Seems to work after doing some basic tests. Thank you all for the great input.
În ziua de Mie 17 Apr 2013, la 13:06:31, Olle E. Johansson a scris: 17 apr 2013 kl. 13:04 skrev Bogdan-Andrei Iancu <[email protected][1]>: > > Hello Arthur, > > > > The OpenSIPS script allows you to implement whatever logic you want, so the answer is : yes, you can do that. > > > > Reuse the part for handling the sequential requests from the default > > opensips script and for initial requests (after handling CANCEL and > > retransmissions) you can simply do : > > $du = "sip:asterisk_ip:asterisk_port"[2]; t_relay; > > > > This will send the INVITE to asterisk without changing the RURI at all. > > > Just to add a little bit more: > > > Depending upon the version of Asterisk, you might want to select transport > as well, like > > > $du = _"sip:asterisk_ip:asterisk_port;transport=udp"_; > Hello, I'm rather new to opensisps. From what I read this is not possible but I thought I should ask just to make sure. Is there a configuration setup so that opensips doesn't handle users/extensions but just forwards everything that matches the configured domains to an asterisk gateway? Something like [email protected][4] should be allowed to go to asterisk and let asterisk to deal with auth. Thank you all for any input. -- Arthur Titeica _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
