[Users] Re: [Devel] openser and asterisk

Bogdan-Andrei Iancu bogdan at voice-system.ro
Thu Sep 29 17:52:13 CEST 2005


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
>
>
>
> _______________________________________________
> Devel mailing list
> Devel at openser.org
> http://openser.org/cgi-bin/mailman/listinfo/devel
>





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