So I took your advice and decided to use * to identify sip 2 sip calls. However, theres something wrong with my routing. I added route(6) to get authorize. Because when I try to dial sip to sip I get 407 proxy authentication required. Still after adding route(6), I still get the 407 proxy authentication required message. What is wrong? Route (1) is just the default message handler This is what I have:

route[3] {

    # -----------------------------------------------------------------
    # INVITE Message Handler
    # -----------------------------------------------------------------

    if (!proxy_authorize("","subscriber")) {
        proxy_challenge("","0");
        return;
    } else if (!check_from()) {
        sl_send_reply("403", "Use From=ID");
        return;
    };

    consume_credentials();

    if (nat_uac_test("19")) {
        setflag(6);
    }

    lookup("aliases");
    if (uri!=myself) {
        route(4);
        route(1);
        return;
    };

    if (uri=~"^sip:\*[0-9]*@"){
        xlog("Sip 2 Sip\n");
        strip(1); #strip the * because we dont need it
        route(4);
        route(6);
        route(1);
        return;

    };

    if (!lookup("location")){

        if (uri=~"^sip:[0-9]*@") {        # International PSTN
            xlog("PSTN Gateway\n");
            route(4);
            route(5);
            return;
        };

        sl_send_reply("404", "User Not Found");
        return;
    };

    route(4);
    route(1);
}

route[4] {

    # -----------------------------------------------------------------
    # NAT Traversal Section
    # -----------------------------------------------------------------

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

route[5] {

    # -----------------------------------------------------------------
    # PSTN Handler
    # -----------------------------------------------------------------
    xlog("Routed to route 5\n");
    rewritehostport("pstn.gateway:5060");

    avp_write("i:45", "inv_timeout");

    route(1);
}

route[6] {

    if (!proxy_authorize("","subscriber")) {
        proxy_challenge("","0");
        return;
    } else if (!check_from()) {
        sl_send_reply("403", "Use From=ID");
        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();
    };
}

Bogdan-Andrei Iancu <bogdan@voice-system.ro> wrote: Hi,

that's right. For example SIPURA ATAs with two lines but online one
terminal use # for line selection....
you better use a digit that does not overlap with the PSTN dialling plan.

regards,
bogdan

Glenn Dalgliesh wrote:

>Well I would becarefull using # since some UA's use # to terminate digit input and dial..... Not positive but I think * would be a better choice.
>---------------------
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>
>-----Original Message-----
>From: Kenny Chua
>Date: Wednesday, Jun 28, 2006 10:56 pm
>Subject: [Users] Using # for Sip 2 Sip calls
>
>Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls?
>
> I came up with something like this:
> lookup("aliases");
> if (uri=~"^sip:#[0-9]*@"){
> xlog("Sip 2 SIP\n");
> route(4);
> route(1);
> return;
> };
>
> Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you.
>
>
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>--0-591390942-1151549737=:48905
>Content-Type: text/html; charset=iso-8859-1
>Content-Transfer-Encoding: 8bit
>
>Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls?

I came up with something like this:
    lookup("aliases");
    if (uri=~"^sip:#[0-9]*@"){
        xlog("Sip 2 SIP\n");
        route(4);
        route(1);
        return;
    };

Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you.

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