[OpenSIPS-Users] Drouting relay issue

[Volkan Oransoy]

Hi all,

I have an interesting issue with one of my test setups. I have a simple routing script which gets the gateway id directly from the header originating from a Freeswith box. The system finds and sets the request URL as anticipated. But even if I can see the request on the proxy, I can't see the traffic on the destination. Interestingly, the same proxy can register to the same destination with uac_registrant as a UAC. And I can receive calls from the same destination. Is there anything missing to route this traffic correctly?

Thanks in advance.

route[to_gateway] {
        if ( route_to_gw($hdr(X-GWID)) ) {
                route(relay);
        }
}
route[relay] {
        if (is_method("INVITE")) {
                t_on_branch("per_branch_ops");
                t_on_reply("handle_nat");
                t_on_failure("failure");
        }
        if (!t_relay()) {
                send_reply(500,"Internal Error");
        }
        exit;
}

The database structure is as follows;

opensips=# select * from dr_gateways;
 id | gwid | type |           address            | strip | pri_prefix | attrs | probe_mode | state | socket | description
----+------+------+------------------------------+-------+------------+-------+------------+-------+--------+-------------
  6 | 5    |    2 | sip:testgw.bulutfon.net:5060 |     0 |            |       |          0 |     0 |        | 5

Here is the INVITE request sent to the destination, which fails as in the screenshot.

INVITE sip:905551234567 at testgw.bulutfon.net:5060 SIP/2.0
Record-Route: <sip:2.2.2.2;lr;ftag=1jcyr93emrjDQ>
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK372.d935ecb2.0
Via: SIP/2.0/UDP 1.1.1.1:6080;received=1.1.1.1;rport=6080;branch=z9hG4bK99mtDaevmN7Nj
Max-Forwards: 67
From: "+908508850000" <sip:908508850000 at testgw.bulutfon.net:5060>;tag=1jcyr93emrjDQ
To: <sip:905551234567 at testgw.bulutfon.net:5060>
Call-ID: aefb8c60-426b-123b-8ca8-82a722ba4f75
CSeq: 51067506 INVITE
Contact: <sip:mod_sofia at 1.1.1.1:6080>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 621
Remote-Party-ID: "+908508850000" <sip:+908508850000 at testgw.bulutfon.net:5060>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1651210438 1651210439 IN IP4 1.1.1.1
s=FreeSWITCH
c=IN IP4 1.1.1.1
t=0 0 ...


--

Volkan Oransoy
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