AsteriskWin32 Version 0.60 Released

Apparently Asterisk is not Linux-only. Version 0.60 of the Windows version of Asterisk has been released recently. This comes from Asterisk build 1.2.14. According to asteriskwin32.com, the following are the features of this new release:

  • Loadable Modules Support : no longer a standalone application (unload supported)
  • Native sound support for MOH: mpg123 no longer requested (but still supported)
  • AEL support (Asterisk Extension Language v1)
  • DUNDI support
  • CAPI: upgraded to v0.7.0 + added new feature Remote CAPI support : ISDN Router w/Remote CAPI support
  • TAPI: upgraded to v0.2.0
  • CELLIAX: Cellular Network connection via audio drivers (soundcard & bluetooth dongle)
  • Voicemail: send only 1 voicemail while installed as service application
  • SIP Channel bug with IP PHONE : audio confusing
  • TAPI: problem with dialogic hardware
  • GPL Compliant: usage of GNU Readline & Interoperability Key (for loadable modules)
  • New Management Application: PBX MANAGER F.E (Multi-lingual)

AsteriskWin32 will work over PSTN through voice-enabled modems installed on your computer, ISDN via ISDN controllers, or through a mobile network via GSM adaptor.

Making Google Talk Work With Asterisk

Products based on proprietary technologies sometimes suck! That’s because you don’t have any choice but to subscribe to the service as the provider defines it. Most of the time, you cannot interface these with your own services or produts. In telephony, for instance, most proprietary systems cannot be easily interfaced with others. Skype, for example, cannot easily be interfaced with Asterisk without complicated set-ups (i.e., actually running Skype on one machine, or going through analog connections).

Some services that are based on open-source technologies (whether in full or loosely) will be easier to use, on the other hand. For example, Google Talk used to be a closed network, but then it opened itself to other Jabber networks (Google Talk is Jabber-based) so you can chat with other Jabber clients.

Since version 1.4, Asterisk has had support for Jingle, which is a protocol needed to “talk” with Google Talk. VoIP Info has a guide to setting up Asterisk for Google Talk here. This requires ikesmel, which can be downloaded here.

jabber.conf

This is where you set your gmail/gtalk account info and will register you with the google server.

[general]
debug=yes
autoprune=no
autoregister=no

[gtalk_account]
type=client
serverhost=talk.google.com
username=username@gmail.com/Talk
secret=*****
port=5222
usetls=yes
usesasl=yes
buddy=buddyusername@gmail.com
statusmessage=”This is an Asterisk server”
timeout=100

gtalk.conf

This is where the settings for the actual calls are made:

[general]
context=google-in
allowguest=yes

;
[guest]
disallow=all
allow=ulaw
context=google-in

[buddy]
username=buddyusername@gmail.com

disallow=all
allow=ulaw
context=google-in
connection=gtalk_account

extensions.conf


[google-in]
exten => s,1,NoOp( Call from Gtalk )
exten => s,n,Set(CALLERID(name)=”From Google Talk”)
exten => s,n,Dial(SIP/my_sip_phones)

[google-out]
exten => 200,1,Dial(gtalk/gtalk_account/buddyusername@gmail.com)

It is suggested that you create a new Gmail/Google account that Asterisk will use.

Trixbox 2.0 Released

A few weeks back, Fonality released version 2.0 of its popular Trixbox 2.0 telephony system. One main highlight of this release is a new GUI manager that adds ease to the setting up the system.

trixbox 2.0 comes with a new point-and-click package manager which lets installers, via simple clicks of their mouse, decide which applications they want to install with trixbox. The advantage to the package manager is two-fold: first, it let’s you choose how lean or rich of a deployment you need. Secondly, it informs the installer, over time, of any new updates to any packages within their trixbox installation as vendors release them.

The installation includes the usual Apache, modified/enhanced Asterisk, FreePBX, Flash Operator, MySQL, phpMyAdmin and SugarCRM packaged with Trixbox. An enhancement in this new version includes detailed call reports, an endpoint manager, VoIP service provier wizards, better integration with SugarCRM, and drivers for Sangoma and Rhino voice cards. Aside from English, version 2.0 also has support for multiple languages such as German, Portugese and Spanish. Other language packs will be released in the future.

Fonality also said that this coming January 30th, Keith Garrison, Trixbox senior product manager, will be conducting the first trixbox.org Webinar, which is meant to deliver information to IT managers and integrators who intend to install Trixbox 2.0 on their systems. The session will give an overview of the new version, and shall discuss what’s included in the package, the system requirements, hardware required, and skills needed for successful deployment, installation and maintenance.

Using The MV-370 GSM Gateway With Asterisk

It’s great to get cheap or even free calls through VoIP. But it’s even better if you can route calls to and from your mobile phone. This way you can go around those costly long-distance and international rates when calling from your mobile.

