Voice 2.0 Anyone? (Part 1)

The coming of “Web 2.0″ has ushered in a whole lot of online services that we could only dream of when the World Wide Web was first conceived and commercialized. There were the blogs, where just about anyone can voice out their thoughts. Then there were the social networking sites, where you got to link up with friends. Then there came social bookmarkers, which let people share their bookmarks and places of interest on the Web. Then came the web applications. Software started to become web-based instead of client-based. Even the big players like Google started introducing (or acquiring) web-based word processors, spreadsheets, and productivity applications like calendars, to-do lists and wikis.

The point of “Web 2.0″ was seemingly the collaborative nature of everything, and as well the user-generated aspect of applications. The development and management of content is no longer centralized but is now distributed among the users.

What about “Voice 2.0″ then? VoIP News has this summary of what Voice 2.0 is all about.

“Voice 2.0” is an umbrella term for a loosely defined set of technologies and ideas that let people transmit voice, data, video and instant messages via IP, anytime, from anywhere. It also implies a world where users, rather than a central authority, will have much greater control over who they communicate with, how and when.

Successful Voice 2.0 applications could lead to large cost savings in telecom, and greater efficiencies in other areas.

Included in this umbrella term are various services and applications that users can utilize. The important point is that control is no longer in the hands of centralized organizations (like telcos), but in the hands of the user. Here are some features of this new concept.

Voice, video and data convergence. Applications today are increasingly able to transmit just about anything over the Internet. Communication, whether instant messaging, voice calls or video conferences, can now go through the same pipes instead of different connections. The need for Voice 2.0, then, is for all these to travel through IP via a common standard or protocol.

Mobile and fixed line convergence. The idea is to create all-in-one devices that can service both fixed line and mobile needs. While traditionally one would have separate gadgets for landline and cellular communications, Voice 2.0 endeavors that a user would be able to access either service from whichever device he chooses. For instance, if one is expecting a call via landline, ideally the system can be set up such that the call can be rerouted via SIP to his VoIP phone.

Your Telco Might Be Ripping You Off

VoIP is traditionally thought of as an alternative to the plain-old-telephone system (a.k.a. “POTS” or public switched telephone network). True, it’s usually cheap and would require none of the usual expensive switching equipment that telephone companies usually use. However, users of traditional telephone systems may not be aware that their telcos are ripping them off by routing calls via the Internet but still charging access fees.

Long distance and overseas calls are usually expensive because of these reasons: For international calls, international gateway facilities charge your local telephone company access fees whenever you dial someone from outside the counrty. For long distance, the same system applies, but this time, the telephone company of the person you’re calling charges your telco for access–or they both have access charges against each other.

The reason they feel they can justify charging expensive fees is because the international gateway facilities (IGFs) are expensive to set up and maintain, and they usually have to comply with regulatory requirements (such as franchises, permits, and the like) before they can run their business. In contrast, Internet service providers are usually viewed as value-added service providers instead of utilities–hence lower barriers and costs to entry.

However, with the onset of VoIP as a popular alternative, telcos have started to route their calls through the Internet, too. I have personally seen several prepaid services (that use calling cards) marketing themselves as good alternatives to the regular telephone system, but not exactly as VoIP solutions. You still have to contact your correspondents by dialing through your local phone system.

What I do notice when I try these services for calling overseas is that the voice quality is different. I mean, you will definitely be able to distinguish between a phone call that’s routed through an analog circuit, and one that’s packet-based. So what does this mean? My telephone company is actually routing my calls through VoIP.

What’s worse is that some telcos I know are charging the same rates for calls, but they’re routing the voice calls via the Internet. This means they’re no longer paying the IGF charges, but still charging us end-users the same rates!

Choosing a VoIP Gateway for Your Company (Part 2)

We gave a brief introduction on VoIP gateways recently, and now we move on to factors that businesses should consider in choosing a VoIP gateway.

VoIP gateways come in both hardware and software forms. However, for businesses, hardware-based solutions are more widely adopted, but can be more expensive. Many prefer hardware-based gateways as they’re considered more reliable, and provide their built-in interfaces. These also don’t consume computer processing power since they have their own internal processors. These can be available as stand-alone boxes, chassis cards or modules.

