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Posted to user@openmeetings.apache.org by "seba.wagner@gmail.com" <se...@gmail.com> on 2012/07/18 09:33:25 UTC
Re: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone
asterisk Sebastian
*how does all of this connect back into the telephone system*
The red5Sip create connections to both ends, the Asterisk server and the
red5 server. On Asterisk side it registers as SIP user on OpenMeetings side
it registers as a Flash user using the RTMP protocol.
Red5Sip does then transcode any incoming audio data from the Flash
AudioCodec (Nellymoser) to SIP codec (I think PCMA but not sure here).
We should interview Timur for further questions.
About enabling the integration: Alexei told you the wrong config key:
"sip.enable" should be "no" and "red5sip.enable" "yes"
Sebastian
2012/7/18 George Kirkham <gk...@co2crc.com.au>
> Sebastian,****
>
>
> Thanks for the quick reply.****
>
> ** **
>
> A question to start off with, in brief terms what does Asterisk do, and
> how does all of this connect back into the telephone system so that you can
> dial into OpenMeeting or dial out of Openmeetings, say to a mobile phone or
> landline phone? During the install/configuration of Asterisk I could not
> see where this would happen. Connecting to the public telecommunications
> system is what SIP is all about, or so I believed?****
>
> ** **
>
> Maybe if I read the documentation you directed me to, this would explain?*
> ***
>
> ** **
>
> Thanks,****
>
> ** **
>
> George Kirkham****
>
> ** **
>
> ** **
>
> *From:* seba.wagner@gmail.com [mailto:seba.wagner@gmail.com]
> *Sent:* Wednesday, 18 July 2012 5:11 PM
> *To:* openmeetings-user@incubator.apache.org
> *Subject:* Re: Help with OpenMeetings, SIP, VIOP and Internode's
> NodePhone asterisk****
>
> ** **
>
> About the questions on "how does it work", you might review those docs:
>
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
> and that one:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/red5SIP
>
> It is intend to be more like a developer documentation however feel free
> to ask any questions. The author of the SIP Gateway was Timur he should be
> around somewhere too.
>
> Sebastian****
>
> 2012/7/18 seba.wagner@gmail.com <se...@gmail.com>****
>
> Hi George,
>
> the SIP integration is based on Asterisk and it requires an Asterisk
> plugin that creates database and writes data to the OpenMeetings tables:
> meetme, extensions, sipusers.
>
> If you want to integrate OpenMeetings via SIP/VoIP with any VoIP server
> you will either have to simulate the functionality of the Asterisk plugin
> or need access to the VoIP Providers infrastructure to install that plugin.
>
> Additionally you might need to find a new approach on the existing SIP
> integration however it will require you to modify code or wirte some hooks
> that trigger the OpenMeetings SOAP/REST Gateway.
>
> Sebastian****
>
> ** **
>
> 2012/7/18 Alexei Fedotov <al...@gmail.com>****
>
> George,****
>
> I know that the first step is to set sip.enable to "yes"****
>
> ** **
>
> ** **
>
> ** **
>
> On Wed, Jul 18, 2012 at 1:56 PM, George Kirkham <gk...@co2crc.com.au>
> wrote:****
>
> Hi all,****
>
> ****
>
> If you have knowledge on Openmeetings with SIP and can assist me, I would
> really appreciate it.****
>
> ****
>
> I have followed the instructions on the Openmeetings web site but I am
> guessing there is a lot of more information that I require as I have
> extremely limited knowledge of SIP****
>
> http://incubator.apache.org/openmeetings/red5sip-integration.html****
>
> ****
>
> I use Internode (http://www.internode.on.net) for my Internet Service
> Provider (ISP). They provide a SIP service called Nodephone. I am
> expecting that it would not be hard to configure OpenMeetings to use
> Internode’s Nodephone server/services. However this is an area that I have
> not worked with before and I am not familiar with the terminology or
> technology.****
>
> ****
>
> Via Internode I have one SIP phone line/connection which I believe can be
> used in conjunction with OpenMeetings. ???****
>
> ****
>
> Any tips, hints or help you could give me are greatly appreciated. ****
>
> ****
>
> ****
>
> I have following the instructions for SIP and Openmeetings but I have a