The MV-370 gateway is manufactured by Portech and will cost you only $150. Aside from being affordable, what’s great with the MV-370 is that it bridges GSM directly with SIP, and it’s not a GSM to FXS (meaning an adaptor that bridges mobile calls to PSTN). This means it connects directly to your VoIP gateway via the local area network.

The MV-370 uses its own GSM subscriber identitiy module (SIM) card, and therefore has its own telephone number. The typical setup would be for the MV-370 to be connected to your local area network, to which your Asterisk gateway is also connected. You just call the MV-370′s phone number, and it gives you an access dial tone. Just dial the desired destination phone number, and Asterisk takes care of connecting to the claled party via your VoIP provider.

The benefits? In some instances, calling a number directly from your mobile phone might be cheaper than having to call home to your GSM-to-Asterisk gateway. However, using a GSM gateway would be ideal in these instances: First, if your mobile network has “family” calling plans, which lets you call certain numbers for lower tarrifs, or even for free. Second, if you’re calling overseas, then calling through VoIP can definitely be cheaper. Third, using a GSM gateway adds in mobility. Say you run an office, and you use Asterisk as your VoIP gateway. You can then have calls made to your office trunk line routed to the mobile phones of your employees–very useful when on the field or when telecommuting!

The MV-370 comes with substandard documentation, though. VoIP info has a walkthrough for setting up the gateway with Asterisk.

VoIP: Why Businesses Should Switch

For a business, switching to new technologies or adopting new infrastructure is not always easy. Think of it as similar to transferring your office across state lines (or even to another country). There are costs involved. But then again, there may be benefits that can transcend the monetary value of doing the switch. In the case of VoIP, we earlier discussed whether businesses can actually save on costs with a switch to VoIP, and which particular business types are ideal for this scenario.

Hardware costs. Of course switching from a traditional phone system to VoIP would entail hardware costs. You would have to purchase your own gateway and VoIP handsets. And even if you have an existing local area network, you might have to upgrade your system to support the extra IP traffic brought about by voice calls.

However, consider that VoIP would also let your company save on maintenance and other set-up costs, particularly because your telephone units are no longer hardwired to the PBX by circuit. This means you can move around the office and still have the same telephone number. So when an employee moves to another office, there will be no need to physically move or switch cables. Or if you’re setting up new rooms or offices, you can even use wireless networking so you won’t have to rewire.

More importantly, using VoIP will let you set up remote offices more easily. You can have employees set up their workstations at home or even in other countries, and still have the same telephone number (this is very useful for work-at-home setups involving customer service).

Software. Switching to VoIP would sometimes mean having to install and maintain software, especially if you’re using a software-based gateway. This is not as costly as hardware, though.

VoIP packages usually come with unified messaging. This includes instant messaging, fax and even data transfer. In contrast, using a traditional telephone system would mean you have to have separate lines for fax, voice and data. True, sometimes you can combine these with digital subscriber lines (DSL), but you would still need separate physical terminals: the fax machine, the telephone handset and a computer. But with unified messaging, a single line (the IP network) can carry these. What’s more important is that you can use the same system to receive and transmit facsimile and even data.

Unified messaging will also make call management easier since everything is on one network. For instance, voice messages no longer have to be recorded on separate machines. Instead, these can be recorded at the VoIP gateway, and then forwarded to the users via email. Same with fax messages.

(continued …)

VoIP: Cost Savings or Not?

Much has been said about how switching from traditional telephone exchange systems to VoIP can save companies lots of money. True, business can save up on long distance charges and even mobile phone subscriptions. Businesses can also save up on setting up separate networks for telephony. The difference in infrastructure requirements of VoIP systems can make it easy for offices or staff to transfer–or even travel across the globe–without even having to change numbers.

However, there is no simple answer to whether switching to VoIP would entail cost savings. The answer is maybe.

For one, the size of a business would dictate how easy it is to determine whether VoIP would help the bottomline. Smaller companies would usually need more basic services, and the cost savings would be in terms of long distance calls. However, the cost advantages may not necessarily be great, since long distance costs these days are increasingly dwindling. Also, consider that purchasing your company’s own VoIP hardware would still entail a great upfront cost.

More complex and larger organizations, meanwhile, might be better able to harness the rich features that VoIP offers, and hence the cost savings are more identifiable. While upfront costs may be greater for bigger businesses, this would most likely be negligible in comparison to total costs to the company (and perhaps even telecommunication costs).

Bigger organizations might be able to save more on infrastructure, compared to maintaining a traditional telephone system. Since computers are already connected using an IP network, then installing VoIP equipment on workstations would require minimal added cost.

