[SR-Users] if (t_check_status("486|408"))

Daniel Tryba daniel at pocos.nl
Tue Jun 4 18:33:03 CEST 2013


On Tuesday 04 June 2013 12:07:35 hiro wrote:
> Sometimes it also seemed that kamailio was sending the INVITE to the
> phone instead of to freeswitch, or when i played around between
> changing $du or $ru the INVITE gets sends to freeswitch but with the
> wrong URI pointing to the phone instead of 127.0.0.1:5070 which is
> where freeswitch is listening.
> I guess it would be easier to reproduce if that random factor wasn't
> there, but at least it's failing most of the time, only in different
> ways.
> I had hoped I could get at least confirmation that it "works here" to
> keep me going :P
> I will test with xlog when I can test at home again which would at
> least exclude NAT issues.

It works here :)

I know your pain. I spend days figuring out the magic trick was to set $du to 
null (which I stumbled upon by accident). Without $du=$null traffic was being 
routed (seemingly random) to either the registered phone or the actual 
voicemail server.

# route to voicemail server
route[TOVOICEMAIL] {
        if(!is_method("INVITE"))
                return;

        # check if VoiceMail server IP is defined
        if (strempty($sel(cfg_get.voicemail.srv_ip))) {
                xlog("SCRIPT: VoiceMail routing enabled but IP not 
defined\n");
                return;
        }

        if($avp(dst_voicemail))
        {
                $du=$null;
                $ru = "sip:tovm-" + $avp(dst_voicemail) + "@" + 
$sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port);

                route(RELAY);

                exit;
        }

        return;
}

failure_route[MANAGE_FAILURE] {
....
        # serial forking
        # - route to voicemail on busy or no answer (timeout)
        if (t_check_status("486|408")) {
                route(CALLREDIRECT);
                route(TOVOICEMAIL);
                exit;
        }
....
}

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