When choosing a hardware solution, one could usually spot the packet processing capacity judging from the size of the chassis–bigger usually means more powerful. Higher packet processing capacity is preferred, so you can avoid poor voice quality (and potentially lost business).

As for capacity, you should choose a gateway based on the simultaneous VoIP calls it can handle. When switching from a traditional phone system to VoIP, It’s important that your gateway can handle your network’s existing load plus some allowance. One good rule of thumb to follow is that your VoIP gateway should have at least 20% greater capacity than the current network load. This way, you have room for expansion and you have some allowance–your gateway can accommodate growth before you find the need to upgrade or replace it.

The number and types of interfaces are also important to interoperability. An adequate number and variety of ports will make connecting devices to your gateway easier. These devices come in different forms, such as billing systems, network management systems, and yes, even your interface to the traditional phone system.

Choosing a VoIP Gateway for Your Company (Part 1)

Gateways are essential aspects of any enterprise VoIP system. These transfer voice (and other traffic) between the Public Switched Telephone Network (PSTN) and the IP (Internet Protocol) network. This means VoIP gateways should be able to do much more than traditional PBXes that only interface your internal analog network with the PSTN. As such, you should also expect the gateway to handle other tasks, such as call management, and routing of voice traffic, and translation across the various VoIP protocols, when needed.

Organizations looking into adopting a VoIP solution may be doing so for several reasons. For some, it’s a way to mitigate expensive long-distance and even overseas charges, particularly if offices and branches are located far away from each other (spanning continents, for instance). For others, VoIP gateways offer a more feature-rich network than possible with traditional telephone exchanges.

VoIP gateways basically offer the following features and functionalities: packetization (translation of analog signals to digital packets), compression and decompression (“codecs”), control signaling and voice/packet routing. If you intend to buy a gateway for your company, your decision should go beyond these basics. For one, you should consider the ease of integration of the gateway with your existing PSTN and PBX. You should also consider the level of support that the vendor is provding. Then, of course, you should consider whether the gateway is compatible with your existing VoIP equipment and infrastructure (if any). Finally, there are the added stability and usability features that you might want to have on your system, such as PSTN failover (you can move to the analog line if the VoIP connection fails), H323 and SIP survivability, multiplexing, NAT transversal (if you’re working behind a corporate firewall).

Skype Unveils Unlimited Call Plan for North America

North American Skype users have a lot to be thankful for. First, there’s the ongoing free North American calls promo. If you’re anywhere within North America–meaning the United States, Canada, and territories–you can call any landline and mobile phone within the region for free using SkypeOut. This means you can save a lot on toll charges, especially when calling distant states or regions.

However, Skype has made it clear that this promo will only be valid for 2006. Luckily, they have unveiled a new SkypeOut calling plan, and while it’s not free it’s still a good deal.

Skype Unlimited is an unlimited calling plan to the North Americas that will require a one-time payment. This is only valid if the subscriber is in the US or Canada (and you will be notified by your Skype client if you’re eligible for this, when you sign up).

For $29.95, you get unlimited calls (at no additional charge) for 12 months after purchase. If you buy within January 2007, you only pay $14.95. The Skype blog even says you can even save more if you buy before 2007 starts, since you only get to pay $14.95, and your unlimited plan lasts up to end of 2007. However, I realized that North America to North America calls are still actually free up to end of 2006.

Still, this looks like a good deal. Or if you already have other calling plans, you might want to gift friends and relatives who are Skype users with unlimited SkypeOut calling plans this Christmas.

Paris Hilton Hacks Voicemail Using Asterisk

Sounds like an interesting title, right? Well, apparently it’s an issue that just brought the world of Hollywod closer to geekdom. However, Paris Hilton did not exactly have to know the inner workings of Asterisk to conduct her “hack attack.”

Last August, Paris allegedly spoofed fellow celeb Lindsay Lohan’s caller ID to retrieve the latter’s voice mail. She was also accused of hacking into about 50 other accounts. Or at the very least, someone who had used Paris Hilton’s name was being on a run of mischief.

It seems some mobile networks these days do not bother to ask customers to key in passwords or PINs to retrieve voicemail. All that a user needs to do is to call the network from his/her mobile phone, and the network will connect to the appropriate voice mailbox based on the number that registers on caller ID.

