31 Aug
2017
31 Aug
'17
12:40 p.m.
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