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

[Daniel-Constantin Mierla]

Hello,

using a proxy in front of the pbx has some side effects if your phones 
are behind the nat. First, for registration, you have to enable with 
Path in the proxy and the pbx has to support it. AFAIK, Asterisk got 
just recently a patch for supporting Path.

Then, the asterisk has to be also changed a bit, because it will match 
all phone registrations coming from the same IP, which can create 
troubles if you match by source IP (not sure if this is still in latest 
Asterisks, but with some versions I run in such troubles, but since I 
was not the one handling the Asterisk, I cannot tell more).

If you really need to stick to existing asterisk deployment, then maybe 
looking at next tutorial will help a bit:

http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

If you can redesign the platform, you may want to change the logic so 
that kamailio takes care of registration and signaling and asterisk of 
media related services. This is more scalable and easier to handle in 
long term.

Cheers,
Daniel

On 8/1/13 12:44 AM, Rafael Carvallo wrote:
> 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.
>
>
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-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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