Re: [Serusers] mod_init: bind_db failed..

Hi I'm running trying out latest CVS checked out today with the following configuration and am having the following error: May 6 00:56:54 hostname /usr/local/sbin/ser[30030]: ERROR:acc:mod_init: bind_db failed...did you load a database module? May 6 00:56:54 hostname /usr/local/sbin/ser[30030]: init_mod(): Error while initializing module acc However I can connect to mysql just fine, from the command line. In addition I just ran ser_mysql.sh create. Anyone know I am getting this error? Attached is my ser.cfg with no modifications ( this is stock from ser.cfg.m4) regards, Andy ------------------------------------------------------------------------ # # $Id: rules.m4,v 1.1 2004/01/13 19:42:56 janakj Exp $ # # (c) 2003 iptel.org # # Rules to process ser.cfg templates # ### m4 macros to make the configuration easier ### End of m4 macro section # # $Id: ser.cfg.m4,v 1.1 2004/01/13 19:42:56 janakj Exp $ # # ser.cfg m4 template # # # Set the following in your CISCO PSTN gateway: # sip-ua # nat symmetric role passive # nat symmetric check-media-src # fork=yes port=5060 log_stderror=no fifo="/tmp/ser_fifo" # uncomment to enter testing mode /* fork=no port=5064 log_stderror=yes fifo="/tmp/ser_fifox" */ debug=3 memlog=4 # memlog set high (>debug) -- no final time-consuming memory reports on exit mhomed=yes listen=192.168.0.223 alias="ser.hostname.com" check_via=yes dns=yes rev_dns=no children=16 # if changing fifo mode to a more restrictive value, put # decimal value in there, e.g. dec(rw|rw|rw)=dec(666)=438 fifo_mode=0666 loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/enum.so" loadmodule "/usr/local/lib/ser/modules/domain.so" #loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("usrloc|acc|auth_db|group|msilo", "db_url", "sql://ser:heslo@localhost/ser") # -- usrloc params -- /* 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #modparam("auth_db", "use_rpid", 1) modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@GW_IP_3>;party=calling;id-type=subscriber;screen=yes;privacy=off") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # report ACKs too for sake of completeness -- as we account PSTN # destinations which are RR, ACKs should show up modparam("acc", "report_ack", 1) modparam("acc", "log_level", 1) # if BYE fails (telephone is dead, record-routing broken, etc.), generate # a report nevertheless -- otherwise we would have no STOP event; => 1 modparam("acc", "failed_transactions", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) # Usage of flags is as follows: # 1 == should account(all to gateway), # 3 == should report on missed calls (transactions to iptel.org's users), # 4 == destination user wishes to use voicemail # 6 == nathelper # modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_missed_flag", 3) # report to syslog: From, i-uri, status, digest id, method modparam("acc", "log_fmt", "fisum") # -- tm params -- modparam("tm", "fr_timer", 20) modparam("tm", "fr_inv_timer", 90) modparam("tm", "wt_timer", 20) # -- msilo params modparam("msilo", "registrar", "sip:registrar@ser.fonomni.com") # -- enum params -- modparam("enum", "domain_suffix", "e164.arpa.") # -- multi-domain modparam("domain", "db_mode", 1) # NAT features turned off -- smartnat available only in nat-capable release # We will you flag 6 to mark NATed contacts modparam("registrar", "nat_flag", 6) # Enable NAT pinging modparam("nathelper", "natping_interval", 15) # Ping only contacts that are known to be behind NAT modparam("nathelper", "ping_nated_only", 1) # --------------------- request routing logic ------------------- route { if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483", "Alas Too Many Hops"); break; }; if (msg:len >= max_len) { sl_send_reply("513", "Message too large"); break; }; # special handling for natted clients; first, nat test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding used); also, # the received test should, if complete, should check all # vias for presence of received if (nat_uac_test("3")) { # allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || !search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed append_to_reply("P-NATed-Caller: Yes\r\n"); }; }; # anti-spam -- if somene claims to belong to our domain in From, # challenge him (skip REGISTERs -- we will chalenge them later) if (search("(From|F):.