Transcript Asterisk@Home Tutorial

Asterisk@Home Tutorial
Kerry Garrison
Director of Technical Services
Tech Data Pros
(949) 502-7819 (888) I-DO-VOIP
[email protected]
http://techdatapros.com
Publisher
http://VOIPSpeak.net
Asterisk@Home
http://asteriskathome.sourceforge.net
What is a PBX?
 Private Branch Exchange
 Connects office telephony equipment to PSTN (Public
Switched Telephone Network)
 Manages internal extensions
 Voicemail / 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
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
What is Asterisk@Home
 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
AAH vs Competition
 Fonality PBXtra
 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
 Asterisk@Home
 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)
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
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)
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
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
Accessing Voicemail
 Asterisk’s voicemail is called Comedian Mail
 Alison
 From any extension or when dialing into the
system, dial *98 to enter the voicemail 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
Asterisk@Home
Extension Demonstration
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
Asterisk@Home
Ring Group Demonstration
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
Asterisk@Home
Queue Demonstration
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
Asterisk@Home
Trunk Demonstration
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
Asterisk@Home
Outbound Rules Demonstration
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
Asterisk@Home
DiD Demonstration
Basic Functions – Auto Attendant
 Most companies will want an autoattendant 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
Asterisk@Home
Auto Attendant Demonstration
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
Asterisk@Home
Incoming Calls Demonstration
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
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 email 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
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
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
Asterisk@Home
MeetMe Demonstration
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
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
SugarCRM
 SugarCRM is the premier commercial
open source customer relationship
management application provider,
breaking the rules set by conventional
CRM solutions.
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
Reporting (CDR)
 AAH Contains a good Call Data Reporting
system
 Add-ons include account codes
Questions & Answers
Thank you for coming
Kerry Garrison
Director of Technical Services
Tech Data Pros – http://techdatapros.com
(949) 502-7819 (888) I-DO-VOIP
[email protected]
Publisher
http://VOIPSpeak.net