Incoming Faxes

• With the fax software installed, you can enter a default fax extension for each channel group or SIP provider so when faxes are received, they are routed to this extension.
• You can set up an incoming call route so that one extension can handle both voice calls and faxes.
• You can set up an incoming call route so that all activity on an extension is treated as a fax and sent to the same extension. This is useful if you want a dedicated incoming fax extension.

Outgoing Faxes

• With the fax software installed, if you want to use a fax machine, you can dedicate its extension to just sending faxes.
• In the User Suite under Fax Options, there is a setting that can be enabled to treat all outgoing calls as faxes.
• If this is set to YES, Switchvox handles all outgoing activity as a fax. It receives a fax-file, puts it in the Fax Outbox folder in that extension’s mailbox, and sends the fax.
• If this is set to NO, this indicates that this extension may make voice calls and send faxes.

Summary

By treating incoming and outgoing calls as faxes, Switchvox immediately processes the call as a fax instead of waiting to detect fax tones before processing the call as a fax.

Fax License

There is one free fax license reserved for every Switchvox registration code. This can be accessed in the Admin Suite via the Digium Addon Products link.

You must have your 16-digit registration code to access the fax license.

This information was originally posted on the VoIP Insider blog.

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New hire trainings, especially ones designed to teach them about VoIP technology and solutions, are always interesting for me (even though I’ve covered the same or similar material dozens of times before). These trainings are interesting because they provide me with fresh perspectives as to how those unfamiliar with VoIP view the technology and how to go about implementing it.

Like how the majority of people totally forgot the grand importance networks play in VoIP. In fact most people look at me a little dumbfounded when we start our trainings out with a thorough dive into networking.

You see the thing with VoIP is that implementations that provide easy on-going management and consistent, continuous call quality are more a function of your networks (LAN and WAN), than your VoIP system.

High quality VoIP is more about what you don’t see: your network preparedness, your bandwidth availability and the reliability of your service providers than what you do see. That VoIP phone you’re salivating over is really nothing more than a figure head; symbolic of the system, but rather powerless to make the use or experience with VoIP easier.

So if you’re considering making the switch to VoIP and haven’t had a site assessment complete or considered your network preparedness, please do. Because a successful VoIP deployment by an large is all about your network.

P.S. Check out our FREE guide to VoIP networks for more information on getting your network ready for VoIP.

This information was originally posted on the VoIP Insider blog.

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More often than not, a customer will say to me, Chris we are implementing this brand new IP PBX and IP phones and we currently have an existing analog based paging system in place we wish to incorporate into the new VoIP solution, how can we accomplish this?

The costs of replacing this equipment may be high (since IP paging equipment is somewhat costly) and the overhead of installing this new equipment may require:

• New cable runs
• New mounting hardware
• Possible modifications to electrical runs

These additions and modifications may not be possible so incorporating existing paging systems into new IP Based solutions is a must.

Lucky for you, Cyberdata offers a few legacy-based paging devices that will help address this need.

Cyberdata Paging Gateway

The first is the Cyberdata Paging Gateway. Like any gateway does in the IP world, it integrates 2 different networks together. Most commonly, this is done over a voice circuit where voice gateways connect to standard RJ-11 analog POTS lines or analog telephones and allow communication to an IP network or IP based PBX. The same is true for the Cyberdata Paging gateway however this gateway is used to connect older legacy analog paging systems to an IP network.

To do this, Cyberdata uses an FXO port. Just like in standard telephony, FXO (Foreign Exchange Office) compliments FXS  (Foreign Exchange Station) devices. This FXO port on the paging gateway will interface with existing TAM interface, Zone controller, or other legacy paging equipment via a standard analog FXS port.

Remember, FXO uses FXS signaling and FXS uses FXO signaling. Once the communication and connection is made between the existing paging equipment with the analog FXS/FXO combination, the gateway now merely acts as another SIP extension on the IP PBX and communicates with the IP PBX over the IP network via its LAN port. All administrators need to do is simply configure a page group on their IP PBX, register the gateway via its designated SIP extension to the IP PBX, then add that extension to the page group.

When users dial the page group extension, the IP PBX will ring the SIP extension registered to the paging gateway and it will send the page to the analog paging equipment attached to it.

Here is a good look at how this solution works:

Cyberdata Zone Controller

The second device used in the paging world to incorporate older analog paging systems with new IP based PBX’s is the Cyberdata Zone Controller. Unlike the paging gateway, the zone controller contains (4) standard audio out ports which are used to connect to existing analog amplifiers. In turn, speakers or horns would then be connected to those amplifiers.