> number of questions.****
>
> 1) How does Openmeetings or red5sip or asterisk connect to a SIP
> server? From what I understand there must be a SIP server somewhere in the
> configuration which will provide a link between ViOP and the
> telecommunications network?****
>
> 2) What is the purpose of red5sip ? I guess there is configuration
> information that I need to know about?****
>
> 3) What is the purpose of asterisk ? I guess there is configuration
> information that I need to know about?****
>
> 4) How do you access the SIP dialler in OpenMeetings conference
> meeting room? Or how do you dial into an openmeeting meeting? I am
> guessing that I would associate a SIP phone number (or a phone number
> range) with a meeting room. If this is so what software is joining these
> two together?****
>
> 5) Below is information about the Nodephone service and below that
> is the text from the installation instructions and highlighted in colour
> (not that I expect that you will see this after the email has been
> redirected), are the parts where the instructions did not exactly define
> what to do and what I did to complete the installation process. Then below
> is an image of my understanding of the SIP settings in OpenMeetings.****
>
> ****
>
> ****
>
> ****
>
> I have attached the information about Internode’s Nodephone system.****
>
> ****
>
> And I have been experimenting with my own Internode Nodephone SIP phone
> number, which is working well on a IP Phone, see images pasted into this
> email. I set up the SIP IP phone using these instructions;
> http://www.internode.on.net/support/guides/nodephone/sipura_spa_3000/****
>
> ****
>
> ****
>
> ****
>
> [image: cid:part1.09030901.07010105@internode.on.net]****
>
> ****
>
> [image: cid:part2.02000806.04010301@internode.on.net]****
>
> ****
>
> At this time I do not know what to enter into OpenMeetings configuration
> settings for SIP, and if there are any other software components that I
> need to install ?****
>
> ****
>
> ****
>
> ****
>
> ****
>
> ****
>
> *SIP-Transport Integration* ****
> ------------------------------
>
> Here is instruction how-to set up red5sip transport integration with
> OpenMeetings on Ubuntu 10.04. ****
>
> ****
>
> *Setup Asterisk* ****
> ------------------------------
>
> Run the commands ****
>
> *sudo apt-get update
> sudo apt-get install asterisk asterisk-mysql *****
>
> Ubuntu 12.04 has broken asterisk-mysql version. For other distribution
> next commands not needed: ****
>
> *aptitude purge asterisk-mysql
> cd /tmp
> apt-get build-dep asterisk-mysql
> apt-get -b source asterisk-mysql
> dpkg -i asterisk-mysql_1.6.2.0-1_i386.deb *****
>
> * *****
>
> The above created;****
>
> dpkg -i asterisk-mysql_1.8.10.1~dfsg-1ubuntu1_amd64.deb****
>
> Enable asterisk mysql realtime module:
>
> Add string ****
>
> *load => res_config_mysql.so* ****
>
> to the /etc/asterisk/modules.conf into the "modules" section. ****
>
> Configure mysql realtime module:
>
> Create file /etc/asterisk/res_mysql.conf and add lines: ****
>
> *[general]
> dbhost=127.0.0.1
> dbname=openmeetings
> dbuser=root
> dbpass=
> dbport=3306 *****
>
> touch /etc/asterisk/res_mysql.conf
> nano /etc/asterisk/res_mysql.conf****
>
> Add next lines into the /etc/asterisk/extconfig.conf: ****
>
> *[settings]
> sipusers => mysql,general,sipusers
> sippeers => mysql,general,sipusers
> extensions => mysql,general,extensions
> meetme => mysql,general,meetme *****
>
> Add next lines into the /etc/asterisk/extensions.conf: ****
>
> *[rooms]
> switch => Realtime/@ *****
>
> Restart asterisk: ****
>
> *service asterisk restart* ****
>
> Insert, for example, SIP user with name 'test': ****
>
> *mysql -u openmeetings -p openmeetings*****
>
> *INSERT INTO sipusers (allow, context, disallow, host, name, secret)
> VALUES ('ulaw' , 'rooms', NULL, 'dynamic' , 'test', '12345'); *****
>
> ****
>
> * *****
>
> *Setup red5sip transport* ****
> ------------------------------
>
> Download red5sip from****
>
> http://red5phone.googlecode.com/svn/branches/red5sip****
>
> Build with Apache Ant ****
>
> mkdir /usr/adm/sip
> cd /usr/adm/sip
> svn checkout
> http://code.google.com/p/red5phone/source/browse/branches/red5sip
> ...
