You need to use the call append_branch() on the failure_route before calling t_relay(). Check the following sample config and adapt the failure logic for your needs: http://www.voipembedded.com/resources/openser_cr.cfg
Regards, Ovidiu Sas
On Fri, Nov 28, 2008 at 9:44 AM, Tyler Brauer tbtbrauer@gmail.com wrote:
Thank you Klaus for the suggestions. I've implemented the configuration below. When the first t_relay fails, Kamailio generates the 408 Request Timeout and doesn't attempt to try an INVITE to the second address. This what the syslog says:
Nov 28 06:11:13 server kamailio[1315]: Attempting Carrier 1 Nov 28 06:11:28 server kamailio[1323]: Attempting Carrier 2 Nov 28 06:11:28 server kamailio[1323]: ERROR:tm:t_forward_nonack: no branch for forwarding Nov 28 06:11:28 server kamailio[1323]: ERROR:tm:w_t_relay: t_forward_nonack failed Nov 28 06:11:28 server kamailio[1317]: Attempting Carrier 1
And here's the watered-down version of my configuration.
# - - - - - - - - - - - - - - - - - - - -
route[0] { if($si=='10.10.10.10') { # Allowed Host route(1); } else { sl_send_reply("603", "Decline"); exit; } }
# - - - - - - - - - - - - - - - - - - - -
route[1] { if(msg:len > 2048) { sl_send_reply("513", "Message Too Big"); exit; }
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } if (method==OPTIONS) { if ((uri==myself) && (! uri=~"sip:.*[@]+.*")) { options_reply(); exit; } } if (!method=="REGISTER") { record_route(); } if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); } if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } route(2);
}
# - - - - - - - - - - - - - - - - - - - -
route[2] { xlog("L_INFO", "Attempting Carrier 1"); t_on_failure("3"); t_relay("udp:1.1.1.1:5090","0x2"); exit; }
# - - - - - - - - - - - - - - - - - - - -
failure_route[3] { xlog("L_INFO", "Attempting Carrier 2"); t_on_failure("4"); t_relay("udp:2.2.2.2:5060"); exit; }
# - - - - - - - - - - - - - - - - - - - -
failure_route[4] { # I would like to know how to redirect the # original call destined to extension "100" # to 2015551234, but the originating server # will get nervous if the username@ part # of its SIP request is changed. How could }
# - - - - - - - - - - - - - - - - - - - -
Any further suggestions would be appreciated. Thanks!
Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users