[Asterisk-Dev] sip call routing

[mark spowage]

there must be a way to script how Asterisk will route calls, or just MODIFY
the source code ?
 
so far searching "asterisk sip call routing" as not given up any clues.
 
when an INVITE comes from a client ..then asterisk should be able to run
a script to determine which client to send the call. SER provides some nice
scripting to handle this kind of "smart" forwarding.
 
why such a basic function is "blind sided" is beyond me
 
thx
 

		
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