Hi Sreeram,

Unfortunately the ladder diagram is not enough as I cannot set the details of all the messages :(. The it looks, the 200 OK coming from 110.46.1.106:5060 may contain bogus routing information (the dialog route set), like a wrong Contact hdr point to that EXTERNAL_IP....

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 11/21/23 12:13 PM, Sreeram Narayanan wrote:
Hi,
Thanks for your response.

I've added the network trace here <https://pastebin.com/raw/Lxi0SRZ4>. I've masked some of the IPs for security. This <https://pastebin.com/raw/rCLdemG4> is what the ACK looks like from the OpenSIPs server. Please let me know if I need to share more information.

On Wed, Nov 15, 2023 at 5:05 PM Bogdan-Andrei Iancu <[email protected] <mailto:[email protected]>> wrote:

    Hi,

    Ideally you should provide a network capture (pcap) from the
    OpenSIPS server, covering both incoming and outgoing traffic -
    this is the only way to understand what is wrong with the call.

    As attachments are limited to 40K here, consider using some
    pastebin or other file sharing service.

    Regards,

    Bogdan-Andrei Iancu

    OpenSIPS Founder and Developer
       https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
       https://www.siphub.com  <https://www.siphub.com>

    On 11/13/23 1:26 PM, Sreeram Narayanan via Users wrote:
    Hello,

    I am trying to use OpenSIPs with the load_balancer module to
    balance inbound calls between 2 Asterisk servers. The setup sits
    behind a NAT. The OpenSIPs server has a public IP and a private
    IP. When an INVITE arrives, it can forward it to one of the
    Asterisk servers and Asterisk responds with a 200 OK. The problem
    starts when I receive the ACK (from Twilio). The ACK starts
    bouncing between the public IP and Private IP of the OpenSIPs
    server. It doesn't reach the Asterisk server and eventually times
    out. I hope someone can help me with this. Thanks in advance.

    Here is my configuration:

        ####### Routing Logic ########
        route {

            if (is_method("INVITE")) {
                rtpproxy_engage();
            }

            if ($rm=="INVITE") {

                lb_start_or_next(1,"pstn");
            }

            t_check_trans();
            record_route();

            t_on_failure("GW_FAILOVER");

            # route the request
            if (!t_relay()) {
                sl_reply_error();
            }

            exit;
        }

        route[RELAY] {
            if (!t_relay()) {
                sl_reply_error();
            }
            exit;
        }

        failure_route[GW_FAILOVER] {
            if (t_was_cancelled()) {
                exit;
            }
            # failure detection with redirect to next available trunk
            if (t_check_status("(408)|([56][0-9][0-9])")) {
                xlog("Failed trunk $rd/$du detected \n");
            }
        }


-- - Sreeram

    _______________________________________________
    Users mailing list
    [email protected]  <mailto:[email protected]>
    http://lists.opensips.org/cgi-bin/mailman/listinfo/users  
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>



--
- Sreeram

_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Reply via email to