Thursday, April 25, 2013

Asterisk –Debian based Asterisk 11, Freepbx 2.11 on VMware / Virtualbox (Asterisk VM/Asterisk Ready Virtual Machine)

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[UPDATED: 03 FEB 2015]

Here’s a VMDK image to run a full featured Asterisk PaBX with FreePBX as the management UI using our default and secure install practices. No registrations, no username/password, no signing up for newsletter.

Get it from Sourceforge: https://sourceforge.net/projects/debianasterisk/[Select SWSterisk11 folder, then download the zip file therein]

 

After extracting, You either need VirtualBox or VMPlayer/VMWare or any Virtualization products that supports VMDK files or if you’re using Hypervisor, convert the image to VHD using MVMC from here.  This is to give you a feel of Asterisk with FreePBX without worrying about installation etc., its plug and play, literally. Just start up to your VirtualBox/VMplayer nd get it up and running in seconds. Go in to FreePBX and start creating extensions and enable other features. This image is free from any lockdowns or customizations that you cannot reverse or disable or enable as you wish. It is completely FREE from any personal restrictions. This image does not trace usage, or “dials home” or anything strange like that. Totally clean, totally lean and totally fast. It is functional and you can hook it up to a real production environment and you almost have a full fledge PBX, just add a Digium VoIP Gateway or another IP based PSTN.

IMPORTANT

  • While it is enterprise ready, it should rather be used for “playing” or “testing”….
  • DISCLAIMER: By using this VIRTUAL MACHINE IMAGE, i disclaim any sorts of liability whatsoever. What you do with this image is purely your choice/actions.
  • This is not "another disto", nothing proprietary, i don't claim any copyrights, just make it look and feel like its mine for fun, but of course any of those customizations can be reversed. All other trademarks are properties of their respective owners. All rights reserved.
Here’s some information about the VM image you just downloaded
  • It’s in ZIP compression, just get WinRAR or 7-ZIP to extract. After extracting, there should be one vmdk just mount the vmdk into VMWare/VMPlayer or Virtualbox and start the image
  • Username/password
  • OS
    - Username: root (the other non root user is swsterisk with same password as below)
    - Password: asteriskrocks (change this!)
  • FreePBX(admin), MySQL(root), AMI(admin): usernames and passwords;
    username: admin
    password: @steriskRocks1 (change this, here’s a good guide to start you off with http://www.freepbx.org/support/documentation/installation/first-steps-after-installation)
  • REMEMBER REMEMBER REMEMBER: CHANGE PASSWORDS!
  • The network adapter is set to auto on eth0.
  • Image needs at least 384M memory (or more if you have more)
  • All source files except kernel-headers are removed to save disk space for downloading, you need to download them manually
  • Be sure your image can access internet when starting otherwise NTP and EXIM will start slow, don’t blame me!

OS features/settings

  • Debian 6.0.7 64bit (Source AMD64 netinstall) – UPDATED, Bash Vulnerability Fixed with latest patch no33, SSLV3 disabled and Ghost Vulnerability fixed. All binaries are retrieved from Debian’s 6 LTS repos. So they are up to date.
  • The interface, extX, is set to use DHCP, so be sure to hook up DHCP or manually. In case you can’t bring the interface up, run #ifconfig –a . Then edit the file in /etc/network/interfaces and set all values to correspond to the interface shown when you run ifconfig –a (not loopback of course)
  • IPV6 disabled
  • MySQL backend (performance tuned)
  • Webmin installed but not started (# /etc/init.d/webmin start , then access using https://<ipaddress>:10000)  - UPDATED!
  • Apache as webserver with enforced HTTPS
  • MySQL administration with Adminer in https://<ipaddress>/dbmanager  - UPDATED!
  • DHCP and TFTP server downloaded, not installed
  • Firewalled with IPTables (be sure to see /bin/wallfire.sh) – UPDATED and fixed wallfire.sh script …can be stopped and started #wallfire stop #wallfire start
  • Time i.e NTP autosyncs with ntp.org daily, when starting and when stopping
  • Exim4 (mailserver) configured to relay, configure your email appropriately #dpkg-reconfigure exim4-config
  • fail2ban for Asterisk and SSH with enhancement to the log checking facility which includes asterisk security channel inside messages log (modify notification email here /etc/fail2ban/jail.conf) – UPDATED to 0.9.1!
  • Munin for monitoring in https://<ipaddress>/munin
  • Phpsysinfo for server information in https://<ipaddress>/phpsysinfo
  • Many CLI tools for troubleshooting like tcpdump, ntop, htop…
  • Astribank support [if ever u need it]
  • Removed VirtualBox OSE support to make it more cross platform compatible.
Asterisk features
FreePBX features
  • https://<ipaddress> to access FreePBX
  • FreePBX 2.11.0.38 (with only basic modules pre-installed) - UPDATED
  • Enhanced FreePBX security built in
  • SIP defaults to NAT yes (avoid all one way audio issue)
  • Security basic hardening in extensions
  • CEL support in FreePBX CDR
  • Enabled g729, speex and silk (enabled for IAX and SIP)
  • Most services are started with /etc/init.d/btelsvc

