[SR-Users] No dispatcher failover

[Alex Balashov]

Hello Gertjan,

The "ingredients" of dispatcher failover are as follows:

1. A "flags" setting that includes the value 2, which enables failover
support:

https://kamailio.org/docs/modules/5.1.x/modules/dispatcher.html#dispatcher.p.flags

2. A failure_route to actually handle your failover logic:

   route[MAIN] {
      ...

      if(!ds_select_dst("2", "8")) {
          sl_send_reply("503", "Out of gateways");
	  exit;
      }

      t_on_failure("DISPATCHER_FAILOVER");

      if(!t_relay())
         sl_reply_error();
   }

   failure_route[DISPATCHER_FAILOVER] {
      if(t_is_canceled())
         exit;

      if(!ds_next_dst()) {
         send_reply("503", "Out of gateways");
	 exit;
      }

      t_on_failure("DISPATCHER_FAILOVER");
      t_relay();
   }

-- Alex

On Wed, Sep 26, 2018 at 02:15:22PM +0200, Gertjan Wolzak wrote:

> Hello Kamailions, 
> 
> I want to send traffic to an Asterisk box but when that box fails traffic should be routed to the second Asterisk box. 
> 
> The dispatcher params are as follows: 
> 
> # ----- Dispatcher params ------ 
> 
> modparam("dispatcher", "ds_probing_mode", 1) 
> modparam("dispatcher", "ds_ping_interval", 20) 
> modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") 
> modparam("dispatcher", "ds_inactive_threshold", 4) 
> modparam("dispatcher", "ds_ping_latency_stats", 1) 
> modparam("dispatcher", "ds_probing_threshold", 1) 
> 
> My dispatcher.list is: 
> 
> #DBBS PBX 
> 
> # SBC's 
> 
> 1 sip:ip_address_sbc:5060 0 0 weight=100 
> 
> #PBX 
> 
> 2 sip:ip_pbx_01:5060 0 10 weight=100 
> 2 sip:ip_pbx_02:5060 0 5 weight=100 
> 
> Verry simple request route: 
> 
> request_route { 
> 
> if (is_method("OPTIONS")) { 
> sl_send_reply("200", "OK"); 
> exit; 
> } 
> 
> #Call from SBC to PBX 
> #Get gateway from dispatcher list 
> 
> if($si=="ip_address_sbc1" |$si=="ip_address_sbc2" ) 
> { 
> ds_select_dst("2", "8"); 
> $var(new_uri) = "sip:" + $tU + "@" + $dd + ":5060"; 
> $ru=($var(new_uri)); 
> route(RELAY); 
> exit; 
> } 
> 
> 
> # Call from pbx to SBC 
> # Get sbc ip from dispatcher list 
> ds_select_dst("1","4"); 
> $var(new_uri) = "sip:" + $tU + "@" + $dd + ":5060"; 
> $ru=($var(new_uri)); 
> route(RELAY); 
> } 
> 
> When I send a call to the kamailio its routed correctly to the first pbx. But when I kill the asterisk process on the first pbx, an invite is still send by Kamailio to the first pbx. 
> 
> Maybe I am assuming incorrectly that when the dispatcher does not react to the ds_ping it will be set to "inactive" and not be selected by the ds_select_dst command... 
> 
> Or am I thinking to easy and do I need to configure some actions in the failure route? 
> 
> Pretty sure I'm doing something wrong, but cant see where and have not been able to find the information. 
> 
> Tips are appreciated. 
> 
> Rgds, 
> Gertjan Wolzak 
> 
> 
> 

> _______________________________________________
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> sr-users at lists.kamailio.org
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-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/