[OpenSIPS-Users] Freeswitch vs Asterisk

[Michael Collins]

Paul,

It sounds like you've done a fair amount of dev work with both projects. I
respect your viewpoint and your experience. Perhaps it's best if we agree to
disagree. In any case, I refer back to the mantra of the FreeSWITCH
developers: "Use what works for your situation." If you've got something
that is working for you then by all means keep using it!

In fact, if you've been doing telephony development with open source tools
you may wish to come to ClueCon (www.cluecon.com) to rub elbows with fellow
OSS telephony devs. It's a great opportunity to talk tech with important
people in the OSS telephony arena. Why I even had the pleasure of having
dinner with the man who wrote OpenSIPS! (Bogdan, hope we can talk tech some
more at ClueCon 2011. :)

-MC

On Thu, Dec 9, 2010 at 7:26 AM, paul.gore.j at gmail.com <paul.gore.j at gmail.com
> wrote:

> I just want to reply to mr Collins with FS: your post looks very much like
> advertisement, and I have seen that "fs is so much better than *" all over
> internet from people connected to fs. That is unethical to say the least.
> In fact we have exprerienced fs crashes with core dump at least  once in 6
> months and we process just under 40K calls/month.
> As to "nat tools" which you mentioned they just do not work. In fact
> usually * box works much better for natted users.
> As to xml curl interface - we do use it, and it's a pathetic way to feed a
> dialplan to a switch, since it's inefficient resource wise, but there was no
> other way available for real time solution where's * supports real time db
> out of the box.
> Trust me we do have development experience with both * socket interface and
> fs one, and in my opinion * solution is far better and has far less bugs.
>
>
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