However, inexpensive prepaid services like SpoofCard.com lets users call a toll-free number, key in whatever number they want to appear on caller ID, and dial the desired destination number. This means users can also spoof those numbers, and retrieve the voicemail as if they were the owner of that number.

SpoofCard.com uses Asterisk to run its telephone network. The fake caller ID service provider says there are legitimate uses for its services. For instance, this could be very useful for employees who need to dial into their company telephone networks, but could only gain access if they call in from a number that is recognized being from within the corporate network.

Still, this just goes to show that Asterisk’s outbound caller ID can be misused.

[via TMC]

Back to the Basics – VoiceMail

The voicemail protocol is quite a bit of fun and super easy to use. Let’s just jump right into it.

From the Voip-Info respository, here is how the command will go in your dialplan.


You will insert the VoiceMail command after the “exten =>” (as per usual)  then unleash all the excitement at the end. Now, there are so many things that you can do with this command that I won’t really get into a lot of it. There is also so much more once you get into the actual voicemail.conf, which we will touch on later.

Where it says “flags” you can place a couple of things. If you put “s” in there, it will skip the usual “Please leave your message after the tone” recording. If you use “u” it will say the user is unavailable. Then “b” is short for busy. You could come up with a script to change this realtime I’m sure.

After your flags, you will put the number of the VM box that you are looking to send a message to. If you leave this portion blank, then you will get a prompt asking for the box number. Simple stuff.

As for the context section, I’ve NEVER had to use that, so we won’t really bother with it.

What about Voicemail.conf you ask? Well that’s another whole article right there. We will touch base really quickly though. Let’s say that you are just trying to set up a basic voicemail box within your Asterierk PBX that doesn’t have outgoing call capabilities. Let’s save as much money as we can :)

Just for ease of use (and because I don’t need to create large networks) I keep all of my voicemail box information under the default context. You can find this under [default] just like you would in the dialplan. Here is a short example of what your box information would look like.

100 => 321,Users Name,email@address.com,pager@address.com,saycid=yes|review=yes|operator=yes

Let’s tear it apart now.

100 – First it the box number, 100. Nothing more to say here.

321 – This is the password for the email box. It can be as long or short as you feel like.

Users Name – This is for CID. What you put here will show up as the callers name.

email@address.com – This one is easy… the email address of the user.

pager@address.com -  This one will go to their cell/pager.

saycid=yes – When the user calls up and checks his VMB, then this will enable the box to read off the callers ID so if they did not leave a number on the message, they will be able to identify the caller.

review=yes -  If the user wants to review the messages again, they can.

operator=yes – This enables the option menu after the messages are reviewed.

Well guys, that’s all we are going to talk about on voicemail today, but check back soon because we will be doing another article on voicemail.conf which is going to be a very informative one. Check back!

Free SkypeOut Calls via VPN? Maybe Not Such a Good Idea

One great thing about VoIP is that it’s free. Well, usually at least. If you’re calling from computer-to-computer, or from VoIP phone to another VoIP phone (or even computer to VoIP phone), you can usually call at zero cost. Of course, if you have your own VoIP to Public Switched Telephone Network (PSTN) link, you can do this for free or at least for the cost of local calls. But once you start calling regular telephones through established VoIP providers, then you would expect to be charged.

Some providers allow for free local calls through their system. One of the larger ones, Skype, will let you call any US or Canada number free via SkypeOut as long as you’re logged in within US and Canadian borders.

However, there are hacks that allow even those from outside US and Canadian territory to call these countries free. This involves pretending you are in those countries even if you’re somewhere else–and this can be done by connecting to the Skype server via virtual private networking (VPN) or a proxy server that’s located in the US or Canada.

However, this would be going against the Skype terms of service, which says you should not use anonymizing software or pretend to be in other regions.

If you’re using any service, proxy or other devices preventing us from locating you (for example by allocating you an anonymous IP address), Skype reserves the right to charge your calls to US and Canadian mobile and landline telephone numbers at the normal rates, regardless of your real location.

So if you are using a Skype with a SkypeOut account when doing this, better take care, or Skype might charge you for that call.