Once the standard audio connection is made between the zone controller and existing amplifiers, the zone controller will communicate to the IP PBX via its LAN port over the IP network. The zone controller contains 4 page zones and allows for up to 15 zone groups.

Here is a good look at how this product fits in the IP scheme:

Incorporating Existing Systems

As you can see, incorporating existing paging systems into new IP based solutions can be very easily accomplished without the need to add any additional overhead costs.

Part 4 of 4 of Designing and Implementing an IP Paging System is coming up next time.

This information was originally posted on the VoIP Insider blog.

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Part 1 is here.

To start off our second segment on IP Paging we will continue to talk about the two methods that IP Paging is delivered in:

• Unicast
• Multicast

In both of these paging environments, two types of paging can be performed. Below we will talk about the two types of paging and offer suggestions to products that “fit the bill” for each type of paging.

Desktop Paging

The first type of paging is desktop paging. Desktop paging is exactly what it sounds like, paging at the desktop. This usually involves an IP Phone that supports paging. During a desktop page, the IP PBX will usually be configured for a page group or groups depending on the scenario, and a user will simply dial into that page group to initiate the desktop page. The SIP extensions of the users inside that page group in the IP PBX are then desktop paged either via a unicast or multicast stream.

Desktop pages are delivered directly to the IP Phone, triggering it to immediately go off-hook and present the page of the speakerphone. So at bare minimum, from a hardware standpoint, you need an IP Phone that supports a speakerphone, which is about 95 percent of them on the market today. If you are unsure, please don’t hesitate to ask one of our dedicated sales representatives.

Desktop Paging Configuration

Desktop paging can also be configured to “play a beep” first before the page comes through the speaker to alert the user that a page is being delivered. Also, pages can be configured to interrupt a user on a call or wait until the user has completed the call to page through. In cases where a page will interrupt a user during a phone conversation, the party that this user is speaking with will not be able to hear the page; however this may cause a distraction to all users on the conversation.

Administrators configuring the page groups must be aware of the users inside the page group as well as the importance of each page. For instance, a car dealership may opt to mute the page until the sales man or parts department is off the phone, however a hospital may opt to interrupt the call because they are delivering time sensitive information. Most all IP Phones and SIP Based IP PBX’s support desktop paging. Please be aware that not all SIP DECT cordless handsets support paging (as they don’t have the ability to go-off hook automatically and/or don’t support a speakerphone). If you are unsure, please confirm with your VoIP experts first.

Overhead Paging

The second type of IP Paging is overhead paging. Great advancements have been made in this department over the past few years to cater to new IP based overhead paging solutions. Companies such as Cyberdata and Valcom have products such as SIP based overhead ceiling speakers, SIP based paging amplifiers and loudspeakers with horns to accompany them. Most of these products are powered via POE so installation is simplified with 1 cable.

There are also options for Wi-Fi connectivity but keep in mind; you still need to power it which may require an electrician to hard wire these units.  Also, configuration is very easy as most of these endpoints are based upon SIP standards and are easily configured with a SIP extension (just like you would with an IP Phone) via the products web GUI configuration. Simply add these endpoints SIP extensions to a new or existing page group on your SIP PBX and dial the page group extension.

Audio is transmitted through the overhead speaker or amplifier horn combination. For applications that require a louder horn, such as a noisy warehouse or manufacturing facility, Cyberdata offers a Loudspeaker for noisy environments which adds more decibels to the volume level allowing pages to be heard properly.

Of course, mounting options may be of concern, so Cyberdata offers a few types of mounts for their overhead paging speakers, such as a wall mount, flush ceiling mount, and wall mount with clock (nice for hallways in a hospital or school).

Overhead Paging Illustrations

Below are 2 illustrations (thanks to Cyberdata) that depict how each type of overhead paging endpoint interfaces with an IP PBX.

 Overhead

Loudspeakers

Next Segment: Part 3 of 4

Tune in next time for Part 3 of IP paging system requirements.

This information was originally posted on the VoIP Insider blog.

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In many new or existing voice over IP deployments, integrating existing paging systems or deploying new IP based paging systems is becoming a much needed necessity.

