[Users] Re: * sip response 407 when using dispatcher (Help Please)

For what ever reason I can't get this to work with Asterisk 1.0.x release, but it work on Asterisk 1.2.x release. They both are using the same configuration and the only difference is Asterisk 1.0.x release was using AST_DATA to talk to Postgres, and Asterisk 1.2.x is using Unix_odbc to talk to Postgres. Anyway I am not very sure why one work and one doesn't but it work great now. =)

On 3/1/06, Chan, Ka Lun kchan1028@gmail.com wrote:

Hi All,

I am trying to set up SER with Dispatcher to loadbalancing the traffic

to 2 * boxes. SER was able to select the * IPs from the dispatcher.list, but * SIP response back authentication required. It work perfectly if I i use rewritehostport instead of using the dispatch module. I am pulling my hair now and still don't know where the problem at.

openser.cfg if (uri=~"sip:+?[1-9][0-9]*@.*") { ds_select_dst("2", "0"); route(4); route(5); return; };

route[4] {

    if (isflagset(6)) {
            force_rport();
            fix_nated_contact();
            force_rtp_proxy();
    };

}

route[5] {

    setflag(1);
    t_on_reply("1");
    forward(uri:host, uri:port);
    append_hf("P-hint: main PSTN route\r\n");
    t_on_failure("1");
    if (!t_relay()) {
            sl_reply_error();
            return;
    };

}

onreply_route[1] {

    if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
            if (!search("^Content-Length:[ ]*0")) {
                    force_rtp_proxy();
            };
    };

    if (nat_uac_test("1")) {
            fix_nated_contact();
    };

}

failure_route[1] {

   append_hf("P-hint: backup PSTN route\r\n");
   rewritehost("x.x.x.x");
   rewriteport(   "5060");
   append_branch();
    t_relay();

}

SIP.conf form * [general] host=dynamic bindaddr=0.0.0.0 port=5060 useragent=x context=default disallow=all allow=g729 allow=ulaw autocreatepeer=yes dtmfmode=rfc2833 qualify=no nat=yes canreinvite=no

Retransmitting #5 (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP x.x.x.x;branch=0 Via: SIP/2.0/UDP 192.168.1.107:5060;received=x.x.x.x;branch=z9hG4bK263724 From: "testing" <sip:testing@64.127.123.100 >;tag=5318 To: < sip:exten@x.x.x.x>;tag=as1cf1692c Call-ID: 1141227578-724-TF-GIXXER@192.168.1.107 CSeq: 813 INVITE User-Agent: x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: < sip:exten@x.x.x.x> Proxy-Authenticate: Digest realm="asterisk", nonce="40a9764f" Content-Length: 0