TMC Labs: pbxnsip's CS 410 IP PBX

TMC Labs got an exclusive peek at pbxnsip's CS 410 IP PBX "all-in-one" appliance, which features a "mini" Session Border Controller (SBC), four built-in analog (FXO) PSTN ports, voicemail, and auto attendant, as well as support for up to ten SIP-based IP stations (hard phones, softphones) and supports up to ten simultaneous calls. The CS 410 actually includes all available features from the other pbxnsip PBX editions that run on Windows or Linux boxes. This includes standard features like voicemail, auto attendant or conferencing, but also advanced features like call barge in and call forking to cell phones. Like other pbxnsip versions, the CS 410 also supports advanced routing functions, such as paging groups, hunt groups, and agent groups.

The pbxnsip (news - alert) appliance reminds us of two other popular Linuxbased IP PBX appliances in the VoIP industry, namely Digium's Asterisk Appliance and Fonality's trixbox appliance. Just comparing these two to the pbxnsip applicance on price alone, the trixbox appliance (4 FXO ports) is $1499, and the Asterisk Appliance (4 FXS, 2 FXO) is $2195, while the CS 410 comes in at just $999. Of course, each of the appliances has features the other two don't, so it really depends on what features you need. Even with the lowest cost, the CS 410 doesn't skimp on features and unlike the other two, it features integration with Microsoft Exchange Server 2007 UM.

Targeting small to mid-sized businesses (SMBs), the CS 410 is a solidstate appliance device with no moving parts, no fans, which results in very minimal heat to ensure long-term reliability of this phone system. In fact, we noticed very minimal heat when we touched the unit's plastic casing. TMC Labs took the CS 410 for a spin and were very impressed with its easy plug-and-play installation and easy-touse web admin.

In lieu of a hard drive, the CS 410 appliance sports 256MB of Flash and 128MB of RAM running Debian Linux. This only gives you about one hour of voice mail storage, but you can easily get around this storage limit by leveraging Microsoft's Unified Messaging capabilities in Exchange 2007. The CS 410 can send all the voice mail to the Exchange 2007 Server. The CS 410 is unique in the industry in that it is the first IP PBX appliance to directly integrate with Microsoft Exchange Server 2007 leveraging SIP. The latest version of Microsoft's Exchange Server features built-in unified messaging capabilities, including voice mail storage/playback, text-tospeech reading of email/calendar, and more. Used in conjunction with each other, you can achieve a rich user experience. For example, we were able to dial into the CS 410 and then enter our extension, followed by * and then our PIN to log on to our personal mailbox. From that point, we could play back voicemail, but also email using text-to-speech (TTS). Similarly, we also had remote access to our calendar, which also leverages TTS.

We setup some Exchange 2007 extensions and configured the CS 410 (See Figure below) to transfer these extensions to the Exchange Server. We simply had to configure a new trunk on the CS 410, set the trunk to "SIP Gateway," and point the IP address to connect to the Exchange Server. On the CS 410, we also had to define a new dial plan for routing incoming calls to the Exchange Server's built-in SIP gateway, which would then accept the call and play the extension's outgoing message. On the Exchange Server side, we also had to create a new dial plan and configure the new Unified Messaging IP gateway (the CS 410's IP address or FQDN). Finally, we had to activate Unified Messaging for each Exchange mailbox and assign an extension number and a PIN. Voila! We now had fully integrated unified messaging for each of the CS 410 extensions we created.

Some other nice features of the CS 410 include paging and music on hold audio connectors. In fact, the system can also send RTP multicast traffic and use multicast- enabled devices for office audio paging. (This is actually in the new snom 370 IP phone.) Another security-related feature is full support for TLS and SRTP to secure the voice. pbxnsip mentioned to TMC Labs that it is thinking about adding a built-in firewall/router to support TOS tagging to ensure QoS for time-sensitive voice packets. Since the unit runs on Linux, pbxnsip says it shouldn't be difficult to add that functionality. The CS 410 also supports a built-in conference bridge, which is great for SMBs looking to save money by avoiding paid hosted conferencing services.

Another feature worth noting is the agent waiting queues. You can record up to ten announcements and have the music on hold mix in these announcements. The queues also feature agent recovery time, call pickup from queue, call escalation, day/night mode, holidays, and Web-based queue status display.

We really liked the Exchange 2007 voice mail integration. Even without Exchange Server 2007 integration, you can still have the system send you voice mail notification via email. We liked how this appliance integrates the analog telephone lines within the appliance without the need for a separate PC with telephony cards or a separate PSTN gateway. The dial plans are especially powerful and easy to configure. The dial plans allow SMBs to take advantage of VoIP's low per-minute costs, while also giving SMBs the option to utilize their existing phone lines. For instance, local calls can be terminated through the FXO lines while international calls can be sent to an ITSP using VoIP. It's important to note that it has built-in session border controller functionality for connecting with remote offices. The CS 410 which supports ten IP endpoints can also be easily upgraded to a CS 425 (simply by changing the license key), which supports 25 registrations and 15 concurrent calls. Overall, TMC Labs was quite impressed with pbxnsip's CS 410, which was easy to configure, a breeze to administer, and sports more features than IP PBXs that cost four times as much.

pbxnsip’s CS 410 IP PBX

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