In this series we will first detail the two common methods that paging is delivered over an IP network utilizing, then discuss how these methods are delivered over network infrastructure such as switches and switches, an IP PBX such as asterisk, and paging endpoints such as IP Phones for desktop paging and overhead speakers and amplifiers that are SIP enabled for much bigger paging solutions.

After we discuss this, we will offer a few product suggestions designed to meet your needs for both desktop paging and overhead paging applications, and last but not least, how to integrate an existing analog based paging system with a new Voice over IP Phone system.

Unicast IP Paging

To fully understand how IP Paging functions, we must first look at the two types of paging that is supported on most IP based PBX’s. The first is called “unicast”, more commonly termed as a unicast page. Unicast pages are delivered on a one to one basis and in our case; the IP PBX is the source for this page. In most common unicast paging setups, administrators will setup a single or numerous page groups in the IP PBX configuration. They will do this by creating a page group by extension. For our example, let’s use a single page group and assign it extension 400.

Paging group 400 then contains (20) SIP extensions that belong to individual users on the system and these extensions are physical SIP endpoints such as an IP Phone. In unicast paging environments, when a user dials paging group 400 from their phone, the IP PBX sets up and maintains 20 individual SIP calls between itself and each IP Phone belonging to the page group 400.The PBX sends SIP and RTP audio traffic to and from each one of these endpoints individually. Now can anyone tell me why this method might not be the most beneficial to use?

OK, you guessed it, since unicast paging is designed to page on a one to one basis; the number of users contained within each page group is the number of simultaneous SIP calls the PBX has to initiate and maintain at that specific moment in time. As you can see, this can get quite taxing on the IP PBX’s CPU and processing power. If the CPU is taxed too much, it can possibly lock up the entire system causing detrimental impact to the server hardware, inevitably drop calls, and affect overall user’s productivity.

In our case, our PBX has to make 20 SIP calls simultaneously at that given time that the paging extension (400) is dialed. In larger applications where paging groups range from 50-100+ users, unicast paging is strongly advised against for this reason. Here is nice diagram of how unicast paging functions:

As you can see, unicast paging has some downfalls and definitely some limitations. In most cases, a typical IP PBX’s can handle around 20 or so concurrent calls without “working to much overtime”, and for those environments, unicast paging will function correctly without many issues. But what if you are one of those environments where you do have 50 or so users in a single page group, let’s say a college or hospital, or large manufacturing warehouse, unicast is just not going to cut it. For this type of environment, we must like multi-cast paging.

Multicast IP Paging

Multicast paging achieves the same outcome as unicast paging but delivers each page much differently. Multicast paging is based upon a one too many architecture. In most multi-cast paging environments, administrators will specify a single multi-cast paging address in the IP PBX setup. They will then configure each paging endpoint such as an IP Phone, overhead speaker, or amplifier, to “listen in” on that multicast addresses.

In a multicast scenario, when a user initiates a page, the page originates from the IP PBX, just like unicast paging however the PBX only sets up a single SIP and RTP audio path to the multicast paging addresses.

The IP Phones or other paging endpoints are always “listening” to this address and when RTP packets or audio is heard, the phone or paging endpoints merely play that audio stream. As you can see, paging groups that contain, oh let’s say 500 users could very easily receive a page using multicast as the PBX only makes 1 SIP call for its 500 users in the multicast page group. This also preserves and protects the IP PBX’s CPU and system resources to maintain a preferable operating environment. Here is a good look at how multi-cast paging takes place:

Unicast vs Multicast IP Paging

The easiest way I try to describe this method is radio stations. Radio stations stream their broadcast, and you, the listener, tune in to the desired radio station to hear what they are playing, one audio stream with possibly thousands of listeners tuning in. As you can see, multicast has its benefits over unicast paging, but there is a catch, not all paging endpoints such as IP Phones can support multicast paging.

Next Segment:  Part 2 of 4

Tune into our next segment where we will discuss the two different types of paging, desktop and overhead paging as well as suggest a few products to these individual paging requirements.

This information was originally posted on the VoIP Insider blog.

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If you are new to the VoIP game or are a seasoned veteran, we all know that in business…we need to be able to make and receive phone calls. Most businesses rely on this type of communication to handle the day to day business activities. Also, it is required in most cases of an emergency.  With the introduction of IP PBX’s and VoIP, we all know that there are great ways to reduce costs as they relate to your recurring  traditional phone “bills”, but what happens if your  IP PBX decides it wants to crash and fail. In either case, this is a recipe for disaster. What if you can’t make phone calls? Can you run your business? How much down-time can you afford? These are all questions you must ask yourself and consider when deploying a VoIP solution…because these are all very real scenarios that have happened and could happen to you, that is, if you don’t protect yourself.

