeswitch-users] event inconsistency
>>>
>>> Hi there,
>>>
>>> Has anyone ever had problems with events not being sent out over socket
>>> connections reliably? I am noticing that sometimes event notifications
>>> are sent, and sometimes they are not. For example, I have subscribed to
>>> CHANNEL_HANGUP, but depending on the timing of when I hangup the call
>>> (or when the dial plan hangs up), the event is not received. I confirmed
>>> with tcpdump that it is not even sent over the socket.
>>>
>>> Has anyone else seen his? What might be the problem?
>>>
>>> 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
>>>
>>> !DSPAM:4dde73da32761960016355!
>>>
>>>
>>>
>>
>> --
>> Adam Kelloway
>> Software Engineer
>> NewPace Technology Development Inc.
>> adam.kelloway at newpace.ca
>> +1 902-406-8375 x1031
>> www.newpace.ca
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
Adam Kelloway
Software Engineer
NewPace Technology Development Inc.
adam.kelloway at newpace.ca
+1 902-406-8375 x1031
www.newpace.ca



From peter.olsson at visionutveckling.se  Fri May 27 17:55:15 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 27 May 2011 15:55:15 +0200
Subject: [Freeswitch-users] event inconsistency
In-Reply-To: <4DDFA95D.9020200@newpace.ca>
References: <4DDE7368.8060904@newpace.ca>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6301@cooper>
	<4DDEA14C.4070000@newpace.ca>
	<BANLkTinPK1zSv4NJSxEgN8oTfi2+OeJNrw@mail.gmail.com>
	<4DDFA95D.9020200@newpace.ca>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE5489E5@cooper>

Yes, fs_cli uses the esl lib as well, so it's done in the exact same way. Default connection is through 127.0.0.1 though, so I'm not sure you're dumping these packets in tcpdump? Anyway, anything sent to fs_cli will definately be sent using the socket.

I have an IVR application using ESL, with about 60-120 concurrent calls, and I've never seen this issue, so there must be something else going on there.

I've never tried the Java libs, and I don't really know if everything there is up-to-date in that lib, but I guess it it should...

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Adam Kelloway
Skickat: den 27 maj 2011 15:39
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] event inconsistency

Thank you for that suggestion,

I am running the latest git FS (from two days ago) with Ubuntu 10.10. My 
event-listening app uses ESL (the org.freeswitch.esl java library) to 
listen for the events. I see the same behaviour when I run the app on 
the local host or on a separate machine. Currently, I am doing the 
latter. Running tcpdump on the FS host shows that all traffic to the 
port that my app is listening on is reliably received.

Listening for the event from fs_cli shows that event does get fired 
consistently, but tcpdump does not show it as being sent over the 
socket. Again, the behaviour is not consistent. Sometimes I do see the 
event sent out (both in tcpdump and in my app) and sometimes I don't. 
There doesn't appear to be any differences in the FS logs.

Is the event notification to the fs_cli done in the same way as an esl 
notification?

Thanks,

Adam

On 3:59 PM, Anthony Minessale wrote:
> how are you consuming events? are you using ESL?
>
> I recommend using fs_cli as a control to rule out faulty client logic.
>
> start fs_cli  and issue the events command from there
>
> /events channel_hangup
> /log 0
>
> then watch for the events there.
>
> P.S. you did not mention your platform or any other pertinent details
> about your setup.
>
>
>
> On Thu, May 26, 2011 at 1:51 PM, Adam Kelloway<adam.kelloway at newpace.ca>  wrote:
>> I got the latest git yesterday from scratch just to make sure. Is there
>> any way I can see debug output of the send operations?
>>
>> I didn't notice this behaviour when I was running the 1.0.6 tar, it is
>> only since I started using git that I saw it.
>>
>> On 3:59 PM, Peter Olsson wrote:
>>> Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.
>>>
>>> Besides that I've never seen any problems.
>>>
>>> /Peter
>>> ________________________________________
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Adam Kelloway [adam.kelloway at newpace.ca]
>>> Skickat: den 26 maj 2011 17:36
>>> Till: FreeSWITCH Users Help
>>> ?mne: [Freeswitch-users] event inconsistency
>>>
>>> Hi there,
>>>
>>> Has anyone ever had problems with events not being sent out over socket
>>> connections reliably? I am noticing that sometimes event notifications
>>> are sent, and sometimes they are not. For example, I have subscribed to
>>> CHANNEL_HANGUP, but depending on the timing of when I hangup the call
>>> (or when the dial plan hangs up), the event is not received. I confirmed
>>> with tcpdump that it is not even sent over the socket.
>>>
>>> Has anyone else seen his? What might be the problem?
>>>
>>> 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
>>>
>>> 
>>>
>>>
>>>
>>
>> --
>> Adam Kelloway
>> Software Engineer
>> NewPace Technology Development Inc.
>> adam.kelloway at newpace.ca
>> +1 902-406-8375 x1031
>> www.newpace.ca
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
Adam Kelloway
Software Engineer
NewPace Technology Development Inc.
adam.kelloway at newpace.ca
+1 902-406-8375 x1031
www.newpace.ca


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

!DSPAM:4ddfaa9f32761727713633!



From yungwei at resolvity.com  Fri May 27 18:41:36 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 27 May 2011 10:41:36 -0400
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <1306499299750-6410887.post@n2.nabble.com>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>

No. 

I just verified that making an outbound call to my cell phone still works. 
I even recorded the session just to make sure, and in the recording I hear things both ways.
I first dial 9911 from my SIP client (behind freeswitch), and this leads to the javascript program below.
   session.answer();
   var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
   session.hangup(16); // disconnects the session between the SIP client and freeswitch
   new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
   while (new_session.ready())
   {
       new_session.streamFile("/path/to/local/wav/file");       
   }

Now back to the case I'm having problem with.
In this case, I first make a call from a landline to freeswitch through my sip provider, and then a javascript program takes over.
I want to transfer the call to a cell phone so that the landline and the cell phone can communicate with each other.
Here's the javascript program:
   session.answer();

   if (session.ready())
   {
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
   }

So what am I missing here?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
Sent: Friday, May 27, 2011 7:28 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Transferring to a cell phone


Yungwei Chen wrote:
> 
> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
After switching to using bridge function, does this also happen when you
make an outbound call to your cell phone using your javascript?

-----
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/Transferring-to-a-cell-phone-tp6408246p6410887.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 msc at freeswitch.org  Fri May 27 19:08:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 27 May 2011 08:08:10 -0700
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
Message-ID: <BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>

You need to put a debug log on pastebin.freeswitch.org. Be sure to use
"FreeSWITCH Log" for the syntax highlighting.
-MC

On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> No.
>
> I just verified that making an outbound call to my cell phone still works.
> I even recorded the session just to make sure, and in the recording I hear
> things both ways.
> I first dial 9911 from my SIP client (behind freeswitch), and this leads to
> the javascript program below.
>   session.answer();
>    var new_session = new
> Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>    session.hangup(16); // disconnects the session between the SIP client
> and freeswitch
>   new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
>   while (new_session.ready())
>   {
>       new_session.streamFile("/path/to/local/wav/file");
>   }
>
> Now back to the case I'm having problem with.
> In this case, I first make a call from a landline to freeswitch through my
> sip provider, and then a javascript program takes over.
> I want to transfer the call to a cell phone so that the landline and the
> cell phone can communicate with each other.
> Here's the javascript program:
>   session.answer();
>
>   if (session.ready())
>   {
>       session.execute("bridge",
> "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>   }
>
> So what am I missing here?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
> Sent: Friday, May 27, 2011 7:28 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>
>
> Yungwei Chen wrote:
> >
> > Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> > anything both ways. Any idea?
> After switching to using bridge function, does this also happen when you
> make an outbound call to your cell phone using your javascript?
>
> -----
> 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/Transferring-to-a-cell-phone-tp6408246p6410887.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 pooja27taneja at gmail.com  Fri May 27 12:47:29 2011
From: pooja27taneja at gmail.com (pooja)
Date: Fri, 27 May 2011 01:47:29 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
Message-ID: <1306486049569-6410226.post@n2.nabble.com>

has anyone used mod_mp4 to play a video.
if i use play_mp4 it says invalid application play_mp4.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6410226.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From clive18 at webmail.co.za  Fri May 27 13:22:08 2011
From: clive18 at webmail.co.za (clive engelberg)
Date: Fri, 27 May 2011 11:22:08 +0200
Subject: [Freeswitch-users] Virtual Migration Detected! - Crit errors in CLI
Message-ID: <8a53a73f30614e79903924e0bb411b6e@www.webmail.co.za>

Hi
I hope someone can help me with this issue.
It seems that there are clock slippages due to overworked CPU with heavy load.
I am getting these errors with 2 or 3 channels only.

On closer inspection, it seems that the server (dell 1950) is using only 1 CPU.
The interrupts show that the other CPU's are hardly being used. (see the
cat/proc/interrupts below).

my gut feeling is that the APIC is not working as it should. Would changing
irq's in the bios help this issue? I am running Centos 5.5

Any suggestions will be appreciated.
Thanks in advance
Clive

# cat /proc/interrupts
           CPU0       CPU1       CPU2       CPU3       CPU4       CPU5 CPU6    
  CPU7
  0:  782410882          0          0          0          0          0         
0          0    IO- APIC-edge  timer
  1:          2          0          0          0          0          0         
0          0    IO-APIC- edge  i8042
  8:          1          0          0          0          0          0         
0          0    IO-APIC- edge  rtc
  9:          0          0          0          0          0          0         
0          0   IO-APIC- level  acpi
12:          4          0          0          0          0          0         
0          0    IO- APIC-edge  i8042
14:        238        194          0          0    7005401      15210         
0          0 IO-APIC-edge  ide0
66:         23          0          0          0          0          0         
0          0   IO- APIC-level  ehci_hcd:usb1, uhci_hcd:usb2, uhci_hcd:usb4
74:          0          0          0          0          0          0         
0          0   IO-APIC- level  uhci_hcd:usb3
82:       5426     137633          0     147604    1635602          0         
0 0   IO-APIC-level  megasas
90:    6118176          0          0          0          0          0         
0          0 PCI-MSI  eth0
98:    3457897          0          0          0          0          0         
0          0 PCI-MSI  eth1
NMI:       6258       2735       3482       3453       3253       2513      
3379 2652
LOC:  782410253  782410193  782410118  782410043  782409968 782409893 
782409818  782409736
ERR:          0
MIS:          0

____________________________________________________________
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From william.nishio at voicetechnology.com.br  Fri May 27 19:12:55 2011
From: william.nishio at voicetechnology.com.br (William Kendi ...)
Date: Fri, 27 May 2011 12:12:55 -0300
Subject: [Freeswitch-users] mod_fsv, record and playback options
In-Reply-To: <AANLkTi=f-+Y8XqY6pwD=c4hD0VCLQXQHqib6yayZEjww@mail.gmail.com>
References: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
	<AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>
	<AANLkTim0Zm1GLpbU5q2dvsm=nguwuMr6xY4ZUQAMWGX0@mail.gmail.com>
	<AANLkTinAwkmV4R=RLrhASWkymL8Crb7MRn5aJWoTM1tY@mail.gmail.com>
	<AANLkTi=f-+Y8XqY6pwD=c4hD0VCLQXQHqib6yayZEjww@mail.gmail.com>
Message-ID: <BANLkTikOmQJNXgKZ64RXYySS+eCM1Hbttg@mail.gmail.com>

I did it again!

My old mod_fsv customization has been reported to be broken in the lastest
FreeSWITCH builds. But after more meddlings I managed to get the module
working again.

My FSV decoder for FFMPEG got updated too.

Thanks in advance.

2011/2/21 William Kendi ... <william.nishio at voicetechnology.com.br>

> I did it!
>
> After some tears and sweats, I finally managed to create a working FSV
> demuxer module for the FFMPEG project!
>
> With this module, files in the FSV format now can be converted to any other
> format through the FFMPEG project!
>
> To install:
>
> 1). Put the "fsvdec.c" file in the "libavformat" directory.
> 2). Insert the line "REGISTER_DEMUXER  (FSV, fsv);" in the file
> "allformats.c" also in the "libavformat" directory.
> 3). Insert the line "OBJS-$(CONFIG_FSV_DEMUXER)               += fsvdec.o"
> in the file "Makefile" also in the "libavformat" directory.
> 4). Build the FFMPEG project using "make install".
>
> Now the FSV format seems to be more usable and I am trying to figure how to
> use the FreeSWITCH JIRA.
>
>
> 2011/1/21 Anthony Minessale <anthony.minessale at gmail.com>
>
>> Sure, send it to Jira and we'll get it in.
>> Though, I'm surprised you would not want to use the mod_mp4 now that
>> it exists =D  the FSV was sort if a hack I made up on a whim.
>>
>>
>>
>> On Fri, Jan 21, 2011 at 5:17 PM, William Suffill
>> <william.suffill at gmail.com> wrote:
>> > Best to add the patches/details into Jira [http://jira.freeswitch.org]
>> so it
>> > can be tracked and reviewed for being added to the source tree.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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  Fri May 27 19:17:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 27 May 2011 08:17:51 -0700
Subject: [Freeswitch-users] mod_fsv, record and playback options
In-Reply-To: <BANLkTikOmQJNXgKZ64RXYySS+eCM1Hbttg@mail.gmail.com>
References: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
	<AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>
	<AANLkTim0Zm1GLpbU5q2dvsm=nguwuMr6xY4ZUQAMWGX0@mail.gmail.com>
	<AANLkTinAwkmV4R=RLrhASWkymL8Crb7MRn5aJWoTM1tY@mail.gmail.com>
	<AANLkTi=f-+Y8XqY6pwD=c4hD0VCLQXQHqib6yayZEjww@mail.gmail.com>
	<BANLkTikOmQJNXgKZ64RXYySS+eCM1Hbttg@mail.gmail.com>
Message-ID: <BANLkTi=jNMeSTO30bBRF0P38_ELB4y-H2g@mail.gmail.com>

Hi five! Do you need someone to drop this into the git repo?
-MC

On Fri, May 27, 2011 at 8:12 AM, William Kendi ... <
william.nishio at voicetechnology.com.br> wrote:

> I did it again!
>
> My old mod_fsv customization has been reported to be broken in the lastest
> FreeSWITCH builds. But after more meddlings I managed to get the module
> working again.
>
> My FSV decoder for FFMPEG got updated too.
>
> Thanks in advance.
>
> 2011/2/21 William Kendi ... <william.nishio at voicetechnology.com.br>
>
>> I did it!
>>
>> After some tears and sweats, I finally managed to create a working FSV
>> demuxer module for the FFMPEG project!
>>
>> With this module, files in the FSV format now can be converted to any
>> other format through the FFMPEG project!
>>
>> To install:
>>
>> 1). Put the "fsvdec.c" file in the "libavformat" directory.
>> 2). Insert the line "REGISTER_DEMUXER  (FSV, fsv);" in the file
>> "allformats.c" also in the "libavformat" directory.
>> 3). Insert the line "OBJS-$(CONFIG_FSV_DEMUXER)               += fsvdec.o"
>> in the file "Makefile" also in the "libavformat" directory.
>> 4). Build the FFMPEG project using "make install".
>>
>> Now the FSV format seems to be more usable and I am trying to figure how
>> to use the FreeSWITCH JIRA.
>>
>>
>> 2011/1/21 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> Sure, send it to Jira and we'll get it in.
>>> Though, I'm surprised you would not want to use the mod_mp4 now that
>>> it exists =D  the FSV was sort if a hack I made up on a whim.
>>>
>>>
>>>
>>> On Fri, Jan 21, 2011 at 5:17 PM, William Suffill
>>> <william.suffill at gmail.com> wrote:
>>> > Best to add the patches/details into Jira [http://jira.freeswitch.org]
>>> so it
>>> > can be tracked and reviewed for being added to the source tree.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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 william.nishio at voicetechnology.com.br  Fri May 27 19:29:43 2011
From: william.nishio at voicetechnology.com.br (William Kendi ...)
Date: Fri, 27 May 2011 12:29:43 -0300
Subject: [Freeswitch-users] mod_fsv, record and playback options
In-Reply-To: <BANLkTi=jNMeSTO30bBRF0P38_ELB4y-H2g@mail.gmail.com>
References: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>
	<AANLkTi=e_n9T3NFaB=Y1BE6_NAtFhAAan7m96WoKOM6t@mail.gmail.com>
	<AANLkTim0Zm1GLpbU5q2dvsm=nguwuMr6xY4ZUQAMWGX0@mail.gmail.com>
	<AANLkTinAwkmV4R=RLrhASWkymL8Crb7MRn5aJWoTM1tY@mail.gmail.com>
	<AANLkTi=f-+Y8XqY6pwD=c4hD0VCLQXQHqib6yayZEjww@mail.gmail.com>
	<BANLkTikOmQJNXgKZ64RXYySS+eCM1Hbttg@mail.gmail.com>
	<BANLkTi=jNMeSTO30bBRF0P38_ELB4y-H2g@mail.gmail.com>
Message-ID: <BANLkTimYOLsEXxbYLvx43=++ZcqidWyF3A@mail.gmail.com>

That should be good.

As Minessale asked, I also created a issue in the FreeSWITCH JIRA.

2011/5/27 Michael Collins <msc at freeswitch.org>

> Hi five! Do you need someone to drop this into the git repo?
> -MC
>
>
> On Fri, May 27, 2011 at 8:12 AM, William Kendi ... <
> william.nishio at voicetechnology.com.br> wrote:
>
>> I did it again!
>>
>> My old mod_fsv customization has been reported to be broken in the lastest
>> FreeSWITCH builds. But after more meddlings I managed to get the module
>> working again.
>>
>> My FSV decoder for FFMPEG got updated too.
>>
>> Thanks in advance.
>>
>> 2011/2/21 William Kendi ... <william.nishio at voicetechnology.com.br>
>>
>>> I did it!
>>>
>>> After some tears and sweats, I finally managed to create a working FSV
>>> demuxer module for the FFMPEG project!
>>>
>>> With this module, files in the FSV format now can be converted to any
>>> other format through the FFMPEG project!
>>>
>>> To install:
>>>
>>> 1). Put the "fsvdec.c" file in the "libavformat" directory.
>>> 2). Insert the line "REGISTER_DEMUXER  (FSV, fsv);" in the file
>>> "allformats.c" also in the "libavformat" directory.
>>> 3). Insert the line "OBJS-$(CONFIG_FSV_DEMUXER)               +=
>>> fsvdec.o" in the file "Makefile" also in the "libavformat" directory.
>>> 4). Build the FFMPEG project using "make install".
>>>
>>> Now the FSV format seems to be more usable and I am trying to figure how
>>> to use the FreeSWITCH JIRA.
>>>
>>>
>>> 2011/1/21 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> Sure, send it to Jira and we'll get it in.
>>>> Though, I'm surprised you would not want to use the mod_mp4 now that
>>>> it exists =D  the FSV was sort if a hack I made up on a whim.
>>>>
>>>>
>>>>
>>>> On Fri, Jan 21, 2011 at 5:17 PM, William Suffill
>>>> <william.suffill at gmail.com> wrote:
>>>> > Best to add the patches/details into Jira [http://jira.freeswitch.org]
>>>> so it
>>>> > can be tracked and reviewed for being added to the source tree.
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> 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 cmrienzo at gmail.com  Fri May 27 19:55:50 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 27 May 2011 11:55:50 -0400
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <1306486049569-6410226.post@n2.nabble.com>
References: <1306486049569-6410226.post@n2.nabble.com>
Message-ID: <BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>

***Disclaimer: I've never used this module...

I think the module isn't loaded.  Check if
/usr/local/freeswitch/mod/mod_mp4* is in installed.  Also make sure it's
loaded in /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml.

If so, then check the trace in /usr/local/freeswitch/log/ to see why it
didn't load.

If it's not installed, then install it:
1. Edit modules.conf and add applications/mod_mp4 to it.
2. make
3. make install
4. Edit conf/autoload_configs/modules.conf.xml and add this line:
   <load module="mod_mp4"/>



On Fri, May 27, 2011 at 4:47 AM, pooja <pooja27taneja at gmail.com> wrote:

> has anyone used mod_mp4 to play a video.
> if i use play_mp4 it says invalid application play_mp4.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6410226.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 yungwei at resolvity.com  Fri May 27 20:34:13 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 27 May 2011 12:34:13 -0400
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
	<BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>

Here's the debug log, http://pastebin.freeswitch.org/16398

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, May 27, 2011 10:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

You need to put a debug log on pastebin.freeswitch.org<http://pastebin.freeswitch.org>. Be sure to use "FreeSWITCH Log" for the syntax highlighting.
-MC
On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:
No.

