[Serusers] SER parallel forking
Daniel-Constantin Mierla
daniel at iptel.org
Thu May 20 12:51:46 CEST 2004
Hello,
you can add a permanent contact (you can use 'serctl ul add ...') for
each user to asterisk and SER will automatically fork.
.Daniel
On 5/20/2004 10:24 AM, Peter Gradwell wrote:
> 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
>
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