Re: [SR-Users] Issue with Asterisk interconnection for VoiceMail

And $ru is OK while sending to wrong (initial) IP? Did you try to set/check $du too?

2015-05-15 11:30 GMT+03:00 Igor Potjevlesch igor.potjevlesch@gmail.com:

Hello,

I experienced a strange issue with some of VoIP accounts.

When the INVITE comes into MANAGE_FAILURE, after timeout, the config identifies, with "dialplan", the right Asterisk instance that should handle the call for voicemail.

This part is okay, and results in a new INVITE with the Request-URI formed with the right domain (eg. sip:<NUMBER>@asterisk3). Then, the request goes to RELAY. Here is the issue: sometimes, the request is forwarded to the IP of the UA (the one initially contacted) instead of the IP of Asterisk.

I can't figure out the difference between a succeeded call and a failed one.

If someone has an idea. Here is the config that handles the VoiceMail:

failure_route[MANAGE_FAILURE] {

[…]

if (isflagset(24)) {

                    $avp(s:inv_timeout) = "5";

                    t_set_fr($avp(s:inv_timeout)*1000);

                    if

(avp_db_load("$to/username","$avp(s:vm_uri)/usr_vm")) {

                               resetflag(24);

                               avp_pushto("$ruri","$avp(s:vm_uri)");

                               # Dynamic routing

                               if

(avp_db_load("$ruri/username","$avp(s:client)/usr_fai")) {

                                           if

(dp_translate("2","$avp(s:client)/$avp(s:dest)") == 1) {

                                                       $ru = "sip:" +

$rU + "@" + $avp(s:dest);

                                           } else {

                                                       # Load default

voicemail

                                                       $avp(s:client)

= "DEFAULT_VM";

dp_translate("2","$avp(s:client)/$avp(s:dest)");

                                                       $ru = "sip:" +

$rU + "@" + $avp(s:dest);

                                           };

                               } else {

                                           # Load default voicemail

                                           $avp(s:client) =

"DEFAULT_VM";

dp_translate("2","$avp(s:client)/$avp(s:dest)");

                                           $ru = "sip:" + $rU + "@" +

$avp(s:dest);

                               }

                    } else {

                               xlog("L_WARN","time=[$Tf] call id=[$ci]

call seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], User have no mail box\n");

                               exit;

                    };



                    prefix("710");

                    xlog("L_WARN","time=[$Tf] call id=[$ci] call

seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], failure route to Voice Mail\n");

                    route(RELAY);

                    exit;

        }

Regards,

Igor.

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users