I just verified that making an outbound call to my cell phone still works.
I even recorded the session just to make sure, and in the recording I hear things both ways.
I first dial 9911 from my SIP client (behind freeswitch), and this leads to the javascript program below.
  session.answer();
  var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  session.hangup(16); // disconnects the session between the SIP client and freeswitch
  new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
  while (new_session.ready())
  {
      new_session.streamFile("/path/to/local/wav/file");
  }

Now back to the case I'm having problem with.
In this case, I first make a call from a landline to freeswitch through my sip provider, and then a javascript program takes over.
I want to transfer the call to a cell phone so that the landline and the cell phone can communicate with each other.
Here's the javascript program:
  session.answer();

  if (session.ready())
  {
      session.execute("bridge", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  }

So what am I missing here?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of mazilo
Sent: Friday, May 27, 2011 7:28 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Transferring to a cell phone


Yungwei Chen wrote:
>
> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
After switching to using bridge function, does this also happen when you
make an outbound call to your cell phone using your javascript?

-----
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/Transferring-to-a-cell-phone-tp6408246p6410887.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

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From yudha2008 at gmail.com  Fri May 27 20:42:41 2011
From: yudha2008 at gmail.com (baskar)
Date: Fri, 27 May 2011 09:42:41 -0700 (PDT)
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
	<BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
	<BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>
	<BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>
Message-ID: <1306514561494-6411726.post@n2.nabble.com>

Hi Mike,

Yes i dialed in from the outside, got to the IVR, and then dialed extension
is 5107.  

Yes i have two endpoint in my bridge line. One is keying extension and other
is default extension. 

But this dialplan send by David what i tried. 
<entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
param="execute_extension
set:continue_on_fail=BUSY,bridge:sofia/internal/$1%XX.XX.XX.XX,bridge:sofia/internal/5998%XX.XX.XX.XX
"/>

My actual conduction is i dialed in from the outside, got to the ivr, where
after keying in the extension number the call gets transferred to the
appropriate extension and if the extension is busy it should be hunted to a
default extension. 

--
Thanks with Regards,
N.Baskar


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inbound-dialplan-tp6404380p6411726.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri May 27 20:43:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 27 May 2011 09:43:53 -0700
Subject: [Freeswitch-users] matching word
In-Reply-To: <BANLkTi=KtPFKMSPjd0D-iLpPT4yXsOxD=g@mail.gmail.com>
References: <BANLkTinfe5rvkz_qQDDPT617pjcLsTswtw@mail.gmail.com>
	<1306405611055-6406420.post@n2.nabble.com>
	<BANLkTimBXuivtsVJSrX0Ck_STsQvZD2ozA@mail.gmail.com>
	<BANLkTinai5FLezOrHGSv2kAU7wbUKNVU3Q@mail.gmail.com>
	<BANLkTi=KtPFKMSPjd0D-iLpPT4yXsOxD=g@mail.gmail.com>
Message-ID: <BANLkTi=Y-LX8tmZQ1k2UtN9AfY2SpUYA5A@mail.gmail.com>

On Thu, May 26, 2011 at 11:41 PM, Sam <u2nsam at gmail.com> wrote:

> Is my dialplan correct ?
>

I suppose that depends on what you're trying to do. I'm assuming you want to
send the call into the voicemail box for "sam". If so then your voicemail
action should be something like this:

 <action application="voicemail" data="default ${domain} $1"/>

(Note that you don't need $$ for any channel variables; a single $ will
suffice.)

HTH,
MC


>
> <extension name="public_extensions">
>       <condition field="destination_number" expression="^(sam)$">
>       <action application="info"/>
>     <action application="voicemail" data="default $${domain}
> ${sip_to_host}"/>
>       </condition>
>     </extension>
>
>
> I want to match three any alphanumeric 3 digit/letters
>
> Also i wan to store the voicemail into the folders where the herachy would
> be      :::          Domain --> Username --> Voicemail file
>
> Regards
> Sam
>
>
>
>
> On Fri, May 27, 2011 at 12:19 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Your regex is only going to match a single alphanumeric character. Are you
>> trying to match an actual email address or what? Once you know what you hope
>> to match it will be easier to know which regex to use. For the record a
>> semi-proper email grabbing regex is like this:
>>
>> ^([a-zA-Z0-9._%+-]+@[a-zA-Z0-9.-]+\.[a-zA-Z]{2,6})$
>>
>> Matching emails is an interesting exercise. If you're not all into "making
>> it perfect" and just want to capture "foo at bar" then do something like
>> this:
>>
>> ^([^@]+ at .*)$
>>
>> -MC
>>
>>
>> On Thu, May 26, 2011 at 10:02 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Thanks for that,
>>>
>>> I want to use that when someone with username sam at gmail.com will be
>>> routed to a voicemail dialplan, with domain partitioning.
>>>
>>> here still i am not able to achiveve it with domain.
>>>
>>> <domain name="gmail.com">
>>>
>>>     <extension name="public_extensions">
>>>       <condition field="destination_number" expression="^(\w)$">
>>>           <action application="info"/>
>>>         <action application="voicemail" data="default $${domain_name}
>>> $1"/>
>>>       </condition>
>>>     </extension>
>>>
>>> </domain>
>>>
>>> here above gmail.com is examplary .
>>>
>>> when there is an invite coming from different server to FS with
>>> sam at gmail.com and it is routed directly on voicemail context to route
>>> via public.xml
>>>
>>> here first the call is not recognized to the context gmail.com but when
>>> i remove that it goes to voicemail directly properly and stores the
>>> voicemail.
>>>
>>> but the voicemail is not stored under domain name gmail.com which should
>>> store it ideally under domain name as folder, which it is just storing it
>>> under user sam folder.
>>>
>>> This is what I want to achieve.
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>>
>>>
>>> On Thu, May 26, 2011 at 3:56 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:
>>>
>>>>
>>>> samir wrote:
>>>> >
>>>> > Friends,
>>>> >
>>>> > How can i match words in expressions,
>>>> >
>>>> > is it by expression="^(\y\w+\y)$"  or could be something else.
>>>> >
>>>> > Regards
>>>> > Sam
>>>> I understand ^(\w)$ will match any word, but not with \y. Perhaps, if
>>>> you
>>>> tell us exactly what you want to accomplish, someone will chime in for a
>>>> better solution.
>>>>
>>>> -----
>>>> 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/matching-word-tp6406170p6406420.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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yudha2008 at gmail.com  Fri May 27 20:45:43 2011
From: yudha2008 at gmail.com (Baskar)
Date: Fri, 27 May 2011 12:45:43 -0400
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
	<BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
	<BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>
	<BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>
Message-ID: <BANLkTimPiU-ndstQS3MFuWLheEyR6_g4GA@mail.gmail.com>

Hi Mike,

Yes i dialed in from the outside, got to the IVR, and then dialed extension
is 5107.

Yes i have two endpoint in my bridge line. One is keying extension and other
is default extension.

But this dialplan send by David what i tried.
<entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
param="execute_extension
set:continue_on_fail=BUSY,bridge:sofia/internal/$1%XX.XX.XX.XX,bridge:sofia/internal/5998%XX.XX.XX.XX
"/>

My actual conduction is i dialed in from the outside, got to the ivr, where
after keying in the extension number the call gets transferred to the
appropriate extension and if the extension is busy it should be hunted to a
default extension.

-- 
Thanks with Regards,
N.Baskar

On Thu, May 26, 2011 at 9:33 PM, Michael Collins <msc at freeswitch.org> wrote:

> Okay, walk us through what you did to get here. You dialed in from the
> outside, got to the IVR, and then dialed 5107? Please confirm. Also, I see
> that you have two endpoints in your bridge line. Are you deliberately trying
> to dial two separate endpoints at the same time?
>
> -MC
>
>
> On Thu, May 26, 2011 at 3:45 PM, Baskar <yudha2008 at gmail.com> wrote:
>
>> HI Mike,
>>
>> I have pasted my log in path http://pastebin.freeswitch.org/16388
>>
>> <http://pastebin.freeswitch.org/16388>--
>> Thanks with Regards,
>> N.Baskar
>>
>> On Thu, May 26, 2011 at 6:33 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Pastebin the debug output of a call to a busy phone.
>>> -MC
>>>
>>>
>>> On Thu, May 26, 2011 at 3:02 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>
>>>> HI Mike,
>>>>
>>>> I am looking at a condition where if the extension that i enter is busy
>>>> then it should automatically route to the extension i specify in the
>>>> ivr_inbound xml.
>>>>
>>>> --
>>>>
>>>> Thanks with Regards,
>>>> N.Baskar
>>>>
>>>> On Thu, May 26, 2011 at 5:43 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> That's because you are telling it to go to the default context:
>>>>>     <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>>> param="transfer $1 XML default"/>
>>>>> Change that "default" to whatever context you want the destination
>>>>> number to be in...
>>>>>
>>>>> -MC
>>>>>
>>>>> On Thu, May 26, 2011 at 2:34 PM, Baskar <yudha2008 at gmail.com> wrote:
>>>>>
>>>>>> Hi David,
>>>>>>
>>>>>> I tried the dial plan suggested by you several times but the moment i
>>>>>> key in the extension it take my input but exits the ivr_inbound.xml and goes
>>>>>> to default.xml. It does not bridge me to the extension i key in but instead
>>>>>> comes out of the xml and connects me to the extension in the default.xml.
>>>>>> Below is the output log of the tests that i have made.
>>>>>>
>>>>>>
>>>>>> Output Log:
>>>>>>
>>>>>> 2011-05-26 17:24:59.065884 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>>> 5:840
>>>>>> 2011-05-26 17:24:59.341669 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>>> 1:720
>>>>>> 2011-05-26 17:24:59.745697 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>>> 0:960
>>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF
>>>>>> 7:1120
>>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:376 digits '5107'
>>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:470 action regex
>>>>>> [5107] [/^([1-6][0-9][0-9][0-9])$/] [2]
>>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:488 IVR action on
>>>>>> menu 'ivr_inbound' matched '5107' param
>>>>>> 'execute_extensionset:continue_on_fail=true,bridge
>>>>>> sofia/internal/5107%XX.XX.XX.XX,bridge sofia/internal/5998%XX.XX.XX.XX'
>>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:492
>>>>>> switch_ivr_menu_execute todo=[2]
>>>>>> 2011-05-26 17:25:00.041720 [DEBUG] switch_ivr_menu.c:580 exit-sound
>>>>>> 'voicemail/vm-goodbye.wav'
>>>>>>
>>>>>> --
>>>>>>
>>>>>> Thanks with Regards,
>>>>>> N.Baskar
>>>>>>
>>>>>> On Thu, May 26, 2011 at 1:49 AM, David Ponzone <
>>>>>> david.ponzone at ipeva.fr> wrote:
>>>>>>
>>>>>>> Baskar,
>>>>>>>
>>>>>>> If you want this behavior only for calls only coming through the
>>>>>>> IVR, I think you will have to use bridge rather than transfer.
>>>>>>> it would look like this:
>>>>>>>
>>>>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>>>>> param="execute_extension
>>>>>>> set:continue_on_fail=BUSY,bridge:sofia/profile/$1%domain,bridge:sofia/profile/1003%domain"/>
>>>>>>>
>>>>>>> or something like it :)
>>>>>>>
>>>>>>>  David Ponzone  Direction Technique
>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>> tel:      01 74 03 18 97
>>>>>>> gsm:   06 66 98 76 34
>>>>>>>
>>>>>>> Service Client IPeva
>>>>>>> tel:      0811 46 26 26
>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>>
>>>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et
>>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>>>>>> susceptible d'alt?ration. **IPeva** d?cline toute responsabilit? au
>>>>>>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
>>>>>>> pas destinataire de ce message, merci de le d?truire imm?diatement et
>>>>>>> d'avertir l'exp?diteur.*
>>>>>>> *
>>>>>>> *
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Le 26/05/2011 ? 00:27, Baskar a ?crit :
>>>>>>>
>>>>>>> Hi Michael,
>>>>>>>
>>>>>>> Thanks for your quick reply. Below is the code for the dial plan
>>>>>>> inbound ivr. Can you please specify where i should be inserting the
>>>>>>> continue_on_fail=true line in the code.
>>>>>>>
>>>>>>> Default.xml
>>>>>>>
>>>>>>>     <extension name="inbound">
>>>>>>>       <condition field="destination_number"
>>>>>>> expression="^(XXXXXXXXXX)$" />
>>>>>>>        <condition wday="2-6" hour="8-17">
>>>>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>         <action application="set" data="accountcode=Inbound"/>
>>>>>>>         <action application="answer"/>
>>>>>>>         <action application="start_dtmf" />
>>>>>>>         <action application="ivr" data="ivr_inbound"/>
>>>>>>>         <action application="record_session"
>>>>>>> data="$${base_dir}/inbound/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.gsm"/>
>>>>>>>         <action application="transfer" data="1018 XML default"/>
>>>>>>>       </condition>
>>>>>>>     </extension>
>>>>>>>
>>>>>>> ivr_inbound.xml
>>>>>>>
>>>>>>> <menu name="ivr_inbound"
>>>>>>>
>>>>>>>  greet-long="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/abc_pe/abc_pe_bus.wav"
>>>>>>>      invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>>>>>      exit-sound="voicemail/vm-goodbye.wav"
>>>>>>>      confirm-macro=""
>>>>>>>      confirm-key=""
>>>>>>>      tts-engine="flite"
>>>>>>>      tts-voice="rms"
>>>>>>>      confirm-attempts="3"
>>>>>>>      timeout="10000"
>>>>>>>      inter-digit-timeout="2000"
>>>>>>>      max-failures="3"
>>>>>>>      max-timeouts="1"
>>>>>>>      digit-len="4">
>>>>>>>    <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
>>>>>>> param="transfer $1 XML default"/>
>>>>>>>    <entry digits="0" action="menu-exec-app" param="transfer 6525 XML
>>>>>>> default"/>
>>>>>>> </menu>
>>>>>>>
>>>>>>> *--
>>>>>>> Thanks with Regards,
>>>>>>>
>>>>>>> N.Baskar
>>>>>>>
>>>>>>> *
>>>>>>> On Wed, May 25, 2011 at 5:25 PM, Michael Collins <msc at freeswitch.org
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> If I understand correctly you're wanting to handle a bridge scenario
>>>>>>>> where the target extension is busy, etc. Most likely you just need to set
>>>>>>>> continue_on_fail=true so that your dialplan continues in the case of the
>>>>>>>> target extension not picking up.
>>>>>>>>
>>>>>>>> -MC
>>>>>>>>
>>>>>>>> On Wed, May 25, 2011 at 1:27 PM, Baskar <yudha2008 at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Hi All,,
>>>>>>>>>
>>>>>>>>> In my inbound dial plan xml file  i have set two conditions
>>>>>>>>> enabled
>>>>>>>>>
>>>>>>>>> Condition1:
>>>>>>>>>
>>>>>>>>> In inbound dial plan callers are given an option to key in the
>>>>>>>>> extension number and reach the appropriate extension (Example: 1001 or 1002
>>>>>>>>> or 1003 etc).
>>>>>>>>>
>>>>>>>>> Condition2:
>>>>>>>>>
>>>>>>>>> The second condition routes call to a default extension in
>>>>>>>>> scenarios where the caller does not specify any extension number (Example:
>>>>>>>>> Default extension is 1007).
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Both the above conditions are working fine.
>>>>>>>>>
>>>>>>>>> Now I need to set up another condition where after keying in the
>>>>>>>>> extension number the call gets transferred to the appropriate extension and
>>>>>>>>> if the extension is busy(Example: say extension 1003 is busy) it should be
>>>>>>>>> hunted to a default extension (example: 1007). How can we set up this
>>>>>>>>> condition in dial plan?
>>>>>>>>>
>>>>>>>>> Can any one guide me.
>>>>>>>>> --
>>>>>>>>> Thanks with Regards,
>>>>>>>>>
>>>>>>>>> N.Baskar
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>  _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> -
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pooja27taneja at gmail.com  Fri May 27 22:28:51 2011
From: pooja27taneja at gmail.com (pooja)
Date: Fri, 27 May 2011 11:28:51 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
Message-ID: <1306520931787-6412076.post@n2.nabble.com>

I did as you said. now it doesnt say invalid application play_mp4. but
instead gives an error :
[ERR] mod_mp4.cpp:505 snookerpup_fhnom3b7.mp4:Missing audio/video track.

but the mp4 file is there in the location.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6412076.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri May 27 22:35:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 27 May 2011 11:35:22 -0700
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <1306514561494-6411726.post@n2.nabble.com>
References: <BANLkTin5yRuRGrNe7_GknZLj467873HCSQ@mail.gmail.com>
	<BANLkTikuXH+tMik4sMkgsv0DF+jWFhpkNA@mail.gmail.com>
	<BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
	<BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
	<BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>
	<BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>
	<1306514561494-6411726.post@n2.nabble.com>
Message-ID: <BANLkTinbdfaEHUzxoND6UCcYSSRCK_4cmQ@mail.gmail.com>

On Fri, May 27, 2011 at 9:42 AM, baskar <yudha2008 at gmail.com> wrote:

> Hi Mike,
>
> Yes i dialed in from the outside, got to the IVR, and then dialed extension
> is 5107.
>
> Yes i have two endpoint in my bridge line. One is keying extension and
> other
> is default extension.
>
> But this dialplan send by David what i tried.
> <entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
> param="execute_extension
>
> set:continue_on_fail=BUSY,bridge:sofia/internal/$1%XX.XX.XX.XX,bridge:sofia/internal/5998%XX.XX.XX.XX
> "/>
>
> My actual conduction is i dialed in from the outside, got to the ivr, where
> after keying in the extension number the call gets transferred to the
> appropriate extension and if the extension is busy it should be hunted to a
> default extension.
>
> Well, it looks like you are trying to execute_extension on an inline
dialplan without actually specifying "inline" in your param. I've never
tried this, so I'm shooting from the hip here. Try this:

<entry action="menu-exec-app" digits="/^([1-6][0-9][0-9][0-9])$/"
param="execute_extension
set:continue_on_fail=BUSY,bridge:sofia/internal/$1%XX.XX.XX.XX,bridge:sofia/internal/5998%XX.XX.XX.XX
inline
"/>

-OR-
Put these 3 actions into an actual dialplan extension, perhaps even their
own context and execute that extension...

-MC

FYI, inline dialplan stuff is discussed here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_InlineDialplan
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From mgende at gendesign.com  Fri May 27 23:31:04 2011
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 27 May 2011 14:31:04 -0500
Subject: [Freeswitch-users] ptime and voice/call quality problem
Message-ID: <BANLkTikPmKZ1X_O=Rwh0UQaRcnkNvBBdhQ@mail.gmail.com>

Hey Folks,

Kind of strange issue here, tied to some other stuff I've posted about.
Anyway, does anyone know how to set ptime or where that is configured?

Please forgive if this is staring me in the face somewhere, I've not found
anything satisfactory just looking around. But, I am slow sometimes.

What's up?  We have an FS that has two incoming numbers. One of them, just
recently, started doing the following (from our VoIP ISP tech guy):

All calls that work (to the second number) network originates the call with
maxptime=20, we respond with ptime=20 and everything works fine.  The calls
that do not work (primary number) network originates the call with
maxptime=30, we respond with ptime=30 then immediately re-invite with
ptime=20.  I suspect somewhere in this re-invite either your switch or our
provider?s switch is getting confused and not sending or processing RTP
based on the re-invite.

Result? The caller can't be heard by the call taker at our FS. Plus, the
sound quality, even of the ring, is bad.

Sorry so long a post! Right now, we're forwarding the "bad" number to the
"good". A band-aid. Any ptime experts here? We did not change this
parameter, but did modify the call_timeout up to 19 from 15 trying to give
'em another ring. Then back down once this started to no avail.

Thanks for reading this far if you did. Any advice appreciated.
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From yudha2008 at gmail.com  Sat May 28 00:08:50 2011
From: yudha2008 at gmail.com (baskar)
Date: Fri, 27 May 2011 13:08:50 -0700 (PDT)
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <BANLkTinbdfaEHUzxoND6UCcYSSRCK_4cmQ@mail.gmail.com>
References: <BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
	<BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
	<BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>
	<BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>
	<1306514561494-6411726.post@n2.nabble.com>
	<BANLkTinbdfaEHUzxoND6UCcYSSRCK_4cmQ@mail.gmail.com>
Message-ID: <1306526930492-6412425.post@n2.nabble.com>

Hi Mike,

I tried out this inline command and it really works well as per my
requirement!!Thanks a lot mate..  

Thanks,
N.Baskar

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inbound-dialplan-tp6404380p6412425.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From spencer at 5ninesolutions.com  Sat May 28 00:35:59 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 27 May 2011 15:35:59 -0500
Subject: [Freeswitch-users] Received in VIa
Message-ID: <B277B4AD-C4FA-42C3-BF02-F708BD329D43@5ninesolutions.com>

Hello all,
We are trying to use Freeswitch as a SBC in conjunction with Opensips.  The call flow looks like this

User UA -> Opensips -> ITSP
                      |             |
                     \/            /\ 
                    Freeswitch

Opensips acts as a registrar and select the outbound route and listens on X.X.X.X:5060.   Freeswitch is a B2BUA and signaling server with media and listens on X.X.X.X:5070.

Basically a user sends an INVITE and immediately following auth, Opensips forwards to Freeswitch which sends a new call back to opensips which is then routed to our ITSPs.

