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@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