When I try to route calls through my SER using sipp, I get there errors

May 11 17:17:34 proxy-ser ser[23159]: ERROR: udp_send: sendto(sock,0x283555b8,650,0,0x28355960,16): Can't assign requested address(49) May 11 17:17:34 proxy-ser ser[23159]: msg_send: ERROR: udp_send failed May 11 17:17:34 proxy-ser ser[23159]: ERROR: t_forward_nonack: sending request failed

Google didn't turn up anything interesting on them. Anyone here know whats up with them?

OS: FreeBSD 5.3 SER: version: 0.8.14 (i386/freebsd)

Also, I have accounting enabled to a database but I can not get it to log Bye messages. I know they are going through my server from some debug messages but they are not making it into the database. partial ser.cfg shown below. Any other suggestions on it are also appreciated.

# Accounting loadmodule "/usr/local/lib/ser/modules/acc.so"

# -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) #modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1)

route{ # xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n");

# initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops -- probebly a loop or something"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; };

# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route();

/* grant Route routing if route headers present */ # loose-route processing # if (loose_route()) { # t_relay(); # break; # };

# if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if(!(src_ip==x.x.x.x || src_ip==127.0.0.1)) { if (method=="REGISTER") { if (!www_authorize("proxy-ser.voip.aebc.com", "subscriber")) { www_challenge("proxy-ser.voip.aebc.com", "1"); break; };

save("ser_location"); break; } else if (method=="INVITE"){ if (!proxy_authorize("proxy-ser.voip.aebc.com", "subscriber")) { proxy_challenge("proxy-ser.voip.aebc.com", "1"); break; };

if(!check_from()) { sl_send_reply("403", "You can't lie about who you are"); }

} }

#Accounting if a call is started or ended if (method=="INVITE" || method=="BYE" || method=="CANCEL") { # xlog("L_NOTICE", "Accounting INVITE/BYE/CANCLE\n"); setflag(1); acc_db_request("", "acc"); }

# native SIP destinations are handled using our USRLOC DB if (lookup("ser_location")) { xlog("L_NOTICE", "Routing - Incommming call \n"); if (!t_relay()) { sl_reply_error(); }; break; };

} else { if(!(method=="ACK")) { sl_send_reply("403", "I only relay for my own customers"); } };

# authorize only for INVITEs . RR/Contact may result in weird # things showing up in d-uri that would break our logic; our if (method=="INVITE") { ## ## Conditional relaying using if(uri=~ and rewritehostport omitted for space and security ## }

# forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); };