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

[Chan, Ka Lun]

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 at 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 at 64.127.123.100 >;tag=5318
> To: < sip:exten at x.x.x.x>;tag=as1cf1692c
> Call-ID: 1141227578-724-TF-GIXXER at 192.168.1.107
> CSeq: 813 INVITE
> User-Agent: x
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: < sip:exten at x.x.x.x>
> Proxy-Authenticate: Digest realm="asterisk", nonce="40a9764f"
> Content-Length: 0
>
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