As I can see, your problem is to establish a call between two sip
phones, which does not need the usage of asterisk. Make sure you save
the contact address of the sip phones and you lookup them when the call
is initiated. The default configuration script of openser
(http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/etc/openser.cfg?rev=1.3&view=auto)
manages such case.
Daniel
On 09/29/05 18:05, 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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