I have, however, worked on a new application and that is a ClickBank store. ClickBank does not offer a store front, so I created one instead. You can see the product which is now offered as a service under Facebook from this site: http://www.facebook.com/apps/application.php?id=201650473205523. You can find an example of the embedded store under this Facebook page, see http://www.facebook.com/pages/Be-Healthy/211733492192178.

To gauge its popularity, I am only opening the service to a limited number of affiliates. Should this service proves popular, then I will move it to faster servers with network connectivity of higher bandwidth.

[sniplet Adsense 4Post]

The store front is not limited to embedding in a page under Facebook. It can also be embedded to any website, as can be seen here: Zhink ClickBank Store. The only difference is that for website embedding, the store cannot be customized. This will change in subsequent releases.

ClickBank Store

Installation of FreePBX with Asterisk

• You can skip this step if you already have FreePBX running on a computer somewhere. If not, then first do a fresh installation of Ubuntu 10.04. You can download a CD of this for the installation from here.
• Once you have Ubuntu 10.04 running, there is a script, found under this article, that can be used to set up FreePBX and Asterisk. Though the script is for Ubuntu 9.10, I think it should apply also for 10.04. Only use the second script. There are a few flaws with the first script. Do read the full article to make sure you comply to some of the security issues.
• With 10.04 some php functions have been deprecated, so you need to make some corrections to the FreePBX codes. This is easily done. Read the errors, go to the FreePBX codes and modify appropriately. For example, split can be replaced by explode and so on.

[sniplet Adsense 4Post] Configure SSGWPE Trunk via FreePBX

• Point your browser to the FreePBX web admin page.
• Click “Setup->Trunks->Add SIP Trunk”. Enter :
• Trunk Description: ssgwpe
• Maximum Channels: 1
• Outgoing Settings->TrunkName: ssgwpe
• Outgoing Settings->PEER Details:
  • username=ssgwpe
  • type=friend
  • secret=password
  • host=dynamic

  These username and secret values should correspond to the values entered under “SIP” tab for the ssgwpe configuration.

• Do nothing for Incoming Settings.

Configure Outbound Route Through This Trunk

• Click “Setup->Outbound Routes->Add Route”. Enter:
• Route Name: Skype
• Dial Patterns: 70|.
• Trunk Sequence: SIP/ssgwpe

Once this is configured, you should then be able to call a Skype user from a SIP client (like XLite) by dialing 70 follow by the destination Skype ID. This is neat. The destination Skype address could also be a trunk into another PBX. This then facilitates inter PBX communication. A side benefit to  using XLite is that it supports recording of  conversations which means you now have a means to record your Skype conversations. Asterisk allows dial patterns that detect non numeric characters which means the preceding 70 can be replaced with more appropriate patterns. However, this cannot be done because of FreePBX limitation. You can choose to modify FreePBX if you know what you are doing. Configure Inbound Routes Through This Trunk Here we shall configure 4 incoming  routes from the trunk. They are:

• skype2ivr which supports incoming calls from Skype to IVR of PBX,
• skype2disa which supports incoming calls from Skype to DISA service,
• skype2echo which supports incoming calls from Skype to echotest service and
• skype2ext which supports incoming calls from Skype to a SIP extension.

Once you get the idea, you can customize your own incoming routes. The names of the routes can then be used under the “Skype” tab of ssgwpe user interface. Skype callers are then directly by ssgwpe to the appropriate incoming routes to Asterisk. Before we do this, we need to add needed modules to FreePBX.

• Click “Setup->Module Admin->Check for updates online”.
• Choose to install “DISA” and “Misc Destinations”.
• At the bottom of the page, press “Process”.

[sniplet Adsense 4Post] Configure skype2ext Assuming that you have a PSTN trunk from your Asterisk and you intend to have some Skype callers directed to your mobile number. This is how you can do it.

• Click “Setup->Misc Destinations->Add Misc Destination”. Enter:
• Description: My Mobile
• Dial: XXXXXX, whatever your dial string setup to your mobile number
• Once changes are submitted,
• Click “Setup->Inbound Routes->Add Incoming Route”. Enter:
• Description: Skype 2 Ext
• DID Number: skype2ext
• Set Destination: check Misc and select “My Mobile”
• Submit changes and apply configuration changes.