*@((192\.168\.0\.223)|(ser\.fonomni\.com))")) { # invites forwarded to other domains, like FWD may cause subsequent # request to come from there but have iptel in From -> verify # only INVITEs (ignore FIFO/UAC's requests, i.e. src_ip==fox) if ((method == "INVITE" || method == "SUBSCRIBE") && !((src_ip == 192.168.0.223) || ((src_ip == 192.168.0.2) || (src_ip == 192.168.0.2)))) { if (!(proxy_authorize("ser.fonomni.com", "subscriber"))) { proxy_challenge("ser.fonomni.com", "0"); break; }; # to maintain outside credibility of our proxy, we enforce # username in From to equal digest username; user with # "john.doe" id could advertise "bill.gates" in From otherwise; if (!check_from()) { log("LOG: From Cheating attempt in INVITE\n"); sl_send_reply("403", "That is ugly -- use From=id next time (OB)"); break; }; # we better don't consume credentials -- some requests may be # spiraled through our server (sfo@iptel->7141@iptel) and the # subsequent iteration may challenge too, for example because of # iptel claim in From; UACs then give up because they # already submitted credentials for the given realm #consume_credentials(); }; # non-REGISTER from other domain } else if ((method == "INVITE" || method == "SUBSCRIBE" || method=="REGISTER" ) && !(uri == myself || uri =~ "(@((192\.168\.0\.2)|(192\.168\.0\.2))([;:].*)*)")) { # and we serve our gateway too (we RR requests to it, so that # its address may show up in subsequent requests after loose_route sl_send_reply("403", "No relaying"); break; }; # By default we record route everything except REGISTERs if (!(method=="REGISTER")) record_route(); # if route forces us to forward to some explicit destination, do so # # loose_route returns true in case that a request included # route header fields instructing SER where to relay a request; # if that is the case, stop script processing and just forward there; # one could alternatively ignore the return value and treat the # request as if it was an outbound one; that would not work however # with broken UAs which strip RR parameters from Route. (What happens # is that with two RR /tcp2udp, spirals, etc./ and stripped parameters, # SER a) rewrites r-uri with RR1 b) matches uri==myself against RR1 # c) applies mistakenly user-lookup to RR1 in r-uri if (loose_route()) { # check if someone has not introduced a pre-loaded INVITE -- if so, # verify caller's privileges before accepting rr-ing if ((method=="INVITE" || method=="ACK" || method=="CANCEL") && uri =~ "(@((192\.168\.0\.2)|(192\.168\.0\.2))([;:].*)*)") { route(3); # Forward to PSTN gateway } else { append_hf("P-hint: rr-enforced\r\n"); # account all BYEs if (method=="BYE") setflag(1); route(1); # Generic forward }; break; }; # ------- check for requests targeted out of our domain... ------- if (!(uri == myself || uri =~ "(@((192\.168\.0\.2)|(192\.168\.0\.2))([;:].*)*)")) { # ... and we serve our gateway too (we RR requests to it, so that # its address may show up in subsequent requests after # rewriteFromRoute append_hf("P-hint: OUTBOUND\r\n"); route(1); break; }; # ------- now, the request is for sure for our domain ----------- # registers always MUST be authenticated to # avoid stealing incoming calls if (method == "REGISTER") { /* if (!allow_register("register.allow", "register.deny")) { log(1, "LOG: alert: Forbidden IP in Contact\n"); sl_send_reply("403", "Forbidden"); break; }; */ # prohibit attempts to grab someone else's To address # using valid credentials; if (!www_authorize("ser.fonomni.com", "subscriber")) { # challenge if none or invalid credentials www_challenge("ser.fonomni.com", "0"); break; }; if (!check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id in REGISTERs"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; m_dump(); break; }; # some UACs might be fooled by Contacts our UACs generate to make MSN # happy (web-im, e.g.) -- tell its urneachable if (uri =~ "sip:daemon@") { sl_send_reply("410", "Daemon is gone"); break; }; # aliases # note: through a temporary error in provisioning interface, there # are now aliases 905xx ... they take precedence overy any PSTN numbers # as they are resolved first lookup("aliases"); # check again, if it is still for our domain after aliases if (!