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

[Michael S Collins]

A few questions for you if I may:
FreeSWITCH doesn't yet have a GUI -are you okay with XML config files?

Do you have TDM circuits for your outbound traffic or are you using a  
SIP provider?

BTW, mod_vmd is used to detect an answering machine beep, but it does  
not detect human vs. machine. For that you'll need mod_amd which isn't  
free but is available at a reasonable price. (email consulting at FreeSWITCH.org 
)

FYI, detecting SIT tones is always a challenge if you telco forces you  
to listen inband. You'll need a little processing power and the  
tone_detect app. I've done it on a PRI and cheap Tormenta 2 clone and  
it actually works pretty well.

-MC

Sent from my iPhone

On Jul 5, 2009, at 3:29 PM, geoffreymina at gmail.com wrote:

> 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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