Recently, I had the pleasure of attending a Xorcom hands-on technical training in Las Vegas, NV. Yes, I actually did pay attention in class and wasn’t distracted by those chirping slot machines just a short walk down the hall. If you don’t know of Xorcom, or they are new to you, you can find more information about them here.

During the training, Xorcom demonstrated the 1^st 100 percent reliable complete failover solution, they call Twinstar. Twinstar is Xorcom’s method of delivering a complete failover solution without any personal involvement. That means that you don’t have to get out of bed at 4am to cater to an alarm that your IP PBX is having issues. Rest assured, with the solution below, you will be 100 percent covered when it comes to making phone calls. Trust me, I seen it in action and you will too.

The Xorcom Twinstar solution comprises of (3) required hardware components and (1) software requirement. First, you need to have the following:

-A Xorcom Astribank. An Astribank is a USB peripheral device that connects to the PBX via USB. The Astribank will allow your VoIP solution to connect to the outside world, either via FXO PSTN, or Digital T1. The Aastribanks are modular and can support PRI signaling either via T1 or E1, and FXO/FXS PSTN RJ-11 connections. You will want to ensure you also suit your Astribank with the dual USB ports for Twinstar failover Support. You can check out the lineup of Astribanks here.

- (2) Xorcom XR2000 or XR3000 IP PBX’s. Xorcom manufacturer’s three models of IP PBX appliances, two of which are suitable for Twinstar failover. The Xorcom XR2000 appliance supports up to 200 users, 60 concurrent calls, with 1GB of RAM and optional RAID support on its hard drives. More details on the XR2000 can be found here. The XR3000 is Xorcom’s “big daddy” server that can support up to 1000 users, 300 concurrent calls, with 1GB of RAM with the option to upgrade to 4GB, optional RAID support on its hard drives, and an Intel Core 2 Duo processor with the option to upgrade this as well, to a Quad Core processor. More information on the XR3000 can be found here.

- Xorcom Twinstar LC0016 – This ensures your Astribanks come with the required 2^ND USB Port and necessary USB cables to connect to your Xorcom XR PBX, to support Twinstar. This also ensures your Astribanks come with the necessary firmware to support Twinstar.

- Complete PBX or Xorcom’s Elastix based Software- Lastly, in order to take advantage of the Twinstar failover solution, both XR appliances must be pre-loaded with Xorcom’s Complete PBX or Xorcom’s Elastix based software. The Elastix-based software is free of charge. The Complete PBX version is paid commercial software developed and offered only through Xorcom directly. When you are talking to your VoIPSupply sales rep, tell them that you want either Xorcom’s Elastix based software or their Complete PBX software loaded. If you would like the Complete PBX software, we can have a Xorcom representative contact you immediately after your hardware purchase through VoIPSupply to discuss pricing and feature support.

Now that we have covered your hardware and software requirements, let’s talk about how the solution works.

Step 1:

Let’s start with the Astribank suited for twinstar failover with dual USB ports. You will first connect your PRI (T1/E1) connections, PSTN RJ-11 FXO connections, and any analog telephones to your RJ-11 FXS connections on your Astribank. This will satisfy your means of making and receiving calls to and from the outside world.

Step 2:

Setup both of your XR servers whether they are the XR2000 or XR3000 with the SAME IDENTICAL ASTERISK CONFIGURATION. No my caps lock is not stuck. This is very important as it relates to the failover portion. Remember, your XR servers will need complete PBX software. Once you have loaded the identical configuration to each server, you will want to designate 1 server as the Primary server and the other as the secondary. Both servers will be on and running simultaneously, but we need to tell the Astribank which one is live (primary) and which one is the backup (secondary). Also, keep in mind, when you make an asterisk configuration change on the main server, you will want to copy or replicate those changes to the secondary server. You should do this daily as you may never know when an incident or problem may happen. You can do this via DRBD Synchronization. More on this setup can be found in the technical guide linked below.

Step 3:

Connect your primary XR server to the Astribank’s USB (MAIN) Port. Then connect your secondary or backup server to the 2^nd USB Port or the (BACKUP) Port as pictured below.

