We are old and grey enough to remember when IBM with ROLM were a major force in the PBX market but after stellar success they faded and were overtaken by the competition. It seems IBM are dipping a toe back in the PBX market by pairing up with Digium and offering a version of the popular Opensource Asterisk PBX system as an add on to their Smart Cube office in a box package.
Customers purchase the Asterisk application from IBM’s Smart Market, and then go to IBM for support. Digium support staff is on call to IBM for tier 2 support, but customers will dealy directly with IBM. The PBX software is sold in two sizes, 20 and 40 concurrent calls.
Smart Cube is sold as a hardware platform with a base set of applications on it and a second set of applications available for customers to buy via Smart Market to address their specific business needs. The Asterisk software can be configured and managed via IBM’s Smart Desk management dashboard, which gives a common look to management of all the Smart Market applications.
Lets see how successful this is and maybe IBM will produce their own flavour of Asterisk in the future.
In a previous post we discussed about Fax support not being dead on IP systems and we see that Asterisk has now added improved fax working to it’s portolio of solutions with the addition of Fax for Asterisk software.
Users can download software from Digium’s online store. It is free for installations requiring only one fax session at a time, while multiple session licence is available for extra cost per simultaneous channel. The software allows faxing to and from the PSTN and IP telephony networks.
If the Asterisk IP PBX in question uses analogue PSTN line cards, the software sends faxes through those. It supports V.17, V.27 and V.29 fax modems, and operates at speeds of up to 14.4 Kbps. If the IP PBX installation uses an IP telephony service, the software will then use the T.38 protocol (if the provider supports it). Users send and receive faxes in the form of TIFF image files.
Fax is one of those technologies that will not go away so IP PBX vendors are being pushed by market forces to produce a solution and it is good to see Digium respond.
In the past we mentioned that Fonality caused a few waves in the Opensource Community by developing their own version of FreePBX to work more closely with their Trixbox product, it seems that FreePBX has now entered the hands of bandwidth.com by hiring the project’s main developer as its Open Source Community Developer and is said to be committing significant resources and effort to expand the scope of the project.
An informal announcement of bandwidth.com’s commitment to FreePBX came through the main developer Philippe Lindheimer’s blog at www.freepbx.org on 14th November in a post called “A Bright Future for FreePBX.” Lindheimer said he had “joined forces” with bandwidth.com as its Open Source Community Director and indicated both he and bandwidth.com would work to expand the scope of FreePBX and to assure it remains “open and strong.”
Lindheimer cited bandwidth.com’s efforts in purchasing the FreePBX trademark and its efforts with FreeSwitch as areas where the company has been helpful to the open source community. Since bandwidth.com sells VoIP and data services not software then there appears to be no conflict of interest.
This is the latest coup for FreePBX; the web based GUI provides preprogrammed functionality and ease of use on top of Asterisk, including features such as follow me, ring groups with calls confirmation, music on hold, conferencing, and paging and intercom functionality for many SIP phones. Digium incorporated FreePBX into its compilation of the AsteriskNow 1.5 turnkey release in October.
See the blog entry here.
Skype have been trying to push hard to penetrate the business market and especially the medium to large enterprise type market, we see that Digium and Skype now plan to work together to Skype enable Asterisk.
The Skype for Asterisk connector will enable a presence on the Skype network while getting all the functionality of a PBX. A general Skype login name could be routed into an Asterisk system for call processing and delivery to the next available person in a queue, a single Skype ID could be used for sales, technical support or other types of customer service queues. So in essence a Skype connection will be treated in the same way as a regular trunk or CO line.
While the two companies have had various discussions over the years on how they could work together, a serious effort came to a head at a June meeting at Digium’s Headquarters in Huntsville USA, since then Digium’s software engineers have worked to produce the solution, making it more robust and suitable for a formal beta, but as yet there is no firm date for release. The software is slated to run with all versions of Asterisk including trixbox.
We will watch this one with interest to see if it elevates Skype to be a serious tool for many businesses.