> A red5sip/src/java/org/openmeetings/utils/PropertiesUtils.java
> A red5sip/settings.properties
> A red5sip/red5sip.iml
> A red5sip/build.xml
> Checked out revision 68.
> cd red5sip/
> /usr/adm/apache-ant-1.8.4/bin/ant clean.all
> /usr/adm/apache-ant-1.8.4/bin/ant -Ddb=mysql
> ...
> BUILD SUCCESSFUL
> Total time: 15 seconds
> cd ..
> mv red5sip /opt/red5sip****
>
> Install jsvc: ****
>
> *apt-get install jsvc* ****
>
> Set the JAVA_HOME environment variable in red5sip.sh****
>
> nano /opt/red5sip/red5sip.sh****
>
> JAVA_HOME=/usr/lib/jvm/jdk1.6.0_32****
>
> Insert proper values to the /opt/red5sip/settings.properties ****
>
> *red5.host - red5 server address (127.0.0.1)
> sip.obproxy - asterisk adderss (127.0.0.1)
> sip.phone - sip phone number (test)
> sip.authid - sip auth id (test)
> sip.secret - sip password (12345)
> sip.realm - sip realm, "asterisk" by default
> sip.proxy -
> rooms - ids of openmeetings rooms, can be, for example, 2,3,5,6 *****
>
> nano /opt/red5sip/settings.properties
> was
> red5.host=192.168.56.101
> sip.obproxy=192.168.56.101
> sip.phone=test
> sip.authid=test
> sip.secret=12345
> sip.realm=asterisk
> sip.proxy=
> rooms=7
> ====================================
> Changed to ;
> red5.host=127.0.0.1
> sip.obproxy=127.0.0.1
> sip.phone=test
> sip.authid=test
> sip.secret=12345
> sip.realm=asterisk
> sip.proxy=
> rooms=7
> =======================================****
>
> Add red5sip to autostart: ****
>
> *ln -s /opt/red5sip/red5sip.sh /etc/init.d/red5sip
> chmod a+x /etc/init.d/red5sip
> update-rc.d /etc/init.d/red5sip defaults *****
>
> Start openmeetings ****
>
> *service red5 start* ****
>
> Start red5sip ****
>
> *service red5sip start* ****
>
> ****
>
> root@om64u-1:/opt/red5sip# Unable to redirect to
> /opt/red5sip/logs/jsvc_red5sip.err
> SLF4J: Class path contains multiple SLF4J bindings.
> SLF4J: Found binding in
> [jar:file:/opt/red5sip/lib/logback-classic-0.9.26.jar!/org/slf4j/impl/StaticLoggerBinder.class]
> SLF4J: Found binding in
> [jar:file:/opt/red5sip/lib/red5.jar!/org/slf4j/impl/StaticLoggerBinder.class]
> SLF4J: See http://www.slf4j.org/codes.html#multiple_bindings for an
> explanation.****
>
> ****
>
> ****
>
> ****
>
> ****
>
> ****
>
> ****
>
> ****
>
> Thanks,****
>
> ****
>
> George Kirkham****
>
> ** **
>
>
>
> ****
>
> --
> Sebastian Wagner
> https://twitter.com/#!/dead_lock
> http://www.openmeetings.de
> http://www.webbase-design.de
> http://www.wagner-sebastian.com
> seba.wagner@gmail.com****
>
>
>
>
> --
> Sebastian Wagner
> https://twitter.com/#!/dead_lock
> http://www.openmeetings.de
> http://www.webbase-design.de
> http://www.wagner-sebastian.com
> seba.wagner@gmail.com****
>
--
Sebastian Wagner
https://twitter.com/#!/dead_lock
http://www.openmeetings.de
http://www.webbase-design.de
http://www.wagner-sebastian.com
seba.wagner@gmail.com
Re: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone
asterisk Sebastian
Posted by "seba.wagner@gmail.com" <se...@gmail.com>.
Once "red5sip.enable" is yes you have a new menu entry in the conference's
room menu "SIP Dialer" to Dial-Out.
To Dial-In I don't know at this moment how the mapping of the phone number
from Asterisk to SIP works.
Sebastian
2012/7/18 seba.wagner@gmail.com <se...@gmail.com>
> *how does all of this connect back into the telephone system*
> The red5Sip create connections to both ends, the Asterisk server and the
> red5 server. On Asterisk side it registers as SIP user on OpenMeetings side
> it registers as a Flash user using the RTMP protocol.
