[Kamailio-Users] Trying to deliver a call across multiple peers with t_relay

[Ovidiu Sas]

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 at 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!
>
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