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Junior Member
How to establish Flash-Skype conference
The goal of this article is establish audio conference between any number of Flash clients and Skype accounts. Flash client mean applications played by Adobe Flash Player.
As a result, we will get a Flash-Skype voice conference.
Establishing and tests require this software.
1. Linux Centos 5.x. – free
2. Adobe Flash Player 10 – Free
3. Flashphoner v1.0.0.109 – Free 10-connects Developer Version
4. Asterisk v1.6.2.10 – Free Asterisk
5. Skype v4.2.0.169 – Free
6. Digium Skype For Asterisk plug-in (hereafter SFA) – $66
7. Xlite v3.0 – Free
8. JDK v1.6_21 – Free
9. Wowza Media Server v2.1.2 – Free 10-connects developer license
Plan:
1. Asterisk installation.
2. JDK installation.
3. Wowza Media Server installation.
4. Flashphoner installation.
5. SFA installation.
6. Creation of manager account in Skype.
7. Xlite installation.
8. Test of Skype to Xlite call.
9. Test of Xlite to Skype call.
10. Test of Skype to Flash call.
11. Test of Flash to Skype call.
12. Test of Xlite+Xlite+Skype conference.
13. Test of Flash+Flash+Skype conference.
14. Audio Codecs
15. Resume
1. Asterisk installation.
– Download the tar-archive here
www asterisk org/downloads/asterisk/releases/asterisk-1.6.2.10.tar.gz
– Install it in usual way: configure, make, make install
Some libraries which are necessary for functioning of Asterisk are listed below:
- openssl, openssl-dev
- ncurses-devel
- zlib-devel
- libxml2-devel
- g++(gcc-c++)
– Configure sip.conf and extensions.conf files. They must look like this:
——————————-
sip.conf
[general]
bindport=5060
bindaddr=101.226.102.61
context=default
allow=all
[2000]
type=friend
secret=2000
host=dynamic
canreinvite=no
[2001]
type=friend
secret=2001
host=dynamic
[2002]
type=friend
secret=2002
host=dynamic
——————————-
extensions.conf
[default]
;exten=>2001,1,Dial(Skype/myaccount@my_personal_skype_account)
;exten=>myaccount,1,Dial(SIP/2000)
exten=>2001,1,Answer
exten=>2001,2,ConfBridge(1,Ma)
exten=>myaccount.1,1,Answer
exten=>myaccount.1,2,ConfBridge(1,Ma)
——————————
We’ll return to the description of these configuration files, when the rest of the software will be installed.
2. JDK installation
– Download the last JDK version here:
www oracle com/technetwork/java/javase/downloads/jdk6-jsp-136632.html
This need for Wowza Media Server work.
– Install it.
3. Wowza Media Server installation
– Download Wowza Developer Edition here – www wowzamedia com/store.html
– Install it
If you use rpm.bin distribution, Wowza installed by running the downloaded file ./WowzaMediaServer-2.1.2.rpm.bin
4. Flashphoner installation
This is a server software, allows you to develop flashphones and click2call buttons. It can connect any application written in Flash with any SIP client. In other words, Flashphoner – a Flash-VoIP gateway, which allows you call from Flash to landline and mobile phones.
– Download Flashphoner here www flashphoner com
– check whether you have installed Wowza Media Server, JDK and gcc.
– Install Flashphoner
5. SFA installation
SFA (Skype For Asterisk) is a paid plug-in to Asterisk. The price of a single Skype to Asterisk licencse is $66.
– Read the documentation and learn about pricing policy here
www digium com/en/products/software/skypeforasterisk.php
– Purchase an SFA license. The key will be sent you by e-mail.
Now you must register the key for your server. After that the key will be tied to your server hardware by MAC-adress. Please note that now, whenever your Mac address will change (change the NIC or move to a new server) you should re-register your SFA-lines to new hardware.
—— Start registration ——
– Download the registration program here
www downloads.digium com/pub/register/
– Choose the executable file which is suitable for your system’s architecture and run it
– Enter your contact and personal information such as Name, Address, Phone, E-mail, etc.
– Enter the license key obtained previously.
After registration will finished successfuly, the registration program will create a *.lic-file in the /var/lib/asterisk/licenses/ directory
—— End registration ——
– Make a backup of your directory with licenses.
– Download the plug-in itself directly, choosing the necessary system’s architecture here: www downloads.digium com/pub/telephony/skypeforasterisk/
– Install plugin in usual way: make, make install
– Load SFA modules into Asterisk. This may be done with this commands:
a) Enter to the Asterisk console
$asterisk -r
b) Load here two modules by commands.
*CLI> module load res_skypeforasterisk.so
*CLI> module load chan_skype.so
– Congrats! SFA installed and ready for work.