Additional reading

 

As usual do give me your feedback. ==> sanjay(the at symbol)astiostech.com

Thanks!
Sanjay Willie

11 comments:

Anonymous said...

Thanks. Downloaded and worked fine. One question how do i duplicate this? Is possible?

Admin said...

Yes you can, as long as the network interface is reset.

Shunil said...

Thanks so much for providing this image!!! I'm completely new to asterisk/freepbx and now I can start playing w/ it immediately w/o having to go through the installation process (which I should still do some time).

I got the VM working under VM Fusion and just had to add a new network interface in Fusion and I was good to go.

Thanks again!

http://www.Gabriellai.com said...

Hi,

How can I configure to work with my local telephone line? so that I can make calls to local people using local landline.

Thanks

http://www.Gabriellai.com said...

Hi,

Can I configure this to work with local telephone line? How can I achieve that so that I can call local people with local landline from Asterisk

Sanjay Willie (Astiostech Account) said...

Yes but you need a SIP account to telco or a SIP gateway device that will then connect to an analog or PRI end to telco.

Raviprasad Mangalore said...


Really good article....Thanks a lot...But if you don't mind, may i know how you customized boot screen? Which files i need to edit to put my name on the login screen OR boot screen?

Anonymous said...

Thank you very much, Sanjay. This is very usefull for beginners.

Omid Mohajerani said...

Great Sanjay .

Im Using Webrtc2sip in with asterisk and video Calling is working great as well .

Matthew Ogden said...

I tried taking this distro, and setting up a extension to be fax enabled, and then an inbound route to pass to that Fax Recepient > Extension.

It doesn't work, it says:
-- Executing [s@from-trunk:5] Set("SIP/FAX-0000000c", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@from-trunk:6] Set("SIP/FAX-0000000c", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [s@from-trunk:7] Goto("SIP/FAX-0000000c", "ext-fax,9002,1") in new stack
-- Goto (ext-fax,9002,1)
[2013-12-13 23:21:13] WARNING[15110][C-0000003d]: pbx.c:6390 __ast_pbx_run: Channel 'SIP/FAX-0000000c' sent to invalid extension but no invalid handler: context,exten,priority=ext-fax,9002,1

Matthew Ogden said...

I downloaded this distro and setup a trunk. I added a extension that had fax enabled. I added an inbound route to push to Fax Recipient -> Exten 9002

But it doesn't seem to work, I get the below error, is there something wrong in the distro?
-- Executing [s@from-trunk:5] Set("SIP/FAX-0000000c", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@from-trunk:6] Set("SIP/FAX-0000000c", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [s@from-trunk:7] Goto("SIP/FAX-0000000c", "ext-fax,9002,1") in new stack
-- Goto (ext-fax,9002,1)
[2013-12-13 23:21:13] WARNING[15110][C-0000003d]: pbx.c:6390 __ast_pbx_run: Channel 'SIP/FAX-0000000c' sent to invalid extension but no invalid handler: context,exten,priority=ext-fax,9002,1