With FS not listening on the default port, we get a received in the Via: message.

Via: SIP/2.0/UDP X.X.X.X:5070;received=X.X.X.X;rport=5070;branch=z9hG4bKpU7KvSaeKSc5D

Is there any way to remove the received header?

Thanks,
Spencer

From djbinter at gmail.com  Sat May 28 00:48:49 2011
From: djbinter at gmail.com (DJB International)
Date: Fri, 27 May 2011 13:48:49 -0700
Subject: [Freeswitch-users] Received in VIa
In-Reply-To: <B277B4AD-C4FA-42C3-BF02-F708BD329D43@5ninesolutions.com>
References: <B277B4AD-C4FA-42C3-BF02-F708BD329D43@5ninesolutions.com>
Message-ID: <BANLkTi=7ZU2h9LNYYXP9t8gws56WYyskVw@mail.gmail.com>

Maybe, you might be able to use the application unset to the variable
sip_full_via.  Just an idea.  You need to try it.

-djbinter


On Fri, May 27, 2011 at 1:35 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> We are trying to use Freeswitch as a SBC in conjunction with Opensips.  The
> call flow looks like this
>
> User UA -> Opensips -> ITSP
>                      |             |
>                     \/            /\
>                    Freeswitch
>
> Opensips acts as a registrar and select the outbound route and listens on
> X.X.X.X:5060.   Freeswitch is a B2BUA and signaling server with media and
> listens on X.X.X.X:5070.
>
> Basically a user sends an INVITE and immediately following auth, Opensips
> forwards to Freeswitch which sends a new call back to opensips which is then
> routed to our ITSPs.
>
> With FS not listening on the default port, we get a received in the Via:
> message.
>
> Via: SIP/2.0/UDP
> X.X.X.X:5070;received=X.X.X.X;rport=5070;branch=z9hG4bKpU7KvSaeKSc5D
>
> Is there any way to remove the received header?
>
> 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 jgalaz at yx.cl  Fri May 27 23:58:56 2011
From: jgalaz at yx.cl (Javier Galaz Jeria)
Date: Fri, 27 May 2011 15:58:56 -0400
Subject: [Freeswitch-users] Problem receiving calls from E1 to TE122p
Message-ID: <20110527195856.GA3362@jgalaz-desktop>

Hi,

I'm using FS git, on Ubuntu 11.04, with Digium TE122p card (from now on srv1), 
and FS git, on Ubuntu Server 11.04 with Sangoma A101 card (from now on srv2), 
both currently configured by vendor's scripts.

When I make a call from srv1 to srv2, everything goes well, I can hear both ways 
no problem, however when I try it the other way around (srv2->srv1) the 
softphone registered to srv2 gets "User Not Available", hardphone registered to 
srv1 gets INVITE and just after that CANCEL. The FS log: [1]

Looking at the logs I can see that after the RINGING goes to EARLY and tries to 
play ringback tone, but it can't write and it can't read from channel because 
it's not opened. So it hangs up. [lines 520-528]

I've searched the wiki, asked on IRC, but can't solve my problem, anyone who has 
configured TE122p could give some pointers?

This are my config files:

-----TE122p-----
/etc/dahdi/system.conf
loadzone    = es
defaultzone = es

span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31


/usr/local/freeswitch/conf/freetdm.conf
[general]
cpu_monitor => no
cpu_monitoring_interval => 1000
cpu_set_alarm_threshold => 80
cpu_reset_alarm_threshold => 70
cpu_alarm_action => warn

[span zt TE122p]
trunk_type => E1
b-channel => 1-15
d-channel => 16
b-channel => 17-31


/usr/local/freswitch/conf/freetdm.conf.xml
<configuration name="freetdm.conf" description="FreeTDM Configuration">
	<settings>
		<param name="debug" value="0"/>
		<param name="sip_headers" value="true"/>
	</settings>

	<libpri_spans>
		<span name="TE122p">
			<param name="node" value="network"/>
			<param name="switch" value="euroisdn"/>
			<param name="dialplan" value="XML"/>
			<param name="context" value="default"/>
		</span>
	</libpri_spans>
</configuration>


------A101-----
/etc/wanpipe1.conf
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE_API, Comment

[wanpipe1]
CARD_TYPE 	= AFT
S514CPU 	= A
CommPort 	= PRI
AUTO_PCISLOT 	= NO
PCISLOT 	= 4
PCIBUS  	= 2
FE_MEDIA	= E1
FE_LCODE	= HDB3
FE_FRAME	= NCRC4
FE_LINE		= 1
TE_CLOCK 	= NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE	= NO
TE_RX_SLEVEL    = 430
LBO 		= 120OH
FE_TXTRISTATE	= NO
MTU 		= 1500
UDPPORT		= 9000
TTL		= 255
IGNORE_FRONT_END	= NO
TDMV_SPAN		= 1
TDMV_DCHAN		= 16
TE_AIS_MAINTENANCE = NO         #NO: defualt  YES: Start port in AIS Blue Alarm and keep line down
                                #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode
								#wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode  
TDMV_HW_DTMF		= YES		# YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT		= NO		# YES: receive fax 1100hz events from hardware
HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation enabled with nlp (default) 
										# OCT_SPEECH: improves software tone detection by disabling NLP (echo possible)
										# OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. 
HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the line - could break fax
HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo cancelation
HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software tone detection (possible echo)   
HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio level to be maintained (-20 default)  
HWEC_TX_GAIN            = 0		# 0: disable   -24-24: db values to be applied to tx signal
HWEC_RX_GAIN            = 0		# 0: disable   -24-24: db values to be applied to tx signal

[w1g1]
ACTIVE_CH	= ALL
TDMV_HWEC	= YES
MTU 		= 80

/usr/local/freeswitch/conf/freetdm.conf 
[span wanpipe wp1]
trunk_type => e1
group=1
b-channel => 1:1-15
b-channel => 1:17-31
d-channel => 1:16

/usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
<configuration name="freetdm.conf" description="Freetdm Configuration">
	<settings>
		<param name="debug" value="0"/>
		<!--<param name="hold-music" value="$${moh_uri}"/>-->
		<!--<param name="enable-analog-option" value="call-swap"/>-->
		<!--<param name="enable-analog-option" value="3-way"/>-->
	</settings>
	<config_profiles>
		<profile name="my_pri_te_1">
			<param name="switchtype" value="euroisdn" />
			<param name="interface" value="pri_cpe"/>
		</profile>
	</config_profiles>
	<sangoma_pri_spans>
		<span name="wp1" cfgprofile="my_pri_te_1">
			<param name="dialplan" value="XML"/>
			<param name="context" value="default"/>
		</span>
	</sangoma_pri_spans>
</configuration>


How can I fix this? Any help would be greatly appreciated. If you need more info 
just tell me.

[1] http://pastebin.freeswitch.org/16397


Best Regards

-j-


From msc at freeswitch.org  Sat May 28 01:02:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 27 May 2011 14:02:37 -0700
Subject: [Freeswitch-users] ptime and voice/call quality problem
In-Reply-To: <BANLkTikPmKZ1X_O=Rwh0UQaRcnkNvBBdhQ@mail.gmail.com>
References: <BANLkTikPmKZ1X_O=Rwh0UQaRcnkNvBBdhQ@mail.gmail.com>
Message-ID: <BANLkTiko7Fge5HvCNVGTyz5aR1j8OoyU3A@mail.gmail.com>

Any chance you can get a SIP trace of working vs. not working calls? Put
them on pastebin.freeswitch.org and the community will take a look.

-MC

On Fri, May 27, 2011 at 12:31 PM, Michael Gende <mgende at gendesign.com>wrote:

> Hey Folks,
>
> Kind of strange issue here, tied to some other stuff I've posted about.
> Anyway, does anyone know how to set ptime or where that is configured?
>
> Please forgive if this is staring me in the face somewhere, I've not found
> anything satisfactory just looking around. But, I am slow sometimes.
>
> What's up?  We have an FS that has two incoming numbers. One of them, just
> recently, started doing the following (from our VoIP ISP tech guy):
>
> All calls that work (to the second number) network originates the call with
> maxptime=20, we respond with ptime=20 and everything works fine.  The calls
> that do not work (primary number) network originates the call with
> maxptime=30, we respond with ptime=30 then immediately re-invite with
> ptime=20.  I suspect somewhere in this re-invite either your switch or our
> provider?s switch is getting confused and not sending or processing RTP
> based on the re-invite.
>
> Result? The caller can't be heard by the call taker at our FS. Plus, the
> sound quality, even of the ring, is bad.
>
> Sorry so long a post! Right now, we're forwarding the "bad" number to the
> "good". A band-aid. Any ptime experts here? We did not change this
> parameter, but did modify the call_timeout up to 19 from 15 trying to give
> 'em another ring. Then back down once this started to no avail.
>
> Thanks for reading this far if you did. Any advice appreciated.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmrienzo at gmail.com  Sat May 28 01:22:50 2011
From: cmrienzo at gmail.com (Chris Rienzo)
Date: Fri, 27 May 2011 17:22:50 -0400
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <1306520931787-6412076.post@n2.nabble.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306520931787-6412076.post@n2.nabble.com>
Message-ID: <576485B0-53B9-4B03-A09E-8A6D76BA6CE5@gmail.com>

That's all the help I can give.  I have no experience using mod_mp4.


On May 27, 2011, at 14:28, pooja <pooja27taneja at gmail.com> wrote:

> I did as you said. now it doesnt say invalid application play_mp4. but
> instead gives an error :
> [ERR] mod_mp4.cpp:505 snookerpup_fhnom3b7.mp4:Missing audio/video track.
> 
> but the mp4 file is there in the location.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6412076.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 yungwei at resolvity.com  Sat May 28 01:27:29 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 27 May 2011 17:27:29 -0400
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
	<BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>
Message-ID: <33095823FD21DF429B481B5163264B7950AC3ACA1B@VMBX102.ihostexchange.net>

The whole picture looks like the following:
A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone

Is there any configuration setting that needs to be enabled or set differently so that those 2 endpoints can talk to each other?
Btw, the same scenario works with Asterisk so my SIP provider shouldn't be the problem.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Friday, May 27, 2011 11:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

Here's the debug log, http://pastebin.freeswitch.org/16398

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, May 27, 2011 10:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

You need to put a debug log on pastebin.freeswitch.org<http://pastebin.freeswitch.org>. Be sure to use "FreeSWITCH Log" for the syntax highlighting.
-MC
On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:
No.

I just verified that making an outbound call to my cell phone still works.
I even recorded the session just to make sure, and in the recording I hear things both ways.
I first dial 9911 from my SIP client (behind freeswitch), and this leads to the javascript program below.
  session.answer();
  var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  session.hangup(16); // disconnects the session between the SIP client and freeswitch
  new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
  while (new_session.ready())
  {
      new_session.streamFile("/path/to/local/wav/file");
  }

Now back to the case I'm having problem with.
In this case, I first make a call from a landline to freeswitch through my sip provider, and then a javascript program takes over.
I want to transfer the call to a cell phone so that the landline and the cell phone can communicate with each other.
Here's the javascript program:
  session.answer();

  if (session.ready())
  {
      session.execute("bridge", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  }

So what am I missing here?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of mazilo
Sent: Friday, May 27, 2011 7:28 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Transferring to a cell phone


Yungwei Chen wrote:
>
> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
After switching to using bridge function, does this also happen when you
make an outbound call to your cell phone using your javascript?

-----
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/Transferring-to-a-cell-phone-tp6408246p6410887.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
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From msc at freeswitch.org  Sat May 28 03:21:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 27 May 2011 16:21:00 -0700
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC3ACA1B@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
	<BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950AC3ACA1B@VMBX102.ihostexchange.net>
Message-ID: <BANLkTi=4fMHUkHjoCZ8exuw6Qgg5Pgj6Cw@mail.gmail.com>

Something is not right with the outbound leg of the call. It looks like
there is an immediate hangup after the b leg answers. Get a siptrace of that
traffic and look to see what is causing the hangup.

-MC

On Fri, May 27, 2011 at 2:27 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> The whole picture looks like the following:
>
> A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone
>
>
>
> Is there any configuration setting that needs to be enabled or set
> differently so that those 2 endpoints can talk to each other?
>
> Btw, the same scenario works with Asterisk so my SIP provider shouldn't be
> the problem.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yungwei Chen
> *Sent:* Friday, May 27, 2011 11:34 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Transferring to a cell phone
>
>
>
> Here's the debug log, http://pastebin.freeswitch.org/16398
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, May 27, 2011 10:08 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Transferring to a cell phone
>
>
>
> You need to put a debug log on pastebin.freeswitch.org. Be sure to use
> "FreeSWITCH Log" for the syntax highlighting.
>
> -MC
>
> On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com>
> wrote:
>
> No.
>
> I just verified that making an outbound call to my cell phone still works.
> I even recorded the session just to make sure, and in the recording I hear
> things both ways.
> I first dial 9911 from my SIP client (behind freeswitch), and this leads to
> the javascript program below.
>   session.answer();
>
>   var new_session = new
> Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>
>   session.hangup(16); // disconnects the session between the SIP client and
> freeswitch
>   new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
>   while (new_session.ready())
>   {
>       new_session.streamFile("/path/to/local/wav/file");
>   }
>
> Now back to the case I'm having problem with.
> In this case, I first make a call from a landline to freeswitch through my
> sip provider, and then a javascript program takes over.
> I want to transfer the call to a cell phone so that the landline and the
> cell phone can communicate with each other.
> Here's the javascript program:
>   session.answer();
>
>   if (session.ready())
>   {
>       session.execute("bridge",
> "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>   }
>
> So what am I missing here?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
> Sent: Friday, May 27, 2011 7:28 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>
>
> Yungwei Chen wrote:
> >
> > Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> > anything both ways. Any idea?
> After switching to using bridge function, does this also happen when you
> make an outbound call to your cell phone using your javascript?
>
> -----
> 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/Transferring-to-a-cell-phone-tp6408246p6410887.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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Sat May 28 05:47:41 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 28 May 2011 03:47:41 +0200
Subject: [Freeswitch-users] Inbound dialplan
In-Reply-To: <1306526930492-6412425.post@n2.nabble.com>
References: <BANLkTi=u=RpF7PAiwv3h0t6gax3oMbq=1A@mail.gmail.com>
	<741C0F00-F48A-43DD-89BC-255AD5FDD4BE@ipeva.fr>
	<BANLkTi=Af6QV0eyPSLT5O-=dcP=+B4=r_A@mail.gmail.com>
	<BANLkTi=F9rKDaJFGWO0Bzk-qeTT6XZFKDw@mail.gmail.com>
	<BANLkTim=4prR18C0dGbh6sBs3UE06zoVpA@mail.gmail.com>
	<BANLkTinufCxDOLh_KWBKezza88PsYzN9iA@mail.gmail.com>
	<BANLkTik43Rcabrh5qY8+A9OR6j-r7O1U9A@mail.gmail.com>
	<BANLkTi=ojFV5d9=O0iz2pfSTC9BSMQNudw@mail.gmail.com>
	<1306514561494-6411726.post@n2.nabble.com>
	<BANLkTinbdfaEHUzxoND6UCcYSSRCK_4cmQ@mail.gmail.com>
	<1306526930492-6412425.post@n2.nabble.com>
Message-ID: <5CF0198C-D0F6-45AB-BA92-C6EB355B1291@ipeva.fr>

Sorry, that was my fault.
I fucked up my copy/paste, and the final "inline" got lost in the config I sent you 2 days ago.

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

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

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




Le 27/05/2011 ? 22:08, baskar a ?crit :

> Hi Mike,
> 
> I tried out this inline command and it really works well as per my
> requirement!!Thanks a lot mate..  
> 
> Thanks,
> N.Baskar
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inbound-dialplan-tp6404380p6412425.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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From spencer at 5ninesolutions.com  Sat May 28 06:11:25 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 27 May 2011 21:11:25 -0500
Subject: [Freeswitch-users] Codec Negotiation Help
In-Reply-To: <90BEEBF8-041B-4AC1-B8BE-33A5391EFB33@5ninesolutions.com>
References: <42E549A5-8F9F-4C57-B358-755A02EDDDDD@5ninesolutions.com>
	<BANLkTi=wjx5JuFVBbzHcw-aSE7SW4AdrKA@mail.gmail.com>
	<BE3CBF8A-CA92-482D-A994-C9FC0FBA1FBB@5ninesolutions.com>
	<BANLkTin-Z6gjD_QqxVWPJ229NoX4JU_YZQ@mail.gmail.com>
	<90BEEBF8-041B-4AC1-B8BE-33A5391EFB33@5ninesolutions.com>
Message-ID: <CE349DED-0D0B-4502-8F42-D9CFF5C6FD15@5ninesolutions.com>

After some testing,
A little clarification on the matter...

[DEBUG] switch_ivr_originate.c:404 Codec string PCMU at 8000h@20i not supported on sofia/internal/5000 at pbx.myserver.com, skipping inheritance

So.. If the codec isn't supported Freeswitch transcodes :-)  You guys rock, who knew something could actually make sense.

Spencer


On May 26, 2011, at 4:20 PM, Spencer Thomason wrote:

> That's what I was thinking.  I don't actually have any devices like this but a few of our phones do support iLBC and we have be toying with the idea of using that on a few links that are slower.  Is there any option to setting inherit_codec=true that would "fall back" to transcoding if need be?
> 
> Spencer
> 
> 
> On May 26, 2011, at 4:05 PM, Michael Collins wrote:
> 
>> I believe in the scenario you describe that the call would fail since the incoming call cannot inherit the only codec supported by the b leg device (GSM).
>> 
>> -MC
>> 
>> On Thu, May 26, 2011 at 10:56 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> Thanks for you help, I'll try that.  So I'm a little confused with the negotiation process with two profiles.  Lets say for  example that I have a device that only supports GSM on the internal profile and a call comes in on the external profile as ULAW, G729.  If I set  would Freeswitch then transcode?  How does the disable-transcoding option work in regards to two profiles?  I.e. Only one profile has transcoding disabled but a call traverses both of them.. (2 legs, one bridge).  Which profile would the transcoding need to be enabled (or not disabled rather)?
>> 
>> Thanks,
>> Spencer
>> 
>> 
>> On May 26, 2011, at 11:47 AM, DJB International wrote:
>> 
>>> Your profile has late negotion enabled.  I believe you can set inherit_codec=true, so that it will force A leg to use the same codec as B leg offered.
>>> 
>>> 
>>> On Thu, May 26, 2011 at 9:04 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>>> Hello all,
>>> I have a problem regarding the codec negotiation on an outbound call.  My setup is like this:
>>> 
>>> Polycom IP 650 (1-n) -NAT-> FS  --> Our Signaling Proxy --> ITSP Proxy ---> ITSP Cisco GW
>>> 
>>> I'd like to use different codecs for different call paths (in order of pref), g729 in passthru only:
>>> IP-650         ->  IP-650   G722, PCMU, G729
>>> Inbound      ->  IP-650   PCMU
>>> IP-650         ->  Outbound PCMU,G729
>>> 
>>> I have two sofia profiles, internal, public IPv4:5060 and external, public:IPv4:5080.
>>> 
>>> The phones use the internal profile and the external profile only communicates with our signaling proxy (no media proxy).
>>> On the internal one:
>>> CODECS IN               G722,PCMU,G729,GSM
>>> CODECS OUT              G722,PCMU,G729,GSM
>>> NOMEDIA                 false
>>> LATE-NEG                true
>>> 
>>> External:
>>> CODECS IN               PCMU,G729
>>> CODECS OUT              PCMU,G729
>>> NOMEDIA                 false
>>> LATE-NEG                true
>>> 
>>> I have inbound-codec-negotiation set to greedy on both profiles and on outbound calls set absolute_codec_string=PCMU,G729 to prevent transcoding.  Note that mod_g729 is enabled for passthru only.
>>> 
>>> The problem I have is this:
>>> We use the dynamic routing module in OpenSIPS to select an outbound provider/GW, all support PCMU and G729.  On one of the routes, the Cisco IOS GW on this route has G729, PCMU configured as its codec pref.
>>> 
>>> I have included a ladder diagram to better illustrate the problem but in a nutshell, the polycom negotiates PCMU with FS, FS asks for both PCMU and G729, the cisco GW sends G729 and FS sends a 488 because it can't transcode.  I would like to keep G729 in the outbound prefs because some routes might not support PCMU.  Should I set one of the profiles to generous, and if so which one?
>>> 
>>> When someone makes an outbound call the following happens (ladder diagram):
>>> http://pastebin.freeswitch.org/16380
>>> 
>>> 
>>> Sorry for the novella, :-)
>>> 
>>> 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
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From babak.freeswitch at gmail.com  Sat May 28 14:24:48 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 28 May 2011 14:54:48 +0430
Subject: [Freeswitch-users] connecting freeswitch to a panasonic tde 100
Message-ID: <BANLkTi=-0Go65iTAGDB2U2ENZpon3KLoXQ@mail.gmail.com>

Hi
Is it possible to connect a tde100 to freeswitch ? As I found out it
seems possible with v-sipgw16 (virtual cards), but if it is so how
should I do it to make and get calls in freeswitch?