> Red5Sip does then transcode any incoming audio data from the Flash
> AudioCodec (Nellymoser) to SIP codec (I think PCMA but not sure here).
> We should interview Timur for further questions.
>
> About enabling the integration: Alexei told you the wrong config key:
> "sip.enable" should be "no" and "red5sip.enable" "yes"
>
> Sebastian
>
>
>
> 2012/7/18 George Kirkham <gk...@co2crc.com.au>
>
>> Sebastian,****
>>
>>
>> Thanks for the quick reply.****
>>
>> ** **
>>
>> A question to start off with, in brief terms what does Asterisk do, and
>> how does all of this connect back into the telephone system so that you can
>> dial into OpenMeeting or dial out of Openmeetings, say to a mobile phone or
>> landline phone? During the install/configuration of Asterisk I could not
>> see where this would happen. Connecting to the public telecommunications
>> system is what SIP is all about, or so I believed?****
>>
>> ** **
>>
>> Maybe if I read the documentation you directed me to, this would explain?
>> ****
>>
>> ** **
>>
>> Thanks,****
>>
>> ** **
>>
>> George Kirkham****
>>
>> ** **
>>
>> ** **
>>
>> *From:* seba.wagner@gmail.com [mailto:seba.wagner@gmail.com]
>> *Sent:* Wednesday, 18 July 2012 5:11 PM
>> *To:* openmeetings-user@incubator.apache.org
>> *Subject:* Re: Help with OpenMeetings, SIP, VIOP and Internode's
>> NodePhone asterisk****
>>
>> ** **
>>
>> About the questions on "how does it work", you might review those docs:
>>
>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>> and that one:
>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/red5SIP
>>
>> It is intend to be more like a developer documentation however feel free
>> to ask any questions. The author of the SIP Gateway was Timur he should be
>> around somewhere too.
>>
>> Sebastian****
>>
>> 2012/7/18 seba.wagner@gmail.com <se...@gmail.com>****
>>
>> Hi George,
>>
>> the SIP integration is based on Asterisk and it requires an Asterisk
>> plugin that creates database and writes data to the OpenMeetings tables:
>> meetme, extensions, sipusers.
>>
>> If you want to integrate OpenMeetings via SIP/VoIP with any VoIP server
>> you will either have to simulate the functionality of the Asterisk plugin
>> or need access to the VoIP Providers infrastructure to install that plugin.
>>
>> Additionally you might need to find a new approach on the existing SIP
>> integration however it will require you to modify code or wirte some hooks
>> that trigger the OpenMeetings SOAP/REST Gateway.
>>
>> Sebastian****
>>
>> ** **
>>
>> 2012/7/18 Alexei Fedotov <al...@gmail.com>****
>>
>> George,****
>>
>> I know that the first step is to set sip.enable to "yes"****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> On Wed, Jul 18, 2012 at 1:56 PM, George Kirkham <gk...@co2crc.com.au>
>> wrote:****
>>
>> Hi all,****
>>
>> ****
>>
>> If you have knowledge on Openmeetings with SIP and can assist me, I would
>> really appreciate it.****
>>
>> ****
>>
>> I have followed the instructions on the Openmeetings web site but I am
>> guessing there is a lot of more information that I require as I have
>> extremely limited knowledge of SIP****
>>
>> http://incubator.apache.org/openmeetings/red5sip-integration.html****
>>
>> ****
>>
>> I use Internode (http://www.internode.on.net) for my Internet Service
>> Provider (ISP). They provide a SIP service called Nodephone. I am
>> expecting that it would not be hard to configure OpenMeetings to use
>> Internode’s Nodephone server/services. However this is an area that I have
>> not worked with before and I am not familiar with the terminology or
>> technology.****
>>
>> ****
>>
>> Via Internode I have one SIP phone line/connection which I believe can be
>> used in conjunction with OpenMeetings. ???****
>>
>> ****
>>
>> Any tips, hints or help you could give me are greatly appreciated. ****
>>
>> ****
>>
>> ****
>>
>> I have following the instructions for SIP and Openmeetings but I have a
>> number of questions.****
>>
>> 1) How does Openmeetings or red5sip or asterisk connect to a SIP
>> server? From what I understand there must be a SIP server somewhere in the
>> configuration which will provide a link between ViOP and the
