29 Sep
2005
29 Sep
'05
noon
whats is sip debug on asterisk showing
Bogdan-Andrei Iancu wrote:
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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