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

Anthony Minessale anthony.minessale at gmail.com
Mon Jul 6 08:43:57 PDT 2009


The best way to describe event socket to someone familiar with asterisk is
that its a combination of AGI and AMI which can be used bidirectional.

You can:

connect one inbound socket from a client and control every call at once
using events.
connect one inbound socket then latch on to an existing single call and
control it.
connect one outbound socket to your application per call and control it.

In all cases you have the option for full control which allows you to gain
access to log, event, and FSAPI commands (the equiv of cli commands in
asterisk)
You can have your script listen on a dedicated port or use the ivrd example
which is a daemon written in C that gets the desired script name from
a channel variable and executes it on the remote end of the socket using
STDIN/STDOUT as the socket.

The other big difference besides that the single protocol does all these
things is that we have a BSD licensed client library in our source tree
called ESL.
its in the libs/esl directory.  This can be use to write clients in C or
several other higher level languages using swig.  fs_cli that is built with
FS is written using
ESL.  Perl, Ruby, Python, Lua, PHP are all working and there is the
beginning of a JAVA one which is stubbed out but just needs a little bit of
work to finish it off
and you could have that too.



On Sun, Jul 5, 2009 at 5:29 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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>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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