>> telecommunications network?****
>>
>> 2) What is the purpose of red5sip ? I guess there is configuration
>> information that I need to know about?****
>>
>> 3) What is the purpose of asterisk ? I guess there is configuration
>> information that I need to know about?****
>>
>> 4) How do you access the SIP dialler in OpenMeetings conference
>> meeting room? Or how do you dial into an openmeeting meeting? I am
>> guessing that I would associate a SIP phone number (or a phone number
>> range) with a meeting room. If this is so what software is joining these
>> two together?****
>>
>> 5) Below is information about the Nodephone service and below that
>> is the text from the installation instructions and highlighted in colour
>> (not that I expect that you will see this after the email has been
>> redirected), are the parts where the instructions did not exactly define
>> what to do and what I did to complete the installation process. Then below
>> is an image of my understanding of the SIP settings in OpenMeetings.****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> I have attached the information about Internode’s Nodephone system.****
>>
>> ****
>>
>> And I have been experimenting with my own Internode Nodephone SIP phone
>> number, which is working well on a IP Phone, see images pasted into this
>> email. I set up the SIP IP phone using these instructions;
>> http://www.internode.on.net/support/guides/nodephone/sipura_spa_3000/****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> [image: cid:part1.09030901.07010105@internode.on.net]****
>>
>> ****
>>
>> [image: cid:part2.02000806.04010301@internode.on.net]****
>>
>> ****
>>
>> At this time I do not know what to enter into OpenMeetings configuration
>> settings for SIP, and if there are any other software components that I
>> need to install ?****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> *SIP-Transport Integration* ****
>> ------------------------------
>>
>> Here is instruction how-to set up red5sip transport integration with
>> OpenMeetings on Ubuntu 10.04. ****
>>
>> ****
>>
>> *Setup Asterisk* ****
>> ------------------------------
>>
>> Run the commands ****
>>
>> *sudo apt-get update
>> sudo apt-get install asterisk asterisk-mysql *****
>>
>> Ubuntu 12.04 has broken asterisk-mysql version. For other distribution
>> next commands not needed: ****
>>
>> *aptitude purge asterisk-mysql
>> cd /tmp
>> apt-get build-dep asterisk-mysql
>> apt-get -b source asterisk-mysql
>> dpkg -i asterisk-mysql_1.6.2.0-1_i386.deb *****
>>
>> * *****
>>
>> The above created;****
>>
>> dpkg -i asterisk-mysql_1.8.10.1~dfsg-1ubuntu1_amd64.deb****
>>
>> Enable asterisk mysql realtime module:
>>
>> Add string ****
>>
>> *load => res_config_mysql.so* ****
>>
>> to the /etc/asterisk/modules.conf into the "modules" section. ****
>>
>> Configure mysql realtime module:
>>
>> Create file /etc/asterisk/res_mysql.conf and add lines: ****
>>
>> *[general]
>> dbhost=127.0.0.1
>> dbname=openmeetings
>> dbuser=root
>> dbpass=
>> dbport=3306 *****
>>
>> touch /etc/asterisk/res_mysql.conf
>> nano /etc/asterisk/res_mysql.conf****
>>
>> Add next lines into the /etc/asterisk/extconfig.conf: ****
>>
>> *[settings]
>> sipusers => mysql,general,sipusers
>> sippeers => mysql,general,sipusers
>> extensions => mysql,general,extensions
>> meetme => mysql,general,meetme *****
>>
>> Add next lines into the /etc/asterisk/extensions.conf: ****
>>
>> *[rooms]
>> switch => Realtime/@ *****
>>
>> Restart asterisk: ****
>>
>> *service asterisk restart* ****
>>
>> Insert, for example, SIP user with name 'test': ****
>>
>> *mysql -u openmeetings -p openmeetings*****
>>
>> *INSERT INTO sipusers (allow, context, disallow, host, name, secret)
>> VALUES ('ulaw' , 'rooms', NULL, 'dynamic' , 'test', '12345'); *****
>>
>> ****
>>
>> * *****
>>
>> *Setup red5sip transport* ****
>> ------------------------------
>>
>> Download red5sip from****
>>
>> http://red5phone.googlecode.com/svn/branches/red5sip****
>>
>> Build with Apache Ant ****
>>
>> mkdir /usr/adm/sip
>> cd /usr/adm/sip
>> svn checkout
>> http://code.google.com/p/red5phone/source/browse/branches/red5sip
>> ...
