[Serusers] SER parallel forking
Peter Gradwell
peter at gradwell.com
Thu May 20 10:24:14 CEST 2004
Hi,
We're integrating SER and Asterisk and want to use parallel forking
so that calls for sip users to simultaneously call the sip phone and
ser<user>@asterisk.gradwell.net, which will then wait a per-user delay
before answering with the voicemail.
The call does divert, but only after SER's own timer expires. This
might work ok as a last resort, but really we want the parallel forking
as described above to work so that we can have per-user delays, it may
also make supporting call diverts easier, since Asterisk is good at
stuff like that.
As far as I can tell the main logic for doing the branching (using
append_branch) is very similar to SER's example config file onr.cfg in
the distribution's examples directory. However I didn't have much luck
with that either! Our config file is included below.
Any thoughts on how to make this work would be most welcome!
many thanks
peter
=====================================================================
# # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 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=9
fork=no
log_stderror=yes
check_via=no # (cmd. line: -v)
dns=yes # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=193.111.200.106
#mhomed=1
port=5060
children=4
fifo="/tmp/ser_fifo"
alias="ser.gradwell.net"
alias="193.111.200.106"
#fifo_db_url="mysql://ser:********@hostingdb/ser"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
#loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
#loadmodule "/usr/local/lib/ser/modules/enum.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/xlog.so"
# Use a private ENUM space [not yet, we just pretend to be the real thing
#modparam("enum","domain_suffix","enum.go-sip.org.")
# We force all the lookup and registrar related stuff to take the domain
# into account.
# Point everything at sip-auth-adm for DB related stuff
modparam("usrloc", "use_domain", 1)
modparam("registrar", "use_domain", 1)
modparam("group", "use_domain", 1)
modparam("auth_db", "use_rpid", 1)
modparam("auth_db", "rpid_column", "username")
# We do not use persistant storage, this reduces DB overhead,
# however, if we move to a HA pair, then this should be set to 1
# and the seed for generating nonce values must be synchronised.
# NOTE: Actually we HAVE to use 1 anyway as our aliases table is
# in SQL.
modparam("usrloc", "db_mode", 2)
modparam("usrloc","db_url", "mysql://ser:********@hostingdb/ser")
modparam("domain","db_url", "mysql://ser:********@hostingdb/ser")
modparam("domain","db_mode",1)
modparam("group","db_url", "mysql://ser:********@hostingdb/ser")
#modparam("group","db_mode",1)
modparam("auth_db","db_url", "mysql://ser:********@hostingdb/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
#modparam("uri","db_url", "mysql://ser:********@hostingdb/ser")
#modparam("uri","use_uri_table", yes)
modparam("acc","db_url", "mysql://ser:********@hostingdb/ser")
modparam("msilo","db_url","mysql://ser:********@hostingdb/ser")
#modparam("msilo","registrar","sip:registrar at go-sip.com")
modparam("tm", "fr_inv_timer", 15 )
modparam("tm", "fr_timer", 10 )
#modparam("tm", "wt_timer", 2 )
# Useful for some badly behaved clients
modparam("rr", "enable_full_lr", 1)
# Set accounting flags, these are the defaults anyway
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
# Nathelpher
#modparam("nathelper", "natping_interval", 10)
# Which flags mean what...
# 1 - account
# 2 - missed call
# 3 - url reqires enum rewrite
# 4 - user has voicemail accessuser has voicemail access
# 5 - user is online
# 6 - inbout call rtp stream should be proxied:
# 7 - outbound call rtp stream should be proxied:
# 8 - set up voicemail in route[3]
# ------------------------- request routing logic -------------------
# main routing logic
route{
xdbg("*****\n");
xdbg("***** %rm %ru\n");
xdbg("*****\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) {
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();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
if(method=="BYE"){
setflag(1);
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 ( is_uri_host_local() || uri == myself ) {
# We want to deal primarily with numbers rather
# than usernames, this makes life easier in
# voicemail.
#
# Translate any usernames to numbers
if (method=="REGISTER") {
# Leaving the domain blank means that the
# domain from the url will be used.
if (!www_authorize("","subscriber")) {
www_challenge("", "1");
break;
};
save("location");
break;
};
log("**** lookup(aliases)\n");
lookup("aliases");
# We don't deal with presence at the moment
if (method=="SUBSCRIBE" || method == "PUBLISH") {
sl_send_reply("503", "Service Unavailable");
break;
};
# Rewriting for other SIP networks...
if(uri=~"^sip:\*\*.*"){
# We only want our customer relaying through our servers
strip(2);
if (!proxy_authorize("","subscriber")) {
proxy_challenge("", "1");
break;
};
sl_send_reply("404", "Not Found");
break;
};
if(uri=~"^sip:[0-9][0-9][0-9]@.*" ){
t_relay_to_udp("asterisk.gradwell.net","5060");
break;
};
if (uri=~"^sip:0.*") {
if (uri=~"^sip:0.*") {
if (uri=~"^sip:00.*") {
strip(2);
}else{
strip(1);
prefix("44");
};
};
route(6);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (method == "INVITE" || method == "ACK" || method == "MESSAGE") {
if (uri=~"^sip:\*") {
t_relay_to_udp("asterisk.gradwell.net","5060");
break;
};
# Request uri is now invalid or in Username form
if (lookup("location")) {
log("*** found in usrloc\n");
if(method=="MESSAGE"){
# Remote agent may not accept messages, we juststore them.
t_on_failure("1");
t_relay();
break;
};
# they are online
setflag(5);
} else {
if(method=="MESSAGE"){
# Remote agent may not accept messages, we juststore them.
if (m_store("1")) {
t_reply("202", "Accepted");
}else{
t_reply("503", "Service Unavailable");
};
break;
};
};
if (!isflagset(5)) {
log("**** not found in usrloc, diverting to vm\n");
revert_uri();
lookup("aliases");
# User not registered with either username or extension
# Instant Unavailable voicemail
acc_db_request("Unavailable - Offline", "missed_calls");
log("**** lookup(aliases)\n");
lookup("aliases");
prefix("ser");
route(2);
break;
};
setflag(8);
route(3);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
route(3);
}
# Our voicemail route.
route[2]{
append_hf("P-hint: in-route-2\r\n");
log("IDESK: Route 2, Forwarding to Voicemail\n");
xdbg("IDESK: method=%rm, r_uri=%ru, cseq=%cs\n");
rewritehost("asterisk.gradwell.net");
rewriteport("5060");
if (!t_relay()) {
log("Forwarding to Voicemail FAILED\n");
sl_reply_error();
break;
};
break;
}
# Stateful relaying with NAT if it is needed.
# NOTE: One possibel enhancement here is to do the rtp-proxying via another box
# this would move the traffic away from the ser server completely.
# Nat for outbound from idesk
route[3]{
log("IDESK: Route for fixing up outbound\n");
if (isflagset(8)) {
# We know where they are, and they have
# voicemail access, so we fork to their
# voicemail account.
append_branch();
revert_uri();
log("**** setting up vm branch\n");
log("**** lookup(aliases)\n");
lookup("aliases");
prefix("ser");
append_hf("P-hint: known-vm \r\n");
rewritehost("asterisk.gradwell.net");
rewriteport("5060");
};
if (method == "INVITE"){
if (isflagset(6)) {
log("IDESK: Outbound RTP Proxying \n");
log("failure 1, reply 1\n");
t_on_failure("1");
#t_on_reply("1");
} ;
if (isflagset(7)){
log("IDESK: Inbound RTP Proxying \n");
log("failure 1, reply 2\n");
t_on_failure("1");
#t_on_reply("2");
} ;
if (isflagset(5) && !isflagset(6) && !isflagset(7)){
log("failure 1, reply 3\n");
t_on_failure("1");
#t_on_reply("3");
};
};
if (!t_relay()) {
sl_reply_error();
break;
};
}
####
####onreply_route[1]{
#### # Our rtp proxying
#### log("IDESK: Reply Route for fixing up Outboudn nat\n");
#### if(status=~"4[0-9][0-9].*"){
#### setflag(2);
#### };
#### if(status=~"200.*" && search("application/sdp")){
#### if(src_ip=="192.168.254.27"){
#### route(5);
#### };
#### # If it comes back through this route then it needs natting
#### if (search("application/sdp")){
#### force_rtp_proxy_from("192.168.254.26");
#### };
#### };
####}
####
####onreply_route[2]{
#### # Our rtp proxying
#### log("IDESK: Reply Route for fixing up Inbound nat\n");
#### if(status=~"4[0-9][0-9].*"){
#### setflag(2);
#### };
#### if(status=~"200.*" && search("application/sdp")){
#### if(src_ip=="194.130.117.27"){
#### route(5);
#### };
#### # If it comes back through this route then it needs natting
#### if (search("application/sdp")){
#### force_rtp_proxy_from("194.130.117.26");
#### };
#### };
####}
onreply_route[3]{
# Our rtp proxying
log("IDESK: Reply Route for fixing up Inbound nat\n");
if(status=~"4[0-9][0-9].*"){
setflag(2);
};
if(status=~"200.*" && search("application/sdp")){
if(isflagset(5) && ( src_ip == "192.168.254.27" || src_ip
== "194.130.117.27")){
route(5);
};
};
}
failure_route[1]{
if(method=="MESSAGE"){
# Remote agent may not accept messages, we juststore them.
if (m_store("1")) {
t_reply("202", "Accepted");
}else{
t_reply("503", "Service Unavailable");
};
break;
};
append_hf("P-hint: missed \r\n");
setflag(2);
append_hf("P-hint: in-reply-route-1\r\n");
log("IDESK: Failure Route 1\n");
}
# Account as missed
route[5] {
log("IDESK: Route 3\n");
append_hf("P-hint: in-route-3\r\n");
append_hf("P-hint: missed \r\n");
# setflag(2);
acc_db_request("No Answer", "missed_calls");
}
# This is our route out to the PSTN;
route[6]{
append_hf("P-hint: in-route-4-pstn\r\n");
# This really is going to be a PSTN call, so we
# Need to check the credentials
# Leaving the domain blank means that the
# domain from the url will be used.
if (uri=~"^sip:44.*") {
strip(2);
prefix("0");
};
if(method=="INVITE"){
# sip.calluk.com will authenticate...
#if (!proxy_authorize("","subscriber")) {
# proxy_challenge("", "1");
# break;
#};
append_rpid_hf();
log("Forwarding to PSTN\n");
setflag(1);
rewritehost("sip.calluk.com");
if(!t_relay_to_udp("sip.calluk.com","5060")){
sl_reply_error();
};
break;
};
if(method=="CANCEL" || method=="ACK" || method=="BYE"){
setflag(1);
if(!t_relay_to_udp("sip.calluk.com","5060")){
sl_reply_error();
};
break;
}
}
=====================================================================
thanks
peter
--
peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/
-- engineering & hosting services for email, web and voip --
-- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --
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