Tutorial Kerry Garrison Director of Technical Services Tech Data Pros (949) (888) I-DO-VOIP Publisher

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

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  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   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 Voic  Asterisk’s voic 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

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

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

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

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

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

DiD Demonstration

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

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

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 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

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 – (949) (888) I-DO-VOIP Publisher