Hi all…

Well I have made some progress… Bellow is my routing statement:

route{

        if (!mf_process_maxfwd_header("10")) {

                sl_send_reply("483","Too Many Hops");

                break;

        };

        if (msg:len >=  2048 ) {

                sl_send_reply("513", "Message too big");

                break;

        };

        if (!method=="REGISTER") record_route();

        if (loose_route()) {

                append_hf("P-hint: rr-enforced\r\n");

                route(1);

                break;

        };

        if (!uri==myself) {

                append_hf("P-hint: outbound\r\n");

                route(1);

                break;

        };

        if (uri==myself) {

                if (method=="REGISTER") {

                        save("location");

                        break;

                };

                lookup("aliases");

                if (!uri==myself) {

                        append_hf("P-hint: outbound alias\r\n");

                        route(1);

                        break;

                };

        };

        append_hf("P-hint: usrloc applied\r\n");

        route(1);

}

route[1]

{

        if (src_ip==10.98.6.5) {

                if (dst_port==5065) {

                        t_relay_to_tcp("10.98.118.20", "5065");

                }

                else if (dst_port==5066) {

                        t_relay_to_tcp("10.98.118.20", "5066");

                }

                else if (dst_port==5067) {

                        t_relay_to_tcp("10.98.118.20", "5067");

                }

                else {

                        t_relay_to_tcp("10.98.118.20", "5060");

                }

        }

        else {

                t_relay_to_udp("10.98.6.5", "5060");

        };

}

When asterisk sends a call to kamailio, Kamailio then sends the invite to 10.98.118.20 via TCP on port 5061.

INVITE sip:1989@10.98.6.5:5061 SIP/2.0

Record-Route: <sip:10.98.6.5:5065;transport=tcp;r2=on;lr=on>

Record-Route: <sip:10.98.6.5:5061;r2=on;lr=on>

Via: SIP/2.0/TCP 10.98.6.5:5065;branch=z9hG4bK74fd.a4578a84.0

Via: SIP/2.0/UDP 10.98.6.5:5060;branch=z9hG4bK7b9bb22d;rport=5060

From: "1103" <sip:1103@10.98.6.5>;tag=as4ae41ccf

To: <sip:1989@10.98.6.5:5061>

Contact: <sip:1103@10.98.6.5>

Call-ID: 44073b911e86b0a96c9104cb7a5ec389@10.98.6.5

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 69

Date: Fri, 21 May 2010 12:23:58 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Content-Type: application/sdp

Content-Length: 232

P-hint: usrloc applied

The problem is that the Kamailio receives a 302 Moved Temporarily with a contact field of CONTACT: <sip:1989@10.98.6.5:5065;transport=TCP>

I need to have Kamailio, use this contact field and re-send the invite.

How can this be done?

Nelson Pereira

Senior Network Specialist

Protus
npereira@protus.com
phone: 613.733.0000 ext.528
MyFax: 613.822.5083

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