Bigger organizations may also be able to utilize other IP services as well, such as videoconferencing and presence management. Videoconferencing can help save on travel costs–executives and managers of companies with branches and offices in faraway states or countries can hold meetings via videoconferencing, instead of having to travel all the way across countries and continents. Presence management could help reduce staffing costs–for instance, instead of needing secretaries, then perhaps virtual assistants that some VoIP systems offer can be of help.

The greater cost savings would come in terms of the productivity boost that come with the rich IP-based applications that VoIP usually brings in. Compared to traditional telephony systems, VoIP can integrate data and voice applications, such as instant messaging and the aforementioned presence management. VoIP systems can also support sending of voicemail and faxes to the company’s email system, thus saving on equipment costs, and the time needed to route physical fax printouts.

However, having to train managers, personnel and IT staff on the maintenance and use of VoIP systems might also entail additional cost to a business. For a small company, this can be negligible, but for a larger organization, this can be considered a large scale–and hence more costly–activity. Businesses should weigh the cost of the lost productivity attributed to training each manager and staff, against the benefits of the users being able to adequately use the system. For one, added services might be overkill, as some companies don’t really need these (for instance, a small business with five employees who share one common office may not need presence indicators).

Again, there is no sure answer too whether a switch to VoIP can entail cost savings. Businesses should evaluate this adequately, taking into consideration not only the money aspects of it (buying equipment, setting up infrastructure), but also in terms of productivity and manpower.

Source

Buying a VoIP Gateway? Here are 10 Things to Consider (Part 2)

The first part of our 10-point guide in buying a VoIP gateway mentioned cost considerations and capacity. Now we will talk about the next five points, and these are mostly about a few technical aspects you should consider.

6. IP connectivity types. You should choose your VoIP gateway based on the standards your company plans to use. There are two major standards: session initiation protocol (SIP) and the H.323 specification. Your choice of client software or hardware would then also depend on support for either of these protocols that you choose.

7. Compression. This is another important technical aspect to consider. A VoIP gateway essentially converts analog voice signals (audio) into digital signals (data) to transfer through the IP network or the Internet. Digitization usually comes with compression, on which the quality of a call is dependent. If the compression is too high, then the voice quality might be poor. If compression is too low, then there might be delays or latency in transmission (will be observed as a “lag”), hence a tradeoff. Digitization usually results in 64 Kbps data rate, and VoIP gateways can further compress this down to anywhere between 5.3 Kbps to 24 Kbps. Select a gateway that lets you be flexible in controlling compression, so you can determine whether to prioritize bandwidth or quality.

8. Upgrade options. Software-based gateways may be easier to upgrade, since it will be a matter of installing software or updating portions of the software. This is especially useful when new standards are set, or the existing standards are updated to support new protocols or equipment. Hardware-wise, upgrade paths include addition of new ports to support added equipment.

9. Compatibility. Since your VoIP gateway will be communicating with both internal and external devices, you should check whether your planned purchase will work with your existing technologies, including the PBX, automatic call director, and interactive voice response systems. Most importantly, the gateway should work with your PBX.

10. Support. Vendor support is important because you would expect extensive use of your VoIP gateway, and this is mostly for mission/business-critical matters. Downtime could be costly. Make sure your equipment comes with warranty, and be clear with your vendor about the coverage of warranty. Also, you should check whether your vendor offers support via phone, email or other means.

Buying a VoIP Gateway? Here are 10 Things to Consider

We previously wrote a brief introduction about VoIP gateways recently, and from there we learned some basic concepts about Gateways, which handle the task of transferring voice or data traffic from a circuit-switched telephone network to an IP-based network. We also talked in passing about some of the factors you have to consider when choosing a Gateway for your business or company.

Now here’s a ten-point guide that can help you make that decision on what exactly you need.

1. Cost. The question of cost should be the first thing you should address. For one, your company might have a budget for such equipment, and you might be in the market for different brands or product sets with comparable features. Part of this would be the cost of setting up, maintenance, and even support from your vendor.

2. Hardware vs. Software. Hardware-based VoIP gateways are perceived to be more reliable and secure. Further, these run on their own processors, and hence do not drain computing resources away from existing computers. Software-based gateways, meanwhile cost less, and are easier to update, upgrade or modify as the need arises. It’s therefore a tradeoff between reliability and flexibility.

3. Chassis size. If you’re planning to install a hardware-based gateway, the chassis size is usually indicative of the gateway’s packet processing capacity. Slow processing means poor voice quality and low capacity. This will not apply if you’re opting for a software-based gateway. In that case, it’s the processing capabilities of the computer you will be using that will be important.

4. Capacity. This is in terms of number of simultaneous VoIP calls the gateway can handle without being overworked. The gateway should be able to cope with the regular traffic of your network, and should also accommodate traffic spikes and expansion (if your organization is growing, for instance).

