[OpenSIPS-Users] Fwd: opensips server dual network card configuration

kevinfang kevinfang.ah at gmail.com
Mon Jul 27 15:47:54 CEST 2015


HI,
I have a opensips servers in a private network, IP address: 10.34.14.24,
now I'm going to this private network address 1: 1 NAT to public networks,
and modify the opensips.cfg profile. Register now through the public
network clients can conduct video and audio communications, and work very
well.

Now I need to add a network card (IP: 192.168.100.100) on opensips server,
and directly connected to a sip client (IP: 192.168.100.200) through this
card.

How should I modify opensips.cfg, make public sip client the private
network sip client to communicate it?

Attach opensips.cfg I now use:



#
# $Id$
#
# OpenSIPS residential configuration script
#     by OpenSIPS Solutions <team at opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
#   the "Residential" scenario.
# You can enable / disable more features / functionalities by
#   re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* comment the next line to enable the auto discovery of local aliases
   based on revers DNS on IPs */
auto_aliases=no

advertised_address=61.132.137.100
alias=61.132.137.100

listen=udp:10.34.14.24:5060  # CUSTOMIZE ME
#listen=udp:10.34.240.150:5060

disable_tcp=yes

disable_tls=yes


####### Modules Section ########

#set module path
mpath="/usr/local/opensips/lib64/opensips/modules"

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timer", 5)
modparam("tm", "fr_inv_timer", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)






#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", 10)
modparam("usrloc", "db_mode",   0)

#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", 7)

/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", 3)
/* account triggers (flags) */
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)

### Authenticated by MySQL ###
loadmodule "db_mysql.so"
loadmodule "auth.so"
loadmodule "auth_db.so"

modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", "mysql://opensips:opensipsrw@localhost
/opensips")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url", "mysql://opensips:opensipsrw@localhost
/opensips")
modparam("auth_db", "load_credentials", "")
### End Authentivated by MySQL ###

loadmodule "domain.so"
modparam("domain", "db_url","mysql://opensips:opensipsrw@localhost
/opensips")
modparam("domain", "db_mode", 1)   # Use caching

loadmodule "nat_traversal.so"
modparam("nat_traversal", "keepalive_interval", 90)

#### nathelper module
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "sipping_from", "sip:pinger at opensips.org")
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")

#### Modulo rtpproxy (forcar o audio atraves do opensips)
loadmodule "rtpproxy.so"
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7890")
modparam("nathelper", "force_socket", "udp:127.0.0.1:7890")
modparam("rtpproxy", "rtpproxy_autobridge", 1)


####### Routing Logic ########

# main request routing logic

route{

if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}

if (has_totag()) {
# sequential requests within a dialog should
# take the path determined by record-routing
if (loose_route()) {
 if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}


# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {
 if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
}
        #t_on_failure("1");
sl_send_reply("404","Not here");
}
exit;
}

# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}

t_check_trans();

if ( !(is_method("REGISTER")  ) ) {
 if (from_uri==myself)
 {

          ### Authenticated by MySQL ###
                      if(!proxy_authorize("", "subscriber")) {
                         if (!registered("location","$fu")) {
         xlog("L_NOTICE","Auth_error for $fU@$fd from $si cause proxy
authentication required");
   }
                  proxy_challenge("", "0");
                   exit;
                    }
                    if(!db_check_from()) {
         xlog("L_NOTICE","Auth_error for $fU@$fd from $si cause Forbidden
auth ID");
                   sl_send_reply("403", "Forbidden auth ID");
                         exit;
                    }

                consume_credentials();
                      # caller authenticated

             ### End Autheticated by MySQL ###
 } else {
# if caller is not local, then called number must be local
 if (!uri==myself) {
xlog("L_NOTICE","Auth_error for $fU@$fd from $si cause Rely forbidden");
send_reply("403","Rely forbidden");
exit;
}
}

}

# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}

# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();

# account only INVITEs
if (is_method("INVITE")) {
  sl_send_reply("180","ringing");

  #SIP trunk to PSTN
      # if(((uri=~"^sip:[055199999]@*") || (uri=~"^sip:[1]@*")) &&
!lookup("location","m")) {
  if(((uri=~"^sip:[055199999]@*") ) ) {
             xlog("sip trunk");
     rewritehostport("10.34.240.131:5060"); # outbound
             route(1);
       }

setflag(1); # do accounting
}

 if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
}

# requests for my domain
 if (is_method("PUBLISH|SUBSCRIBE"))
{
sl_send_reply("503", "Service Unavailable");
exit;
}

if (is_method("REGISTER"))
{
  ### Authenticated by MySQL ###
    if (!www_authorize("", "subscriber")) {
        www_challenge("", "0");
        exit;
    }
    if (!db_check_to()) {
        sl_send_reply("403", "Forbidden auth ID");
        exit;
    }
    ### End Authenticate by MySQL ###

    # we receive a register request
    # we will execute fix_nated_register nad fix_nated_contact
    fix_nated_register();
    fix_nated_contact();

if (   0 ) setflag(7);

if (!save("location"))
    sl_reply_error();

exit;
}

if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}


# do lookup with method filtering
if (!lookup("location","m")) {
 # voicemail
# xlog("voicemail service\n");
# rewritehostport("10.34.240.129:5080");

}

 # when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}


route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {

t_on_branch("2");
t_on_reply("2");
t_on_failure("1");

    if(nat_uac_test("127")){
      # user identified as behing a nat
      xlog("we are on route relay, user behind nat");
      fix_nated_contact();
    }

    # if we have an application/sdp on our body, so we execute
    # the rtpproxy_offer
    if(has_body("application/sdp")){
       xlog("we have sdp on this $rm");
       rtpproxy_offer("c","61.132.137.100");
    }

}

  # removing the rtpproxy session
  if(is_method("CANCEL|BYE")){
      unforce_rtp_proxy();
  }


if (!t_relay()) {
send_reply("500","Internal Error");
};
exit;
}


branch_route[2] {
xlog("new branch at $ru\n");
}


onreply_route[2] {
 xlog("incoming reply\n");

        # we receive a reply, we need to check about application/sdp
        # on our body, if we have, we answer that
        if(is_method("ACK") && has_body("application/sdp")){
                rtpproxy_answer();
        }else if(has_body("application/sdp")){
                # offering rtpproxy on a non ack message
                rtpproxy_offer("c","61.132.137.100");
        }


        # here we try to identify if the user is behind a nat again
        # but now is the second user (the called user)
        if(nat_uac_test("127")){

               xlog("we are on nat handle , user behind nat, fixing
contact");
               fix_nated_contact();
        }

}


failure_route[1] {

        if(is_method("INVITE")) {
             # call failed - relay to voice mail
     #append_branch();
     #t_relay("udp:10.34.240.131:5080");
        }

if (t_was_cancelled()) {
exit;
}

}
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