From the user interface of SSGWPE, click on the “Skype” tab and enter:

• Click “Add Group” to add a new group to accommodate the Skype callers to your mobile number. Choose any name you like.
• You can use “Add Buddy To Group” or use “<<” to add the appropriate Skype buddies to your new group.
• Enter “sip:skype2ext@XXX.XXX.XXX.XXX” for the “Number to call for the group”. Replace “XXX.XXX.XXX.XXX” which your Asterisk server IP address.
• Enter your welcome message.
• Click “Save” to save the configuration.

The Skype callers in that group when calling this gateway should now be redirected to your mobile number. Configure skype2disa DISA service allows you to get a dial tone to dial out through the Asterisk trunks. To add a DISA service,

• Click “Setup->DISA->Add DISA”. Enter:
• DISA Name: Disa One
• PIN: XXXX -> choose what you like
• Leaves the rest unchanged.
• Submit changes and apply configuration changes.

To configure skype2disa incoming route,

• Click “Setup->Inbound Routes->Add Incoming Route”. Enter:
• Description: Skype 2 DISA
• DID Number: skype2disa
• Set Destination: check DISA and select “Disa One”
• Submit changes and apply configuration changes.

Similarly configure ssgwpe as per previous example to allow a group of Skype callers to have this service. Configure skype2echo and skype2ivr Echo test is written by me as a record and playback more akin to that of Skype EchoTest service. The default echotest service in FreePBX is more for latency test. Similarly, ivr is written as a test for “dtmf” tones from Skype callers. These two examples show how you can manually create your own services through various files used by FreePBX and Asterisk. Download this compressed file that contains the following 3 files:

• pls-enter-echo-message-after-tone.gsm
• dialout.gsm and
• getNumber.php

Uncompress the files and copy the 2 gsm files to “/var/lib/asterisk/sounds”.  Copy “getNumber.php” to “/var/lib/asterisk/agi-bin”.  The copying and changing of ownership to the files require you to have root permission. With root permission, edit “/etc/asterisk/extensions_custom.conf” and add in the following:

• ; echo test for call coming from ssgwpe
  exten = 759731,1,Answer
  ;exten = 759731,n,Wait(2)
  exten = 759731,n,Playback(pls-enter-echo-message-after-tone)
  exten = 759731,n,Record(asterisk-recording%d:ulaw,2,20)
  exten = 759731,n,Playback(${RECORDED_FILE})
  exten = 759731,n,Playback(goodbye)
  exten = 759731,n,Hangup

  ; custom ivr for call coming from ssgwpe
  exten = 759732,1,Answer
  exten = 759732,n,SetMusicOnHold(default)
  exten = 759732,n,WaitMusicOnHold(5)
  exten = 759732,n,agi(getNumber.php)
  exten = 759732,n,NoOp(${Number})
  exten = 759732,n,GotoIf($["${Number}" != ""]?dial:end)
  exten = 759732,n(dial),Playback(pls-hold-while-try)
  exten = 759732,n,Goto(from-internal,${Number},1)
  exten = 759732,n(end),Playback(goodbye)
  exten = 759732,n,Hangup

Similar to earlier description on creation of “Misc Destination”, create “Record & Playback” with “Dial” extension as 759731 and “Skype 2 IVR” with “Dial” extension as 759732.

Similar to earlier description on creation of “Inbound Routes”, create these two routes:

• Description: Skype 2 Echotest
• DID Number: skype2echo
• Set Destination: Misc, select “Record & Playback”

and

• Description: Skype 2 Custom IVR
• DID Number: skype2ivr
• Set Destination: Misc, select “Skype 2 IVR”

“Apply configuration changes” so that Asterisk change reload with the updated configuration.

As per earlier example, configure ssgwpe to use these two routes.

Conclusion

Integrating SSGWPE via FreePBX is straight forward. The way SSGWPE is configured is similar to that for configuring Linksys Sipura 3102, albeit simpler. When SSGWPE is properly configured, it can be seen on the flash operator panel under the “Trunk” section.

