We were generally impressed with Fonalitys Connect all-inclusive hosted IP PBX service, which we recently tested over a two- to three-week period.
Fonality, a six-year veteran of the VoIP wars, sells both cloud- and premise-based SIP solutions. (Mark Miller profiled the service here last month.)
Connect, based on our experience with it, is a very strong competitor in this crowded field, which also includes Junction Networks OnSIP (reviewed here earlier this year) and 8x8s Virtual Office (reviewed here last year).
In fact it may be the strongest of the three given the consistently good audio and connection quality. But it is of course not perfect, and call quality may vary depending on a plethora of factors.
The basic bundleConnect bundles VoIP services, IP phones (or softphones) and unlimited local and long-distance calling. The Connect plan, which includes IP desk phone, unlimited calling, and basic PBX features, starts at $30 per seat per month.
Telephony features with the basic Connect plan include auto attendant, personalized greetings for each extension, local phone number, advanced call forwarding, voicemail-to-email, and Web-based management.
Notably missing from this list is find-me-follow-me functionality. It is possible to forward calls to another number and have the call bounce back to Fonality voicemail if not answered. But at least some other hosted PBX services offer full-tilt find-me-follow-me as a standard feature.
Another quibble is that Fonality does not provide exhaustive pricing and feature information at its Web site. You see some information about each package or plan and then have to click a link to "Get a quote," which leads to an online e-mail form.
For example, we doubt very much that the advertised unlimited long distance includes calls to cell phones in Rwanda, but this is not laid out anywhere that we could see.
The package we tested included five extensions, two equipped with Polycom SoundPoint IP 331 IP phones, three with a Fonality-branded audio-only version of CounterPaths EyeBeam softphone coupled with Altec Lansings AHS201i computer-telephone headsets (which AL doesnt advertise at its site).
The Fonality Connect Team Edition service we testedwhich includes an audio conference roomwould have cost a real customer a $45 one-time setup per seat and $45 per seat per month. There are no contracts or time commitments on the Connect service.
Initial set-upSet-up was fairly simple. The Polycom phones came pre-programmed with an extension, DID, and network login configuration. They logged in to the Fonality system automatically on connection. Initial boot-up took a few minutes.
The e-mailed package of customer information included a download link for the Fonality PBxtra EyeBeam software and license codes for each of the softphone extensions.
Softphone set-up on the network was also automatic. After the software installs, it asks for a license code. Once the code has been entered, the software connects to the server and configures itself with the correct extension and DID.
We successfully talked one non-technical test participant through this process in an IM session that took less than ten minutes.
Configuring user informationEach extension did also require some initial set-up in the Web-based management interface. We handled this as an in-house administrator. Its something Fonality would have done for us, but its well within the capabilities of non-technical office managers.
Set-up involved entering information for each 'employee,' including initial passwords and physical addresses, in an e911 dialog. (The e911 information is required to activate extensions for outgoing calling.)
The administrative Web interface, while slow at times, is fairly intuitive, with five main tabs across the top: AutoAnswer (access to auto attendant set-up and administration), Users/Extensions (add, delete, set up, and view settings for extensions and users), ACD (call center auto call distributionan optional feature we did not test), Reporting (call detail recording, billing, call quality), and Options.
Clicking on a tab drops down a sub-menu with links to set-up dialogs, such as Edit call menu, Scheduler, Voice prompts, Music on hold and Sub-menus under the AutoAnswer tab.
Setting up the auto attendant is probably the most complex chore. We have seen simpler, more intuitive online auto attendant configurations. A wizard that walks a non-technical user through this process would be a welcome addition.
The first menu lets you set up a call sequence to determine what the caller hears. You tell it which greetings or prompts followed by how long a wait, followed by what other greeting or prompt or action, and so on.
Creating voice promptsGreetings and voice prompts are selected from a drop-down list of default or previously recorded messages. The default auto attendant message uses standard promptspress 1 to dial by name, etc. You can record your own, but in a different dialog. (Why not keep it all together?)
Its easy to add steps to a sequence, by choosing from a long drop-down list of default options or creating your own (which are then added to the list). And you can add to the default list of keypress optionswhat the system does when the caller presses a key in response to a prompt.
To record a new custom voice prompt, you tell the system which extension youre at, it phones you and prompts you to start recording. The new recording appears in the list of available prompts at the Web page almost immediately.
We could see no way to record a prompt locally and upload it, so audio quality depends on the quality of the connection over which its recorded. When we tried this feature, the result didnt sound optimal.
Call reportingThe reporting is excellent: detailed, clear, easy to access. The call-quality ratings were particularly interesting, showing latency (among other things) for each call. This would be useful for reviewing end user experience in the absence of reliable reporting from users (or as a check against user reports).
One tester did experience an apparent outage, though of uncertain cause and durationwhich was seemingly not reported. Attempts to place PSTN calls resulted in a rapid busy signal.
Our user experienceIn general, though, call quality in all situations, with both internal and PSTN calls, was superior, including when one tester, in the UK, was using a softphone with her laptops built-in speakers and microphone.
The three principal testers all have very high-speed Internet connections: 10 Mbps cable (though with a sub-500Kbps uplink), 8Mbps DSL (though with a history of general VoIP-unfriendliness) and an unknown but seemingly very fast university network.
There was some slight break-up caused by jitter at the beginning of some calls, and just barely noticeable latency on others. Audio quality was reported by most participants as "like a regular phone call" or "better than a cell phone call."
We tested both the conference bridge service and ad hoc conferencing. With the bridge, internal participants call an extension to enter an audio conference room. It can be set up with log-in prompts, but by default dumps callers straight into the conference. The conferences involved one participant in the eastern U.S., one in eastern Canada and one in the UK.
In both cases, the U.S.-based participant (on the historically VoIP-unfriendly Internet connection) sounded slightly muffled to the other two, but was still audible and understandable. Otherwise, call and connection quality were good over one 20-minute and one shorter session. There was little if any break-up even when participants talked over each other.