Now that we have gone through steps 1-3, your hardware solution should look something like this:

Step 4:

OK, now this is where the configuration may get a little technical. Within the asterisk configuration, you will create a cluster IP address that encompasses both of your XR servers. For instance, my primary XR server’s IP address will be 192.168.1.2, my backup XR Server will be 192.168.1.3 and the cluster IP address will be 192.168.1.1. The cluster IP address encompasses both XR servers which you will use to register your IP Phones to. This allows your IP phones to easily failover if a problem is detected. I am aware of certain IP Phones support a backup SIP server option in their configuration but not all manufacturers do. In the twinstar failover solution, you can use any IP phone you want, just ensure you register them to the cluster IP address. Here is a small look at this setup:

Step 5:

Test the Twinstar Failover solution- Once steps 1-4 are complete and you are properly making and receiving phone calls from your IP phones or analog phones connected to Astribank, pull the power plug on your primary server. This will cause twinstar to take action and immediately failover all connections including PRI (T1 or E1), FXO or FXS, SIP Trunks, and your SIP phones to the backup server. This failover switch takes about 10-20 seconds to complete but requires no unplugging and movement of cables or equipment. After the switch, you are now back up and running on a fully functional phone system identical to the one that just failed without a hiccup…well there is one hiccup, if you were on a call during that transition to failover, your call will be dropped. This is next step for Xorcom, to achieve, a fully redundant failover solution without any in-call interruptions.

And that’s how the most reliable failover solution works… Below is a link to a short video presented by Xorcom that describes in a nutshell the twinstar failover solution described above.

http://www.xorcom.com/optional-extras/twinstar.html

A much more technical detail of how the twinstar solution is setup can be found here:

http://www.xorcom.com/images/stories/techdocs/twinstar-how-it-works.pdf

Feel free to give us a call and discuss your solution and options that can be explored with Xorcom and twinstar.

This information was originally posted on the VoIP Insider blog.

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Open source telephony applications have opened many opportunities in the VoIP industry and many companies such as trixbox, elastix, freepbx, pbxinaflash, and rhino, which are just a few, have taken the asterisk source code (licensed under the GNU GPL) and rounded it to their own business applications. Since the topic is “open-source”, the source code of Asterisk is distributed freely among companies, users, administrators, developers, and integrators alike and together have produced the world’s most powerful telephony engine and what that means to you, the VoIP customer is, a freely distributed source code or compiled ISO of a feature rich IP Phone system better termed as the IP PBX.

In the sections below, I will highlight some benefits and pros to using an open-sourced based phone system, also note some of the downsides or struggles you may have if you are not completely familiar with open-sourced telephony solutions.

Open Source Phone System Benefits and Upsides

Here are a few of the benefits/up-sides of open sourced telephony applications.

