A Suprisingly Simple Switch from Cisco Call Manager to Asterisk
The south Texas school is boldly moving thousands of users off a Cisco VoIP platform to an open source VoIP network based on Asterisk.
SHSU is in the process of moving its 6,000 students, faculty and staff off of Cisco CallManager IP PBXs and a legacy Nortel Meridian PBX over to Linux servers running Asterisk, which includes call processing, voicemail and PSTN gateway functionality. The driver for this project was cost, says Aaron Daniel, senior voice analyst at Sam Houston State University.
"We thought that it will be more cost effective in the long run to go with an open source solution, because of the massive amounts of licensing fees required to keep the Cisco CallManager network up and running," says Aaron Daniel, who this week gave a presentation on his migration project at the VON show in Boston. In the Cisco model, each phone attached to the CallManager required a separate annual licensing fee to operate, Daniel says. In SHSU's Asterisk/Cisco model, where it will keep its existing Cisco phones but attach them to Asterisk servers on the back end, the phone licensing costs are eliminated.
SHSU so far has moved 1,600 IP phones from Cisco CallManagers to Asterisk, which runs the IETF-standard version of SIP. The Asterisk functions are spread across six redundant Dell servers: two act as redundant PSTN gateways (and are outfitted with four-port T-1 cards from Digium, which commercially distributes Asterisk); two more servers handle call processing; another set provides voicemail.
More control over the IP PBX software and servers was another reason SHSU made the Asterisk jump, Daniel says. "We felt we were more susceptible to hacks," since only Cisco-approved servers updates and patches could be installed on the Windows Server 2000-based CallManagers, he says. "We have a lot more peace of mind with the open-source system. If a bad exploit is found in SIP, we can fix it ourselves."
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