Hi Matt,
I redirected this email on the users mailing list - it's more appropriate.
the idea seams ok, with couple of comments: 1) be sure that fwd to localhost is ok (instead of a routable IP) 2) doing Record-Route may be a good think.
to debug tour problem, add some log("...") statements into your script to be able to trace the processing. Also a network trace (including on lo device) will be helpful to see what happens - if the messages are received, if they are sent and where. Also watch the log for potential errors.
regards, bogdan
Matt L. Zhu wrote:
has anyone successfully setup openser as the frontend proxy for asterisk? here is my setup
/etc/asterisk/sip.conf [general] context=default port=5065 bindaddr=0.0.0.0 srvlookup=yes
[ser] type=user context=proxy host=192.168.0.10
then i edited openser.cfg to do something like this
if(uri=~"sip:[a-zA-Z.]*@(xxx.xxx.com)|(192.168.0.10)") { forward( localhost, 5065 ); break; };
i connected two sipphones (wengo) in this case to openser, but calls are not going through at all, connecting directly to asterisk works. have anyone worked in this situation?
thanks
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