Asterisk Case Studies

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Enterprise open-source VoIP with Asterisk



A lot of people want to know if it is possible to build an enterprise-grade open-source VoIP solution and if there are any benefits to it.

 

The answer is that it is not only possible, but there are special features that are exclusive Asterisk which is an open-source IP telephony platform.  Asterisk not only serves as an IP call signaling server (sometimes referred to as an IP PBX), but it also serves as a tradition TDM/analog PBX and seamlessly bridges the gap between the two worlds.

In Figure A, we have an enterprise class hybrid IP/TDM/analog telephony solution.  Note that there is no such thing as a "pure" IP solution because there is no way to avoid analog devices and the analog telephony POTS (Plain Old Telephone Service) world.  At some point a company has to support an analog Fax machine or an analog telephone port for some thing.  The only question is how many analog phones versus how many IP phones do you use and who gets which phones.

In my "Open Source VoIP" illustration, the solution is made up of several key components.  They are:

IP PBX (Asterisk servers)
PBX is a very generic term that usually indicates some kind of central box that controls all the telephones.  The term IP PBX is a very loose term and usually means there is some kind of call signaling server that sets up and negotiates IP to IP phone or IP to analog POTS (Plain Old Telephone Service) calls.  The PBX also manages the phone system in general.  In our particular illustration, we have two redundant Asterisk servers that act as the IP PBX solution.  The server hardware for Asterisk in this scenario could be a generic "white box" 1U server costing US$1,000 to US$2,500 or a name brand server from an IBM, Dell, or HP costing US$1,500 to US$5,000.  That's the beauty of Asterisk software, is that it can run on commodity x86 Intel or AMD hardware and the cost savings are enormous compared to proprietary PBX systems that cost 10 times more.

Voicemail and Fax servers
Voicemail and Fax server functionality can be rolled in to a single Asterisk server.  The beauty of this approach is that a commodity server will have massive storage capacity compared to a proprietary voice mail solution.  Voice mails can be emailed.  There are smaller Internet based telephone companies that use Asterisk to host production voice mail today.  Asterisk has the following voicemail features:

Ethernet to T1 Bridge
An Ethernet to T1 Bridge such as the RedFone foneBRIDGE can be used to link multiple servers to 4 T1 devices.  Even though a slightly cheaper quad T1 PCI card can be used directly in an Asterisk server, it forces you to commit those T1 resources to that particular server.  The foneBRIDGE allows you the flexibility to use those T1 resources on multiple servers and be more robust in a server failure scenario.  T1 interfaces connect to the Telephone Company or Channel Banks that can break out to many analog devices.  The foneBRIDGE is priced US$2,200, which costs more than a Quad T1 PCI adapter but it isn't locked to any specific server.  If more than 4 T1s are desired, additional foneBRIDGE devices can be added.  Asterisk's parent company Digium makes PCI T1 and E1 adapters which are natively supported in Asterisk.

Channel banks
Channel banks allow you to break a T1 line in to 24 independent FXS or FXO ports.  So if you bought 1 foneBRIDGE that used up a T1 to connect to the Telco and used 3 T1 connections to link up to 3 24-port Channel Banks, you would be able to serve 72 analog telephone or fax devices.  Here is a great resource that breaks down some of the Channel Bank solutions and vendors.  24 port Channel Banks range in pricing from US$700 to US$1,500.  Considering the cost of Power-Over-Ethernet, powered FXS analog ports are about half the price of a powered IP phone port.  If you don't need so many analog ports and you're not interested in building redundant Asterisk servers, you can skip the foneBRIDGE and the big Channel Bank by going with a USB device like the Astribank 8.

An FXS port on a Channel Bank allows you to plug an analog phone in to it and the FXS will supply power to the phone.  An FXO port on a Channel Bank allows uplink to a telephone company which is providing you an FXS port in the form of an RJ-11 jack.  Either of the following connection scenarios are valid.

Ethernet Switch segments
The thick tubes illustrated in the drawing represent distinct Ethernet segments.  This could be a physically separate switch or an isolated VLAN on a switch.  The internal LAN segment and the VoIP segment are separated by a router/firewall device.  Having the firewall is highly recommended if you don't want your phones and phone system hacked or infected.

The TDMoE (Time Division Multiplexing over Ethernet) LAN segment is for PBX-Voicemail and PBX-Channel Bank communications.  In the past, TDM communications between these devices used expensive T1 cards, but this has been vastly improved by sending TDM communications over inexpensive Ethernet.