• The solutions are free – Well not completely free…You still need a server or hardware to run your software on and may require the use of telephony PSTN or Digital T1 cards to connect to the outside world. The software ISO or asterisk source code however, is completely free. You can obtain a downloadable version of asterisk/Linux or any other open source ISO from the leading players in the market such as trixbox CE, elastix, FreePBX, PBXinaflash, etc. Now you may be asking, what is the difference between Asterisk as just source code, and a software ISO? A Software ISO has been developed by a company such as the ones listed below, which have already compiled a version of Linux on top of a version of source code Asterisk, and also added a nice and easy to manage web GUI interface so administrators don’t need to rely on asterisk command line to administer their phone system. If you are novice to PBX software and asterisk, I would suggest downloading one of these ISO’s that can be found on each manufacturer’s website respectively and also mentioned in my previous post. Free also goes beyond the costs of the software. Most commercialized VoIP solutions require a per user license fee, concurrent call license fee, phone license fee, etc, and on top that may require the purchase of maintenance contracts and support contracts. With open-sourced applications, none of these fees apply and you are free to do as you please with your system, but also keep in mind, if you do need assistance, support is right around the corner, both free and paid options.
• Customizations – Customizability is a big deal breaker when it comes to phone systems. Since our phone systems are IP based, we can now integrate other applications such as databases, CRM tools, email, click to dial, presence applications such as HUD-Lite, and really any application or program you wish to integrate with Asterisk. In a typical open source distribution, administrators have access to the asterisk command line interface where most of this customizability is configured. In relation to this, most proprietary solutions don’t provide you with this access and rely solely on administering the PBX through a web GUI. Now don’t be mistaken here, commercialized VoIP applications still have most of these third party applications built in such as CRM integration, Google maps integration, database query retrievals, and everyone has outlook integration in the form of VM to email etc… but most of these applications are developed by the manufacturer of that system and developers within their particular dev groups. With open source, if you are a company that needs full customization and integration with very unique tools, that maybe even you, yourself have developed, open-source is the route to take.
• Resources available to you…and once again are free -  Open-source lives through it users, administrators, and developers. Information is shared everyday with the public in the form of online WIKI’s, forums, documentation, chat, and the Asterisk IRQ. That just names a few. WiKI’s and Forums pack a huge punch when it comes to information that is user related. Information on fixes, how to’s, issues that arise, and feature requests are all shared, and I always like to think, “If you have a problem or a request, 98 percent of the time, the answer is out there”, you just need to know where to look so below is a few links to a wealth of information on open-sourced VoIP applications.
  • www.asterisk.org
  • www.trixbox.org
  • www.elastix.org
  • www.freepbx.org
  • www.pbxinaflash.org
• Open source + open standards = interoperability – Interoperability with almost every VoIP endpoint in the market, including IP Phones, ATA’s, Gateways, PCI cards, WiFI Phones, SIP DECT solutions, etc… All interoperability is based upon SIP protocol and SIP standards. What this means to you is that, when using your open-sourced solution, you are not tied down to a specific phone model or manufacturer, you can use anything that is SIP standards based, and can easily mix and match phone models and manufacturers to meet end users needs and preferences. For instance, a company may be using Aastra phones for their desktop clients, but need a fully robust conference phone for their training room, no problem; Manufacturers such as Polycom, Snom, and Konftel can easily satisfy their conferencing needs with their conference phone offerings, as Aastra currently does not make a conferencing unit. In another situation, the same company may need a few mobile SIP DECT solutions for their warehouse employees, no problem, any of the Snom M3, Aastra MBU-400, or Siemens Gigaset DECT solutions will also work on their open-sourced based PBX. So as you can see, not only do you have a wide variety of endpoints to choose from, you also have the flexibility to mix and match SIP endpoints to meet every one of your users needs.
• Longevity – Asterisk was created in 1999 by Mark Spencer who later founded Digium. Since 1999, Asterisk has never looked back. It is constantly being updated with the latest feature sets, refined for better performance, and fixed for issues that may be found in earlier releases.  Major releases are constantly being updated and delivered to end users and administrators worldwide. Since the source code of asterisk is always under heavy development, and does rely a lot on field developers and system administrators, it is constantly being updated with the latest and greatest features and bug fixes from sources that have fully tested the solution and are committed 100 percent to its future. At this time, Asterisk has plenty of gas left in the tank to continuously bring us the best to offer if the IP telephony world. Since asterisk as a source code is always being updated, companies like the ones mentioned above are also releasing bigger and better solutions and if you are customer already running an asterisk based open-sourced solution, obtaining these updates on your current system is very easy and you guessed it, free of charge. This simply negates your required update and maintenance packages often times required on commercial and proprietary VoIP solutions.
• Professional Support services are available – So if you are not one of those people that likes to spend a lot of time researching an issue you may be having, or need a fix ASAP to guard against system downtime, professional Asterisk services are available from Digium and respective open-source manufacturers to help get the problem solved quickly, so you are 100 percent covered.

Potential downsides to Open Source Phone System Solutions

While there are plenty of upsides to using an open-source based phone system compared to a commercial or proprietary VoIP solution, there are some downsides to using these types of solutions. Below, we will detail out a few:

