[SR-Users] Routing outgoing call from asterisk to registered SIP provider

[Bruno Salzano]

Hello,
i've DID provider that require to register to it's server. I've been able
to configure this provider using kamailio and uac and to route all inbound
calls from DID provider to internal asterisk media server. All is working
great.

Now i need help to configure call routing from internal asterisk to the
external DID provider.

I've done something like:

route[LOCATION] {
if(is_method("INVITE")) {
route(FROMASTERISK);
route(TOASTERISK);
exit;
}
[...]
}

route[FROMASTERISK] {
if (ds_is_from_list()) {
xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server Cluster\n");
rtpengine_manage("direction=internal direction=external replace-origin
replace-session-connection trust-address");
$ru = "sip:TESTING_NUMBER at EXTERNAL_PROVIDER_IP";
route(RELAY);
exit;
}
}

It seems that when i route the call outbound, i get a 407. So i would like
to react to this using uac_auth() as described here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()

Is this the right way? Can someone help me, giving hints and some working
example?

Using kamailio 4.3.3 with rtpengine 4.1.0

Best regards,
Bruno
-- 
The life is short.. live at your best!
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