SSGWPE affords you or your company a Skype internet identity, much like a PSTN number, where you and your employees can be reached easily even when they are not connected to the internet, but connected via GSM or PSTN networks.

Skype Sip Gateway, Personal Edition:

• To understand more about Skype Sip Gateway, Personal Edition, read this.

Release Summary:

• Runs on Ubuntu 10.04. No special configuration for Ubuntu.
• Use only Pulse Audio server.
• Connection is fast and reliable.
• Audio quality is excellent.
• New graphical user interface that shows status and other enhancements.
• Easily integrate as a trunk to FreePBX.
• Quick recovery when problems do arise, for example, when Pulse Audio server dies.
• Does NOT USE Jackd for audio connection management. There were problems with using Jackd in the previous release.
• Uses DBUS for communication with Skype instead of X11. This reduces segmentation problem tremendously.

[sniplet Adsense 4Post]

Installation:

• Install Ubuntu: Use a relatively fast computer. This is because Pulse Audio and Skype are quite CPU intensive. When the two programs are active, they consistently top the various processes as shown under “top” program. I am using an “Acer Aspire L320″, which uses Intel Core 2 Duo, CPU 2140 @ 1.6GHz with 1.5 GB memory. I have tested on old P4 and the performance is not good.
• Download and install Ubuntu 10.04 Desktop Edition. Do not use the server edition.
• Install Skype: Once you have Ubuntu up and running, download and install Skype for Ubuntu. The latest version can still run under 10.04. Hopefully Skype can release a newer copy soon.
• Log into your Skype account and test that it is working properly. Ensure the following for Skype:
  • The box “Sign me in when Skype starts” should be checked.
  • Under “Options->Sound Devices”, check that PulseAudio is used for microphone, speakers and ringing. You can choose to uncheck the box for “Allow Skype to automatically adjust my mixer levels”.
  • Under “Options->Public API”, ensure “DBUS” is checked. You can leave “X11″ checked too.
  • Under “Options->Video Devices”, disable Skype video. SSGWPE does not currently support video routing.
  • These are optional. “Options->General”, change the values in the boxes for “Show me as Away when I am inactive for” and “Show me as Not Available when I am inactive for” to “Off”. This is supposed to keep your Skype status constant. I think this does not work very well. “Options->Privacy”, check the box for “Allow my status to be shown on the web”.

[sniplet Adsense 4Post]

• Install SSGWPE: Created a directory, say “release” under your home directory. Now download SSGWPE and saved it in “release” directory.
• Unzip the downloaded file and extract all files and saved under “release” directory.
• Open a “terminal” can change directory to “release” directory. Now run “sudo ./install.sh”. This should get and install the necessary software and files.
• Now run “./safe_ssgwpe.sh”. This should run ssgwpe which should automatically run Skype too.
• Configure ssgwpe as follows:
  • Under “Registration” tab, enter the “Skype Login ID” that corresponds to the Skype account that you are using. In trial mode, you do not need to enter “Registration Key”. Without the key, you get to test the gateway but calls are limited to 1 minute time span. When Skype indicates that a program “skypegw” wants to connect to it, allow it to connect with the box “Remember this selection” checked.
  • Under “SIP” tab, you need to make ssgwpe a SIP client to a server. You should have already created a corresponding SIP account in the server for this gateway. If the server is also on this computer and it uses 5060, then make sure that “Sip UDP Port” for this gateway should not be 5060. You can use 5061.
  • Under “Skype” tab, you can configure the destinations that groups of “Skype callers” should be directed to.
• That’s it. You can now start testing the gateway.

Integration to Free PBX

Read the article here.

When this errors occur, SSGWPE attempts at recovery by re-starting the various processes involved in the gateway. Most times the problem disappears, but sometimes the problem is more persistent. In the latter case, the repeated re-starting can be a rather painful experience to the user.

Having tested SSGWPE on a number of PCs, it is found that the above mentioned error is very much reduced on more powerful PCs. With more powerful PCs, recovery from such errors is also faster. It appears that Jackd needs a steady state before receiving audio connections, and with faster PCs, that steady state is reached earlier, resulting in less likelihood for the error occurring.

