[Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch

[geoffreymina at gmail.com]

Hello,
I have been reading through the on-line info as well as some reviews of the  
FreeSwitch platform. I am fairly convinced at this point that FreeSwitch is  
at least something I need to carefully look into.

Our company utilizes asterisk to support our SaaS ACD/VPD/IVR platform. We  
currently support many thousands of concurrent agents (inbound and  
outbound). I have spent a lot of time trouble shooting bugs and working  
through 'issues' with asterisk. While I have tamed the beast, I am still  
not thrilled with the performance, nor am I very excited about the  
direction the project appears to be heading. It seems like every time  
a 'fix' is committed to SVN, it breaks something else. It's kind of like  
the wild-wild-west over there... and it certainly doesn't give me the  
warm/fuzzies when thinking about the future of my company.

One of the benefits of our architecture is that our business logic is  
completely abstracted from the asterisk system. We use a combination of  
FastAGI and AMI to control channels on the asterisk server. We have a Java  
based server which interfaces with the higher level call routing engines.  
It looks to me like the Mod_event_socket would probably satisfy my  
requirements for controlling the calls via an external process, although it  
doesn't look as cut/dry as the FastAGI model. I haven't seen anything which  
would let me know the equivalent of the FastAGI 'script' being requested.

The other thing I haven't seen is how to dynamically create conferences on  
the fly and redirect channels into them. We use app_conference on asterisk  
to avoid the ztdummy issue. Once the higher level intelligence engine  
determines two channels need to speak with each other, they are both  
redirected via AMI Redirect into a dynamic Conference created just for that  
particular call.

Also - what is the status of call progress on FreeSwitch? Some things that  
are important to me are answering machine detection as well as detecting  
SIT intercept tones in the early media stream... any love here?

I have a ton more questions, but this seems like a good start.

Thanks!
Geoff
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