1. Asterisk@Home Tutorial
Kerry Garrison Director of Technical Services Tech Data Pros (949) (888) I-DO-VOIP Publisher

3. What is a PBX? Private Branch Exchange
Connects office telephony equipment to PSTN (Public Switched Telephone Network)
Manages internal extensions
Voic / Message Indicators
Transfers / Hold / Conf Calls
Typically large box hanging on a wall somewhere in “the phone room”
Expensive
Difficult to manage (have to call the phone guy)
Very limited in choices of telephones

4. Asterisk PBX Open Source Software (Free) Runs on standard PC hardware
Uses inexpensive cards to connect to PSTN, T1/E1, ISDN
Ability to use ITSP’s
Uses standard protocols (SIP, IAX)
Lots of telephone choices
By itself, is not very easy to maintain

5. What is
Complete ISO image that installs CentOS Linux and Asterisk PBX
AAH is a FULL VERSION of Asterisk and is not limited in any way!
Installs in about an hour
Includes web-based management tools
AMP (Asterisk Management Portal)
Handbook project is under way
Lots of community support
Geek Gazette
Nerd Vittles
Slashdot
VOIPSpeak.net

6. AAH vs Competition Fonality PBXtra SwitchVox Asterisk@Home
Pre-packaged system ready to install
Limited telephone support
Good for small systems
SwitchVox
Excellent interface
Limited hardware support
System is locked down except via web interface
Interface is not very attractive (AMP)
Will run on wide variety of hardware (not always a good thing)
Full access to config files and CLI (command line interface)

7. AAH Hardware Compatibility
Server requires minimum hardware specs
We have run it on PIII 500mhz 384mb RAM
Softphone
X-Lite
SJPhone
IAXComm
Hard Phone
Sipura SPA-841
Grandstream GXP-2000
Polycom VOIP Phones
Cisco VOIP Phones
SNOM SIP Phones
Zultys VOIP Phones
Many others
Analog Telephone Adapter
Sipura ATA’s
Grandstream ATA’s
Cisco ATA’s
Digium IAXy
Others

8. Telephony Connectivity
ITSP Service
BroadVoice
IAX.cc
VoicePulse
VoipJet
Many, many others
PSTN Connection
Intel Chipset modem (X100P Cards)
Digium FXO/FXS, T1, E1, etc
Sipura SPA-3000 (PSTN Connection)

9. Telephone Connectivity
A brief word on using ITSP’s
Our company has tested over a dozen and so far have all been very reliable with Broadvoice being the primary exception
If you are using your ITSP DID phone number as your primary number, what happens when your connectivity is down or your ITSP is down? Build for this scenario!!!
Do not share your data traffic with your phone traffic, use a dedicated broadband connection for your phones, downloading a Windows update onto a workstation is enough to destroy your phone service
Don’t put all your eggs into one basket, get setup with at least two ITSP’s so you have some level of failover
How does using an ITSP save you money?
Most do not have monthly service charges, this can save you hundreds of dollars a month right there
Rates are usually 1.5 – 2 cents per minute, this can be a minor cost savings
Elimination of long distance charges across the US and often into dozens of other countries. Depending on your phone usage, this can be a massive savings

10. Basic Functions - Extensions
An extension is an individually addressable location
Mailbox
Telephone
Mailboxes and telephone devices may be tied together via the AMP interface
Ring Group
Queue

11. Accessing Voicemail Asterisk’s voicemail is called Comedian Mail
Alison
From any extension or when dialing into the system, dial *98 to enter the voic system.
You will be given voice prompts telling you what to do
Using *97 will take you directly to the voice mailbox of the extension you are on
You will then be asked for your password

12. Extension Demonstration
Extension Demonstration

13. Basic Functions – Ring Groups
A ring group is a group of extensions tied together under one parent extension
When a ring group extension is dialed, all of the phones in that ring group ring at the same time, the first to pick up takes the call
Ring groups can consist of external phone numbers such as cell phones
A ring group has several settings to determine how the calls are handled

14. Ring Group Demonstration
Ring Group Demonstration

15. Basic Functions - Queues
A queue is a holding area for inbound calls so that callers can sit on hold waiting for someone to answer instead of getting a busy signal or being forced to immediately leave a message
The Asterisk queue system can tell callers their place in the queue and the estimated wait time
Agents must be logged into the queue for calls to be routed to them

