<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><html>On Mar 29, 2008, at 7:57 AM, kokoska rokoska wrote:</html><br class="Apple-interchange-newline"><blockquote type="cite"><br>Hi all!<br><br>I'm very new to Freeswitch and thus I'm looking for advices/hints for <br>painless start :-)</blockquote><div><br></div><div>Nothing is painless. </div><br><blockquote type="cite"><br>I have a lot of experience with Asterisk and OpenSER, but the philosophy <br>od Freeswitch differs...<br><br></blockquote><div><br></div><div>Thats an understatement. ;)</div><div><br></div><br><blockquote type="cite">What is going on:<br>I like to deploy PBX/Switch for a lot of SIP users wich registers with <br>it and - also - with larg number of SIP gateways the PBX/Switch should <br>regester with. Users population/gateways/call routing have to be dynamic <br>(database-driven, like I'm accustomed form Asterisk and OpenSER) with <br>quite standard features (conditional/unconditional forwarding, <br>voice-mail, call-waiting, resource limits etc.) and especially with good <br>over-all performance.<br><br>Like i red in docs, dynamic SIP users could by done with mod_xml_curl <br>directory but I like to ask: Is it "fast enough"?</blockquote><div><br></div><div>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.</div><br><blockquote type="cite"><br>Couldn't be better direct DB lookups? If yes, how to accomplish that?<br>BTW: Is there a way how to share "registered" users between independant <br>Freeswitch boxes like I do with OpenSER<br>(single registrar, many proxies)?<br><br></blockquote><div><br></div><div><span class="Apple-style-span" style="font-family: Times; font-size: 16px; "><span class="Apple-style-span" style="font-family: -webkit-sans-serif; font-size: 13px; line-height: 19px; "></span></span></div><div>./configure --enable-core-odbc</div><br><blockquote type="cite">The second thing I'm thinking about is dynamic call-routing rules (aka <br>dialplan) with a lots of "destination numbers mangling" I have to d<span class="Apple-style-span" style="-webkit-text-stroke-width: -1; ">o.</span></blockquote><blockquote type="cite">What is better way - using mod_xml_curl and try to serve exact <br>"extension" based on db lookups and followed processing or using an <br>event socket (in outbound mode I think) and completly control call-flow <br>in Freeswitch from remote deamon. Or should I look to another scenario?</blockquote><div><br></div><div>use xml_curl.</div><br><blockquote type="cite"><br><br>Like I wrote, now I'm using Asterisk (together with OpenSER) with <br>realtime users and whole my dialpan looks<br>like:<br>exten=> _X.,1,AGI(routing.bin)<br>exten=> _X.,2,Hangup<br>where routing.bin is my simple "ANSI C" application doing all I need and <br>"from time to time" communicating with "underlying" Asterisk :-)<br><br>I know I could study in-depth all source code and experiment with <br>various deployment scenarios, but it is distressful and long, long way <br>I'm trying to aviod. That is why I ask you, the ones with much deeper <br>knowledge of Freeswitch, what is the best point to start.<br></blockquote><div><br></div><div>Better get ready to dive in.</div><div><br></div><div><a href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a> or #freeswitch on irc.freenode.net</div><div><br></div><br><blockquote type="cite">Any suggestions, recommendation or hints are very appreciated! :-)<br><br></blockquote><br></div><div>be like Nike and JUST DO IT! ;)</div><div><br></div><div>/b</div><div><br></div></body></html>