Connect OpenSIPS to TelecomsXchange

OpenSIPS often fronts a voice network as an SBC or least-cost router. This guide sends calls from it to TelecomsXchange and accepts TelecomsXchange's calls for routes you sell or numbers you bought. For every TelecomsXchange value in one place, see Connect a SIP platform.

Before you start#

• A SIP account with OpenSIPS's public IP, for example 203.0.113.10, or a VoIP login and password from Edit Account → Add SIP Digest Auth.
• One or more purchased routes and their tech prefixes from My Interconnections, for example 80039#.
• OpenSIPS 3.x with a database (db_mysql or db_postgres) and the tm, drouting, uac, uac_auth and permissions modules.
• To sell: a route with IP Address:Port set to OpenSIPS, for example 203.0.113.10:5060, and the values in Interconnect Info.

How do I send calls to TelecomsXchange?#

1. Add one gateway per purchased route. They share the address and differ in pri_prefix, the tech prefix that drouting puts in front of the number:

-- type 1 is a label of your choice for TCXC gateways
INSERT INTO dr_gateways (gwid, type, address, strip, pri_prefix, description)
VALUES ('tcxc_80039', 1, 'sip01.telecomsxchange.com:5060', 0, '80039#', 'TCXC route 80039');

-- send UK numbers (44...) in routing group 1 to that gateway
INSERT INTO dr_rules (groupid, prefix, timerec, priority, gwlist, description)
VALUES ('1', '44', '', 0, 'tcxc_80039', 'UK via TCXC');

2. Load the modules in opensips.cfg:

loadmodule "drouting.so"
loadmodule "uac_auth.so"
loadmodule "uac.so"
loadmodule "permissions.so"

modparam("drouting", "db_url", "mysql://opensips:PASSWORD@localhost/opensips")
modparam("drouting", "use_domain", 0)
modparam("permissions", "db_url", "mysql://opensips:PASSWORD@localhost/opensips")
# digest variant: the account's VoIP login, the realm TCXC challenges with (its platform IP), the VoIP password
modparam("uac_auth", "credential", "VOIP_LOGIN:174.136.44.213:VOIP_PASSWORD")

3. Normalise the number, route, and handle the challenge:

route[to_tcxc] {
    # strip + or 00 so the rules see plain E.164
    if ($rU =~ "^\+") {
        $rU = $(rU{s.substr,1,0});
    } else if ($rU =~ "^00") {
        $rU = $(rU{s.substr,2,0});
    }
    # do_routing() picks the gateway and prepends its pri_prefix: 80039#447700900123
    if (!do_routing(1)) {
        send_reply(404, "No TCXC route");
        exit;
    }
    t_on_failure("tcxc_auth");
    t_relay();
    exit;
}

failure_route[tcxc_auth] {
    if (t_was_cancelled()) exit;
    if (t_check_status("401|407")) {
        if (uac_auth()) {      # adds Authorization / Proxy-Authorization
            t_relay();
        }
        exit;
    }
}

Call route(to_tcxc) from your main route for INVITEs your customers send to international numbers, after authenticating those customers. Reload rules with opensips-cli -x mi dr_reload.

With IP authentication only, TelecomsXchange doesn't challenge and uac_auth() is never called; leave the credential line out.

Using dispatcher instead. For a single route, put sip:sip01.telecomsxchange.com:5060 in a dispatcher set, set $rU = "80039#" + $rU; after normalising, then ds_select_dst(1, 4) and relay with the same failure route.

How do I receive calls from TelecomsXchange?#

Calls on routes you sell, and on numbers you bought, come from 174.136.44.213 over UDP, from source ports 5060 to 5072, with the number in E.164 format such as 14165551234.

1. Allow 174.136.44.213 on your firewall for UDP 5060.
2. Add it to the address table as group 10, then run opensips-cli -x mi address_reload:

INSERT INTO address (grp, ip, mask, port, proto, context_info)
VALUES (10, '174.136.44.213', 32, 0, 'any', 'tcxc');

3. Near the top of your main route, before customer authentication:

if (is_method("INVITE") && check_source_address(10)) {
    # from TelecomsXchange: E.164 number in $rU, e.g. 14165551234
    # send it to your media servers, here drouting group 2
    if (!do_routing(2)) {
        send_reply(404, "Not here");
        exit;
    }
    t_relay();
    exit;
}

If your route has an Outgoing Prefix or strips digits (Edit Route → Outgoing Number Translation Parameters), expect the number in that format.

How do I receive calls on numbers I bought?#

Set the number's SIP Endpoint (Primary) in DID Settings to OpenSIPS, for example sip:14165551234@203.0.113.10:5060. The INVITE comes from 174.136.44.213, matches check_source_address(10), and is routed by the number in $rU.

How do I test it?#

1. Check the gateways and addresses are loaded: opensips-cli -x mi dr_gw_status and opensips-cli -x mi address_dump.
2. Start a capture: sngrep port 5060.
3. Place a call and check the INVITE to sip01 carries 80039#447700900123 (or 80039%23447700900123) and the reply code. If you see a 401 or 407, note its realm for the credential line.
4. In the portal, watch the call in Live Calls and find its SIP code in Logs → CDR (Voice).

What if something goes wrong?#

References#

• OpenSIPS drouting module
• OpenSIPS dispatcher module
• OpenSIPS uac module
• OpenSIPS uac_auth module
• OpenSIPS permissions module