• Personal Knowledge – If you are person that is new to the whole asterisk, open source thing, I strongly suggest doing your research and be prepared to research information on your own. If you are not that kind of person, you may want to look at a commercial VoIP solution such as 3CX or Switchvox. If you are not familiar with the asterisk source code, and command line interface better known as the CLI, get used to it. If you need professional assistance in learning this information, Digium offers great hands on training classes all over the world that will help you succeed at this. Once you are satisfied with your knowledge, you can take the Digium DCAP exam and become a Digium Certified Asterisk Professional. More details on these trainings can be found here. This will help you immensely in your personal knowledge as well as assist you in administering your asterisk or open-source IP PBX.
• Testing (in general) – Test the solution, before you implement the new phone system in a production environment. It is often suggested to setup a small lab and use this as a test “sandbox” per say. Test the features of asterisk, its functionality, get used to administering it both from the CLI and Web GUI if applicable. This way if you run into requests down the road, you know exactly where to look. Even after fully testing a system, bugs can arise… hey not everybody is perfect. The problem with this is you may have to wait till a new distribution and patch is released for a fix, whereas on a commercial VoIP solution, you usually don’t run into issues like these.
• Testing your hardware – See my previous post for recommended hardware to run your asterisk or open source deployment. With commercial VoIP applications, in most cases, you purchase a known tested server already preloaded with software, and is a known given that everything is going to run smoothly. If you are building your own server and plan on installing asterisk or an open-source ISO, you need to ensure that the server is meeting your needs. Once again, see my previous post that explains most of these needs as it relates to hardware requirements as well as compatible motherboards and chipsets for your software to function correctly.  Once again, if you are not comfortable with doing this, give VoIP Supply a call and we can suggest a known compatible server for your open sourced asterisk software and we can even pre-install that load or ISO for you. No need to compile, burn to CD, then install.
• Getting your hands dirty…per say – Open source solutions do require a good amount of know-how as you can see from above, time which is spent on learning the code and how to administer the system, and also time spent on the hardware build, software download and compilation if you wish to choose the asterisk command line distribution. There may be some bumps in the road during all this and you will need to keep an open-mind towards it. Yes, even I get frustrated with solutions, but you need to know where to look to solve your problems, and learn from any mistakes or mishaps along the way. This inadvertently makes you a better Asterisk administrator, so leave your egos at the door and plan to dig in and expect issues.  If you simply just want a phone system, and don’t really care to know all of the ins and outs of asterisk, then open source is not for you, and you should go with a commercial VoIP solution which will offer you the same feature sets as Asterisk as in an open source world, without some of the hassle.

Stay tuned for my next installment where I will explain some of the pros and cons to a commercial VoIP solution, as well as speak about Proprietary solutions and how these compare to open-sourced solutions.

This information was originally posted on the VoIP Insider blog.

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Since the last time you’ve looked into a phone system, though, a lot has changed. Today you have more choice than ever.

Despite the choice and after dragging a fine-tooth over the feature sets and costs savings of VoIP vs. Analog phone systems, you’ll likely find that moving your phone system to VoIP is a conclusive no-brainer.

But that doesn’t mean that the PSTN isn’t useful, nor does it mean that you necessarily have to completely replace your existing phone system – thanks to VoIP Gateways.

VoIP gateway overview

VoIP gateways can come across as a bit confusing but hold in there as I will guide you via layman’s terms through these devices powerful capabilities and functionality.

VoIP gateways are a piece of hardware that have the standard purpose of converting Time Division Multiplexing (TDM) telephony traffic from the PSTN (Plain ‘ol Analog Lines) into digital IP packets for transport over an IP network. VoIP gateways can also be used to translate digital IP packets into TDM telephony traffic for transport across the PSTN.

That’s great but….how do VoIP gateways actually work and what does that mean to me?

VoIP gateways are actually rather simple. Think of a VoIP gateway as a bridge between the IP land and the Analog world. Depending on the origination of the voice traffic, a VoIP gateway will convert the voice traffic into proper form to be received by the destined network (IP or Analog). Meaning, Analog phones can be integrated into your VoIP system as can PSTN (POTS) lines.

VoIP gateway applications

Basically, there are three primary applications for a VoIP gateway (which also happen to be the most popular and common uses for a VoIP gateway).

• VoIP connectivity for your Analog PBX system – It is possible that you had just purchased an analog PBX and or your organization does not want to spring for a new VoIP system or is too large to execute a wholesale tear out. Using an FXS analog VoIP gateway or a digital VoIP gateway you can VoIP enable your existing PBX system. An FXS gateway affords the opportunity of the benefits of VoIP calling without the costs associated with installing a new VoIP System.
• Using Analog Endpoints (traditional telephones) with your VoIP system – For most organizations the phones sitting on desks represent the biggest investment made when purchasing a phone system. That’s why you may not want to replace your existing desktop phones. While analog phones clearly cannot shine a shoe to IP phones in terms of functionality, there is no need to waste a recent investment you have made into Analog phones or if they work, replace them. Enter the FXS gateway. In using a FXS gateway, you can connect traditional analog phones to your VoIP phone system giving you the ability to further the life of your investment on analog phones.
• PSTN connectivity for your VoIP phone system – The final use for a VoIP gateway is one that is used by those who have decided to replace their existing phone system with a VoIP system. Your ability to send and receive phone calls is critical to your business. With VoIP, if a network connection is down or internet connectivity is lost… you will not be able to send or receive calls (including emergency dials). Luckily, you can create a fail-safe for this scenario by implementing a FXO or digital VoIP gateway to connect to the PSTN giving you failover and life-line capabilities.