>> A red5sip/src/java/org/openmeetings/utils/PropertiesUtils.java
>> A red5sip/settings.properties
>> A red5sip/red5sip.iml
>> A red5sip/build.xml
>> Checked out revision 68.
>> cd red5sip/
>> /usr/adm/apache-ant-1.8.4/bin/ant clean.all
>> /usr/adm/apache-ant-1.8.4/bin/ant -Ddb=mysql
>> ...
>> BUILD SUCCESSFUL
>> Total time: 15 seconds
>> cd ..
>> mv red5sip /opt/red5sip****
>>
>> Install jsvc: ****
>>
>> *apt-get install jsvc* ****
>>
>> Set the JAVA_HOME environment variable in red5sip.sh****
>>
>> nano /opt/red5sip/red5sip.sh****
>>
>> JAVA_HOME=/usr/lib/jvm/jdk1.6.0_32****
>>
>> Insert proper values to the /opt/red5sip/settings.properties ****
>>
>> *red5.host - red5 server address (127.0.0.1)
>> sip.obproxy - asterisk adderss (127.0.0.1)
>> sip.phone - sip phone number (test)
>> sip.authid - sip auth id (test)
>> sip.secret - sip password (12345)
>> sip.realm - sip realm, "asterisk" by default
>> sip.proxy -
>> rooms - ids of openmeetings rooms, can be, for example, 2,3,5,6 *****
>>
>> nano /opt/red5sip/settings.properties
>> was
>> red5.host=192.168.56.101
>> sip.obproxy=192.168.56.101
>> sip.phone=test
>> sip.authid=test
>> sip.secret=12345
>> sip.realm=asterisk
>> sip.proxy=
>> rooms=7
>> ====================================
>> Changed to ;
>> red5.host=127.0.0.1
>> sip.obproxy=127.0.0.1
>> sip.phone=test
>> sip.authid=test
>> sip.secret=12345
>> sip.realm=asterisk
>> sip.proxy=
>> rooms=7
>> =======================================****
>>
>> Add red5sip to autostart: ****
>>
>> *ln -s /opt/red5sip/red5sip.sh /etc/init.d/red5sip
>> chmod a+x /etc/init.d/red5sip
>> update-rc.d /etc/init.d/red5sip defaults *****
>>
>> Start openmeetings ****
>>
>> *service red5 start* ****
>>
>> Start red5sip ****
>>
>> *service red5sip start* ****
>>
>> ****
>>
>> root@om64u-1:/opt/red5sip# Unable to redirect to
>> /opt/red5sip/logs/jsvc_red5sip.err
>> SLF4J: Class path contains multiple SLF4J bindings.
>> SLF4J: Found binding in
>> [jar:file:/opt/red5sip/lib/logback-classic-0.9.26.jar!/org/slf4j/impl/StaticLoggerBinder.class]
>> SLF4J: Found binding in
>> [jar:file:/opt/red5sip/lib/red5.jar!/org/slf4j/impl/StaticLoggerBinder.class]
>> SLF4J: See http://www.slf4j.org/codes.html#multiple_bindings for an
>> explanation.****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> Thanks,****
>>
>> ****
>>
>> George Kirkham****
>>
>> ** **
>>
>>
>>
>> ****
>>
>> --
>> Sebastian Wagner
>> https://twitter.com/#!/dead_lock
>> http://www.openmeetings.de
>> http://www.webbase-design.de
>> http://www.wagner-sebastian.com
>> seba.wagner@gmail.com****
>>
>>
>>
>>
>> --
>> Sebastian Wagner
>> https://twitter.com/#!/dead_lock
>> http://www.openmeetings.de
>> http://www.webbase-design.de
>> http://www.wagner-sebastian.com
>> seba.wagner@gmail.com****
>>
>
>
>
> --
> Sebastian Wagner
> https://twitter.com/#!/dead_lock
> http://www.openmeetings.de
> http://www.webbase-design.de
> http://www.wagner-sebastian.com
> seba.wagner@gmail.com
>
--
Sebastian Wagner
https://twitter.com/#!/dead_lock
http://www.openmeetings.de
http://www.webbase-design.de
http://www.wagner-sebastian.com
seba.wagner@gmail.com