So, do try out SSGWPE on faster PCs if you encounter the above problem.

Finally, remember that some sound cards to not like the “-s” option for Jackd. So instead of “/usr/bin/jackd -dalsa -dhw:0 -r48000 -p1024 -n2 -s” use “/usr/bin/jackd -dalsa -dhw:0 -r48000 -p1024 -n2″.

[sniplet Adsense 4Post]

Connect your server/PC to a network that offers DHCP service. With simple networks, the DHCP service is usually provided by the router cum switch. Login into your router/switch, you should be able to see the IPs of the connected computers.

Put the Vasuntu CD into the server/PC and allow the server/PC to boot up Vasuntu. Once Vasuntu is ready, you should be able to see the IP address of Vasuntu from the router/switch.

Point your browser to the IP address of Vasuntu and you should be presented with a web interface. You can login using default username: vasuntu and password: vasuntu. Click on the “Remote Admin Using VNC” and you should get a remote display of Vasuntu.

You can then proceed to install Vasuntu into the server/PC by clicking the “Install” icon on the desktop. Another tip here … should you not be interested in language packs, you should choose to skip this during installation.

[sniplet Adsense 4Post]

From the desktop interface, click System->Preferences->Sessions. You are presented with a “Sessions” window. The first tab has “Startup Programs”. Click on “New” button to add a new startup program. In the window enter the following:

• Name:  Safe Skype Gateway
• Command: gnome-terminal -x /usr/local/bin/ssgwpe/safe_skypegw.sh

Click the “OK” button.

Going back to the desktop interface, click on System->Administration->Login Window. Enter your root/admin password when prompted. You are now presented with a “Login Window Preferences” window.

Click on the “Security” tab. Check the field “Enable Automatic Login” and choose the user name that shall automatically login when the system boots. This is also the user running the SSGWPE.

When the above is done, and upon system reboots, SSGWPE shall start up automatically. It is important to note that though this comes handy, it also presents a security risk. There is a short window of time after start up where the auto login user can be compromised. So, do use with care.
[sniplet Adsense 4Post]

• Show me as ‘Away’ when I am inactive for Off minutes
• Show me as  ‘Not Available’ when I am inactive for Off minutes

Just enter ‘Off’ into the boxes for the above two fields.

[sniplet Adsense 4Post]

The file is named skypegw.xml. This file is updated when you make changes via the user interface of the SSGWPE. If the file is corrupted, SSGWPE may fail to start.

You can also manually edit this file, but before doing that, it is best to make a backup copy of it.  As a practice, it is always good to have a backup copy. The copy allows you to restore your configuration when you need to.

From the SSGWPE interface, to add buddies to the groups, you need first to add a contact as buddy from the Skype interface first, and then followed by updating the buddies to SSGWPE by waiting till the next refresh or pressing “Refresh” button under “From Skype” tabbed panel of the SSGWPE interface. Once the buddy is in the list, you can then add the buddy to a group.

Should you find the preceding step convoluted, and choose to bypass this, you can manually edit the configuration file instead. Just ensure the right format and right Skype identities are added. As mentioned before, wrongly editing this file may cause the SSGWPE to fail.

[sniplet Adsense 4Post]

This release is a live installable CD. Booting from the CD allows you to test drive the software and allows you to install the software to your PC if you choose to.

Vasuntu is an integrated software stack featuring Asterisk, Voiceone, Flash Operator Panel, Skype Sip Gateway (Personal Edition), Skype and remote access services.

Version 1.0

This version is built on Ubuntu 7.04 with linux kernel version 2.6.20-17-generic.

Main Configuration

1. Asterisk

• Asterisk 1.4.20

• Zaptel 1.4.11

• Libpri 1.4.7

• AddOns 1.4.7

2. Oslec

• Patch for Zaptel 1.4.11

3. Voiceone

• 0.7.1

4. Skype Sip Gateway (Personal Edition)

• 2.0

5. Flash Operator Panel

• 0.29

6. Skype

• 2.0.0.72

Download

To download click here.