In closing, migrating to VoIP can be done using a VoIP gateway. Stay tuned as next week’s blog will describe in more detail how to leverage your existing handsets with a VoIP gateway.  If this guide has helped you further dive into the process of purchasing a VoIP gateway ot you have further questions, please do not hesitate to give me a call at 716.250.1990. I will do my best to further educate with honest and accurate information.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

Wait no longer. The Sangoma Fax Sync Cable can help.

All of you VoIP engineers, installers and end users have heard it, “Keep your faxing off of your VoIP system”

And for good reasons. Fax over IP historically works about 50% of the time, leaving most users scratching their heads as to why.

Let’s put it this way, when someone sends an email and the packets are interrupted, say one or two drop out, the email recipient simply sends a request to the sender, asking for the lost packets, the sender sends those packets again, the recipient re-arranges them and you have your email.

With VoIP on a SIP call, if packet loss occurs, the users may hear a bit of broken silence on the line for short periods of time, simply because voice traffic is considered “real time”. With faxing, if packet loss occurs, since fax machines are considered “dumb terminals” in the VoIP world, many times the fax never even makes it to the fax machine with no alert or notification, or the fax machine displays an error instead of the fax.

For companies conducting their main portion of their business over faxing, this is unacceptable. For health care applications where faxes may be considered “mission critical”, this may be detrimental to a patient’s health.

As you can see there are plenty of applications that are using VoIP and trying to successfully integrate their fax machines into their VoIP system and make it work every time. Sangoma technologies have designed a Sangoma Fax Sync Cable to do just that. Sangoma states that their “Faxing solution works 99 % of the time.”

To start, you’ll need a few items:

• IP PBX Server (Phonebochs, trixbox, Asterisk, ect…)
• Sangoma Digital PCI card. You can use the A101, 102, 104, 108 and so on and so forth. Or if need be, you can use BRI or any other form of Digital communication. (It is HIGHLY recommended to use echo cancellation on this card)
• Sangoma Analog PCI card. You can use any of the combination’s of FXS/ FXO cards you wish (A200 or A400 Series), but you must have at least 1 FXS Mod (2 ports total) to hook up to the fax machine. (It is not required, but still recommended to get echo cancellation on this card)
• The latest version of Drivers for the Analog and Digital PCI cards. (Trixbox and Phonebochs already have these.)

Here’s how it works:

• Faxes coming into a VoIP PBX system over T1 PRI lines are taken in by a Sangoma A101D (Single T1 PCI card with echo cancellation).
• Also included with the VoIP PBX is a Sangoma A200 (RJ-11) or A400 (DB-25 which may require a breakout box and the use of a Sangoma Y cable) series FXS card. This card is where you will physically connect your analog fax machines into.
• Lastly, connect the Sangoma Fax Sync cable from the A101D T1 card to the Sangoma A200/A400 FXS card via each card’s jumper connections. This cable needs no additional configuration and simply put by Sangoma,“All T1/E1 & BRI cards supporting this feature will automatically provide the clock, and therefore no additional configuration is necessary.”

You’re all set!!!, the picture below describes this setup using a Sangoma A400 series card:

For more information on the Fax Sync Faxing over VoIP solution, please visit Sangoma’s WIKI located here or contact VoIP Supply at 800.398.8647.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

First add your WIP310 as a device (Other Device)

Trixbox will then assign a SIP Username and Password (noted after you have added the WIP310 as a device)

Also, in the paragraph above the “add a device a section”, you will notice a domain name address to point your WIP310 to. There are both a local and non-local domain name to use. You MUST use either of the two depending on if your WIP310 is local to the PBX or not. (See Figure 1 Below)

Figure 1

Next add your extension to the Trixbox PRO system (See Figure 2 Below)

Figure 2

Once completed, use the document below to complete your setup, making the following substitutions in the registration section (Step 18):

Click Here to Download Linksys WIP 310 Basic SIP Registration Instructions

Proxy: sxxxxxxx.trixbox.fonality.com (from the paragraph in the device phones tab)
Outbound Proxy: sxxxxxxx.trixbox.fonality.com (from the paragraph in the device phones tab)
Display Name: (From device phone section)
User ID: (From device phone section)
Password: (From device phone section)

Save all changes. That should do it!

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!