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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