17 apr 2013 kl. 13:04 skrev Bogdan-Andrei Iancu <[email protected]>:
> 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"; > 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"; /O > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 04/17/2013 12:51 PM, Arthur Titeica wrote: >> >> 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] should be allowed to go to asterisk >> and >> let asterisk to deal with auth. >> >> Thank you all for any input. >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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