-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Robert La Ferla Subject: [Freeswitch-users] Freeswitch?
I am an Asterisk user but I hate the monolithic design. I would like to know more about the Freeswitch design and roadmap. I know that it is basically a lower level yet modular system. I have read the FAQ and wiki.
I basically want to know what will it take to put together a Freeswitch system that:
1. Routes calls from my SIP extensions to either a SIP service provider (like Broadvoice) AND an PSTN provider (like Veriizon)
Our sip support is coming along quite nicely. It's well worth trying it out and seeing if it fits your needs. Our TDM card support is still in its infancy, but hopefully more on this soon.
2. Has a voicemail system
It does not have a hard application for voicemail, and we will likely not write a hard C module for this. What it does have is extensive ivr capabilities that are designed to make complicated IVR systems such as voicemail. I know of at least one of these that has been written. It is not in tree at this time, although I think something along those lines is likely to be there at some point soon. We also need to do a bit of work to get a proper MWI abstraction in tree. If anyone is interested in sponsoring this work, feel free to chime in.
3. Has dial and Caller ID rules for dialing (dial plan), answering, etc...
Our functionality in this regard is very advanced. You can do full regex matching on pretty much any call element you could think of.
Are these components in place yet? If not, will they be soon? Are there other open source packages that interoperate w/Freeswitch to accomplish the above?
There has already been some work on sipx to integrate FreeSWITCH(tm) into their configuration system for conferencing. I would be happy to see more cross project cooperation like this in the future. Time will tell. Overall, I would say we have still yet to have a feature freeze or release, so be aware that there are still things that may change both in api and configuration (we had a major change in the sip module usage today), but other than that caveat, by all means please try it out and see if it meets your needs. Mike