Category Archives: PBX

Asterisk PBX Tips

Tribox 2.4 Beta Released

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The latest version of Tribox 2, which is the most popular Asterisk-based telephony platform in use now, was released by Fonality last week. Tribox 2.4 is includes the latest releases of CentOS 5, Asterisk 1.4 and FreePBX 2.3. Tribox 2.4 is still just a beta version with the general release set to be announced shortly after the latest release of Aterisk 1.4 comes out and they are able to incorporate needed changes. The newest release boasts an overall better performance especially in terms of choice/flexibility, compatibility, and usability. Applications included in Tribox 2.4 Beta include Apache, Asterisk (version 1.4), FreePBX, Flash Operator Panel, MySQL, phpMyAdmin and SugarCRM.

According to Andrew Gills, the founder of Tribox,they are just beginning to develop and deliver according to their user’s demand. As he said, “Listening to the trixbox community is our first priority. They wanted to see Asterisk 1.4 incorporated into trixbox, so we are delivering just that. This beta release is just the first step and by having trixbox 2.4 based on CentOS 5, trixbox is compatible with the newest hardware on the market.” So while it seems that the latest version has already increased Tribox’s compatibility we can be pretty sure that the compatibility issue is just a step towards a larger goal.

For more information on Tribox please visit the Tribox website.

Simicomm Announces Easy Speak PBX

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Simicomm has just released its Easy Speak PBX, a pure software-based PBX solution that can be easily installed via a bootable cd solution. Simicomm promises an installation that can be done in 15 minutes.

Based on the open source Asterisk PBX platform, the Easy Speak PBX is aimed towards enterprise customers who wants to slowly migrate to a VOIP telecommunications infrastructure but does not have the technical expertise or the big budget to deploy other costly IP PBX solutions.

Standing on the shoulder of the Asterisk platform, Easy Speak offers a full-featured IP PBX Solution with the following features:

PBX Features:
Web Access to Voicemail
Music on Hold
Blind Transfer
Call Detail Records
Call Forward on Busy
Call Forward
Call Parking
Call Queuing
Call Routing (DID & ANI)
Call Transfer
Call Waiting
Caller ID
Caller ID Blocking
Caller ID on Call Waiting

Call Features:
Roaming Extensions
Assisted Transfer
Three-way Calling
Voicemail:

  • Visual Indicator for Message Waiting (on phone)
  • Stutter dial tone for Message Waiting
  • Voicemail to email
  • Voicemail Groups
  • Web Voicemail Interface

From Simicomm’s official press release:

“EasySpeak PBX (300 MB file) is downloaded from www.simicomm.com, and burned to a CD. The CD is loaded onto a (dedicated) server, which will run through the installation steps automatically. Just plug the phones and Internet into the Ethernet ports (1 and 0 respectively) and you’re ready to go. To use analog phones, a line card (i.e. Digium analog telephony devices) is required. Telephones will automatically register and be assigned extensions in the order they are connected. Changing of extensions and management of other features can be made by logging into the web-based toolbox.”

Apart for a line card for supporting analog phones, Easy Speak PBX deployment is pure software solution, making it a very compelling solution with minimum hardware peripheral requirements.

As mentioned in Simcomm’s official press release, the beta version of Easy Speak can be downloaded and be used for a trial period of 14 days. This gives small businesses (the target market for this product) a chance to test drive the product and see if it fits with their VOIP Solution needs without really spending any money on it.

Easy Speak is a very attractive IP PBX solution because of its low cost and reliability, being based on the popular and highly-touted Asterisk open source telecommunications platform.

Digium and Vyatta Enters New Partnership

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Here’s some news for all you Asterisk lovers out there. Asterisk’s parent company, Digium, just entered into a partnership with Vyatta, a company that specializes in open source routing, firewall and VPN solutions. It’s a collaboration that could have a big impact in the world of open source networking.

Digium and Vyatta have announced a partnership to collaborate on open-source voice and data networks. The two companies will work together to make it easier for customers to purchase, deploy, and maintain high-quality, integrated voice and data platforms.

The partnership is aimed at helping SMBs and large enterprises. SMB and enterprise customers are increasingly seeking telecom and data communications solutions that are more flexible, efficient, and tailored to their specific needs. Open source-based products are uniquely capable of rapid integration and feature flexibility, making them an ideal choice as “unified communications” move up on the IT and business priority list.

The partnership involves both technology and marketing initiatives, and includes efforts to improve VoIP quality of service and security features in both Digium and Vyatta products. The companies will also focus on making it easier for customers to install and configure a secure, integrated voice and data environment using Digium’s Asterisk and Vyatta’s open-source networking solutions that include routing, firewall, and VPN functionality.

Speaking to CXOtoday.com, Dave Roberts, vice president of strategy and marketing at Vyatta said, “Presently, various VoIP don’t interact well with traditional security devices. For instance, Session Initiation Protocol (SIP) doesn’t go through firewalls and Network Address Translation (NAT) devices. Traditionally, firewalls try to detect what a VoIP session is trying to do. In most cases they are detected, but in the case of SIP it is hard. Essentially, our collaboration would enable PBX to notify the firewall of the exact happenings.” Roberts also added, “Voice over IP is now a mainstream technology, and we are poised to do the same with open-source networking. By combining the efforts of Vyatta and Digium, the two technology leaders in our respective areas, and working together to expand our marketing reach, we can drive rapid adoption of our solutions.”

Digium also commented, “A partnership between Vyatta and Digium is a natural fit that will leverage our very complementary core competencies to enhance a common mission – to provide SMBs and enterprises with open-source alternatives to expensive and proprietary solutions,” said Bill Miller, vice president of marketing and business development.

Choosing a VoIP Gateway for Your Company (Part 2)

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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)

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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).

Job Postings

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If you are a corporate client looking for someone to work on a PBX for you in the cheapest and most efficient manner, you have come to the right place. Take a look at our forum and post a job offer that our freelancing Asterisk-ers can take a look at. It’s easy, free, and confidential.

On the other hand, if you are a freelancing VoIP telephony specialist, you may want to post an excerpt of your resume in our freelancer section of the forum. As said before, it’s easy, free, and totally confidential!

So before you sit down and worry about cold calling those 100 businesses that you had in mind, why not publish your resume or job offer here and let us do all the hard work for you?