5. Foreign exchange office (FXO) ports. The primary function of VoIP gateways is to convert signals from the public switched telephone network into IP packets. For analog PSTN lines, FXO ports are needed. Small businesses and remote offices would usually need at least four FXO ports.

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VoIP Predictions for 2007

VoIP News cites top ten predictions in the VoIP industry for 2007. These predictions are taken from all around the blogosphere and tech industry news sites. Mostly, the predictions relate to how VoIP will further strengthen its hold on consumers (business and personal alike) as a service of choice, on top of, or as an alternative to traditional telephony. Here are some highlights.

1. Jon Arnold thinks cable companies will continue to dominate, but some independent providers–such as Vonage–will survive. This is in terms of consumer-oriented VoIP services. For one, cable companies have the advantage, already having their foot in the door. They who already service consumers at their residences as ISPs have an advantage in offering value-added services at little added cost. Arnold also thinks vendors will continue to consolidate this year.

2. VoIP watch thinks the FCC will start seeing VoIP as a telephony service rather than a value added service, according to VoIP watch. With this, restrictions and rules applying to telephone companies are likely to be enforced. One of these is E911, which I think is just reasonable. Emergency services have always been a problem seen with VoIP systems, especially since you can be anywhere in the world and use the same number and service–hence being difficult to track down. VoIP watch also predicts that the Gizmo project will be acquired.

3. Ted Wallingford of Signal-to-Noise predicts that providers that offer pure VoIP services will move on to provide more consolidated services, like TalkPlus and GrandCentral, which offer integration between VoIP and other telephony systems (TalkPlus enables regular mobile phone users to call Skype users via their own Skype names, for instance, and not via SkypeIn). Wallingford also predicts increased uptake of VoIP as part of customer-service systems for enterprises and publishers, particularly with ubiquitous “call me” links or buttons on websites, for instance.

4. Ken Camp of Digital Common Sense predicts that companies that have relevance engines will be on the rise–these are systems that work like presence indicators, but also tell users other relevant information like who their contacts are in conference/meeting with. Camp also predicts radical changes in how phone handsets are to be designed–perhaps this would be due to the increasing proliferation of data that phones are able to transmit these days.

5. Mark Evans predicts that security in VoIP and its applications will be taken more seriously, especially with the increased uptake of VoIP systems by enterprises.

6. MobileCrunch thinks that the sheer number of new users signing up for Skype every day, particularly from China, can be defined as a “disruption.” Of course, with Skype’s peer-to-peer model, this might not necessarily pose a problem in terms of capacity.

Whatever the predictions are, the common denominator seems to be the increased rate of adoption of VoIP, whether for consumers or enterprises. This is certainly good news to those banking on various VoIP-related business models.

Voice 2.0 Anyone? (Part 2)

Voice 2.0 entails the concept of persons being able to transmit voice, data and video from anywhere, anytime, using their tools of choice. It’s more about convergence and accessibility to services using a common set of standards. We previously discussed the basics, and here are more concepts that fall under “Voice 2.0.”

Presence management. VoIP clients and IM networks usually have presence indicators that tell users which contacts are online or unavailable. Voice 2.0 endeavors to have this feature across all devices that a person has. One enhancement would be for the presence management system to determine whether a user is online or available on his desktop, VoIP phone, laptop, PDA, mobile phone, or office landline, and route calls and messages accordingly.

Another enhancement of the envisioned presence management systems would be that these would automatically schedule conferences and meetings based on the online patterns of members of groups and organizations. For instance, virtual meetings are best held when everyone is online, or at least a good majority can be in attendance. The presence management system would make sure of this.

One other aspect of presence management would be the ability to use different calling numbers from one device–this can even mean anonymity with the use of disposable contact numbers. Advantages of this can include the ability for one to call using his home phone, but appear on caller ID as having dialled from his office number. Or, one can use his mobile phone to call, but the other end would see a caller ID coming from a land line number.

Facsimile. Traditional faxes are incompatible with VoIP because these require circuit-switched connections. There are existing services that act as fax receiving services, and then send the digitized document via email. But these still require circuit-switched connections. New technologies are being developed such that traditional fax machines can call IP devices to transmit data, and vice versa.

Other issues. The concept Voice 2.0 also envisions to address issues like open source vs. proprietary technologies. Good examples of this are the use of Skype as the most popular consumer-grade VoIP solution (proprietary), and Asterisk, a software-based corporate PBX solution (open source).

The main point behind “Voice 2.0″ is the change of the structure in telecommunications, from centralized to user-oriented. The control over the flow of data, voice and video is increasingly moving towards the user rather than centralized operators. This is largely because of the increasing popularity of IP-based solutions, which does away with the need for expensive exchanges and circuits, which are replaced by the Internet pipes, which have already been laid out. The concern now would be to determine set standards that can be used to run the various applications.