[SR-Users] _w_t_relay_to(): t_forward_noack failed when forwarding call

[Konstantinos Merentitis]

I would like to forward all incoming calls that are not answered from kamailio, to my asterisk voicemail server. - I have not setup any Realtime Integration.
I am using the "Redirected Dialed Number Information Service" header and calling "554 at 192.168.1.134" which is my voicemail server.
Forwarding succeeds when the subscriber is not registered to kamailio:

Nov  7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============User sip:6297 at 192.168.1.131 not found here, going to voicemail=============
Nov  7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: #012 RDNIS = sip:297 at 192.168.1.134 
Nov  7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297 at 192.168.1.134  =========== 

I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded:

Nov  7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: transaction canceled
Nov  7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: #012 RDNIS = sip:297 at 192.168.1.134 
Nov  7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297 at 192.168.1.134  =========== 
Nov  7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed
Nov  7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL)


failure and voicemail routes are slightly modified from the default config:

route[LOCATION] {

#!ifdef WITH_SPEEDDIAL
        # search for short dialing - 2-digit extension
        if($rU=~"^[0-9][0-9]$") {
                if(sd_lookup("speed_dial")) {
                        route(SIPOUT);
                }
        }
#!endif

#!ifdef WITH_ALIASDB
        # search in DB-based aliases
        if(alias_db_lookup("dbaliases")) {
                route(SIPOUT);
        }
#!endif

        $avp(oexten) = $rU;
        if (!lookup("location")) {
                xlog("=============User $ru not found here, going to voicemail=============\n");
                $var(rc) = $rc;
                route(TOVOICEMAIL);
                t_newtran();
                switch ($var(rc)) {
                        case -1:
                        case -3:
                                send_reply("404", "Not Found");
                                exit;
                        case -2:
                                send_reply("405", "Method Not Allowed");
                                exit;
                }
        }

        # when routing via usrloc, log the missed calls also
        if (is_method("INVITE")) {
                setflag(FLT_ACCMISSED);
        }

        route(RELAY);
        exit;
}

route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
        if(!is_method("INVITE|SUBSCRIBE")) 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(is_method("INVITE")) {
                $var(RDNIS)=$di;
                xlog("\n RDNIS = $var(RDNIS) \n");
                if($avp(oexten)==$null) return;
#               $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp")
#                               + ":" + $sel(cfg_get.voicemail.srv_port);
                xlog("=============forwarding to voicemail1, RDNIS: $di  =========== ");
                $ru = "sip:554 at 192.168.1.134:5060";
        } else {
                if($rU==$null) return;
#              $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp")
#                              + ":" + $sel(cfg_get.voicemail.srv_port);
                xlog("=============forwarding to voicemail2, RDNIS: $di  =========== ");
                $ru = "sip:554 at 192.168.1.134:5060";

        }
        route(RELAY);
        exit;
#!endif

        return;
}

# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
        route(NATMANAGE);

        if (t_is_canceled()) {
                #modified here
                xlog("transaction canceled\n");
                route(TOVOICEMAIL);
                #exit;
                #/modified
        }
        
#!ifdef WITH_BLOCK3XX
        # block call redirect based on 3xx replies.
        if (t_check_status("3[0-9][0-9]")) {
                t_reply("404","Not found");
                exit;
        }
#!endif

#!ifdef WITH_BLOCK401407
        # block call redirect based on 401, 407 replies.
        if (t_check_status("401|407")) {
                t_reply("404","Not found");
                exit;
        }
#!endif

#!ifdef WITH_VOICEMAIL
        # serial forking
        # - route to voicemail on busy or no answer (timeout)
        if (t_check_status("486|408")) {
                $du = $null;
                xlog("============= Route failure, forwarding to voicemail =============");
                route(TOVOICEMAIL);
                exit;
        }
#!endif
}