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