From anton.vazir at gmail.com  Sat May 28 15:24:16 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 28 May 2011 16:24:16 +0500
Subject: [Freeswitch-users] connecting freeswitch to a panasonic tde 100
In-Reply-To: <BANLkTi=-0Go65iTAGDB2U2ENZpon3KLoXQ@mail.gmail.com>
References: <BANLkTi=-0Go65iTAGDB2U2ENZpon3KLoXQ@mail.gmail.com>
Message-ID: <BANLkTik8DCecH_JE7p7J=59NiK6SB18UFw@mail.gmail.com>

Get an e1 module

2011/5/28 babak yakhchali <babak.freeswitch at gmail.com>:
> Hi
> Is it possible to connect a tde100 to freeswitch ? As I found out it
> seems possible with v-sipgw16 (virtual cards), but if it is so how
> should I do it to make and get calls in freeswitch?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From tang.du at hotmail.com  Sat May 28 18:43:06 2011
From: tang.du at hotmail.com (tangdu)
Date: Sat, 28 May 2011 07:43:06 -0700 (PDT)
Subject: [Freeswitch-users] load mod_gsmopen error
In-Reply-To: <1306294731594-6401282.post@n2.nabble.com>
References: <1306294731594-6401282.post@n2.nabble.com>
Message-ID: <1306593786312-6414343.post@n2.nabble.com>

Thanks for your help?
I will take another model cellphone to test it?
Best Regards
Tang Du

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From sireeps at gmail.com  Sat May 28 19:49:05 2011
From: sireeps at gmail.com (Kamen)
Date: Sat, 28 May 2011 08:49:05 -0700 (PDT)
Subject: [Freeswitch-users] setting up forward incoming calls to
	external gateway
In-Reply-To: <BANLkTikdyLNY-PSwpd1As9aAEU833bXOVw@mail.gmail.com>
References: <1305488599172-6366400.post@n2.nabble.com>
	<BANLkTi=mu8sfw6p-Mw6sp6VawVBXyk6V1g@mail.gmail.com>
	<1306019791787-6390451.post@n2.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58FDA53D4D@cooper>
	<1306087718406-6392049.post@n2.nabble.com>
	<E22C41E5-108E-44E7-96CE-F7C6E6D1867D@gmail.com>
	<1306164331568-6394908.post@n2.nabble.com>
	<BANLkTinvAzxVVTBGi-c3D8_hci6i2maBpA@mail.gmail.com>
	<1306186596224-6396358.post@n2.nabble.com>
	<BANLkTikdyLNY-PSwpd1As9aAEU833bXOVw@mail.gmail.com>
Message-ID: <1306597745279-6414512.post@n2.nabble.com>

Alright folks!

I have figured the problem.  The solution was kind of simple. My SIP
provider does not ring back while caller is trying to connect to the number. 
After 10 seconds of silence the line is disconnected on caller side. I do
not know why it takes longer than 10 seconds to connect, but that is another
issue.  In any case after I added

<action application="ring_ready"/>	

to my default extension the caller can hear rings and the system does not
hangup.  Eventually the call gets through.

Thanks anyone who tried to help me, anyway.

Regards,
Sergei Kamen




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From sireeps at gmail.com  Sat May 28 20:48:24 2011
From: sireeps at gmail.com (Kamen)
Date: Sat, 28 May 2011 09:48:24 -0700 (PDT)
Subject: [Freeswitch-users] collectInput does not call callback function
Message-ID: <1306601304658-6414679.post@n2.nabble.com>

Hi folks!

I seem to have an odd problem, since it works for all others.  From my java
script I try to collect user input:

session.collectInput( mycb, "digits", 7000, 0);

where mycb is a regular function collecting user input:


function mycb( session, type, obj, arg ) 
{
   console_log( "info", "collect digits \n" );
   try {
       if ( type == "digits" )  {
         console_log( "info", "digit: "+obj.digit+"\n" );
         if ( obj.digit == "#" ) {
           console_log( "info", "detected pound sign.\n" );
           exit = true;
           return( false );
         }

         dtmf.digits += obj.digit;

         if ( dtmf.digits.length >= digitmaxlength ) {
           exit = true;
           return( false );
         }
       }
   } catch (e) {
       console_log( "err", e+"\n" );
   }
   return( true );
} //end function mycb


The very first line is a console info message, so I would know the function
is actually called. It never happens.  Right before collectInput call I have
a message playing a prompt, which I can hear.  So workflow definitely goes
into the collectInput function, and then after 7 seconds it continues with
dtmf.digits.length=0 even though I punched in the numbers, of course.  Thats
another proof the callback function never was called.

Has anyone experienced this problem before? I run FS in Windows.  Appreciate
your help.

Regards,

Sergei Kamen



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From mgende at gendesign.com  Sun May 29 00:46:30 2011
From: mgende at gendesign.com (Michael Gende)
Date: Sat, 28 May 2011 15:46:30 -0500
Subject: [Freeswitch-users] ptime and voice/call quality problem
In-Reply-To: <BANLkTiko7Fge5HvCNVGTyz5aR1j8OoyU3A@mail.gmail.com>
References: <BANLkTikPmKZ1X_O=Rwh0UQaRcnkNvBBdhQ@mail.gmail.com>
	<BANLkTiko7Fge5HvCNVGTyz5aR1j8OoyU3A@mail.gmail.com>
Message-ID: <BANLkTimJX_t_riSjm3qzWhzqxjsp8US=7w@mail.gmail.com>

Yes, I believe I can get some packet captures from our VoIP ISP. I'll post
them to the group you suggest on Tuesday. Thanks for the suggestion!

Mike G.

On Fri, May 27, 2011 at 4:02 PM, Michael Collins <msc at freeswitch.org> wrote:

> Any chance you can get a SIP trace of working vs. not working calls? Put
> them on pastebin.freeswitch.org and the community will take a look.
>
> -MC
>
> On Fri, May 27, 2011 at 12:31 PM, Michael Gende <mgende at gendesign.com>wrote:
>
>> Hey Folks,
>>
>> Kind of strange issue here, tied to some other stuff I've posted about.
>> Anyway, does anyone know how to set ptime or where that is configured?
>>
>> Please forgive if this is staring me in the face somewhere, I've not found
>> anything satisfactory just looking around. But, I am slow sometimes.
>>
>> What's up?  We have an FS that has two incoming numbers. One of them, just
>> recently, started doing the following (from our VoIP ISP tech guy):
>>
>> All calls that work (to the second number) network originates the call
>> with maxptime=20, we respond with ptime=20 and everything works fine.  The
>> calls that do not work (primary number) network originates the call with
>> maxptime=30, we respond with ptime=30 then immediately re-invite with
>> ptime=20.  I suspect somewhere in this re-invite either your switch or our
>> provider?s switch is getting confused and not sending or processing RTP
>> based on the re-invite.
>>
>> Result? The caller can't be heard by the call taker at our FS. Plus, the
>> sound quality, even of the ring, is bad.
>>
>> Sorry so long a post! Right now, we're forwarding the "bad" number to the
>> "good". A band-aid. Any ptime experts here? We did not change this
>> parameter, but did modify the call_timeout up to 19 from 15 trying to give
>> 'em another ring. Then back down once this started to no avail.
>>
>> Thanks for reading this far if you did. Any advice appreciated.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From garmt.noname at gmail.com  Sun May 29 01:18:48 2011
From: garmt.noname at gmail.com (Grmt)
Date: Sat, 28 May 2011 23:18:48 +0200
Subject: [Freeswitch-users] collectInput does not call callback function
In-Reply-To: <1306601304658-6414679.post@n2.nabble.com>
References: <1306601304658-6414679.post@n2.nabble.com>
Message-ID: <4de166bd.81320e0a.7135.7e06@mx.google.com>

s/"digits"/"dtmf" ?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kamen
Sent: Saturday, 28 May, 2011 18:48
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] collectInput does not call callback function

Hi folks!

I seem to have an odd problem, since it works for all others.  From my java
script I try to collect user input:

session.collectInput( mycb, "digits", 7000, 0);

where mycb is a regular function collecting user input:


function mycb( session, type, obj, arg ) 
{
   console_log( "info", "collect digits \n" );
   try {
       if ( type == "digits" )  {
         console_log( "info", "digit: "+obj.digit+"\n" );
         if ( obj.digit == "#" ) {
           console_log( "info", "detected pound sign.\n" );
           exit = true;
           return( false );
         }

         dtmf.digits += obj.digit;

         if ( dtmf.digits.length >= digitmaxlength ) {
           exit = true;
           return( false );
         }
       }
   } catch (e) {
       console_log( "err", e+"\n" );
   }
   return( true );
} //end function mycb


The very first line is a console info message, so I would know the function
is actually called. It never happens.  Right before collectInput call I have
a message playing a prompt, which I can hear.  So workflow definitely goes
into the collectInput function, and then after 7 seconds it continues with
dtmf.digits.length=0 even though I punched in the numbers, of course.  Thats
another proof the callback function never was called.

Has anyone experienced this problem before? I run FS in Windows.  Appreciate
your help.

Regards,

Sergei Kamen



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From anton.vazir at gmail.com  Sun May 29 02:46:30 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 29 May 2011 03:46:30 +0500
Subject: [Freeswitch-users] Please advice,
	Unable to continue dialplan if legB fails on early-media
Message-ID: <BANLkTi=wtE39ev7Pv6DSt3-_Sww3rQ7J8w@mail.gmail.com>

I want to play to leg-a some 'Your party does not answer' if leg-b
hangs-up without answering, but can't.

I use outbound ESL
When receiving a call (leg-A) I do 'originate &park' (leg-B)
than I do uuid_bridge A and B legs when I get early_media from leg_b

If leg-B answers, and than hangs up - everything is OK, I can play
extra tones for leg-A, like: 'Your call duration, bla, bla, bla'
successfully.

But, if leg-B hangs while in EARLY_MEDIA (without answering), leg A
hangs up too (closes ESL connection) , ignoring all variables

hangup_after_bridge=false
originate_continue_on_timeout=true
continue_on_fail=true
park_after_bridge=true

I also was trying to set

ignore_early_media=true

in this case I receive

-ERR ORIGINATOR_CANCEL

for originate job, but originator does not cancel.

and than ELS connection closes.

If I do not bridge legs on early media, I can play what I want on
receiving "CHANNEL_HANGUP" from leg B, but I want Leg-A to hear native
progress tones from leg B

Please advice


From yehavi.bourvine at gmail.com  Sun May 29 08:31:24 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 29 May 2011 07:31:24 +0300
Subject: [Freeswitch-users] Easy way to count current calls count on a
	specific profile/gateway
Message-ID: <BANLkTin6ESEL9PqnAc=h1cxvXJnMnQUR5A@mail.gmail.com>

Hello,

  Is there an easy eay to count the current number of active calls per
profile or gateway? I would like to have statistics on our external
suppliers usage.

                            Thanks, __Yehavi:
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From avi at avimarcus.net  Sun May 29 09:48:47 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 29 May 2011 08:48:47 +0300
Subject: [Freeswitch-users] Easy way to count current calls count on a
 specific profile/gateway
In-Reply-To: <BANLkTin6ESEL9PqnAc=h1cxvXJnMnQUR5A@mail.gmail.com>
References: <BANLkTin6ESEL9PqnAc=h1cxvXJnMnQUR5A@mail.gmail.com>
Message-ID: <BANLkTin=o9uLDoHQRYP=MgZsM=8u8KCn8A@mail.gmail.com>

There's a few options...
1) "show channels like $var count" should let you filter by something
2) "sofia status" has a "..RUNNING (\d+?)" that shows you the count of
active channels for the entire profile.
3) Set a limit in the dialplan before the bridge without a maximum and then
retrieve the count to count by whichever, gateway or profile.

-Avi

On Sun, May 29, 2011 at 7:31 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello,
>
>   Is there an easy eay to count the current number of active calls per
> profile or gateway? I would like to have statistics on our external
> suppliers usage.
>
>                             Thanks, __Yehavi:
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Sun May 29 10:08:53 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 29 May 2011 09:08:53 +0300
Subject: [Freeswitch-users] Easy way to count current calls count on a
 specific profile/gateway
In-Reply-To: <BANLkTin=o9uLDoHQRYP=MgZsM=8u8KCn8A@mail.gmail.com>
References: <BANLkTin6ESEL9PqnAc=h1cxvXJnMnQUR5A@mail.gmail.com>
	<BANLkTin=o9uLDoHQRYP=MgZsM=8u8KCn8A@mail.gmail.com>
Message-ID: <BANLkTin4f-sKUD5UVx7cbX6jXGhAXawNxA@mail.gmail.com>

Thanks!

                            __Yehavi:


2011/5/29 Avi Marcus <avi at avimarcus.net>

> There's a few options...
> 1) "show channels like $var count" should let you filter by something
> 2) "sofia status" has a "..RUNNING (\d+?)" that shows you the count of
> active channels for the entire profile.
> 3) Set a limit in the dialplan before the bridge without a maximum and then
> retrieve the count to count by whichever, gateway or profile.
>
> -Avi
>
> On Sun, May 29, 2011 at 7:31 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>> Hello,
>>
>>   Is there an easy eay to count the current number of active calls per
>> profile or gateway? I would like to have statistics on our external
>> suppliers usage.
>>
>>                             Thanks, __Yehavi:
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Sun May 29 10:28:49 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 29 May 2011 10:58:49 +0430
Subject: [Freeswitch-users] connecting freeswitch to a panasonic tde 100
In-Reply-To: <BANLkTik8DCecH_JE7p7J=59NiK6SB18UFw@mail.gmail.com>
References: <BANLkTi=-0Go65iTAGDB2U2ENZpon3KLoXQ@mail.gmail.com>
	<BANLkTik8DCecH_JE7p7J=59NiK6SB18UFw@mail.gmail.com>
Message-ID: <BANLkTi=MmoGxQbpGPfKWJHYQM6PWuTnKYw@mail.gmail.com>

Is there anyway to trunk tde 100 to freeswitch using its sip support?


From peter.olsson at visionutveckling.se  Sun May 29 11:19:48 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 29 May 2011 09:19:48 +0200
Subject: [Freeswitch-users] connecting freeswitch to a panasonic tde 100
In-Reply-To: <BANLkTi=MmoGxQbpGPfKWJHYQM6PWuTnKYw@mail.gmail.com>
References: <BANLkTi=-0Go65iTAGDB2U2ENZpon3KLoXQ@mail.gmail.com>
	<BANLkTik8DCecH_JE7p7J=59NiK6SB18UFw@mail.gmail.com>,
	<BANLkTi=MmoGxQbpGPfKWJHYQM6PWuTnKYw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B630C@cooper>

Yes, I've done this in my lab, and it works ok.

As you mentioned earlier, you need to configure a v-sipgw16. I don't remember any specifics out of my head though - but if you have a specific question when you configure it, jus ask me, and I can check in our running config.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r babak yakhchali [babak.freeswitch at gmail.com]
Skickat: den 29 maj 2011 08:28
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] connecting freeswitch to a panasonic tde 100

Is there anyway to trunk tde 100 to freeswitch using its sip support?

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4de1e88b32761843072369!



From babak.freeswitch at gmail.com  Sun May 29 13:05:13 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 29 May 2011 13:35:13 +0430
Subject: [Freeswitch-users] connecting freeswitch to a panasonic tde 100
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B630C@cooper>
References: <BANLkTi=-0Go65iTAGDB2U2ENZpon3KLoXQ@mail.gmail.com>
	<BANLkTik8DCecH_JE7p7J=59NiK6SB18UFw@mail.gmail.com>
	<BANLkTi=MmoGxQbpGPfKWJHYQM6PWuTnKYw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B630C@cooper>
Message-ID: <BANLkTim9wDB44mNs17-rqbJhk0Mp-FojEQ@mail.gmail.com>

Thanks Peter, this is the situation:
            +--------------------------------------------+
            |           Panasonic KX-TDE 100             |
            |                                            |
            |    +----------+          +-----------+     |
CO Lines <--|--->|  LCOT16  |          |  CSLC16   |<----|---->  Phones
            |    +----------+          +-----------+     |
            |         ^                      ^           |
            |         |                      |           |
            |         v                      v           |
            |      +------------------------------+      |
            |      |          V-SIPGW16           |      |
            |      +------------------------------+      |
            |                      ^                     |
            |                      |                     |
            +----------------------|---------------------+
                                   |
                                   v
                          +----------------+
                          |   Fresswitch   |
                          +----------------+
I need to process all calls in freeswitch. so all calls from pstn (comming
from lcot16 lines) and  phones (attached to CSLC16) should go to freeswitch
and fs should be able to distribute them (weather to pstn or phones). If you
are on IRC and of course if you like let me know to contact you there.
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From kris at livecall.com  Sun May 29 13:18:54 2011
From: kris at livecall.com (Kris)
Date: Sun, 29 May 2011 02:18:54 -0700
Subject: [Freeswitch-users] collectInput does not call callback function
References: <1306601304658-6414679.post@n2.nabble.com>
	<4de166bd.81320e0a.7135.7e06@mx.google.com>
Message-ID: <AE11948AEEA849BDBD543F81541BE516@stor1>

I experience a similar problem after exiting a conference and continuing my 
C# App. Here is how it works. Caller is given options in
Session.StreamFile(play_info[i].filename, (int)start_off);

If he presses lets say '1'  Session.DtmfReceivedFunction is called by FS and 
caller gets put into a conference. I have conference maintenance events 
poped in a separate thread.

He presses '#' and gets back to the the same menu, but now if he presses '1' 
or '#'..anything during StreamFile , the Session.DtmfReceivedFunction is no 
longer called by FS. Once it goes into Get Digits after StreamFile a  press 
of the digit is returned ok.

I noticed, if I dont create EventConsumer for the conf maintenace, it works 
ok. I need it though to have more control over the conference. It smells 
like a bug.

If anyone knows where to look it is probable where a DtmfReceivedFunction 
callback during playback should occur if the caller is deleted from the 
conference instead of assuming that the dtmf will be reported by 
conference:maintenance event.

Kris





----- Original Message ----- 
From: "Grmt" <garmt.noname at gmail.com>
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 28, 2011 2:18 PM
Subject: Re: [Freeswitch-users] collectInput does not call callback function


> s/"digits"/"dtmf" ?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kamen
> Sent: Saturday, 28 May, 2011 18:48
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] collectInput does not call callback function
>
> Hi folks!
>
> I seem to have an odd problem, since it works for all others.  From my 
> java
> script I try to collect user input:
>
> session.collectInput( mycb, "digits", 7000, 0);
>
> where mycb is a regular function collecting user input:
>
>
> function mycb( session, type, obj, arg )
> {
>   console_log( "info", "collect digits \n" );
>   try {
>       if ( type == "digits" )  {
>         console_log( "info", "digit: "+obj.digit+"\n" );
>         if ( obj.digit == "#" ) {
>           console_log( "info", "detected pound sign.\n" );
>           exit = true;
>           return( false );
>         }
>
>         dtmf.digits += obj.digit;
>
>         if ( dtmf.digits.length >= digitmaxlength ) {
>           exit = true;
>           return( false );
>         }
>       }
>   } catch (e) {
>       console_log( "err", e+"\n" );
>   }
>   return( true );
> } //end function mycb
>
>
> The very first line is a console info message, so I would know the 
> function
> is actually called. It never happens.  Right before collectInput call I 
> have
> a message playing a prompt, which I can hear.  So workflow definitely goes
> into the collectInput function, and then after 7 seconds it continues with
> dtmf.digits.length=0 even though I punched in the numbers, of course. 
> Thats
> another proof the callback function never was called.
>
> Has anyone experienced this problem before? I run FS in Windows. 
> Appreciate
> your help.
>
> Regards,
>
> Sergei Kamen
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/collectInput-does-not-call-cal
> lback-function-tp6414679p6414679.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 anton.vazir at gmail.com  Sun May 29 13:22:59 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 29 May 2011 14:22:59 +0500
Subject: [Freeswitch-users] Please advice,
	Unable to continue dialplan if legB fails on early-media
In-Reply-To: <BANLkTi=wtE39ev7Pv6DSt3-_Sww3rQ7J8w@mail.gmail.com>
References: <BANLkTi=wtE39ev7Pv6DSt3-_Sww3rQ7J8w@mail.gmail.com>
Message-ID: <BANLkTi=dKkbSQU-=c15TtTCtiibmpOjFUQ@mail.gmail.com>

Resolved, ignore_early_media thing, since api call does not return
until answered in that case, so it's interrupted by the following
command.

But still looking how to resolve leg-a hangup if leg-b fails in
early-media stage...

