Re: [SR-Users] Asterisk and Kamailio as a sip router

Hey everyone, currently i have this configuration set on kamailio: modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 0) #forzado de la reescritura direccion de destino modparam("dispatcher", "flags", 2) #banderas de funcionamiento, 2 significa "soporte para failover" modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) #tiempo que transcurre antes de verificar nuevamente una salida inactiva modparam("dispatcher", "ds_probing_threshhold", 5) #Numero de intentos antes de marcar una salida como inactiva modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) #loadmodule "dispatcher.so" ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { if (is_method("SUBSCRIBE")){ if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){ t_relay(); } route(REGISTRAR); } if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){ t_relay(); } else{ route(ASTERISK); } } route[ASTERISK]{ ds_select_dst("1", "8"); t_relay(); exit(); } Everything else is left with the same basic configuration it had when i installed the software. I want to use it kinda of a sip router, so what Kamailio does is just forward the packets betwen my Asterisk boxes and the Sip Phones. Currently it seems to work (partially) but i have doubts about if this is correctly done (it's the first time using Kamailio and i need this working withing a week at most). As you can see i use the module dispatcher for failover/failback (this is the purpose of using Kamailio, a failover/failback setup). One major problem i've found with this setup is, if the phones are currently connected and working with one of my 2 asterisk boxes and if that box fails, Kamailio starts sending the traffic to the second box (as intended), but the phones don't try to subscribe to the new asterisk box, rather they just keep sending traffic (and obviously kamailio forwarding it). Sometimes one of the phones subscribe to the new box, but that's not always the case, the packets reach the new Asterisk box, but since the phones aren't registered to it, they can't make calls. Other times the behavior of the setup is rather weird, i can call some extensions and some others i can not (even though they are registered within the asterisk box) The traffic gets to the Asterisk box (as shown in the asterisk logs) but the call is shown as "service unavailable". I've checked a lot of times the Asterisk setup and it seems to be fine so i think it has something to do with Kamailio. I know this might be a really bad configuration file, but it's been at most 3 days since i started using Kamailio which i sometimes find kinda hard to understand and i really need this working within a week. To summarize all i want is kamailio forwarding packets between the currently active server and the phones so if it fails, then the packets go to the second one, the phones must re-subscribe to the new active server. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users