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 :-) ============================================================================================
# ----------- 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(); }; } _________________________________________________________________ Chat to your friends for free on selected mobiles http://clk.atdmt.com/UKM/go/174426567/direct/01/ _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users