2011/5/29 Anton VG <anton.vazir at gmail.com>:
> I want to play to leg-a some 'Your party does not answer' if leg-b
> hangs-up without answering, but can't.
>
> I use outbound ESL
> When receiving a call (leg-A) I do 'originate &park' (leg-B)
> than I do uuid_bridge A and B legs when I get early_media from leg_b
>
> If leg-B answers, and than hangs up - everything is OK, I can play
> extra tones for leg-A, like: 'Your call duration, bla, bla, bla'
> successfully.
>
> But, if leg-B hangs while in EARLY_MEDIA (without answering), leg A
> hangs up too (closes ESL connection) , ignoring all variables
>
> hangup_after_bridge=false
> originate_continue_on_timeout=true
> continue_on_fail=true
> park_after_bridge=true
>
> I also was trying to set
>
> ignore_early_media=true
>
> in this case I receive
>
> -ERR ORIGINATOR_CANCEL
>
> for originate job, but originator does not cancel.
>
> and than ELS connection closes.
>
> If I do not bridge legs on early media, I can play what I want on
> receiving "CHANNEL_HANGUP" from leg B, but I want Leg-A to hear native
> progress tones from leg B
>
> Please advice
>


From anton.vazir at gmail.com  Sun May 29 13:34:14 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 29 May 2011 14:34:14 +0500
Subject: [Freeswitch-users] Please advice,
	Unable to continue dialplan if legB fails on early-media
In-Reply-To: <BANLkTi=dKkbSQU-=c15TtTCtiibmpOjFUQ@mail.gmail.com>
References: <BANLkTi=wtE39ev7Pv6DSt3-_Sww3rQ7J8w@mail.gmail.com>
	<BANLkTi=dKkbSQU-=c15TtTCtiibmpOjFUQ@mail.gmail.com>
Message-ID: <BANLkTinZ9TqW6a6q4mD_iqd55msZ0wNmOA@mail.gmail.com>

Resolved, I was not parking the leag-a on leg-b hangup (listened only
on leg-a hangup event, and missing leg-b hangup

2011/5/29 Anton VG <anton.vazir at gmail.com>:
> Resolved, ignore_early_media thing, since api call does not return
> until answered in that case, so it's interrupted by the following
> command.
>
> But still looking how to resolve leg-a hangup if leg-b fails in
> early-media stage...
>
> 2011/5/29 Anton VG <anton.vazir at gmail.com>:
>> I want to play to leg-a some 'Your party does not answer' if leg-b
>> hangs-up without answering, but can't.
>>
>> I use outbound ESL
>> When receiving a call (leg-A) I do 'originate &park' (leg-B)
>> than I do uuid_bridge A and B legs when I get early_media from leg_b
>>
>> If leg-B answers, and than hangs up - everything is OK, I can play
>> extra tones for leg-A, like: 'Your call duration, bla, bla, bla'
>> successfully.
>>
>> But, if leg-B hangs while in EARLY_MEDIA (without answering), leg A
>> hangs up too (closes ESL connection) , ignoring all variables
>>
>> hangup_after_bridge=false
>> originate_continue_on_timeout=true
>> continue_on_fail=true
>> park_after_bridge=true
>>
>> I also was trying to set
>>
>> ignore_early_media=true
>>
>> in this case I receive
>>
>> -ERR ORIGINATOR_CANCEL
>>
>> for originate job, but originator does not cancel.
>>
>> and than ELS connection closes.
>>
>> If I do not bridge legs on early media, I can play what I want on
>> receiving "CHANNEL_HANGUP" from leg B, but I want Leg-A to hear native
>> progress tones from leg B
>>
>> Please advice
>>
>


From anton.vazir at gmail.com  Sun May 29 13:53:46 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 29 May 2011 14:53:46 +0500
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTinFdKhyy2HkRLB8Dohtc5CL0vhbiA@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
	<BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>
	<BANLkTimX6S3rSZF8jALpbk0XF1+sQ0daYg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F9@cooper>
	<BANLkTinFdKhyy2HkRLB8Dohtc5CL0vhbiA@mail.gmail.com>
Message-ID: <BANLkTi=kG-Y0DfQ43Qtkt8u3D2RPji74Tw@mail.gmail.com>

Have you resolved the issue?

2011/5/25 Anton VG <anton.vazir at gmail.com>:
> Anthony, I do use the same scheme, and did not experienced the
> problem. But I do use park everywhere instead of ValetPark - can it be
> the reason?
>
> 2011/5/25 Peter Olsson <peter.olsson at visionutveckling.se>:
>> You need to provide the entire debug trace from this, nor just only after calling uuid_bridge.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Antonio Teixeira [eagle.antonio at gmail.com]
>> Skickat: den 25 maj 2011 10:32
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
>>
>> Good Morning List As Promised.
>>
>> I Have tested with G711 the problem remains and appears not be affected by the codec change.
>>
>> Here i have my debug log :
>> http://pastebin.freeswitch.org/16369
>>
>> Just a short explanation :
>>
>> the call XXXX1010 is Leg A and is currently valet_park()
>>
>> call XXXX371 is Leg B and is currently on park()
>>
>> Strangely FS says LEG A is out of order but i can hear MOH and sometimes with this exact software it connects fine.
>>
>> Hope you can help me out i will keep testing during the day.
>>
>> Regards
>> A/T
>>
>>
>> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>>
>> Ok Anthony.
>>
>> I will provide you with data regarding the use of G711 and full debug logs.
>> Will keep an eye on the logs.
>>
>> Regards
>> A/T
>>
>>
>> 2011/5/24 Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
>> The +OK only means the attempt to bridge was successful, if something
>> else goes wrong after that, you will not know because it happens
>> later.
>>
>> As suggested, look at the cause of the hangup on the failed bridge.
>>
>>
>> On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
>> <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>> wrote:
>>> Hello List , Good Morning.
>>>
>>> In my fight to get ESL & Python & Freeswitch all to behave properly i
>>> noticed a possible BUG ( need the veterans to confirm).
>>>
>>> Scenario :
>>>
>>> Originate & ValetPark () + MOH OR ?Just Park() , Both show the same problem.
>>> Codec G729
>>>
>>> I Make another call:
>>> originate & park()
>>> codec G729
>>>
>>> Now Python ESL Inbound Or FS Console :
>>>
>>> uuid_bridge uuid1 uuid2
>>>
>>> +OK uuid
>>>
>>> Now one of the two things happen :
>>>
>>> 1) One the call gets connected hurray :) , Audio Perfect ?, etc.
>>>
>>> 2) The call gets dropped :(
>>>
>>> In both cases uuid_bridge reports +OK even in the case the call is dropped.
>>> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>>> SYN/ASYNC Problems) ?I still get sometimes ( not always) a dropped call
>>> could this be related to the use of G729 ?
>>>
>>> I have lots of available licenses.
>>>
>>> Regards
>>> Ant?nio Teixeira
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> !DSPAM:4ddcbfed32761527616399!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From sireeps at gmail.com  Sun May 29 19:38:35 2011
From: sireeps at gmail.com (Kamen)
Date: Sun, 29 May 2011 08:38:35 -0700 (PDT)
Subject: [Freeswitch-users] collectInput does not call callback function
In-Reply-To: <4de166bd.81320e0a.7135.7e06@mx.google.com>
References: <1306601304658-6414679.post@n2.nabble.com>
	<4de166bd.81320e0a.7135.7e06@mx.google.com>
Message-ID: <1306683515927-6416704.post@n2.nabble.com>

Good point grmt!

But that is where I am getting confused.  What is that second argument? 
According to collectInput description in wiki (
http://wiki.freeswitch.org/wiki/Session_collectInput
http://wiki.freeswitch.org/wiki/Session_collectInput ) it is a
"callbackArguments" which in the example given is the last argument (is it?
because no more explanation is given):

function mycb( session, type, data, arg )

So I take that "dtmf" of the collectInput is arg of the callback function.

session.collectInput( mycb, "dtmf", 3000, 0)

But why is it a string type in the example?  Isn't dtmf defined as an object
and then used in callback function as an object?
arg.digits += data.digit;

At the same time in callback function example type parameter is apparently
expected to be a string "dtmf". Is it a coincidence or I am missing
something here? If collectInput second parameter is string "dtmf" then it
also looks that in callback function it is apparently a second parameter
(type). So what is it?  Is it an object dtmf which turns as arg parameter in
callback function, or it is a string which in callback function turns into
type parameter?  I took it should be a string turning into "type" of
callback function, so it does not matter what string is as long as it is the
same.  So I made it "digit".  

If I am missing something here, I'd really appreciate if someone let me
know.  I also tried it to be just an object (without quotes, in spite of the
example), or followed the example exactly.  In any case the callback is not
happening - the first line of my callback function is a console prompt,
which is not printed. Thus is my question.

Regards,

Sergei Kamen




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/collectInput-does-not-call-callback-function-tp6414679p6416704.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From cmcureau at gmail.com  Sun May 29 20:37:01 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Sun, 29 May 2011 11:37:01 -0500
Subject: [Freeswitch-users] Forcing SRTP/TLS connections
Message-ID: <BANLkTi==jP5Lpy6p0GdwWT7-SQ7gZcEk5w@mail.gmail.com>

Hi, everyone.

I've begun experimenting with SRTP and TLS connections since the ZRTP
download seems offline for the foreseeable future...sigh.

I've got the SRTP enabled in my conf/vars.xml for both internal and external
connections, but it appears that I can still connect without security from
an unconfigured ATA.  What I want to know is how to enforce a secure
connection at registration time.  Ideally, I'd like to do this both between
two freeswitch servers and between a freeswitch server and an ATA.

Thanks in advance!
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From avi at avimarcus.net  Sun May 29 22:11:13 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 29 May 2011 21:11:13 +0300
Subject: [Freeswitch-users] Forcing SRTP/TLS connections
In-Reply-To: <BANLkTi==jP5Lpy6p0GdwWT7-SQ7gZcEk5w@mail.gmail.com>
References: <BANLkTi==jP5Lpy6p0GdwWT7-SQ7gZcEk5w@mail.gmail.com>
Message-ID: <BANLkTi=QoCcSJ-h3yE1gRwWt6_higGa=qA@mail.gmail.com>

For lack of other responses so far.. these may be hacky:
1) mess with the ports - TLS and regular are on different ports. set 5060
for the tls and disable / firewall the non-tls port.
2) re: srtp you can check if it's on in the dialplan, and if not redirect to
some sort of error message instead of completing the call.
Not sure you would want to completely disable non-tls/srtp though... I
suppose it depends on your usage.

-Avi

On Sun, May 29, 2011 at 7:37 PM, Chris Cureau <cmcureau at gmail.com> wrote:

> Hi, everyone.
>
> I've begun experimenting with SRTP and TLS connections since the ZRTP
> download seems offline for the foreseeable future...sigh.
>
> I've got the SRTP enabled in my conf/vars.xml for both internal and
> external connections, but it appears that I can still connect without
> security from an unconfigured ATA.  What I want to know is how to enforce a
> secure connection at registration time.  Ideally, I'd like to do this both
> between two freeswitch servers and between a freeswitch server and an ATA.
>
> 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 garmt.noname at gmail.com  Mon May 30 00:37:44 2011
From: garmt.noname at gmail.com (Grmt)
Date: Sun, 29 May 2011 22:37:44 +0200
Subject: [Freeswitch-users] collectInput does not call callback function
In-Reply-To: <1306683515927-6416704.post@n2.nabble.com>
References: <1306601304658-6414679.post@n2.nabble.com>
	<4de166bd.81320e0a.7135.7e06@mx.google.com>
	<1306683515927-6416704.post@n2.nabble.com>
Message-ID: <4de2ae9b.0b610e0a.3fd0.1fe0@mx.google.com>

Some clarification from a quick inspection of the code:

The callback function can be a generic function and possibly be used by
other functions that specify a callbackfunction: it can also capture
"events". So callbackfunction can be called with two different types: "dtmf"
or "events". collectInput will result calling your callbackfunction with
type="dtmf". Checking for "digits" is indeed useless ...

Note that the last argument(s) of collectInput are optional.


What is that second argument?
It should be: "an object dtmf which turns as arg parameter in callback
function."

(I'm guessing a bit here, because I don't know javascript or
mod_spidermonkey, and don't know about the quotes and I didn't try it out
and I did only a quick inspection of the code).


Garmt 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kamen
Sent: Sunday, 29 May, 2011 17:39
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] collectInput does not call callback function

Good point grmt!

But that is where I am getting confused.  What is that second argument? 
According to collectInput description in wiki (
http://wiki.freeswitch.org/wiki/Session_collectInput
http://wiki.freeswitch.org/wiki/Session_collectInput ) it is a
"callbackArguments" which in the example given is the last argument (is it?
because no more explanation is given):

function mycb( session, type, data, arg )

So I take that "dtmf" of the collectInput is arg of the callback function.

session.collectInput( mycb, "dtmf", 3000, 0)

But why is it a string type in the example?  Isn't dtmf defined as an object
and then used in callback function as an object?
arg.digits += data.digit;

At the same time in callback function example type parameter is apparently
expected to be a string "dtmf". Is it a coincidence or I am missing
something here? If collectInput second parameter is string "dtmf" then it
also looks that in callback function it is apparently a second parameter
(type). So what is it?  Is it an object dtmf which turns as arg parameter in
callback function, or it is a string which in callback function turns into
type parameter?  I took it should be a string turning into "type" of
callback function, so it does not matter what string is as long as it is the
same.  So I made it "digit".  

If I am missing something here, I'd really appreciate if someone let me
know.  I also tried it to be just an object (without quotes, in spite of the
example), or followed the example exactly.  In any case the callback is not
happening - the first line of my callback function is a console prompt,
which is not printed. Thus is my question.

Regards,

Sergei Kamen




--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/collectInput-does-not-call-cal
lback-function-tp6414679p6416704.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 tculjaga at gmail.com  Mon May 30 01:44:33 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 29 May 2011 23:44:33 +0200
Subject: [Freeswitch-users] 300 message without Diversion header
In-Reply-To: <BANLkTi=bubgQXH54VviNjSzp1-QeVF6z1Q@mail.gmail.com>
References: <BANLkTinw9-XMnE9tW_bcmgJyUpZWjFj89Q@mail.gmail.com>
	<BC87F03A-02BC-4B24-89AE-202A080F0538@freeswitch.org>
	<BANLkTi=bubgQXH54VviNjSzp1-QeVF6z1Q@mail.gmail.com>
Message-ID: <BANLkTin0zesB0qAE5LWcjq_zcKjtckyHow@mail.gmail.com>

so, is ti by design or its a bug.... can someone tell ?

T:

On Fri, May 27, 2011 at 10:17 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

>
>
> On Thu, May 26, 2011 at 9:54 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Check your variables. via uuid_dump/info
>>
>
>
>
> here they are :=)
>
>
> EXECUTE sofia/external/00681038515000402 at 127.0.0.1:5061 info()
> 2011-05-26 10:05:29.230485 [INFO] mod_dptools.c:1203 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-Call-State: [RINGING]
> Channel-State-Number: [4]
> Channel-Name: [sofia/external/00681038515000402 at 127.0.0.1:5061]
> Unique-ID: [17ebc5e5-87cf-4fc9-a5b7-c69c0e682d41]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMA]
> Channel-Read-Codec-Rate: [8000]
> Channel-Read-Codec-Bit-Rate: [64000]
> Channel-Write-Codec-Name: [PCMA]
> Channel-Write-Codec-Rate: [8000]
> Channel-Write-Codec-Bit-Rate: [64000]
> Caller-Direction: [inbound]
> Caller-Username: [00681038515000402]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [00681038515000402]
> Caller-Caller-ID-Number: [00681038515000402]
> Caller-Network-Addr: [10.4.62.88]
> Caller-ANI: [00681038515000402]
> Caller-Destination-Number: [30003016094191500]
> Caller-Unique-ID: [17ebc5e5-87cf-4fc9-a5b7-c69c0e682d41]
> Caller-Source: [mod_sofia]
> Caller-Context: [public]
> Caller-Channel-Name: [sofia/external/00681038515000402 at 127.0.0.1:5061]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1306407929129882]
> Caller-Channel-Created-Time: [1306407929129882]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_direction: [inbound]
> variable_uuid: [17ebc5e5-87cf-4fc9-a5b7-c69c0e682d41]
> variable_sip_local_network_addr: [10.4.62.88]
> variable_sip_network_ip: [10.4.62.88]
> variable_sip_network_port: [5061]
> variable_sip_received_ip: [10.4.62.88]
> variable_sip_received_port: [5061]
> variable_sip_via_protocol: [udp]
> variable_sip_from_user: [00681038515000402]
> variable_sip_from_port: [5061]
> variable_sip_from_uri: [00681038515000402 at 127.0.0.1:5061]
> variable_sip_from_host: [127.0.0.1]
> variable_sip_from_user_stripped: [00681038515000402]
> variable_sip_from_tag: [3]
> variable_sofia_profile_name: [external]
> variable_sip_full_via: [SIP/2.0/UDP 127.0.0.1:5061
> ;branch=z9hG4bK-354-3-0;received=10.4.62.88]
> variable_sip_from_display: [00681038515000402]
> variable_sip_full_from: [00681038515000402 <
> sip:00681038515000402 at 127.0.0.1:5061>;tag=3]
> variable_sip_to_display: [30003016094191500]
> variable_sip_full_to: [30003016094191500 <
> sip:30003016094191500 at 10.4.62.88:5060>]
> variable_sip_req_user: [30003016094191500]
> variable_sip_req_port: [5060]
> variable_sip_req_uri: [30003016094191500 at 10.4.62.88:5060]
> variable_sip_req_host: [10.4.62.88]
> variable_sip_to_user: [30003016094191500]
> variable_sip_to_port: [5060]
> variable_sip_to_uri: [30003016094191500 at 10.4.62.88:5060]
> variable_sip_to_host: [10.4.62.88]
> variable_sip_contact_user: [00681038515000402]
> variable_sip_contact_uri: [00681038515000402 at 127.0.0.1]
> variable_sip_contact_host: [127.0.0.1]
> variable_channel_name: [sofia/external/00681038515000402 at 127.0.0.1:5061]
> variable_sip_call_id: [3-354 at 127.0.0.1]
> variable_sip_via_host: [127.0.0.1]
> variable_sip_via_port: [5061]
> variable_max_forwards: [70]
> variable_sip_h_Diversion: [<sip:030230003038516608363 at 10.4.13.11
> ;user=phone>;reason=deflection;counter=1]
> variable_switch_r_sdp: [v=0
> o=user1 53655765 2353687637 IN IP4 127.0.0.1
> s=-
> c=IN IP4 127.0.0.1
> t=0 0
> m=audio 6000 RTP/AVP 0
> a=rtpmap:0 PCMA/8000
> ]
> variable_remote_media_ip: [127.0.0.1]
> variable_remote_media_port: [6000]
> variable_sip_audio_recv_pt: [0]
> variable_sip_use_codec_name: [PCMA]
> variable_sip_use_codec_rate: [8000]
> variable_sip_use_codec_ptime: [20]
> variable_read_codec: [PCMA]
> variable_read_rate: [8000]
> variable_write_codec: [PCMA]
> variable_write_rate: [8000]
> variable_dtmf_type: [none]
> variable_endpoint_disposition: [RECEIVED]
> variable_intf: [false]
> variable_aPfx: [006810]
> variable_divertTo: [385030230003038516608363]
> variable_aNum: [38515000402]
> variable_IP_ADDR: [10.4.62.88:5060]
> variable_bPfx: [300030]
> variable_bNum: [16094191500]
> variable_caller_id_number: [006810385030230003038516608363]
> variable_my_sid: [[006810 38515000402 -> 300030 16094191500 :
> gid-68ab-5969]]
> variable_red_contact: [<sip:01002016094191500 at 10.4.62.88:5060
> ;user=phone>;q=0.99,<sip:04901016094191500 at 10.4.62.88:5060
> ;user=phone>;q=0.98,<sip:03861016094191500 at 10.4.62.88:5060
> ;user=phone>;q=0.97,<sip:00001016094191500 at 10.4.62.88:5060
> ;user=phone>;q=0.96]
>
> variable_authResult: [0]
> variable_componentStatus: [0:0]
> variable_current_application: [info]
>
>
>
>
> Invite:
>    ------------------------------------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.62.88:5060 SIP/2.0
>    Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-354-3-0
>    Max-Forwards: 70
>    Contact: <sip:00681038515000402 at 127.0.0.1>
>    Diversion: <sip:030230003038516608363 at 10.4.13.11
> ;user=phone>;reason=deflection;counter=1
>    From: 00681038515000402 <sip:00681038515000402 at 127.0.0.1:5061>;tag=3
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.62.88:5060>
>    Call-ID: 3-354 at 127.0.0.1
>    CSeq: 1 INVITE
>    Content-Type: application/sdp
>    Content-Length:   129
>
>    v=0
>    o=user1 53655765 2353687637 IN IP4 127.0.0.1
>    s=-
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMA/8000
>    ------------------------------------------------------------------------
>
>
>
> 300:
>    ------------------------------------------------------------------------
>    SIP/2.0 300 Multiple Choices
>    Via: SIP/2.0/UDP 127.0.0.1:5061
> ;branch=z9hG4bK-354-3-0;received=10.4.62.88
>    From: 00681038515000402 <sip:00681038515000402 at 127.0.0.1:5061>;tag=3
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.62.88:5060
> >;tag=N0pScX59jUjHm
>    Call-ID: 3-354 at 127.0.0.1
>    CSeq: 1 INVITE
>    Contact: <sip:01002016094191500 at 10.4.62.88:5060;user=phone>;q=0.99
>    Contact: <sip:04901016094191500 at 10.4.62.88:5060;user=phone>;q=0.98
>    Contact: <sip:03861016094191500 at 10.4.62.88:5060;user=phone>;q=0.97
>    Contact: <sip:00001016094191500 at 10.4.62.88:5060;user=phone>;q=0.96
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
>
>
>
>
>>
>> /b
>>
>> On May 26, 2011, at 2:42 PM, Tihomir Culjaga wrote:
>>
>> > hello,
>> >
>> > i got a strange issue ... I'm using FS as a redirect server (route
>> server)
>> > and it does work fine ... except if the original INVITE contains a
>> Diversion
>> > header, the same header is lost in the responding 300 or 302 message.
>> > Is this by design or its a bug ?
>> >
>> >
>> > this is a part of the code in sofia where u construct redirect response
>> > message...
>> >
>> >                if (argc > 1) {
>> >                    nua_respond(tech_pvt->nh, SIP_300_MULTIPLE_CHOICES,
>> > SIPTAG_CONTACT_STR(dest),
>> >                                TAG_IF(!zstr(extra_headers),
>> > SIPTAG_HEADER_STR(extra_headers)), TAG_END());
>> >                } else {
>> >                    nua_respond(tech_pvt->nh, SIP_302_MOVED_TEMPORARILY,
>> > SIPTAG_CONTACT_STR(dest),
>> >                                TAG_IF(!zstr(extra_headers),
>> > SIPTAG_HEADER_STR(extra_headers)), TAG_END());
>> >                }
>> >
>> >
>> > so, why the original diversion header is missing ? ... m'I missing
>> something
>> > ?
>> >
>> >
>> > also, should the 3000/302 message always contain a diversion field
>> saying
>> > the call is diverted ?
>> >
>> >
>> > Thanks for your answer,
>> > Tihomir.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 May 30 12:09:49 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 30 May 2011 08:09:49 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTi=kG-Y0DfQ43Qtkt8u3D2RPji74Tw@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
	<BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>
	<BANLkTimX6S3rSZF8jALpbk0XF1+sQ0daYg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F9@cooper>
	<BANLkTinFdKhyy2HkRLB8Dohtc5CL0vhbiA@mail.gmail.com>
	<BANLkTi=kG-Y0DfQ43Qtkt8u3D2RPji74Tw@mail.gmail.com>
Message-ID: <BANLkTi=6=vi+MXR4DMgkzZkhuO7nyN-0cw@mail.gmail.com>

Hello Anton.

