[OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS & Asterisk

[Ross Beer]

Hi,
 
I am using the following config with Opensips and having a problem with one way audio. When connecting the softphone directly to asterisk that runs on the same machine audio passes without any problems. Firewalls are all open and Zoiper/Snom phones connect without issue.
 
If anyone could offer some advice it would be much appreciated as I'm tearing my hear out :-)
 
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# ----------- global configuration parameters ------------------------
debug=3         # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode 
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no           # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
listen=udp:<IP ADDRESS>:5060
children=4
# ------------------ module loading ----------------------------------
#set module path
mpath="/usr/local/lib64/opensips/modules/"
# Uncomment this if you want to use SQL database
loadmodule "db_mysql.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "signaling.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "mediaproxy.so"
loadmodule "xlog.so"
loadmodule "dialog.so"
loadmodule "load_balancer.so"
loadmodule "mi_datagram.so"
# Uncomment this if you want digest authentication
# db_mysql.so must be loaded !
#loadmodule "auth.so"
#loadmodule "auth_db.so"
# !! Nathelper
loadmodule "nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- mi_fifo params --
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_datagram", "socket_name", "/tmp/opensips.sock")
modparam("mi_datagram", "unix_socket_mode", 0600)
modparam("mi_datagram", "socket_timeout", 2000)
# -- 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)
# -- 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")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# !! Nathelper
modparam("usrloc","nat_bflag",6)
modparam("nathelper","sipping_bflag",8)
modparam("nathelper", "ping_nated_only", 1)   # Ping only clients behind NAT
# -- MediaProxy --
modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock")
modparam("mediaproxy", "mediaproxy_timeout", 500)
modparam("dialog", "dlg_flag", 13)
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_url", "<DB URL>")
modparam("load_balancer", "db_url","<DB URL>")
# -------------------------  request routing logic -------------------
# main routing logic
route{
 # 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");
  exit;
 };
 if (msg:len>=  2048 ) {
  sl_send_reply("513", "Message too big");
  exit;
 };
 # !! Nathelper
 # Special handling for NATed clients; first, NAT test is
 # executed: it looks for via!=received and RFC1918 addresses
 # in Contact (may fail if line-folding is used); also,
 # the received test should, if completed, should check all
 # vias for rpesence of received
 if (nat_uac_test("3")) 
 {
  # Allow RR-ed requests, as these may indicate that
  # a NAT-enabled proxy takes care of it; unless it is
  # a REGISTER
  if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
   log("LOG:Someone trying to register from private IP, rewriting\n");
   # This will work only for user agents that support symmetric
   # communication. We tested quite many of them and majority is
   # smart enough to be symmetric. In some phones it takes a 
   # configuration option. With Cisco 7960, it is called 
   # NAT_Enable=Yes, with kphone it is called "symmetric media" and 
   # "symmetric signalling".
   # Rewrite contact with source IP of signalling
   fix_nated_contact();
   if ( is_method("INVITE") ) {
    fix_nated_sdp("1"); # Add direction=active to SDP
   };
   force_rport(); # Add rport parameter to topmost Via
   setbflag(6);    # Mark as NATed
   # if you want sip nat pinging
   # setbflag(8);
  };
 };
 # subsequent messages withing a dialog should take the
 # path determined by record-routing
 if (loose_route()) {
  # mark routing logic in request
  append_hf("P-hint: rr-enforced\r\n"); 
  route(1);
  exit;
 };
 # 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
 if (!is_method("REGISTER"))
  record_route();
 if (!uri==myself) {
  # mark routing logic in request
  append_hf("P-hint: outbound\r\n"); 
  route(1);
  exit;
 };
 # 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 (is_method("REGISTER")) 
  {
   # Uncomment this if you want to use digest authentication
   #if (!www_authorize("siphub.org", "subscriber")) {
   # www_challenge("siphub.org", "0");
   # return;
   #};
   save("location");
   exit;
  };
  # detect resources and do balancing
         #load_balance("1","pstn");
  rewritehostport("<ASTERISK IP ADDDRESS>:5061");
              
     # LB function returns negative if no suitable destination (for requested resources) is found,
         # or if all destinations are full
         #if ($retcode<0) 
  #{
             # sl_send_reply("500","Service full");
              # exit;
         #}
  xlog("Selected destination is: $du\n");
 };
 append_hf("P-hint: usrloc applied\r\n"); 
 route(1);
}
route[1] 
{
 # !! Nathelper
 if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){
  sl_send_reply("479", "We don't forward to private IP addresses");
  exit;
 };
 # if client or server know to be behind a NAT, enable relay
 if (isbflagset(6)) 
 {
  use_media_proxy();
 };
 # NAT processing of replies; apply to all transactions (for example,
 # re-INVITEs from public to private UA are hard to identify as
 # NATed at the moment of request processing); look at replies
 t_on_reply("1");
 # send it out now; use stateful forwarding as it works reliably
 # even for UDP2TCP
 if (!t_relay()) {
  sl_reply_error();
 };
}
# !! Nathelper
onreply_route[1] {
 # NATed transaction ?
 if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") 
 {
  fix_nated_contact();
  use_media_proxy();
 # otherwise, is it a transaction behind a NAT and we did not
 # know at time of request processing ? (RFC1918 contacts)
 }
 else if (nat_uac_test("1")) 
 {
  fix_nated_contact();
 };
}


  		 	   		  
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