I love the fact that there is no GUI. I have never used any GUI for asterisk, so that is certainly not a problem. XML is fine with me. We are a pure VoIP environment. I have many wholesale SIP providers whom I interface with. AMD and SIT detection are very important to me. Because of that, I am exploring a relationship with Sangoma for their SIP based CPD product to satisfy those requirements. There are a couple things which I don't like... namely that it only runs on windows, but I may be able to ignore that for the time being. thanks. On Mon, Jul 6, 2009 at 1:43 AM, Michael S Collins<msc@freeswitch.org> wrote:
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@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@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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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