On Mar 29, 2008, at 7:57 AM, kokoska rokoska wrote:
Hi all!
I'm very new to Freeswitch and thus I'm looking for advices/hints for
painless start :-)
Nothing is painless.
I have a lot of experience with Asterisk and OpenSER, but the
philosophy
od Freeswitch differs...
Thats an understatement. ;)
What is going on:
I like to deploy PBX/Switch for a lot of SIP users wich registers with
it and - also - with larg number of SIP gateways the PBX/Switch should
regester with. Users population/gateways/call routing have to be
dynamic
(database-driven, like I'm accustomed form Asterisk and OpenSER) with
quite standard features (conditional/unconditional forwarding,
voice-mail, call-waiting, resource limits etc.) and especially with
good
over-all performance.
Like i red in docs, dynamic SIP users could by done with mod_xml_curl
directory but I like to ask: Is it "fast enough"?
Direct DB in my opinion is a very bad idea. With xml_curl you can
interface to just about anything and cluster it up and fail over
rather easily with http gets. And no it's NOT slow, that depends on
how fast your web server and db are... trust me it can scream if you
do it correctly.
Couldn't be better direct DB lookups? If yes, how to accomplish that?
BTW: Is there a way how to share "registered" users between
independant
Freeswitch boxes like I do with OpenSER
(single registrar, many proxies)?
./configure --enable-core-odbc
The second thing I'm thinking about is dynamic call-routing rules (aka
dialplan) with a lots of "destination numbers mangling" I have to do.
What is better way - using mod_xml_curl and try to serve exact
"extension" based on db lookups and followed processing or using an
event socket (in outbound mode I think) and completly control call-
flow
in Freeswitch from remote deamon. Or should I look to another
scenario?
use xml_curl.
Like I wrote, now I'm using Asterisk (together with OpenSER) with
realtime users and whole my dialpan looks
like:
exten=> _X.,1,AGI(routing.bin)
exten=> _X.,2,Hangup
where routing.bin is my simple "ANSI C" application doing all I need
and
"from time to time" communicating with "underlying" Asterisk :-)
I know I could study in-depth all source code and experiment with
various deployment scenarios, but it is distressful and long, long way
I'm trying to aviod. That is why I ask you, the ones with much deeper
knowledge of Freeswitch, what is the best point to start.
Better get ready to dive in.
http://wiki.freeswitch.org or #freeswitch on irc.freenode.net
Any suggestions, recommendation or hints are very appreciated! :-)
be like Nike and JUST DO IT! ;)
/b
_______________________________________________
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
