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Posted to user@openmeetings.apache.org by George Kirkham <gk...@co2crc.com.au> on 2012/07/18 08:56:17 UTC
Help with OpenMeetings, SIP, VIOP and Internode's NodePhone asterisk
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/
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/St
aticLoggerBinder.class]
SLF4J: Found binding in
[jar:file:/opt/red5sip/lib/red5.jar!/org/slf4j/impl/StaticLoggerBinder.c
lass]
SLF4J: See http://www.slf4j.org/codes.html#multiple_bindings for an
explanation.
Thanks,
George Kirkham
RE: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone asterisk Sebastian
Posted by Stephen Cottham <St...@robertbird.com.au>.
Asterix is a software PABX, it's a phone system in its own right. You
can install ISDN cards, PSTN cards use VOIP etc.. and connect to your
Telco provider via various trunks (Aus main providers Telstra, Optus,
Skype, internode, exetel etc..) - We currently use Trixbox with an ISDN
card in Sydney which is based on the Asterix core and is probably the
easiest option for you to get an idea for how it works and can be setup.
Check the opensource version here
http://fonality.com/trixbox/
I can assist you with a basic setup to connect to your SIP provider.
Not sure on the openmeetings side of things as I have not looked at that
yet, either way it would be good to get an idea of the Asterix system
before diving in.
Cheers
From: George Kirkham [mailto:gkirkham@co2crc.com.au]
Sent: 18 July 2012 08:18
To: openmeetings-user@incubator.apache.org
Subject: RE: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone
asterisk Sebastian
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+Integratio
n+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/
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/St
aticLoggerBinder.class]
SLF4J: Found binding in
[jar:file:/opt/red5sip/lib/red5.jar!/org/slf4j/impl/StaticLoggerBinder.c
lass]
SLF4J: See http://www.slf4j.org/codes.html#multiple_bindings for an
explanation.
Thanks,
George Kirkham
--
Sebastian Wagner
https://twitter.com/#!/dead_lock <https://twitter.com/#%21/dead_lock>
http://www.openmeetings.de <http://www.openmeetings.de>
http://www.webbase-design.de <http://www.webbase-design.de>
http://www.wagner-sebastian.com <http://www.wagner-sebastian.com>
seba.wagner@gmail.com <ma...@gmail.com>
--
Sebastian Wagner
https://twitter.com/#!/dead_lock <https://twitter.com/#%21/dead_lock>
http://www.openmeetings.de <http://www.openmeetings.de>
http://www.webbase-design.de <http://www.webbase-design.de>
http://www.wagner-sebastian.com <http://www.wagner-sebastian.com>
seba.wagner@gmail.com <ma...@gmail.com>
Stephen Cottham
Group IT Manager (Associate)
Robert Bird Group
Level 5, 333 Ann St
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Phone: +6173 319 2777 (AUS)
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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
Re: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone
asterisk Sebastian
Posted by "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
RE: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone asterisk Sebastian
Posted by George Kirkham <gk...@co2crc.com.au>.
Hmmm, apologies but I don't know what I am really doing here.
I enabled sip (and entered whatever I knew to enter, and then got this
error message "SIP Applet is not ready!"
Thanks,
George Kirkham
From: George Kirkham [mailto:gkirkham@co2crc.com.au]
Sent: Wednesday, 18 July 2012 5:18 PM
To: openmeetings-user@incubator.apache.org
Subject: RE: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone
asterisk Sebastian
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+Integratio
n+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/
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/St
aticLoggerBinder.class]
SLF4J: Found binding in
[jar:file:/opt/red5sip/lib/red5.jar!/org/slf4j/impl/StaticLoggerBinder.c
lass]
SLF4J: See http://www.slf4j.org/codes.html#multiple_bindings for an
explanation.
Thanks,
George Kirkham
--
Sebastian Wagner
https://twitter.com/#!/dead_lock <https://twitter.com/#%21/dead_lock>
http://www.openmeetings.de <http://www.openmeetings.de>
http://www.webbase-design.de <http://www.webbase-design.de>
http://www.wagner-sebastian.com <http://www.wagner-sebastian.com>
seba.wagner@gmail.com <ma...@gmail.com>
--
Sebastian Wagner
https://twitter.com/#!/dead_lock <https://twitter.com/#%21/dead_lock>
http://www.openmeetings.de <http://www.openmeetings.de>
http://www.webbase-design.de <http://www.webbase-design.de>
http://www.wagner-sebastian.com <http://www.wagner-sebastian.com>
seba.wagner@gmail.com <ma...@gmail.com>
RE: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone asterisk Sebastian
Posted by 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+Integratio
n+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/
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/St
aticLoggerBinder.class]
SLF4J: Found binding in
[jar:file:/opt/red5sip/lib/red5.jar!/org/slf4j/impl/StaticLoggerBinder.c
lass]
SLF4J: See http://www.slf4j.org/codes.html#multiple_bindings for an
explanation.
Thanks,
George Kirkham
--
Sebastian Wagner
https://twitter.com/#!/dead_lock <https://twitter.com/#%21/dead_lock>
http://www.openmeetings.de <http://www.openmeetings.de>
http://www.webbase-design.de <http://www.webbase-design.de>
http://www.wagner-sebastian.com <http://www.wagner-sebastian.com>
seba.wagner@gmail.com <ma...@gmail.com>
--
Sebastian Wagner
https://twitter.com/#!/dead_lock <https://twitter.com/#%21/dead_lock>
http://www.openmeetings.de <http://www.openmeetings.de>
http://www.webbase-design.de <http://www.webbase-design.de>
http://www.wagner-sebastian.com <http://www.wagner-sebastian.com>
seba.wagner@gmail.com <ma...@gmail.com>
Re: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone asterisk
Posted by "seba.wagner@gmail.com" <se...@gmail.com>.
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
Re: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone asterisk
Posted by "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
Re: Help with OpenMeetings, SIP, VIOP and Internode's NodePhone asterisk
Posted by 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****
>