Still no i? setting a tottaly different enviroment to test other variables
like network our Voice Provider and also pressure on FS.
Will keep you guys updated.


Regards
A/T

2011/5/29 Anton VG <anton.vazir at gmail.com>

> Have you resolved the issue?
>
> 2011/5/25 Anton VG <anton.vazir at gmail.com>:
> > Anthony, I do use the same scheme, and did not experienced the
> > problem. But I do use park everywhere instead of ValetPark - can it be
> > the reason?
> >
> > 2011/5/25 Peter Olsson <peter.olsson at visionutveckling.se>:
> >> You need to provide the entire debug trace from this, nor just only
> after calling uuid_bridge.
> >>
> >> /Peter
> >> ________________________________________
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Antonio Teixeira [
> eagle.antonio at gmail.com]
> >> Skickat: den 25 maj 2011 10:32
> >> Till: FreeSWITCH Users Help
> >> ?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
> >>
> >> Good Morning List As Promised.
> >>
> >> I Have tested with G711 the problem remains and appears not be affected
> by the codec change.
> >>
> >> Here i have my debug log :
> >> http://pastebin.freeswitch.org/16369
> >>
> >> Just a short explanation :
> >>
> >> the call XXXX1010 is Leg A and is currently valet_park()
> >>
> >> call XXXX371 is Leg B and is currently on park()
> >>
> >> Strangely FS says LEG A is out of order but i can hear MOH and sometimes
> with this exact software it connects fine.
> >>
> >> Hope you can help me out i will keep testing during the day.
> >>
> >> Regards
> >> A/T
> >>
> >>
> >> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com<mailto:
> eagle.antonio at gmail.com>>
> >> Ok Anthony.
> >>
> >> I will provide you with data regarding the use of G711 and full debug
> logs.
> >> Will keep an eye on the logs.
> >>
> >> Regards
> >> A/T
> >>
> >>
> >> 2011/5/24 Anthony Minessale <anthony.minessale at gmail.com<mailto:
> anthony.minessale at gmail.com>>
> >> The +OK only means the attempt to bridge was successful, if something
> >> else goes wrong after that, you will not know because it happens
> >> later.
> >>
> >> As suggested, look at the cause of the hangup on the failed bridge.
> >>
> >>
> >> On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
> >> <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>> wrote:
> >>> Hello List , Good Morning.
> >>>
> >>> In my fight to get ESL & Python & Freeswitch all to behave properly i
> >>> noticed a possible BUG ( need the veterans to confirm).
> >>>
> >>> Scenario :
> >>>
> >>> Originate & ValetPark () + MOH OR  Just Park() , Both show the same
> problem.
> >>> Codec G729
> >>>
> >>> I Make another call:
> >>> originate & park()
> >>> codec G729
> >>>
> >>> Now Python ESL Inbound Or FS Console :
> >>>
> >>> uuid_bridge uuid1 uuid2
> >>>
> >>> +OK uuid
> >>>
> >>> Now one of the two things happen :
> >>>
> >>> 1) One the call gets connected hurray :) , Audio Perfect  , etc.
> >>>
> >>> 2) The call gets dropped :(
> >>>
> >>> In both cases uuid_bridge reports +OK even in the case the call is
> dropped.
> >>> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
> >>> SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped call
> >>> could this be related to the use of G729 ?
> >>>
> >>> I have lots of available licenses.
> >>>
> >>> Regards
> >>> Ant?nio Teixeira
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com<mailto:
> MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:
> PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org<mailto:
> sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<mailto:
> googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> !DSPAM:4ddcbfed32761527616399!
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Mon May 30 13:57:49 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 30 May 2011 15:27:49 +0530
Subject: [Freeswitch-users] Interdomain calls
Message-ID: <BANLkTimZx=vROZzBcOkCftugzOQ1JF6EZg@mail.gmail.com>

Hello,


how would be the dialplan structure for interdomain calls like if, an user
from abc.com is dialing a user abc.net,
how would the dialplan look like in FS.



Also if a call from outside entity (qwe.com) calls abc.com user how to write
a dialplan for this scenario.



Regards
Sam
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From julf at julf.com  Mon May 30 14:05:55 2011
From: julf at julf.com (Johan Helsingius)
Date: Mon, 30 May 2011 12:05:55 +0200
Subject: [Freeswitch-users] Problems with wctdm / ftdm / TDM400P
Message-ID: <4DE36C03.5050103@julf.com>

Hi!

The two TDM channels I have been using on my TDM400P card
seem to have stopped working. No dialtone, no reaction to
trying to make a call.

ftdm list givews:

+OK
span: 1 (DECTTDM)
type: analog
physical_status: ok
signaling_status: UP
chan_count: 1
dialplan: XML
context: dect
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
+OK
span: 2 (DOORTDM)
type: analog
physical_status: ok
signaling_status: UP
chan_count: 1
dialplan: XML
context: doorbell
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
+OK

But immediately afterwards, I get:

2011-05-30 11:59:13.998318 [ERR] ftdm_io.c:4439 0:0: Failed to get alarms
2011-05-30 11:59:13.998318 [ERR] ftdm_io.c:5097 Failed to set channel alarms in
span DOORTDM

	Julf


From william at xofap.com  Mon May 30 14:21:38 2011
From: william at xofap.com (William Alianto)
Date: Mon, 30 May 2011 17:21:38 +0700
Subject: [Freeswitch-users] Bridging Gateway from another server
In-Reply-To: <BANLkTi=XnmTVO-DWqAdcCMY-gWCvrOZCOw@mail.gmail.com>
References: <4DDE2633.2030002@xofap.com>	<BANLkTinGsRnpx_g6V3-Lk1BZT2XwayvRXg@mail.gmail.com>	<4DDF6FFA.9060108@xofap.com>
	<BANLkTi=XnmTVO-DWqAdcCMY-gWCvrOZCOw@mail.gmail.com>
Message-ID: <4DE36FB2.5080802@xofap.com>

I found out that it's codec problem. I enabled the wrong codec, so it 
didn't connect. After I fixed the codec, the call still not successful. 
I didn't see any incoming log on FS2 either.

On 2:59, Tihomir Culjaga wrote:
> yap,
>
> can u provide a siptrace (sofia profile <your profile> siptrace on) 
> along with debug logs on both FS ?
>
> put them on pastebin.
>
> T.
>
> On Fri, May 27, 2011 at 11:33 AM, William Alianto <william at xofap.com 
> <mailto:william at xofap.com>> wrote:
>
>     I tried both scenario, but it return as "Incompatible Destination"
>     instead. Do I missed some step?
>
>
>     On 2:59, Tihomir Culjaga wrote:
>>     on FS1 machine define FS2 as gateway and bridge the call to it...
>>
>>
>>
>>     <extension name="To_FS2">
>>     <condition field="destination_number" expression="(.*)">
>>     <action application="bridge" data="sofia/gateway/FS2/$1"/>
>>     </condition>
>>     </extension>
>>
>>
>>
>>     or without creating any gateways just bridge to FS2 IP address...
>>
>>     <extension name="To_FS2">
>>     <condition field="destination_number" expression="(.*)">
>>     <action application="bridge" data="sofia/profilename/$1 at FS2_IP"/>
>>     </condition>
>>     </extension>
>>
>>
>>
>>
>>
>>     On Thu, May 26, 2011 at 12:06 PM, William Alianto
>>     <william at xofap.com <mailto:william at xofap.com>> wrote:
>>
>>         I'm trying to do an outbound scenario as following :
>>
>>         Client --> FS1 --> FS2 --> SBC
>>
>>         It maybe more simple if I add the SBC as gateway to FS1, but
>>         it only
>>         accept IP from FS2 due to IP restriction from provider. Is
>>         there any
>>         possible solution for this scenario?
>>
>>         _______________________________________________
>>         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,
>
>     William
>
>
>     _______________________________________________
>     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 gcd at i.ph  Mon May 30 15:18:59 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 30 May 2011 19:18:59 +0800
Subject: [Freeswitch-users] Problems with wctdm / ftdm / TDM400P
In-Reply-To: <4DE36C03.5050103@julf.com>
References: <4DE36C03.5050103@julf.com>
Message-ID: <BANLkTi=HjvULHKAiY+wf5bNuv0zjs64UJg@mail.gmail.com>

just to make sure. did u supply power to the interface card? it requires one
just like hard disk/optical  drives.


On Mon, May 30, 2011 at 6:05 PM, Johan Helsingius <julf at julf.com> wrote:

> Hi!
>
> The two TDM channels I have been using on my TDM400P card
> seem to have stopped working. No dialtone, no reaction to
> trying to make a call.
>
> ftdm list givews:
>
> +OK
> span: 1 (DECTTDM)
> type: analog
> physical_status: ok
> signaling_status: UP
> chan_count: 1
> dialplan: XML
> context: dect
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
> +OK
> span: 2 (DOORTDM)
> type: analog
> physical_status: ok
> signaling_status: UP
> chan_count: 1
> dialplan: XML
> context: doorbell
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
> +OK
>
> But immediately afterwards, I get:
>
> 2011-05-30 11:59:13.998318 [ERR] ftdm_io.c:4439 0:0: Failed to get alarms
> 2011-05-30 11:59:13.998318 [ERR] ftdm_io.c:5097 Failed to set channel
> alarms in
> span DOORTDM
>
>        Julf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From julf at julf.com  Mon May 30 15:59:36 2011
From: julf at julf.com (Johan Helsingius)
Date: Mon, 30 May 2011 13:59:36 +0200
Subject: [Freeswitch-users] Problems with wctdm / ftdm / TDM400P
In-Reply-To: <BANLkTi=HjvULHKAiY+wf5bNuv0zjs64UJg@mail.gmail.com>
References: <4DE36C03.5050103@julf.com>
	<BANLkTi=HjvULHKAiY+wf5bNuv0zjs64UJg@mail.gmail.com>
Message-ID: <4DE386A8.6040601@julf.com>

> just to make sure. did u supply power to the interface card? it requires one
> just like hard disk/optical  drives.

I did, and nothing has changed in the hardware setup. Or not
knowingly. Seems one of the modules might have blown,
here is an extract from dmesg:

[   10.131010] Module 0: Installed -- AUTO FXS/DPO
[   10.644001] Unable to do INITIAL ProSLIC powerup on module 1
[   11.156017] Unable to do INITIAL ProSLIC powerup on module 1
[   11.156019] Module 1: FAILED FXS (FCC)
[   11.356308] Module 2: Installed -- AUTO FXO (FCC mode)
[   11.556308] Module 3: Installed -- AUTO FXO (FCC mode)

[   11.604259] Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)

So one of the 4 modules refused to power up.

	Julf



From gcd at i.ph  Mon May 30 16:06:11 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 30 May 2011 20:06:11 +0800
Subject: [Freeswitch-users] Problems with wctdm / ftdm / TDM400P
In-Reply-To: <4DE386A8.6040601@julf.com>
References: <4DE36C03.5050103@julf.com>
	<BANLkTi=HjvULHKAiY+wf5bNuv0zjs64UJg@mail.gmail.com>
	<4DE386A8.6040601@julf.com>
Message-ID: <BANLkTimEGPXwLV8Bkuk6qENK3CetHSLyrw@mail.gmail.com>

you have 2 FXS modules. try to swap modules 0 and 1 to isolate whether the
problem is in the daughter board or on the motherboard. this is far i can
suggest. :-)


On Mon, May 30, 2011 at 7:59 PM, Johan Helsingius <julf at julf.com> wrote:

> > just to make sure. did u supply power to the interface card? it requires
> one
> > just like hard disk/optical  drives.
>
> I did, and nothing has changed in the hardware setup. Or not
> knowingly. Seems one of the modules might have blown,
> here is an extract from dmesg:
>
> [   10.131010] Module 0: Installed -- AUTO FXS/DPO
> [   10.644001] Unable to do INITIAL ProSLIC powerup on module 1
> [   11.156017] Unable to do INITIAL ProSLIC powerup on module 1
> [   11.156019] Module 1: FAILED FXS (FCC)
> [   11.356308] Module 2: Installed -- AUTO FXO (FCC mode)
> [   11.556308] Module 3: Installed -- AUTO FXO (FCC mode)
>
> [   11.604259] Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)
>
> So one of the 4 modules refused to power up.
>
>        Julf
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From julf at julf.com  Mon May 30 16:43:38 2011
From: julf at julf.com (Johan Helsingius)
Date: Mon, 30 May 2011 14:43:38 +0200
Subject: [Freeswitch-users] Problems with wctdm / ftdm / TDM400P
In-Reply-To: <BANLkTimEGPXwLV8Bkuk6qENK3CetHSLyrw@mail.gmail.com>
References: <4DE36C03.5050103@julf.com>	<BANLkTi=HjvULHKAiY+wf5bNuv0zjs64UJg@mail.gmail.com>	<4DE386A8.6040601@julf.com>
	<BANLkTimEGPXwLV8Bkuk6qENK3CetHSLyrw@mail.gmail.com>
Message-ID: <4DE390FA.6070400@julf.com>

Nandy,

> you have 2 FXS modules. try to swap modules 0 and 1 to isolate whether the
> problem is in the daughter board or on the motherboard. this is far i can
> suggest. :-)

Will try that next time I can shut down the system (it is in production
use). I verified that one of the FXS modules is working, but it could
still be a motherboard fault.

	Julf


From mitch.capper at gmail.com  Mon May 30 22:05:34 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 30 May 2011 11:05:34 -0700
Subject: [Freeswitch-users] Forcing SRTP/TLS connections
In-Reply-To: <BANLkTi==jP5Lpy6p0GdwWT7-SQ7gZcEk5w@mail.gmail.com>
References: <BANLkTi==jP5Lpy6p0GdwWT7-SQ7gZcEk5w@mail.gmail.com>
Message-ID: <BANLkTi=feKRsH4gDyaTHNY=3-spnrJtyHw@mail.gmail.com>

You may want to check out my tls patch at:

it features the option for tls_only forcing only the encrypted port to
be open: http://jira.freeswitch.org/browse/FS-3071  in your dialplan
for calls you can set sip_secure_media to true to force it to SRTP.

~Mitch

On Sun, May 29, 2011 at 9:37 AM, Chris Cureau <cmcureau at gmail.com> wrote:
> Hi, everyone.
> I've begun experimenting with SRTP and TLS connections since the ZRTP
> download seems offline for the foreseeable future...sigh.
> I've got the SRTP enabled in my conf/vars.xml for both internal and external
> connections, but it appears that I can still connect without security from
> an unconfigured ATA. ?What I want to know is how to enforce a secure
> connection at registration time. ?Ideally, I'd like to do this both between
> two freeswitch servers and between a freeswitch server and an ATA.
> Thanks in advance!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From andre.rosowski at omigos.de  Mon May 30 12:19:54 2011
From: andre.rosowski at omigos.de (=?ISO-8859-1?Q?Andr=E9_Rosowski?=)
Date: Mon, 30 May 2011 10:19:54 +0200
Subject: [Freeswitch-users] Any way to limit the ability to create
	conference rooms on demand?
Message-ID: <4DE3532A.9000004@omigos.de>

Hi there,

I'm using Bigbluebutton (online conferencing) with Freeswitch so that 
users can call a number and then have to enter the "voicebridge" number 
to be put into a voice conference. The problem, however, is that if a 
user dialed the wrong "voicebridge" number he will be put into a new 
conference thus creating one on demand as stated in the describtion of 
"mod_conference". Is there any way to manually create conference rooms 
and make them static...not allowing for new conference "rooms" to be 
created on demand.

default.xml in dialplan:

<extension name="phone conference">
<condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">
<action application="conference" data="${pin}@wideband"/>
</condition>
</extension>

Thanks for your help : )

Regards,

Realdoe

-- 
Omigos Labs UG (haftungsbeschr?nkt) - Meisenstra?e 96 - 33607 Bielefeld
Fon: +49 521 2997 200 - Fax: +49 521 2997 101
info at omigos.de - www.omigos.de
Gesch?ftsf?hrer: BSc Andr? Rosowski, Benjamin Bittner, Julian Klima
Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255
USt-IdNr.: 305/5860/1585



From infos at madovsky.org  Tue May 31 01:17:39 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 30 May 2011 17:17:39 -0400
Subject: [Freeswitch-users] null/undef variables
Message-ID: <88BD9F808CB64B14BD7CE8334333C458@e1705>

Just updated to today git
so all variables I set to null like
<action application="set" data="myvar="/>

don't work anymore.
is there now another method to set to undef/null variable ?

Thanks
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From philipp.kempgen at amooma.de  Tue May 31 02:35:21 2011
From: philipp.kempgen at amooma.de (Philipp Kempgen)
Date: Tue, 31 May 2011 00:35:21 +0200
Subject: [Freeswitch-users] null/undef variables
In-Reply-To: <88BD9F808CB64B14BD7CE8334333C458@e1705>
References: <88BD9F808CB64B14BD7CE8334333C458@e1705>
Message-ID: <4DE41BA9.5090005@amooma.de>

Madovsky wrote:
> Just updated to today git
> so all variables I set to null like
> <action application="set" data="myvar="/>
> 
> don't work anymore.
> is there now another method to set to undef/null variable ?

I'd give
	<action application="set" data="myvar=_undef_" />
or
	<action application="unset" data="myvar" />
a try.


http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset


    Philipp
-- 
http://twitter.com/kempgen



From kheimerl at cs.berkeley.edu  Tue May 31 03:14:36 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 30 May 2011 16:14:36 -0700
Subject: [Freeswitch-users] CLI Chat Command
Message-ID: <BANLkTikW2Wb5ZUTQdnS9SDNUzg-EbDUXgw@mail.gmail.com>

Hello Freeswitch-users!

I, for the life of me, cannot figure out how to send a SIP MESSAGE to
a connected client. I've pored over a variety of mailing lists
(primarily this one), wiki pages, and source code. It still doesn't
make any sense. This email contains the wide variety of ways to do
this i've seen on these media, but I still have yet to see any actual
messages sent. Any ideas would be appreciated, i've started tearing
apart mod_sofia, and I feel like that's going to take a long long
time.