Analog phones
Analog phones are the regular phones we've been using for a hundred years.  They are simple, inexpensive, and well understood.  They don't have as many features as an IP phone, but they do a great basic job for many people.  An analog phone has an FXO RJ-11 port that connects directly to a power-providing FXS port on a Channel Bank or provided by the Telco on the wall socket.

Analog faxes
While server based faxing has revolutionized faxing in the office place, there are times that a simple fax machine is still needed.  A fax machine like the analog phone has an FXO RJ-11 port that connects directly to a power-providing FXS port on a Channel Bank or provided by the Telco on the wall socket.

CAT-5 Ethernet wiring
CAT-5, CAT-5e, or CAT-6 cabling are used for 10-Base T, 100-Base T, or even gigabit Ethernet.  CAT-5 is the most generic Ethernet cabling.

T1 wiring (red)
T1 wiring is used to connect T1 ports between PBXs, Channel Banks, and the Telco.  A T1 carries 24 standard telephone channels which is why a single T1 port can be split in to 24 individual FXS or FXO ports or a combination of FXO/FXS ports.  A T1 connected to a Telco can carry 24 simultaneous calls.

Standard analog phone cabling
This is usually just a pair of copper cabling with RJ-11 connectors on both ends.  These cables carry enough power to feed a simple analog phone.  These cables are used between the Telco, the analog phone, the analog fax, and the Channel Bank.

Router/Firewall devices
A router connects multiple Ethernet segments.  Having an integrated Firewall or at least a simple router ACL can limit the traffic flow between two Ethernet segments.  For security reasons, it's always a good idea to restrict traffic flow to a minimum.

IP Phones
IP phones are actually special purpose dedicated computers that run VoIP software.  IP phones cost anywhere from US$70 for a very basic unit to US$700 units that have large color LCDs for customer applications.  IP phones can be powered via Ethernet with the 802.3af POE (Power over Ethernet) standard or Cisco's proprietary POE for Cisco IP phones or a separate AC Adapter.  Adding POE to an Ethernet Switch costs an extra US$50 per port over the cost of a regular Ethernet Switch.  Here is a good list of vendors that offer IP phones.

Computers with software IP phones
Software based IP phones or "softphones" run on general purpose computers and are very similar in function to dedicated IP Phones.  Softphone prices range from free to US$50 and here is a good list of softphones.

PDAs with software IP phones
PDA based softphones are identical to PC based software phones only they run on small PDAs instead of full blown laptops and desktop computers.

Telco cloud
The Telco cloud represents any generic telephone company.  Companies can connect to a Telco over a simple single-channel pair of copper cabling via the RJ-11 jack on the wall or they can trunk 24 channels over a single T1 line.  This is still the most common approach in Telco connectivity.

There is a new breed of telephone companies that operate over the Internet and let you connect directly to them using a cheaper Internet connection.  One unique benefit of Asterisk is the fact that it has its own IAX (Inter Asterisk Exchange) protocol which allows the efficient multiplexing of multiple VoIP channels.  This allows you to multiplex over 180 VoIP conversations over a single T1 Internet or Frame Relay connection whereas a normal T1 can only carry 23 voice channels.

Source:  ZDNet Asia

 

Written by Dal on July 10th, 2006 with no comments.
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Consolidation Of Your Business Phone System (PBX) With An Asterisk VoIP Solution

Note:  This is an informative article talking ways Asterisk can help your business with multiple office through consolidation.

Here's the scenario.......

You're planning planning ahead for a consolidation of your business phone systems including a potential move of your headquarters to a new building.

Currently your company has 300 employees and operates in 15 locations:

- 6 warehouse locations with business offices (~30 - 50 employees each)

- 1 small warehouse (5 employees)

- 2 business offices (~10 employees each)

- 7 small stores (3-4 employees each) - 2 share space with warehouse locations

You also have some outside sale folks that work from home most of the time.

Currently you run several disconnected phone systems and some Centrex (store locations). You'd like to standardize on one platform with integrated voicemail for the company. The plan is to do this in the next 1-2 years, whether or not you move to a new building.
 

All of these facilities are connected data-wise via a private routed network served by a Tier 1 carrier. Your headquarters is the hub for these locations and currently hosts all of the data servers.