(uri == myself || uri =~ "(@((192\.168\.0\.2)|(192\.168\.0\.2))([;:].*)*)")) { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); route(1); break; }; # Remove leading + if it is a number begining with + if (uri =~ "^[a-zA-Z]+:\+[0-9]+@") { strip(1); prefix("00"); }; if (!does_uri_exist()) { # Try numeric destinations through the gateway if (uri =~ "^[a-zA-Z]+:[0-9]+@") { route(3); } else { sl_send_reply("604", "Does Not Exist Anywhere"); }; break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-URI", "voicemail")) { setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to UsrLoc to bypass our authorization logic if (uri =~ "(@((192\.168\.0\.2)|(192\.168\.0\.2))([;:].*)*)") { log(1, "LOG: Weird! Gateway address in UsrLoc!\n"); route(3); break; }; # if user is on-line and is in voicemail group, enable redirection /* no voicemail currently activated if (method == "INVITE" && isflagset(4)) { t_on_failure("1"); }; */ # ... and also report on missed calls ... note that reporting # on missed calls is mutually exclusive with silent C timer setflag(3); # we now know we may, we know where, let it go out now! append_hf("P-hint: USRLOC\r\n"); route(1); } # # Forcing media relay if necesarry # route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")) { sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(6)) { if (!is_present_hf("P-RTP-Proxy")) { force_rtp_proxy(); append_hf("P-RTP-Proxy: YES\r\n"); }; append_hf("P-NATed-Calee: Yes\r\n"); }; # nat processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # natted at the moment of request processing); look at replies t_on_reply("1"); if (!t_relay()) { sl_reply_error(); break; }; } onreply_route[1] { # natted transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; # keep Cisco gatweay sending keep-alives if (isflagset(7) && status=~"2[0-9][0-9]") { remove_hf("Session-Expires"); append_hf("Session-Expires: 60;refresher=UAC\r\n"); fix_nated_sdp("1"); }; } # # logic for calls to the PSTN # route[3] { # discard non-PSTN methods if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "OPTIONS" || method == "BYE")) { sl_send_reply("500", "only VoIP methods accepted for GW"); break; }; # turn accounting on setflag(1); # continue with requests to PSTN gateway ... # no authentication needed if the destination is on our free-pstn # list or if the caller is the digest-less gateway # # apply ACLs only to INVITEs -- we don't need to protect other # requests, as they don't imply charges; also it could cause troubles # when a call comes in via PSTN and goes to a party that can't # authenticate (voicemail, other domain) -- BYEs would fail then if (method == "INVITE") { if (!is_user_in("Request-URI", "free-pstn")) { if (!proxy_authorize("ser.fonomni.com", "subscriber")) { proxy_challenge("ser.fonomni.com", "0"); break; }; # let's check from=id ... avoids accounting confusion if (!check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; } else { # Allow free-pstn destinations without any checks route(5); break; }; if (uri =~ "^sip:00[1-9][0-9]+@") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "International numbers not allowed"); break; }; route(5); } else { sl_send_reply("403", "Invalid Number"); break; }; }; # authorized PSTN break; } route[5] { rewritehostport("192.168.0.2:5060"); consume_credentials(); append_hf("P-Hint: GATEWAY\r\n"); # Try alternative gateway on failure t_on_failure("7"); # Our PSTN gateway is symmetric and can handle direction=active flag # properly, therefore we don't have to use RTP proxy t_relay(); } failure_route[7] { rewritehostport("192.168.0.2:5060"); append_branch(); t_relay(); } # ------------- handling of unavailable user ------------------ route[4] { # message store if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; if (t_newtran()) { if (method == "ACK") { log(1, "CAUTION: strange thing: ACK passed t_newtran\n"); break; }; t_reply("404", "Not Found"); }; # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted if (method == "INVITE") { acc_log_request("404 missed call\n"); acc_db_request("404 missed call", "missed_calls"); }; } ------------------------------------------------------------------------ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers