[Serusers] problem with module acc
Marian Danisek
majo at sunteq.sk
Mon Oct 18 10:28:44 CEST 2004
hello,
i have ser version 0.8.14. Everything working fine, but i want to use
accounting with mysql, so i recompile ser a and acc module with
DEFS+=-DSQL_ACC
now ser crash everytime a try to start it, following messagess apears in
log :
Oct 18 10:19:51 mdk10 ser: Listening on
Oct 18 10:19:51 mdk10 ser: 127.0.0.1 [127.0.0.1]:5060
Oct 18 10:19:51 mdk10 ser: 192.168.1.250 [192.168.1.250]:5060
Oct 18 10:19:51 mdk10 ser: Aliases: mdk10:5060 localhost:5060
sunteq.sk:* mdk10.sunteq.sk:*
Oct 18 10:19:51 mdk10 ser: ser startup succeeded
Oct 18 10:19:51 mdk10 /sbin/ser[9609]: connect_db(): Too many connections
Oct 18 10:19:51 mdk10 /sbin/ser[9609]: db_init(): Error while trying to
connect database
Oct 18 10:19:51 mdk10 /sbin/ser[9609]: group:init_child(): Unable to
connect database
Oct 18 10:19:51 mdk10 /sbin/ser[9609]: init_mod_child(): Error while
initializing module group
Oct 18 10:19:51 mdk10 /sbin/ser[9609]: init_children failed
anybody could help, please ?
below is my ser config file
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
alias=mdk10.sunteq.sk
alias=sunteq.sk
#alias=atlas.sunteq.sk
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/lib/ser/modules/mysql.so"
loadmodule "/lib/ser/modules/sl.so"
loadmodule "/lib/ser/modules/tm.so"
loadmodule "/lib/ser/modules/rr.so"
loadmodule "/lib/ser/modules/maxfwd.so"
loadmodule "/lib/ser/modules/usrloc.so"
loadmodule "/lib/ser/modules/registrar.so"
loadmodule "/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/lib/ser/modules/auth.so"
loadmodule "/lib/ser/modules/auth_db.so"
# load the voicemail module
#loadmodule "/lib/ser/modules/vm.so"
# load the enum module
loadmodule "/lib/ser/modules/enum.so"
# load the group module, to verify if a user forwards to voicemail
loadmodule "/lib/ser/modules/group.so"
# load the nathelper module
loadmodule "/lib/ser/modules/nathelper.so"
loadmodule "/lib/ser/modules/acc.so"
# ----------------- setting module-specific parameters ---------------
# -- registrar parameter
# special NAT flag indicates that a registered client is behind NAT
modparam("registrar", "nat_flag", 6)
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser")
modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:heslo@localhost/ser")
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
#modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- voicemail params --
#modparam("voicemail", "db_url","mysql://ser:heslo@localhost/ser")
# -- voicemail params --
#modparam("group", "db_url","mysql://serro:heslo@localhost/ser")
# -- nathelper params --
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)
modparam("tm", "fr_inv_timer", 30 )
#modparam("tm", "fr_inv_timer", 8 )
# ------------------------- request routing logic -------------------
# main routing logic
route{
log(1, "-------------------------------------------\n");
log(1, "entering main loop\n");
if (nat_uac_test("2")) {
log(1, "src address different than via header->NAT
detected\n");
log(1, "force_rport and fix_nated_contact and
setflag(5)\n");
#try NAT traversal, works only if the client is symmetrical
force_rport();
fix_nated_contact();
append_hf("P-hint: fixed NAT contact for request\r\n");
# flag 5 indicates that incoming request is from NATed
client
setflag(5);
};
if (method=="REGISTER")
log(1, "REGISTER message received\n");
if (method=="INVITE")
log(1, "INVITE message received\n");
if (method=="ACK")
log(1, "ACK message received\n");
if (method=="BYE")
log(1, "BYE message received\n");
if (method=="CANCEL")
log(1, "CANCEL message received\n");
if (method=="SUBSCRIBE")
log(1, "SUBSCRIBE message received\n");
if (method=="NOTIFY")
log(1, "NOTIFY message received\n");
if (method=="OPTIONS")
log(1, "OPTIONS message received\n");
if (method=="INFO")
log(1, "INFO message received\n");
if (method=="MESSAGE")
log(1, "MESSAGE message received\n");
if (method=="REFER")
log(1, "REFER message received\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");
break;
};
if (msg:len > max_len) {
#if (len_gt( max_len )) {
sl_send_reply("513", "Message too big");
break;
};
# loose-route processing
if (loose_route()) {
log(1, "loose_route processing\n");
t_relay();
break;
};
# Check for PSTN access
if (uri=~"^sip:0[0-9]*@.*") {
route(3);
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 (method=="REGISTER") {
log(1, "analyzing REGISTER request\n");
# Uncomment this if you want to use digest authentication
if (!www_authorize("mdk10.sunteq.sk",
"subscriber")) {
www_challenge("mdk10.sunteq.sk", "0");
break;
};
if (isflagset(5)) {
#register from nated client, save
nat_flag=6
#in location table
setflag(6);
};
if (!save("location")) {
log(1, "save location error\n");
sl_reply_error();
};
break;
};
lookup("aliases");
#mark transaction for voicemail
if (is_user_in("Request-URI", "voicemail\n")) {
log(1, "requested user is in voicemail group");
setflag(4);
};
# Process Aliases
lookup("aliases");
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# handle user which was not found
log(1, "requested user not found\n");
route(4);
break;
};
};
#add failure route which should be performed if response code >=300
if (method=="INVITE" && isflagset(4)) {
log(1, "invite for voicemail user->initiate
failureroute[1]\n");
t_on_failure("1");
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
route(1);
}
route[1]{
log(1, "-------------------------------------------\n");
log(1, "entering route[1] - relaying SIP message\n");
if ((isflagset(5)) || (isflagset(6))) {
log(1, "at least one of the participants is
NATed->record_route\n");
record_route();
log(1, " -->setting up reply processing
->onreply_route[1]");
t_on_reply("1");
if (method=="INVITE") {
log(1, " INVITE request-->force_rtp_proxy,
set NATED-INVITE flag(7)");
force_rtp_proxy();
append_hf("P-hint: request forced to rtp
proxy\r\n");
setflag(7);
};
};
log(1, "relaying message ...\n");
if (!t_relay()) {
log(1, "t_relay error occured\n");
sl_reply_error();
};
}
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
log(1, "-------------------------------------------\n");
log(1, "onreply_route[1] entered\n");
if (isflagset(6)) {
log(1, "transaction was sent to a NATED client -> fix
nated contact\n");
fix_nated_contact();
append_hf("P-hint: fixed NAT contact for response\r\n");
}
if ( (status=~"100") ) {
log(1, "status 100 received\n");
};
if ( (status=~"180") ) {
log(1, "status 180 received\n");
};
if ( (status=~"202") ) {
log(1, "status 202 received\n");
};
if ( (status=~"200" || status=~"183") ) {
log(1, "status 2xx or 183");
if ( isflagset(7) ) {
log(1, "marked(7) as NATED-INVITE ->
force_rtp_proxy \n");
force_rtp_proxy();
append_hf("P-hint: response forced to rtp
proxy\r\n");
};
};
}
route[3] {
if (method=="INVITE" && (!src_ip==192.168.1.253)) {
if (!proxy_authorize( "mdk10.sunteq.sk","subscriber")) {
proxy_challenge( "mdk10.sunteq.sk", "0");
break;
};
# let's check from=id ... avoids accounting confusion
if(!is_user_in("credentials", "local")) {
sl_send_reply("403", "NO PSTN Privileges...");
break;
};
consume_credentials();
}; # INVITE to authorized PSTN
# if you have passed through all the checks, let your call go to GW!
force_rtp_proxy();
record_route();
t_on_reply("1");
# snom conditioner
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP ephemeral)
t_relay_to_udp("192.168.1.253","5060");
}
route[4]{
log(1, "-------------------------------------------\n");
log(1, "entering route[4] = requested user not online\n");
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL" || method == "REFER" || method == "BYE")) {
log(1, "no invite,ack,cancel,refer->return 404\n");
sl_send_reply("404", "Not Found");
break;
};
# not voicemail subscriber and no echo/conference call
if ( isflagset(4)) {
log(1, "flag(4) active\n");
};
if (uri =~ "conference") {
log(1, "conference call\n");
};
if (uri =~ "echo") {
log(1, "echo call\n");
};
if ( !( isflagset(4) || (uri =~ "conference") || (uri =~
"echo") ) ) {
log(1, "no voicemail subscriber->return 404");
sl_send_reply("404", "Not Found and no voicemail turned
on");
break;
};
if ( isflagset(5) ) {
log(1, "caller is NATed->record_route\n");
record_route();
log(1, " -->setting up reply processing
->onreply_route[1]");
t_on_reply("1");
if (method=="INVITE") {
log(1, " INVITE request-->force_rtp_proxy");
force_rtp_proxy();
};
};
# forward to voicemail now
rewritehostport("192.168.1.253:5060");
log(1, "forward to voicemail\n");
t_relay_to_udp("192.168.1.253", "5060");
}
failure_route[1] {
/* XX: note: unsafe if preloaded routes without username used */
log(1, "-------------------------------------------\n");
log(1, "failureroute[1] entered\");
revert_uri();
rewritehostport("212.17.35.184:5060");
append_branch();
t_relay_to_udp("212.17.35.184", "5060");
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