I'll lead with the current status, as there's likely debugging info
there. I have just one client, IMSI641104878332498 and I want to send
a message to and from him/her from fs_cli.

freeswitch at internal> sofia status profile internal
=================================================================================================
Name             	internal
Domain Name      	N/A
Auto-NAT         	false
DBName           	sofia_reg_internal
Pres Hosts       	192.168.1.144,192.168.1.144
Dialplan         	XML
Context          	public
Challenge Realm  	auto_from
RTP-IP           	192.168.1.144
SIP-IP           	192.168.1.144
URL              	sip:mod_sofia at 192.168.1.144:5060
BIND-URL         	sip:mod_sofia at 192.168.1.144:5060
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS IN        	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT       	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	0
FAILED-CALLS-IN  	0
CALLS-OUT        	0
FAILED-CALLS-OUT 	0

Registrations:
=================================================================================================
Call-ID:    	1562658417 at 192.168.1.144
User:       	IMSI641104878332498 at 192.168.1.144
Contact:    	IMSI641104878332498 <sip:IMSI641104878332498 at 192.168.1.144:5063>
Agent:      	unknown
Status:     	Registered(UDP)(unknown) EXP(2011-05-30 17:52:00) EXPSECS(6706)
Host:       	darth-maul
IP:         	192.168.1.144
Port:       	5063
Auth-User:  	unknown
Auth-Realm: 	192.168.1.144
MWI-Account:	IMSI641104878332498 at 192.168.1.144

Total items returned: 1
=================================================================================================

Firstly, the basic item from the wiki failed:

freeswitch at internal> chat
sip|IMSI641104878332498 at 192.168.1.144|IMSI641104878332498 at 192.168.1.144|foo
Sent
nua: nh_create_handle: entering
freeswitch at internal> nua(0x7fb14c005170): creating handle 0x7fb14c03e300 failed
nua: nua_handle_bind: entering
nua: nua_message: entering
nua: nua_r_message with invalid handle (nil)

I initially hoped that the commands would map from originate, and
tried the following:

freeswitch at internal> chat
sip|user/IMSI641104878332498|user/IMSI641104878332498|foo
Error! Message Not Sent
freeswitch at internal> 2011-05-30 16:02:24.563454 [ERR]
sofia_presence.c:133 Chat proto [dp]
from [user/IMSI641104878332498]
to [user/IMSI641104878332498]
foo
Invalid Profile user

That's fine, user isn't a sip profile. Internal is though.

freeswitch at internal> chat
sip|internal/IMSI641104878332498|internal/IMSI641104878332498|foo
Error! Message Not Sent
freeswitch at internal> 2011-05-30 16:03:02.363461 [ERR]
sofia_presence.c:149 Can't find registered user
IMSI641104878332498 at internal

Why is it taking the profile and making it the host? I found this in
the source code (sofia_presence.c:129), so fine. That's on purpose. A
mailing list entry suggested adding sip: before the user, for some
reason. That seems to help:

freeswitch at internal> chat
sip|internal/sip:IMSI641104878332498|internal/sip:IMSI641104878332498|foo
Sent

or

freeswitch at internal> chat
sip|internal/sip:IMSI641104878332498 at 192.168.1.144|internal/sip:IMSI641104878332498 at 192.168.1.144|foo
Sent

Success...   but not really. No message goes out (confirmed via
wireshark). Instead, nua fails for both of these:
nta outgoing create: invalid URI

What URI? My last option is to hard code everything:

freeswitch at internal> chat
sip|internal/IMSI641104878332498 at 192.168.1.144|internal/IMSI641104878332498 at 192.168.1.144|foo
Sent

Nua fails somewhere different with that one:
nua(0x7fb14c005170): creating handle 0x18577c0 failed
nua: nua_handle_bind: entering
nua: nua_message: entering
nua: nua_r_message with invalid handle (nil)

Lastly, adding the actual port only breaks things worse:

freeswitch at internal> chat
sip|internal/IMSI641104878332498 at 192.168.1.144:5063|internal/IMSI641104878332498 at 192.168.1.144:5063|foo
Error! Message Not Sent
2011-05-30 16:11:47.823468 [ERR] sofia_presence.c:149 Can't find
registered user IMSI641104878332498 at 192.168.1.144:5063

WHAT IS GOING ON?

Any help would be appreciated!


From infos at madovsky.org  Tue May 31 08:19:44 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 31 May 2011 00:19:44 -0400
Subject: [Freeswitch-users] null/undef variables
References: <88BD9F808CB64B14BD7CE8334333C458@e1705>
	<4DE41BA9.5090005@amooma.de>
Message-ID: <1336A6906918468EB7CF4061AA5580E7@e1705>

thanks Philipp,

_undef_ is considered as a string..
I forgot to mention that the var
is initiated with inline in an extension,
and can be modified after in another
with empty value. I tried also to set all
the var as inline but doesn't make sense..

Thanks

----- Original Message ----- 
From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, May 30, 2011 6:35 PM
Subject: Re: [Freeswitch-users] null/undef variables


> Madovsky wrote:
>> Just updated to today git
>> so all variables I set to null like
>> <action application="set" data="myvar="/>
>>
>> don't work anymore.
>> is there now another method to set to undef/null variable ?
>
> I'd give
> <action application="set" data="myvar=_undef_" />
> or
> <action application="unset" data="myvar" />
> a try.
>
>
> http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
>
>
>    Philipp
> -- 
> http://twitter.com/kempgen
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From steveayre at gmail.com  Tue May 31 10:34:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 May 2011 07:34:36 +0100
Subject: [Freeswitch-users] null/undef variables
In-Reply-To: <1336A6906918468EB7CF4061AA5580E7@e1705>
References: <88BD9F808CB64B14BD7CE8334333C458@e1705>
	<4DE41BA9.5090005@amooma.de>
	<1336A6906918468EB7CF4061AA5580E7@e1705>
Message-ID: <BANLkTinngeEHKRE3jR-sx1Y1GQDX94E=ag@mail.gmail.com>

__undef__ (2 underscores either side) only I think affects caller id.

Did you try the unset app?

-Steve



On 31 May 2011 05:19, Madovsky <infos at madovsky.org> wrote:

> thanks Philipp,
>
> _undef_ is considered as a string..
> I forgot to mention that the var
> is initiated with inline in an extension,
> and can be modified after in another
> with empty value. I tried also to set all
> the var as inline but doesn't make sense..
>
> Thanks
>
> ----- Original Message -----
> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
> To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, May 30, 2011 6:35 PM
> Subject: Re: [Freeswitch-users] null/undef variables
>
>
> > Madovsky wrote:
> >> Just updated to today git
> >> so all variables I set to null like
> >> <action application="set" data="myvar="/>
> >>
> >> don't work anymore.
> >> is there now another method to set to undef/null variable ?
> >
> > I'd give
> > <action application="set" data="myvar=_undef_" />
> > or
> > <action application="unset" data="myvar" />
> > a try.
> >
> >
> >
> http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
> >
> >
> >    Philipp
> > --
> > http://twitter.com/kempgen
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Tue May 31 10:42:34 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 31 May 2011 02:42:34 -0400
Subject: [Freeswitch-users] null/undef variables
References: <88BD9F808CB64B14BD7CE8334333C458@e1705><4DE41BA9.5090005@amooma.de><1336A6906918468EB7CF4061AA5580E7@e1705>
	<BANLkTinngeEHKRE3jR-sx1Y1GQDX94E=ag@mail.gmail.com>
Message-ID: <EE067AD842194494BEA94AB379D7F991@e1705>

ooops, ok I used one underscore.
yes I tried unset after the inline but doesn't work.
should I set unset inline also ?
nevermind I changed the dialplan to different way.

Thanks
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, May 31, 2011 2:34 AM
  Subject: Re: [Freeswitch-users] null/undef variables


  __undef__ (2 underscores either side) only I think affects caller id.

  Did you try the unset app?

  -Steve




  On 31 May 2011 05:19, Madovsky <infos at madovsky.org> wrote:

    thanks Philipp,

    _undef_ is considered as a string..
    I forgot to mention that the var
    is initiated with inline in an extension,
    and can be modified after in another
    with empty value. I tried also to set all
    the var as inline but doesn't make sense..

    Thanks


    ----- Original Message -----
    From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
    To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
    Sent: Monday, May 30, 2011 6:35 PM
    Subject: Re: [Freeswitch-users] null/undef variables


    > Madovsky wrote:
    >> Just updated to today git
    >> so all variables I set to null like
    >> <action application="set" data="myvar="/>
    >>
    >> don't work anymore.
    >> is there now another method to set to undef/null variable ?
    >
    > I'd give
    > <action application="set" data="myvar=_undef_" />
    > or
    > <action application="unset" data="myvar" />
    > a try.
    >
    >
    > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
    > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
    >
    >
    >    Philipp
    > --
    > http://twitter.com/kempgen
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org


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




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


  _______________________________________________
  FreeSWITCH-users mailing 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 steveayre at gmail.com  Tue May 31 11:26:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 May 2011 08:26:41 +0100
Subject: [Freeswitch-users] null/undef variables
In-Reply-To: <EE067AD842194494BEA94AB379D7F991@e1705>
References: <88BD9F808CB64B14BD7CE8334333C458@e1705>
	<4DE41BA9.5090005@amooma.de>
	<1336A6906918468EB7CF4061AA5580E7@e1705>
	<BANLkTinngeEHKRE3jR-sx1Y1GQDX94E=ag@mail.gmail.com>
	<EE067AD842194494BEA94AB379D7F991@e1705>
Message-ID: <BANLkTiks+L1fcG6C_E_jqVOAo9feD=o82w@mail.gmail.com>

>
> should I set unset inline also ?
>

Yes, try that. It does support inline.

-Steve


On 31 May 2011 07:42, Madovsky <infos at madovsky.org> wrote:

>  ooops, ok I used one underscore.
> yes I tried unset after the inline but doesn't work.
> should I set unset inline also ?
> nevermind I changed the dialplan to different way.
>
> Thanks
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, May 31, 2011 2:34 AM
> *Subject:* Re: [Freeswitch-users] null/undef variables
>
> __undef__ (2 underscores either side) only I think affects caller id.
>
> Did you try the unset app?
>
> -Steve
>
>
>
> On 31 May 2011 05:19, Madovsky <infos at madovsky.org> wrote:
>
>> thanks Philipp,
>>
>> _undef_ is considered as a string..
>> I forgot to mention that the var
>> is initiated with inline in an extension,
>> and can be modified after in another
>> with empty value. I tried also to set all
>> the var as inline but doesn't make sense..
>>
>> Thanks
>>
>> ----- Original Message -----
>> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
>> To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, May 30, 2011 6:35 PM
>> Subject: Re: [Freeswitch-users] null/undef variables
>>
>>
>> > Madovsky wrote:
>> >> Just updated to today git
>> >> so all variables I set to null like
>> >> <action application="set" data="myvar="/>
>> >>
>> >> don't work anymore.
>> >> is there now another method to set to undef/null variable ?
>> >
>> > I'd give
>> > <action application="set" data="myvar=_undef_" />
>> > or
>> > <action application="unset" data="myvar" />
>> > a try.
>> >
>> >
>> >
>> http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
>> >
>> >
>> >    Philipp
>> > --
>> > http://twitter.com/kempgen
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pooja27taneja at gmail.com  Tue May 31 12:53:56 2011
From: pooja27taneja at gmail.com (pooja)
Date: Tue, 31 May 2011 01:53:56 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <576485B0-53B9-4B03-A09E-8A6D76BA6CE5@gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306520931787-6412076.post@n2.nabble.com>
	<576485B0-53B9-4B03-A09E-8A6D76BA6CE5@gmail.com>
Message-ID: <1306832036825-6421915.post@n2.nabble.com>

okk.. can anyone else please help..??

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6421915.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sid.kshatriya at gmail.com  Tue May 31 13:49:44 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Tue, 31 May 2011 15:19:44 +0530
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <1306832036825-6421915.post@n2.nabble.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306520931787-6412076.post@n2.nabble.com>
	<576485B0-53B9-4B03-A09E-8A6D76BA6CE5@gmail.com>
	<1306832036825-6421915.post@n2.nabble.com>
Message-ID: <BANLkTikozhRaq-VgL+8Kt_K-tC_Th-QB3Q@mail.gmail.com>

Try playing your mp4 in a player. There seems to be something wrong with
it... or try another one -- follow the error message :-)

On Tue, May 31, 2011 at 2:23 PM, pooja <pooja27taneja at gmail.com> wrote:

> okk.. can anyone else please help..??
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6421915.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
>



-- 
Sidharth Kshatriya
www.sidk.info
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From Nabble at slickdeals.endjunk.com  Tue May 31 15:27:56 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 31 May 2011 04:27:56 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
Message-ID: <1306841276439-6422324.post@n2.nabble.com>


Chris Rienzo wrote:
> If it's not installed, then install it:
> 1. Edit modules.conf and add applications/mod_mp4 to it.
I took a look at the build/modules.conf.in file on my local FS git
repository (43a5af7df6089a258e7c2a143c2982ec0f8accbb) and did not find any
mention of mod_mp4. As such, the compilation generated the module.conf file
which has no mention of mod_mp4 in it. Does that mean one needs to manually
edit and add applications/mod_mp4 to the build/modules.conf.in file in order
to compile mod_mp4?

-----
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/how-to-use-mod-mp4-tp6410226p6422324.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chrisg.lists at gmail.com  Tue May 31 15:29:39 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Tue, 31 May 2011 13:29:39 +0200
Subject: [Freeswitch-users] xml_curl xml tag error
Message-ID: <BANLkTimM9D92RLa9Kzw=LxQmVeugu_19Cw@mail.gmail.com>

Hi List,

I am getting the malformed XML errror below:

2011-05-31 13:24:05.161809 [INFO] mod_dialplan_xml.c:331 Processing
1000 <1000>->0112341111 in context lcr_trunks
2011-05-31 13:24:05.182614 [ERR] switch_xml.c:1611 Error[[error near
line 20]: unexpected closing tag </document>]
2011-05-31 13:24:05.182614 [WARNING] mod_dialplan_xml.c:361 Context
lcr_trunks not found
2011-05-31 13:24:05.182614 [INFO] switch_core_state_machine.c:142 No
Route, Aborting


My static XML file being called:
<?xml version="1.0" encoding="utf-8"?>
<include>
        <context name="default" />
</include>

My dynamic XML being served up by xml_curl:
<?xml version="1.0" encoding="UTF-8" standalone="no"?><document
type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
    <context name="lcr_trunks">
      <extension name="Catch carrier test">
        <condition field="destination_number" expression="^(.*)$">
          <action application="lcr" data="$1 peak"/>
          <action application="bridge" data="${lcr_auto_route}"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

Why is the dynamic XML having an issue with "</document>" tag? Looks
legal to me?

Thanks in advance,
Chris


From chrisg.lists at gmail.com  Tue May 31 15:34:16 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Tue, 31 May 2011 13:34:16 +0200
Subject: [Freeswitch-users] xml_curl xml tag error
In-Reply-To: <BANLkTimM9D92RLa9Kzw=LxQmVeugu_19Cw@mail.gmail.com>
References: <BANLkTimM9D92RLa9Kzw=LxQmVeugu_19Cw@mail.gmail.com>
Message-ID: <BANLkTinX5uM2eFBC0kd31TCmc6RyheEQAQ@mail.gmail.com>

Apologies, the error I pasted was incorrect. Corrected it should read
line 11 and not line 20 as below:

2011-05-31 13:32:09.392074 [INFO] mod_dialplan_xml.c:331 Processing
1000 <1000>->0112341111 in context lcr_trunks
2011-05-31 13:32:09.432080 [ERR] switch_xml.c:1611 Error[[error near
line 11]: unexpected closing tag </document>]
2011-05-31 13:32:09.432080 [WARNING] mod_dialplan_xml.c:361 Context
lcr_trunks not found
2011-05-31 13:32:09.432080 [INFO] switch_core_state_machine.c:142 No
Route, Aborting

Chris



On Tue, May 31, 2011 at 1:29 PM, Chris Graham <chrisg.lists at gmail.com> wrote:
> Hi List,
>
> I am getting the malformed XML errror below:
>
> 2011-05-31 13:24:05.161809 [INFO] mod_dialplan_xml.c:331 Processing
> 1000 <1000>->0112341111 in context lcr_trunks
> 2011-05-31 13:24:05.182614 [ERR] switch_xml.c:1611 Error[[error near
> line 20]: unexpected closing tag </document>]
> 2011-05-31 13:24:05.182614 [WARNING] mod_dialplan_xml.c:361 Context
> lcr_trunks not found
> 2011-05-31 13:24:05.182614 [INFO] switch_core_state_machine.c:142 No
> Route, Aborting
>
>
> My static XML file being called:
> <?xml version="1.0" encoding="utf-8"?>
> <include>
> ? ? ? ?<context name="default" />
> </include>
>
> My dynamic XML being served up by xml_curl:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?><document
> type="freeswitch/xml">
> ?<section name="dialplan" description="RE Dial Plan For FreeSwitch">
> ? ?<context name="lcr_trunks">
> ? ? ?<extension name="Catch carrier test">
> ? ? ? ?<condition field="destination_number" expression="^(.*)$">
> ? ? ? ? ?<action application="lcr" data="$1 peak"/>
> ? ? ? ? ?<action application="bridge" data="${lcr_auto_route}"/>
> ? ? ? ?</condition>
> ? ? ?</extension>
> ? ?</context>
> ?</section>
> </document>
>
> Why is the dynamic XML having an issue with "</document>" tag? Looks
> legal to me?
>
> Thanks in advance,
> Chris
>


From steveayre at gmail.com  Tue May 31 16:26:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 May 2011 13:26:39 +0100
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <1306841276439-6422324.post@n2.nabble.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
Message-ID: <BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>

bootstrap.sh will generate modules.conf from modules.conf.in

At that point it contains the default modules to be built. You can then
customise the modules.conf file to add your own module choices or remove
some of the default ones.

-Steve


On 31 May 2011 12:27, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Chris Rienzo wrote:
> > If it's not installed, then install it:
> > 1. Edit modules.conf and add applications/mod_mp4 to it.
> I took a look at the build/modules.conf.in file on my local FS git
> repository (43a5af7df6089a258e7c2a143c2982ec0f8accbb) and did not find any
> mention of mod_mp4. As such, the compilation generated the module.conf file
> which has no mention of mod_mp4 in it. Does that mean one needs to manually
> edit and add applications/mod_mp4 to the build/modules.conf.in file in
> order
> to compile mod_mp4?
>
> -----
> 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/how-to-use-mod-mp4-tp6410226p6422324.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 peter.olsson at visionutveckling.se  Tue May 31 17:07:39 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 31 May 2011 15:07:39 +0200
Subject: [Freeswitch-users] connecting freeswitch to a panasonic tde 100
In-Reply-To: <BANLkTim9wDB44mNs17-rqbJhk0Mp-FojEQ@mail.gmail.com>
References: <BANLkTi=-0Go65iTAGDB2U2ENZpon3KLoXQ@mail.gmail.com>
	<BANLkTik8DCecH_JE7p7J=59NiK6SB18UFw@mail.gmail.com>
	<BANLkTi=MmoGxQbpGPfKWJHYQM6PWuTnKYw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B630C@cooper>
	<BANLkTim9wDB44mNs17-rqbJhk0Mp-FojEQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE549238@cooper>

Sounds like you want to be able to do recording of calls?

Only thing I know in this PBX is how to configure a SIP trunk to work for FreeSWITCH (and to dial that trunk via a prefix), but the rest is dialplan stuff inside the PBX - and I don't know much of that. If you need all calls to get through FS though, I would guess you will need to register the actual phones on different number then users actually dial, and then route the "real" number to FS instead, that can route them back to the Panasonic PBX.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r babak yakhchali
Skickat: den 29 maj 2011 11:05
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] connecting freeswitch to a panasonic tde 100

Thanks Peter, this is the situation:
            +--------------------------------------------+
            |           Panasonic KX-TDE 100             |
            |                                            |
            |    +----------+          +-----------+     |
CO Lines <--|--->|  LCOT16  |          |  CSLC16   |<----|---->  Phones
            |    +----------+          +-----------+     |
            |         ^                      ^           |
            |         |                      |           |
            |         v                      v           |
            |      +------------------------------+      |
            |      |          V-SIPGW16           |      |
            |      +------------------------------+      |
            |                      ^                     |
            |                      |                     |
            +----------------------|---------------------+
                                   |
                                   v
                          +----------------+
                          |   Fresswitch   |
                          +----------------+
I need to process all calls in freeswitch. so all calls from pstn (comming from lcot16 lines) and  phones (attached to CSLC16) should go to freeswitch and fs should be able to distribute them (weather to pstn or phones). If you are on IRC and of course if you like let me know to contact you there.

