Hooking pbx up voip

If you are interested in Voice over IP, are not intimidated by open-source software, and don't want to part with tens of thousands of dollars up front, take a close look at Asterisk (www.asterisk.org). Developed largely by Mark Spencer, this Linux-based, open-source PBX replacement is an excellent appli-cation and can even be considered for use as your permanent VoIP gateway solution.

Asterisk provides most features of traditional PBXs, such as voice mail services, call conferencing, ACD (Automatic Call Distribution), and IVR (Interactive Voice Response), to name just a few. In addition, it can handle call queuing, bridging, and conference calling with the use of virtual conference rooms. With its automatic generation of CDRs (Call Detail Records), the solution also points to Mark Spencer's aspirations of having it become a viable alternative to traditional PBX solutions.

Using Asterisk was very satisfactory, and the overall call quality was excellent. What imperfections we experienced in the VoIP calls could be attributed to poor network conditions rather than Asterisk. The system's performance impressed us, especially considering that we had installed it on a 500-MHz Pentium III system with 256MB of RAM. This was fine for our testing purposes, in which we ran a household telephone system with only four phones. But if you are planning to use this in an office environment, you'll need to scale your hardware to match your performance needs. CPU utilization will quickly increase when multiple calls are in progress.

One challenge to keep in mind from the beginning: Depending on the hardware you select for your system (especially the phones), you might encounter glitches such as poor sound quality that have to be researched and then patched or fixed. This is a major downside to implementing an open-source solution; it usually does not provide an integrated end-to-end package, including hardware and software. While you have more flexibility in the hardware and software selection, you might find yourself having to solve any number of problems.

PROTOCOL AND CODEC SUPPORT

Support for multiple protocols including H.323, SIP (Session Initiation Protocol), and the proprietary protocol IAX2 (Inter-Asterisk eXchange 2) ensures connectivity to many different VoIP devices and applications.

For integrators who want to connect to traditional telephony networks, Asterisk supports a variety of Channel Associated Signaling (CAS) protocols for in-band signaling within the T1 protocol. Asterisk supports the most common standards for those limited to ISDN (Integrated Services Digital Network) PRI (Primary Rate Interface), such as 4ESS, DMS100, and EuroISDN.

Audio-encoding codecs available include G.711 ulaw (common in the U.S.), G.711 alaw (used in Europe), and G.723.1 or G.726 with an appropriate license. The latest development releases also support G.726.

To interconnect with digital or analog lines (aside from VoIP over your network links), Asterisk supports mainly hardware from its sponsor Digium (www.digium.com). This hardware provides connectivity solutions that range from single-port FXO (Foreign Exchange Office) PCI cards for interfacing with your standard analog line to quad-span E1 and T1 channel bank PCI cards for digital connectivity.

ASTERISK: GETTING STARTED

To set up and use Asterisk, you have to know your way around a Linux command line and be comfortable setting up a Linux server. The base for our installation was Red Hat 9. After a minimal installation of Red Hat, including just development applications and kernel development packages, we installed Asterisk using a CVS (Concurrent Versions System) download of the stable code release.

We used a Digium Wildcard X100P ($99.95 direct) for interfacing with an external analog phone line and a Wildcard TDM400P ($337) to support an analog telephone on the internal side. To support an office environment with 30 phone lines, you'll need one of the several types of E1/T1 cards available from Digium, each costing about $1,500. If you do not have any cards, you will be able to use Asterisk only as a VoIP gateway on your existing network connection. For outside connectivity we used a standard POTS circuit and a 1.5/768 DSL connection to the VoIP provider Voicepulse (www.voicepulse.com). Voicepulse offers Voicepulse Connect, a service that allows telephony service providers and developers to connect via the IAX2 protocol. This permits connecting Asterisk as a PBX gateway rather than just hooking up phones as SIP clients.

CONFIGURATION AND USE

Getting Asterisk running means diving head first into the many configuration files. There are two dozen such files (depending on the number of features you want to have available), and all of them can be edited using a standard text editor such as Vi or Emacs. The nature of Asterisk's modular design allows for a great deal of PBX customization. For instance, you can configure dial plans for all your extensions and route traffic as you see fit, to either your VoIP or analog connections.

Asterisk comes preconfigured with over 500 voice prompts, which have been recorded in a professional-sounding female voice and can be used in interactive menus. It records incoming voice mails using the GSM audio codec and in Wave format, and it stores the files in the appropriate user mailbox directory on the file system. If desired, voice mails can be sent to the user's e-mail address and viewed using any player that supports files in Wave format. Asterisk add-ons also allow users to check their voice mail through a Web browser once they're authenticated.

If you manage a large installation, you might want to take advantage of Asterisk's ability to store mailbox and user log-on information in a MySQL or PostgreSQL database. This also allows users to change their mailbox passwords via an interactive voice menu.

Full support for Caller ID information assists Asterisk in storing CDRs either in the local file system or in a database. Administrators can then use account codes on a per-channel or per-user basis for billing calculations.

In our setup, we used a Cisco 7960 IP telephone configured for SIP and an analog cordless phone connected to the Wildcard TDM400P. Asterisk had no problems routing calls from either phone to the analog line or Voicepulse connection, depending on what the dial plan prescribed. Overall call quality was excellent and seldom displayed any of the imperfections often noted with packet-routed voice calls. We experienced hardly any drop-outs, and echo cancellation worked well. Most of our complaints about sound quality involved the tone; VoIP can sound relatively harsh and metallic.

Asterisk has many more features, such as video call support, message-waiting indicators, call parking and transfer, paging, and intercom, among others. For a more detailed overview of Asterisk's setup, configuration, and capabilities, visit http://voip-info.org.

Overall, we were very impressed with Asterisk and the accompanying Digium hardware, especially considering the solution's capabilities and the relatively low cost of rolling it out—if you possess the needed Linux know-how.