University dumps Cisco VoIP for open source Asterisk
<<<... So far, SHSU has been able to operate the Asterisk/Cisco IP phones at one-third the cost of CallManager/Cisco IP phones, Daniel says. When the digital Nortel handsets are migrated to SIP-based Cisco phones, or analogue sets, another large chunk of savings will come just by shutting down the electrical and cooling resources required to keep the old PBX running. "The Meridian takes up an awful lot of power itself. The room it's in has to be cooled to 60 degrees, and it has to have its own generator," Daniel says.
SIP extensions wanted
While Asterisk and the SIP protocol lack some of the more extensive features on the Cisco CallManager, the university community has handled the transition with few glitches.
The only major feature missing in the Asterisk/Cisco phone network is secretarial functions, which allow an administrator to manage and answer phone extensions for multiple end-users. To fix this, Daniel is looking into extensions to the SIP protocol that allow for multiple-line handling, he says. In another potential issue with open-source VoIP, SHSU loses the technical support from Cisco with its Asterisk migration. But Daniel says he has so far been able to keep up with support issues through mailing lists and the online community that develops and supports Asterisk. Dell provides support on the server hardware, and Digium supports the T-1 cards installed in the boxes. "We try to have checks and balances," among the IT staff that supports the Asterisk system, Daniel says. "We try to keep the [the Linux and Asterisk server images] as pristine as possible."