!DSPAM:4de20d9532768868113018!
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From jeff at jefflenk.com  Tue May 31 18:19:26 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 31 May 2011 07:19:26 -0700 (PDT)
Subject: [Freeswitch-users] CLI Chat Command
In-Reply-To: <BANLkTikW2Wb5ZUTQdnS9SDNUzg-EbDUXgw@mail.gmail.com>
References: <BANLkTikW2Wb5ZUTQdnS9SDNUzg-EbDUXgw@mail.gmail.com>
Message-ID: <1306851566813-6422915.post@n2.nabble.com>

Are you using the latest code? There was a fix submitted in the last week or
so with that.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/CLI-Chat-Command-tp6420849p6422915.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue May 31 19:17:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 31 May 2011 10:17:39 -0500
Subject: [Freeswitch-users] null/undef variables
In-Reply-To: <BANLkTiks+L1fcG6C_E_jqVOAo9feD=o82w@mail.gmail.com>
References: <88BD9F808CB64B14BD7CE8334333C458@e1705>
	<4DE41BA9.5090005@amooma.de>
	<1336A6906918468EB7CF4061AA5580E7@e1705>
	<BANLkTinngeEHKRE3jR-sx1Y1GQDX94E=ag@mail.gmail.com>
	<EE067AD842194494BEA94AB379D7F991@e1705>
	<BANLkTiks+L1fcG6C_E_jqVOAo9feD=o82w@mail.gmail.com>
Message-ID: <BANLkTi=RB3QcuO6Xsn-PuZa4eUN=ZG+zqw@mail.gmail.com>

Madovsky,

Please explain why you stubbornly refuse to use JIRA.
You found a real bug.  If I was not paying attention maybe I never
would have read your email.
This is why we have JIRA.....

For goodness sake...... please....


On Tue, May 31, 2011 at 2:26 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> should I set unset inline also ?
>
> Yes, try that. It does support inline.
>
> -Steve
>
>
> On 31 May 2011 07:42, Madovsky <infos at madovsky.org> wrote:
>>
>> ooops, ok I used one underscore.
>> yes I tried unset after the inline but doesn't work.
>> should I set unset inline also ?
>> nevermind I changed the dialplan to different way.
>>
>> Thanks
>>
>> ----- Original Message -----
>> From: Steven Ayre
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, May 31, 2011 2:34 AM
>> Subject: Re: [Freeswitch-users] null/undef variables
>> __undef__ (2 underscores either side) only I think affects caller id.
>>
>> Did you try the unset app?
>>
>> -Steve
>>
>>
>>
>> On 31 May 2011 05:19, Madovsky <infos at madovsky.org> wrote:
>>>
>>> thanks Philipp,
>>>
>>> _undef_ is considered as a string..
>>> I forgot to mention that the var
>>> is initiated with inline in an extension,
>>> and can be modified after in another
>>> with empty value. I tried also to set all
>>> the var as inline but doesn't make sense..
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
>>> To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, May 30, 2011 6:35 PM
>>> Subject: Re: [Freeswitch-users] null/undef variables
>>>
>>>
>>> > Madovsky wrote:
>>> >> Just updated to today git
>>> >> so all variables I set to null like
>>> >> <action application="set" data="myvar="/>
>>> >>
>>> >> don't work anymore.
>>> >> is there now another method to set to undef/null variable ?
>>> >
>>> > I'd give
>>> > <action application="set" data="myvar=_undef_" />
>>> > or
>>> > <action application="unset" data="myvar" />
>>> > a try.
>>> >
>>> >
>>> >
>>> > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
>>> >
>>> >
>>> > ? ?Philipp
>>> > --
>>> > http://twitter.com/kempgen
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 jmoran at secureachsystems.com  Tue May 31 19:24:22 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Tue, 31 May 2011 11:24:22 -0400
Subject: [Freeswitch-users] unimrcp version and setup
Message-ID: <361E98F99D3CC3439EED59BC1924ED69507BB4@SERVER2003.SecuReachSystems.local>

We need to move into better TTS than flite, so we've decided we should
look into mod_unimrcp.

 

As far as setting up unimrcp server itself, are there any "gotchas" or
recommendations?

We were looking to install UniMRCP Release 1.0.0 r1725 for linux along
with unimrcp-deps-1.0.0 Dependency Package 1.0.0 (APR, Sofia-SIP). Are
those the most up-to-date versions that we should be looking to install?
I noticed a newer dependency package, but no newer UniMRCP release after
that June 2010 1.0.0 version.

 

As for mrcp engines, we are thinking about IVONA, Cepstral, Acapela, or
Loquendo. Do each of these engines supply their own mrcp engine that
somehow gets installed and registered with uniMRCP?

 

-Jason Moran

 

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From infos at madovsky.org  Tue May 31 19:30:12 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 31 May 2011 11:30:12 -0400
Subject: [Freeswitch-users] null/undef variables
References: <88BD9F808CB64B14BD7CE8334333C458@e1705><4DE41BA9.5090005@amooma.de><1336A6906918468EB7CF4061AA5580E7@e1705><BANLkTinngeEHKRE3jR-sx1Y1GQDX94E=ag@mail.gmail.com><EE067AD842194494BEA94AB379D7F991@e1705><BANLkTiks+L1fcG6C_E_jqVOAo9feD=o82w@mail.gmail.com>
	<BANLkTi=RB3QcuO6Xsn-PuZa4eUN=ZG+zqw@mail.gmail.com>
Message-ID: <949516EF447B4EB2A5C6DC38F7FDF141@e1705>

Hi Tony,

I use Jira when I'm sure I don't any code mistakes.
I estimated the problem below too obvious to create a Jira, don't you ?
but If you think it can be a bug I will open a Jira.

Thanks

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, May 31, 2011 11:17 AM
Subject: Re: [Freeswitch-users] null/undef variables


Madovsky,

Please explain why you stubbornly refuse to use JIRA.
You found a real bug.  If I was not paying attention maybe I never
would have read your email.
This is why we have JIRA.....

For goodness sake...... please....


On Tue, May 31, 2011 at 2:26 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> should I set unset inline also ?
>
> Yes, try that. It does support inline.
>
> -Steve
>
>
> On 31 May 2011 07:42, Madovsky <infos at madovsky.org> wrote:
>>
>> ooops, ok I used one underscore.
>> yes I tried unset after the inline but doesn't work.
>> should I set unset inline also ?
>> nevermind I changed the dialplan to different way.
>>
>> Thanks
>>
>> ----- Original Message -----
>> From: Steven Ayre
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, May 31, 2011 2:34 AM
>> Subject: Re: [Freeswitch-users] null/undef variables
>> __undef__ (2 underscores either side) only I think affects caller id.
>>
>> Did you try the unset app?
>>
>> -Steve
>>
>>
>>
>> On 31 May 2011 05:19, Madovsky <infos at madovsky.org> wrote:
>>>
>>> thanks Philipp,
>>>
>>> _undef_ is considered as a string..
>>> I forgot to mention that the var
>>> is initiated with inline in an extension,
>>> and can be modified after in another
>>> with empty value. I tried also to set all
>>> the var as inline but doesn't make sense..
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
>>> To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, May 30, 2011 6:35 PM
>>> Subject: Re: [Freeswitch-users] null/undef variables
>>>
>>>
>>> > Madovsky wrote:
>>> >> Just updated to today git
>>> >> so all variables I set to null like
>>> >> <action application="set" data="myvar="/>
>>> >>
>>> >> don't work anymore.
>>> >> is there now another method to set to undef/null variable ?
>>> >
>>> > I'd give
>>> > <action application="set" data="myvar=_undef_" />
>>> > or
>>> > <action application="unset" data="myvar" />
>>> > a try.
>>> >
>>> >
>>> >
>>> > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
>>> >
>>> >
>>> > Philipp
>>> > --
>>> > http://twitter.com/kempgen
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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  Tue May 31 19:51:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 31 May 2011 10:51:52 -0500
Subject: [Freeswitch-users] null/undef variables
In-Reply-To: <949516EF447B4EB2A5C6DC38F7FDF141@e1705>
References: <88BD9F808CB64B14BD7CE8334333C458@e1705>
	<4DE41BA9.5090005@amooma.de>
	<1336A6906918468EB7CF4061AA5580E7@e1705>
	<BANLkTinngeEHKRE3jR-sx1Y1GQDX94E=ag@mail.gmail.com>
	<EE067AD842194494BEA94AB379D7F991@e1705>
	<BANLkTiks+L1fcG6C_E_jqVOAo9feD=o82w@mail.gmail.com>
	<BANLkTi=RB3QcuO6Xsn-PuZa4eUN=ZG+zqw@mail.gmail.com>
	<949516EF447B4EB2A5C6DC38F7FDF141@e1705>
Message-ID: <BANLkTi=4gqWu1eATJ2-uV11zCurbaDc0vw@mail.gmail.com>

its already fixed

On Tue, May 31, 2011 at 10:30 AM, Madovsky <infos at madovsky.org> wrote:
> Hi Tony,
>
> I use Jira when I'm sure I don't any code mistakes.
> I estimated the problem below too obvious to create a Jira, don't you ?
> but If you think it can be a bug I will open a Jira.
>
> Thanks
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, May 31, 2011 11:17 AM
> Subject: Re: [Freeswitch-users] null/undef variables
>
>
> Madovsky,
>
> Please explain why you stubbornly refuse to use JIRA.
> You found a real bug. ?If I was not paying attention maybe I never
> would have read your email.
> This is why we have JIRA.....
>
> For goodness sake...... please....
>
>
> On Tue, May 31, 2011 at 2:26 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>> should I set unset inline also ?
>>
>> Yes, try that. It does support inline.
>>
>> -Steve
>>
>>
>> On 31 May 2011 07:42, Madovsky <infos at madovsky.org> wrote:
>>>
>>> ooops, ok I used one underscore.
>>> yes I tried unset after the inline but doesn't work.
>>> should I set unset inline also ?
>>> nevermind I changed the dialplan to different way.
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> From: Steven Ayre
>>> To: FreeSWITCH Users Help
>>> Sent: Tuesday, May 31, 2011 2:34 AM
>>> Subject: Re: [Freeswitch-users] null/undef variables
>>> __undef__ (2 underscores either side) only I think affects caller id.
>>>
>>> Did you try the unset app?
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 31 May 2011 05:19, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> thanks Philipp,
>>>>
>>>> _undef_ is considered as a string..
>>>> I forgot to mention that the var
>>>> is initiated with inline in an extension,
>>>> and can be modified after in another
>>>> with empty value. I tried also to set all
>>>> the var as inline but doesn't make sense..
>>>>
>>>> Thanks
>>>>
>>>> ----- Original Message -----
>>>> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
>>>> To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, May 30, 2011 6:35 PM
>>>> Subject: Re: [Freeswitch-users] null/undef variables
>>>>
>>>>
>>>> > Madovsky wrote:
>>>> >> Just updated to today git
>>>> >> so all variables I set to null like
>>>> >> <action application="set" data="myvar="/>
>>>> >>
>>>> >> don't work anymore.
>>>> >> is there now another method to set to undef/null variable ?
>>>> >
>>>> > I'd give
>>>> > <action application="set" data="myvar=_undef_" />
>>>> > or
>>>> > <action application="unset" data="myvar" />
>>>> > a try.
>>>> >
>>>> >
>>>> >
>>>> > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
>>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
>>>> >
>>>> >
>>>> > Philipp
>>>> > --
>>>> > http://twitter.com/kempgen
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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 Nabble at slickdeals.endjunk.com  Tue May 31 19:57:31 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 31 May 2011 08:57:31 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
Message-ID: <1306857451501-6423364.post@n2.nabble.com>


Steven Ayre wrote:
> 
> bootstrap.sh will generate modules.conf from modules.conf.in
I just don't see bootstrap.sh will create modules.conf from
build/modules.conf.in to contain applications/mod_mp4 if
build/modules.conf.in doesn't have applications/mod_mp4 in it.

-----
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/how-to-use-mod-mp4-tp6410226p6423364.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Tue May 31 20:01:50 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 31 May 2011 12:01:50 -0400
Subject: [Freeswitch-users] null/undef variables
References: <88BD9F808CB64B14BD7CE8334333C458@e1705><4DE41BA9.5090005@amooma.de><1336A6906918468EB7CF4061AA5580E7@e1705><BANLkTinngeEHKRE3jR-sx1Y1GQDX94E=ag@mail.gmail.com><EE067AD842194494BEA94AB379D7F991@e1705><BANLkTiks+L1fcG6C_E_jqVOAo9feD=o82w@mail.gmail.com><BANLkTi=RB3QcuO6Xsn-PuZa4eUN=ZG+zqw@mail.gmail.com><949516EF447B4EB2A5C6DC38F7FDF141@e1705>
	<BANLkTi=4gqWu1eATJ2-uV11zCurbaDc0vw@mail.gmail.com>
Message-ID: <C157823FA7D94468B857A943220386CE@e1705>

:), ok thanks Tony,
next time I will submit  a Jira... sorry for that

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, May 31, 2011 11:51 AM
Subject: Re: [Freeswitch-users] null/undef variables


its already fixed

On Tue, May 31, 2011 at 10:30 AM, Madovsky <infos at madovsky.org> wrote:
> Hi Tony,
>
> I use Jira when I'm sure I don't any code mistakes.
> I estimated the problem below too obvious to create a Jira, don't you ?
> but If you think it can be a bug I will open a Jira.
>
> Thanks
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, May 31, 2011 11:17 AM
> Subject: Re: [Freeswitch-users] null/undef variables
>
>
> Madovsky,
>
> Please explain why you stubbornly refuse to use JIRA.
> You found a real bug. If I was not paying attention maybe I never
> would have read your email.
> This is why we have JIRA.....
>
> For goodness sake...... please....
>
>
> On Tue, May 31, 2011 at 2:26 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>> should I set unset inline also ?
>>
>> Yes, try that. It does support inline.
>>
>> -Steve
>>
>>
>> On 31 May 2011 07:42, Madovsky <infos at madovsky.org> wrote:
>>>
>>> ooops, ok I used one underscore.
>>> yes I tried unset after the inline but doesn't work.
>>> should I set unset inline also ?
>>> nevermind I changed the dialplan to different way.
>>>
>>> Thanks
>>>
>>> ----- Original Message -----
>>> From: Steven Ayre
>>> To: FreeSWITCH Users Help
>>> Sent: Tuesday, May 31, 2011 2:34 AM
>>> Subject: Re: [Freeswitch-users] null/undef variables
>>> __undef__ (2 underscores either side) only I think affects caller id.
>>>
>>> Did you try the unset app?
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 31 May 2011 05:19, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> thanks Philipp,
>>>>
>>>> _undef_ is considered as a string..
>>>> I forgot to mention that the var
>>>> is initiated with inline in an extension,
>>>> and can be modified after in another
>>>> with empty value. I tried also to set all
>>>> the var as inline but doesn't make sense..
>>>>
>>>> Thanks
>>>>
>>>> ----- Original Message -----
>>>> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
>>>> To: "FreeSwitch Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, May 30, 2011 6:35 PM
>>>> Subject: Re: [Freeswitch-users] null/undef variables
>>>>
>>>>
>>>> > Madovsky wrote:
>>>> >> Just updated to today git
>>>> >> so all variables I set to null like
>>>> >> <action application="set" data="myvar="/>
>>>> >>
>>>> >> don't work anymore.
>>>> >> is there now another method to set to undef/null variable ?
>>>> >
>>>> > I'd give
>>>> > <action application="set" data="myvar=_undef_" />
>>>> > or
>>>> > <action application="unset" data="myvar" />
>>>> > a try.
>>>> >
>>>> >
>>>> >
>>>> > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Setting_a_variable_as_undefined
>>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_unset
>>>> >
>>>> >
>>>> > Philipp
>>>> > --
>>>> > http://twitter.com/kempgen
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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 



From eagle.antonio at gmail.com  Tue May 31 20:08:26 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 31 May 2011 16:08:26 +0000
Subject: [Freeswitch-users] unimrcp version and setup
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED69507BB4@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED69507BB4@SERVER2003.SecuReachSystems.local>
Message-ID: <BANLkTikqGvaT4hJE07M8pKPqUpFuyZ4Grg@mail.gmail.com>

Hello Jason.
I'm a heavy Loquendo User , so here is my input.

Loquendo installs an MRCPV2 or V1 server , than you can configure unimrcp to
use the Loquendo MRCP server ip and well thats it , its highly straight
forward process , regarding the Unimrcp version i simply use the one that
comes with FS.

The process is that you use speak in your TTS , FS will send a SIP INVITE to
Loquendo MRCP with your voice configuration , etc , than it responds with OK
and starts streaming voice ( you don't have to handle any of the low level
negotiation).

Simply it works great , it behaves greatly and above all it handles tons of
simultaneous calls.

If you need more info regarding integration with Loquendo feel free to drop
me an e-mail.

PS Not an Loquendo partner or a salesman but a happy customer :)

Regards
A/T


2011/5/31 Jason Moran <jmoran at secureachsystems.com>

> We need to move into better TTS than flite, so we?ve decided we should look
> into mod_unimrcp.
>
>
>
> As far as setting up unimrcp server itself, are there any ?gotchas? or
> recommendations?
>
> We were looking to install UniMRCP Release 1.0.0 r1725 for linux along with
> unimrcp-deps-1.0.0 Dependency Package 1.0.0 (APR, Sofia-SIP). Are those the
> most up-to-date versions that we should be looking to install? I noticed a
> newer dependency package, but no newer UniMRCP release after that June 2010
> 1.0.0 version.
>
>
>
> As for mrcp engines, we are thinking about IVONA, Cepstral, Acapela, or
> Loquendo. Do each of these engines supply their own mrcp engine that somehow
> gets installed and registered with uniMRCP?
>
>
>
> -Jason Moran
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 at slickdeals.endjunk.com  Tue May 31 20:20:03 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 31 May 2011 09:20:03 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
Message-ID: <1306858803865-6423457.post@n2.nabble.com>

Steven,

Do you know what library package is needed to compile applications/mod_mp4?

-----
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/how-to-use-mod-mp4-tp6410226p6423457.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jan.berger at video24.no  Tue May 31 21:16:46 2011
From: jan.berger at video24.no (Jan Berger)
Date: Tue, 31 May 2011 19:16:46 +0200
Subject: [Freeswitch-users] unimrcp version and setup
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED69507BB4@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED69507BB4@SERVER2003.SecuReachSystems.local>
Message-ID: <612D0573735B43338BE7ECD4F330ABDE@dell9400>

Hi Jason,

 

The main issue with tts/asr engines is the language kits. MRCU will allow
you to connect to various engines, but make sure the engine support the
languages you need.

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason
Moran
Sent: 31. mai 2011 17:24
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] unimrcp version and setup

 

We need to move into better TTS than flite, so we've decided we should look
into mod_unimrcp.

 

As far as setting up unimrcp server itself, are there any "gotchas" or
recommendations?

We were looking to install UniMRCP Release 1.0.0 r1725 for linux along with
unimrcp-deps-1.0.0 Dependency Package 1.0.0 (APR, Sofia-SIP). Are those the
most up-to-date versions that we should be looking to install? I noticed a
newer dependency package, but no newer UniMRCP release after that June 2010
1.0.0 version.

 

As for mrcp engines, we are thinking about IVONA, Cepstral, Acapela, or
Loquendo. Do each of these engines supply their own mrcp engine that somehow
gets installed and registered with uniMRCP?

 

-Jason Moran

 

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From msc at freeswitch.org  Tue May 31 21:35:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 31 May 2011 10:35:54 -0700
Subject: [Freeswitch-users] Call For Help: FreeSWITCH Cookbook
Message-ID: <BANLkTinuuqCh4duDNsnkt33k1aNgqmAZuw@mail.gmail.com>

Hello FreeSWITCHers!

We have a need for some assistance with a few of our recipes for the
cookbook. If you are able to write technical documentation in English and
have first-hand knowledge of these topics then you might be in a position to
help:

CDRs:
Parsing XML CDRs
Handling A and B leg CDRs
Using a web server to handle XML CDRs

Event Socket:
Inbound ESL connections (basic how-to)
Outbound ESL connections (basic how-to)
Launch an outbound call with inbound event socket & ESL
Handle inbound call with socket app & ESL
Reacting to events (events for "my call" vs. system-wide events, etc.)

Misc:
Presence for BLF and SLA
Presence for FIFO agent status

If you think you've got the chops to write one or more recipes for the
cookbook then email me off list and we'll discuss the specifics. I will get
you all the information you need to get started.

Thanks!

-Michael
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From chechmate at gmail.com  Tue May 31 19:09:40 2011
From: chechmate at gmail.com (chechmate at gmail.com)
Date: Tue, 31 May 2011 18:09:40 +0300
Subject: [Freeswitch-users] hi
Message-ID: <4de52ecd.0121440a.67d1.254e@mx.google.com>

 hello, now i'm overwhelmed with joy http://g.msn.com.br/BR9/1369.0?http://cnbc7.com/news


From anton.vazir at gmail.com  Tue May 31 22:12:33 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 31 May 2011 23:12:33 +0500
Subject: [Freeswitch-users] Freeswitch tts russian voice 'ele' or other?
Message-ID: <BANLkTik7ZHmvCXu=3fH3=8AdjE29ic9nJg@mail.gmail.com>

Hi!

In ru.xml it's mentioned

tts-engine="cepstral" tts-voice="elena">

But after hours of googling I found nothing where and how to get the
mentioned russian 'elena' voice for cepstral

Any info?