6. Creation a Skype Manager account
– Create Skype Manager account here www skype com/intl/ru/business
Skype Manager account it the “domain-account”, it will give you a possibility create child accounts and manage it. You need register special manager account even if you already have the personal one (despite the fact that Skype will offer use it)
– Create child account to your Skype Manager account.
Reason that SFA supports operations only with child accounts.
– Configure SFA on your server for work with your new Skype Manager child account.
Config file must look like this
—————————————
chan_skype.conf
[general]
engine_directory=/home/skype
debug=yes
[myaccount]
context=default
secret=myaccount_password
disallow=all
allow=ulaw
—————————————
Where:
myaccount – is a child account of Skype Manager account
myaccount_password – skype password for Skype Manager account
7. Xlite installation
– Download free version of Xlite here www counterpath com
– Install it and run
– Add SIP account in the “SIP Account Settings” menu with the following parameters:
————————————
Display Name: 2000
User name: 2000
Password: 2000
Authorization user name: 2000
Domain: 101.226.102.61
————————————
“101.226.102.61” is an example value. Put here the ip-address of your Asterisk server.
Port 5060 will be used by default.
8. Test of Skype–>Xlite call
– Configure extensions.conf file (see “Asterisk installation” section), make it look like this
——————————–
extensions.conf
[COLOR="RoyalBlue "][default]
exten=>myaccount,1,Dial(SIP/2000)[/COLOR]
——————————–
– Check configure of sip.conf file (see “Asterisk installation” section)
– Check configure of chan_skype.conf file (see “Creation a Skype Manager account” section)
If all this files configures exactly as it appears above, we can from Skype to Skype_Child_Acc and the call goes to the following path:
a. Your_Skype_Acc –> Skype_Child_Acc
b. Skype_Child_Acc –> Your_Asterisk (by chan_skype.conf)
c. Your_Asterisk –> 2000 (by extensions.conf)
d. 2000 –> Xlite (by “SIP Account Settings”)
As a result of the test we must obtain a successful call, for example, from your personal Skype account to Skype_Child_Acc. In this case the call must be picked up by Xlite on
account 2000.
9. Test Xlite –> Skype call
– Configure extensions.conf file (see “Asterisk installation” section), make it look like this
——————————–
extensions.conf
[default]
exten=>2001,1,Dial(Skype/myaccount@my_personal_skype_account)
——————————–
– Check configure of sip.conf file (see “Asterisk installation” section)
– Check configure of chan_skype.conf file (see “Creation a Skype Manager account” section)
If all this files configures exactly as it appears above, we can from Xlite to
Your_Skype_Acc and the call goes to the following path:
a. Xlite –> 2001
b. 2001 –> Your_Asterisk
c. Your_Asterisk –> Skype_Child_Acc (by extensions.conf)
d. Skype_Child_Acc –> Your_Skype_Acc (by chan_skype.conf)
10. Test Skype–>Flash call
– Do everything as in “8. Test of Skype–>Xlite call”, but use Flashphoner (client and server application) instead of Xlite.
11. Test Flash–>Skype call
– Do everything as in “9. Test of Xlite–>Skype call”, but use Flashphoner (client and server application) instead of Xlite.
12. Test Xlite+Xlite+Skype conference
... Read details of this part at www Flashphoner com/blog ...
13. Test Flash+Flash+Skype conference
– Do everything as in “12. Test Xlite+Xlite+Skype conference”, but use Flashphoner (client and server application) instead of Xlite.
14. Audio codecs
Codecs
Speex 16kHz (wideband) codec used on the Flash side.
G.711 used in the Flashphoner<–>Asterisk direction.
G.729 used in the Asterisk<–>Digium<–>Skype direction
15. Resume
So, we configured conference between Adobe Flash Player and Skype.
This function is demanded in flash conference services for the possibility of joining together the Flash and phone Conferences
The possibility for this gives Flashphoner – Flash-SIP server, which allows you to connect Flash and any SIP client.
See more information here - www Flashphoner com/blog
Last edited by Flasphoner; 08-24-2010 at 12:29 AM.
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Isn't there an easier way to do it?
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Junior Member
Really this way is quite simple,
it just looks terrible, because a lot of text
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I did not read the whole article since it is your first post here. Welcome
- So is it completely free, since it is based on Skype and Adobe Flash player?
- Have you tried it with success?
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Junior Member

1. This have some price, because there are some connecting elements between Flash and Skype
Flash --> Flashphoner --> Wowza --> Asterisk --> Skype-for-Asterisk --> Skype
Flash....................Free
Flashphoner...........$995
Wowza..................$995 or $65/mo
Asterisk.................Free
Skype-for-Asterisk...$66/per line
Skype....................Free
2. Yes, we tried it with succes - this article written by real experience.
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