Hello,

not sure I understood properly your issue, but if it is that the BYE is not getting to your dispatching/distribution rules, then all should be fine, because BYE (and other requests withing dialog) should be routed with 'record routing', therefore be sure you do record_route() from initial INVITE of the call and loose_route() for requests within dialog. In case you look at default kamailio.cfg, it has this kind of logic in it.

Cheers,
Daniel

Hello! provider trunks registration on kamailio UAC?! it's clear.

but how does asterisk find out through which trunk the call should be made? 

I tried to set the header from asterisk dialplan. 

for example:

 if (is_method("INVITE")) {

                                 #record_route_preset("109.195.102.122");

        route(DIRECTION);

        setflag(FLT_ACC); # do accounting

        }

#        ------ LOADBALANCE ROUTE ------------ #

if(!ds_is_from_list()) {

route(DISPATCH);

}

route[DISPATCH] {

        #round robin dispatching on gateways group '1'

        if(!ds_select_dst("1", "4"))

                {

                send_reply("404", "No destination");

                exit;

        }

        xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");

        t_on_failure("RTF_DISPATCH");

route(NATMANAGE);

route(DRELAY);

        exit;

}

route[DIRECTION] {

                 if ($hdr(x-trunk)  != $null) {

                 if (!is_method("BYE")){

                                        $fu="";

                                        t_on_failure("MANAGE_FAILURE");

                                        $dlg_ctx(timeout_route) = "DIALOG_END";

                                        $avp(i:10)=43200;

                                        $dlg_ctx(timeout_bye) = 0;

        sql_pvquery("ca", "select l_uuid, auth_username, auth_password, realm, l_domain, r_domain  from uacreg where id='$hdr(x-trunk)'", "$avp(uuid), $avp(uname), $avp(passwd), $avp(realm), $avp(src_ipaddr), $avp(dst_ipaddr)");

                                        t_on_failure("MANAGE_FAILURE");

                                        $dlg_ctx(timeout_route) = "DIALOG_END";

                                        $avp(i:10)=43200;

                                        $dlg_ctx(timeout_bye) = 0;

                $fu="";

                uac_replace_from("sip:$avp(uname)@$avp(dst_ipaddr)");

                $tu="sip:"+$tU+"@"+$avp(dst_ipaddr);

                $ru="sip:"+$tU+"@"+$avp(dst_ipaddr);

                remove_hf("Contact");

                $var(contact)="sip:"+$avp(uname)+"@10.49.9.2:5060";

                insert_hf("Contact: <$var(contact)>\r\n");

                #insert_hf("Contact:  ");

                msg_apply_changes();

                fix_nated_register();

                xlog("L_INFO","Contact header $var(contact) 111111111111111111111111111111111111111 is $ct {$ct}\n");

                route(RELAY);

 }      #### BYE

}       ### XTRUNK

But in this configuration I do not come bye

but when I register providers trunks on asterisk - problem with BYE not visible. 
but I can not register provider-trunks on all the asterisks, because incoming invite arrives at the link + address, and all the asteriscs ring. All my asterisk's behind nat

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