When and if you move to a new facility your boss is considering outsourcing the mainframe and server systems such that all of the equipment is hosted by a separate company. This would relieve you of the considerations of building a server room in the new place. You do currently have a raised-floor server room, where your current phone system is located.

Of course with no server room (if you go that route), this limits your ability to host a PBX (you currently use a ROLM 9751).

Here's the questions you should like ask....and ensure answers for:

1. In a hosted PBX or VoIP solution, or even with a centralized on-site PBX can you still keep local numbers for each location?

2. If your equipment is centrally located, how do local calls work? e.g. - if your phone system is located in Maryland and someone in New Jersey needs to make a local call, is that really a long-distance call since the equipment is in Maryland? How is this typically handled?

3. What about DID numbers? Can you keep these? How are they routed?

4. What would a company do in terms of having a local operator at larger locations? Is there a sort of gatekeeper in place at these locations, or would it all be centralized at one site?

5. Currently you use a different automated attendant setup at a few of your locations. Would this still be possible or even recommended?

6. What is the usual way of connecting multiple sites to a centralized telephone system? What type of backup links are typically used?

7. You figure moving to a completely new system would cost around $1,000 per user (phone equipment, initial setup, new phones, training). Much less for a hosted system, but a high MRC you suppose. Is this estimate in the ballpark?

8. What recommendations can you expect on what type of systems may "fit the bill"? Some features you're looking for are below:

- Outside sales would like to be able to forward their lines to a home/cell phone.

- Internet access to change user settings would be nice (web-based user management).

- You have several Inside Sales queues, so you'd need good ACD capabilities.

- Ability to dial by extension to anyone at another location.

- Distribution lists for voicemail.

- Custom on-hold messages by location (different or store locations).

- Local paging at your warehouse locations (page over intercom).

- Local directions to your supplier truck drivers.

- After-hours/emergency messages need to be customized by location. (For example, if your Pittsburgh office is closed due to snow).

9. What about backup analog lines? Since you have a large inside sales presence, the ability to receive phone calls is critical. What is a good number of lines (percentage of total trunks, perhaps?) that are required and how are they usually setup?
 
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Written by Dal on January 1st, 1970 with no comments.
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Toss your PBX: Why Asterisk may be the VoIP future of your network

Note:  ComputerWorld has this nice write about Asterisk and what its benefits and limitations currently are.  What I really like about this article is the only real limitations they mentioned was that some companies are leery of open source software and the fact that the Asterisk integrator in the article did not has enterprise level experience.  Personally I know of a couple companies that do these types of installation.  It is true it is not for the faint of heart but if you have been working with Asterisk since 2004-05 you should be fine.
 
"Here's your network's dirty little secret: Your private branch exchange (PBX) is old and outdated, and if you want to bring it into the modern era with IP telephony and voice over IP (VoIP), you're going to have to spend a bundle. Specialized switches and hardware and proprietary systems don't come cheap, and they might not even offer all the telephony features you're looking for." 

But there is an alternative, as thousands of businesses and network administrators have discovered. The open-source Asterisk PBX has been gaining a big following, offering surprisingly powerful telephony features on inexpensive hardware. Not only has it been saving companies money, but it has also been able to integrate telephony with network applications in ways that previously might not have been possible.

But Asterisk isn't for everyone. And there are issues you need to confront if you plan a move to Asterisk. So here, in a nutshell, is what you need to know about Asterisk, along with advice from those who have already deployed it.

What is Asterisk?

Let's start with the basics: Asterisk is open-source PBX software that runs on a wide variety of operating systems, including Windows, Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris. It can run on inexpensive, off-the-shelf hardware, and it includes high-end features such as interactive voice response, voice mail, conference calling, and automatic call distribution and routing that have until now only been available on proprietary PBXs.

It's also exceedingly flexible. New functions can be created by writing scripts in Asterisk's language, by writing modules in C, and by writing scripts in Perl or other languages.

Particularly important is that it handles VoIP calls and works with a variety of VoIP protocols, including the Session Initiation Protocol (SIP) and H.323. It also functions as a gateway between IP phones and the public switched telephone network.

All this means that it can be used to create powerful, programmable PBXs at a low cost, says Joshua Stephens, CEO of Switchvox, a San Diego-based integrator and provider of PBX systems, including many built using Asterisk.

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Written by Dal on January 1st, 1970 with no comments.
Read more articles on Asterisk Case Studies.

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