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 [email protected] ) 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, [email protected] 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 > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
