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

[Geoffrey Mina]

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 at 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 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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