Hello List,

i answering myself - just for the records.

For my issues i found an solution.

1.) If sips from broken client, save "sip://" scheme to record-route (like described before) and then reformat it to the old $ru
2.) Attach a FLT_Flag to this
3.) On my dispatcher route i force "transport=udp" to it, if my FLT_Flag there is

The 60 second sound problem was an rfc4028 - session timer issue between UAC and target B2BUA (FreeSWITCH).
I could solve this with 'param name="enable-timer" value="false"' within FreeSWITCH sip-profile.


#!define FLT_SRCSIPS 19

request_route {
...
        remove_hf("Route");
        if (is_method("INVITE|SUBSCRIBE")) {
                if ($ru =~ "^sips:") {
                        route(FIX_SIPS);
                } else {
                        record_route();
                }
        }
...
}

route[FIX_SIPS] {
        if ($ru =~ "^sips:") {
                xlog("L_INFO","---FIX_SIPS-before: ru:($ru)");
                $var(orig_uri) = $ru;
                $ru = "sip:" + $rU + "@" + $td;
                xlog("L_INFO","---FIX_SIPS-now: $rU $td ru:($ru) orig_uri:($var(orig_uri))");
                record_route();
                $ru = $var(orig_uri);
                xlog("L_INFO","---FIX_SIPS-after: ru:($ru)");
                setflag(FLT_SRCSIPS);
        }
}


route[DISPATCHIVR] {
        ...
        if(isflagset(FLT_SRCSIPS)) {
                $var(orig_du) = $du;
                $du = "sip:" + $rd + ":" + $rp + ";transport=udp";
                xlog("L_INFO","reformate ($var(orig_du)) to ($du)");
        }
        ...
}


2017-08-18 13:54 GMT+02:00 Karsten Horsmann <khorsmann@gmail.com>:
Hello List,


of course - best way is not to use sips: uri scheme. But i have to deal with that.

I try to configure an multihomed kamailio (public/private IP) in front of my SIP-Servers.
One of my softphones variants used TLS for connection but the INVITES of this softphone use the "sips" URI scheme.

As i saw in other posts, this seems to be an issue for many people.
In my case it breaks the routing to my internal sip-server (only plain udp).

Most of my config is similar to havfos example + TLS.

[...]



--
Kind Regards
*Karsten Horsmann*