Monthly Archives: May 2008

FreeSWITCH 1.0.0

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I didn’t see any mention of this on the VoIP blogs I read, but it seems to be worth noting.

FreeSWITCH version 1.0.0 was released on May 26 after about three years of development. I would call it a cross between Asterisk and OpenSER. It’s robust like OpenSER but has more base features similar to what Asterisk provides. Makes sense, as the developer is a former Asterisk developer.

Interesting blog post: How does FreeSWITCH compare to Asterisk?

It’s designed as a soft-switch, not a PBX (see FAQ) but you can add the PBX features on to it that you want. I think this will be a real competitor to Asterisk, sipX and whoever else in the free/open-source world.

Junction Networks offers some cool services including hosted conference bridges, which they’re powering with FreeSWITCH.

7906, meet Asterisk

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The Cisco 7906 with the current SIP firmware load (8.3.4 SR1) connected to an Asterisk system performs very much like it does connected to CallManager via SCCP. Cisco has been pushing for feature parity between SIP and SCCP protocols on their IP phones (see this white paper) since they began to really embrace SIP a couple years ago. Because SCCP implements most of its functionality on the PBX end, and SIP puts most of the functionality at the endpoint, Cisco’s SIP firmware has to implement some of the CallManager functionality and look-and-feel on the endpoint device.

7906 with SIP firmware

Biggest drawback when using SIP is the need to setup a client-side dialplan in order to effect PSTN-like (or CallManager-like) dialing: the PBX knows when you’re done dialing a number and immediately begins to process it. With the SIP firmware you have to either set up a client-side dialplan or press a “Dial” softkey after entering a number (like you do on a cell phone). Cisco’s SIP firmware implements KPML (RFC 4730) to make this smoother, but Asterisk does not.

The interface is nearly identical. There’s a DND key on the SIP phone by default. Hold, call waiting, conference (using the phone’s built-in conference bridge), and attended transfer all work. Unattended transfer does not–I’m unclear as to why not.

I would guess that Cisco’s higher-end phones show more disparity between the SCCP and SIP interface and functionality but on the 7906, Cisco’s current low-end single line phone model, they’ve just about reached their goal of feature parity.