16.
Queue Demonstration

17. Basic Functions - Trunks
A trunk is a circuit that defines an inbound or outbound connection configuration.
Zaptel is the standard PSTN trunk
SIP/IAX Trunks are for ITSP connections
Some trunks may handle inbound, outbound, or both

18.
Trunk Demonstration

19. Basic Functions - Outbound Rules
Outbound rules define what paths an outgoing call will take
An outbound rule with multiple trunks assigned acts as a failover in case the preceding trunk is not available
Outbound rules are best used for least-cost routing by sending certain calls over specific trunks that have the most favorable calling rates for the call destination

20. Outbound Rules Demonstration
Outbound Rules Demonstration

21. Basic Functions - DiD DiD stands for Direct In-Dial
Rules set where a call from a phone number will go to
Employees with their own phone numbers
Fax machines
Toll-Free numbers
All inbound lines “should” have a DiD set for future compatibility and maintenance

22.
DiD Demonstration

23. Basic Functions – Auto Attendant
Most companies will want an auto-attendant or “IVR” (Interactive Voice Response) system for inbound calls
Building a basic menu system in AMP is fairly simple
Complex, multi-level IVR systems are also possible with AMP/AAH

24. Auto Attendant Demonstration
Auto Attendant Demonstration

25. Basic Functions – Incoming Calls
The Incoming Calls configuration ties all the inbound configuration together
Sets “day” and “night” hours
Sets where incoming calls go to

26. Incoming Calls Demonstration
Incoming Calls Demonstration

27. Advanced Settings - NAT
There are special considerations to be made when running your PBX behind a router
This really only affects remote extensions and ITSP connectivity
Edit sip.conf and set the localnet and externip settings
Remote extensions must have NAT=yes in their configuation

28. Advanced Settings – Time & Network
Use netconfig to set the IP settings on the server
Use timeconfig to set the current date and time
If you have to send outbound through a specific host (i.e. Cox cable) then edit the sendmail.cf file and set the SmartHost setting to your SMTP server
# "Smart" relay host (may be null) DSsmtp.west.cox.net

29. Advanced Settings – Updating CentOS
Yes, just like Windows, Linux system have regular updates too, be sure and keep your server up-to-date.
yum –y update

30. Advanced Settings – Web Meetme
Web MeetMe is a conference room system for use by all users
Prepend 8 to the extensions to access that extension’s MeetMe room
For extension 200, use 8200
You can control the room via the web interface

31.
MeetMe Demonstration

32. Advanced Settings – Updating Asterisk
In the past, the AAH install included a script to update to the current HEAD version of Asterisk, while this worked in the past, the next version of Asterisk has so many changes, that a simple upgrade script isn’t going to be feasible
With AAH 2.0, which will include the upcoming new version of Asterisk, getting back on a scripted upgrade path is most likely not going to be a problem

33. Advanced Settings – Remote Extensions
Setting up a remote user is no different than setting up a regular user
Take into consideration NAT traversal (localnet, externip on server and nat=yes on extension config)
Difficult configurations can sometimes be overcome by using a STUN server
IAX is less prone to NAT problems than SIP but very few remote devices support IAX today

34. SugarCRM
SugarCRM is the premier commercial open source customer relationship management application provider, breaking the rules set by conventional CRM solutions.

35. Flash Operator Panel Displays status of all connections
Extensions
Queues
Trunks
Enables basic operator functions
Transfer calls: by dragging the phone icon to the destination you want
Hang-up calls: by double clicking on the red button
Originate calls: by dragging an available extension to an available destination
Conference calls: You can add a third person to an existing conversation by dragging an available extension to a leg of an already connected call.
Mute/Unmute meetme members: just double click on the arrow of a meetme participant
Get information about last call: double click on the arrow of an available button

36. Reporting (CDR) AAH Contains a good Call Data Reporting system
Add-ons include account codes

37. Questions & Answers
Thank you for coming
Kerry Garrison Director of Technical Services Tech Data Pros – (949) (888) I-DO-VOIP Publisher