[OpenSIPS-Users] Webrtc client ~> opensips~>astrrisk

[Prathibha B]

I have a webrtc client. I'm using the dispatcher module as a stateless proxy which forwards the request to asterisk.

dispatcher table contains:

+----+-------+-----------------------------------+---------------------+-------+--------+----------+-------+-------------+------------+
| id | setid | destination                       | socket              | state | weight | priority | attrs | description | probe_mode |
+----+-------+-----------------------------------+---------------------+-------+--------+----------+-------+-------------+------------+
|  1 |     1 | sip:10.214.0.18:443;transport=wss | wss:10.214.0.18:443 |     1 | 100    |        1 |       | pbx1        |          0 |
+----+-------+-----------------------------------+---------------------+-------+--------+----------+-------+-------------+------------+

The registration is successful. But when I make a call, it throws 404: Not Found.

Script:

```
route {
        if (!mf_process_maxfwd_header(10)) {
                send_reply(483, "Too Many Hops");
                exit;
        }

        if (!ds_select_dst(1, 0)) {
                send_reply(503, "Service Unavailable");
                exit;
        }
       t_relay();
}
```
Sent from Outlook for Android<https://aka.ms/AAb9ysg>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20250219/b0f4d8af/attachment.html>