[Serusers] Forward to Quintum GW - One way audio

Hi guys, I have a strange problem when tying to forward call to Quintum gw. Callee is hearing what I speak but I don't hear anything. Forwarding to another SER is ok. Both - users with real ips and natted. Somebody can help me? Here is my config file: debug=10 # debug level (cmd line: -dddddddddd) #fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" #fork=no # ------------------ 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/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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.15/ser") modparam("usrloc","user_column","username") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) #modparam("acc", "db_url", "mysql://ser:heslo@192.168.2.15/ser") #modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 3) #modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "db_url", "mysql://ser:heslo@192.168.2.15/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_missed_flag", 1) #modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 1) #modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") #modparam("mediaproxy", "natping_interval", 20) modparam("nathelper","rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) listen=193.2.6.17 # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("70")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (nat_uac_test("3")) { if (method == "REGISTER" || ! search("^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". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("193.2.6.17", "subscriber")) { www_challenge("193.2.6.17", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (uri=~"^sip:[1-9]*@193.2.6.17") { # QUINTUM GW rewritehost("194.24.1.6"); append_branch("194.24.1.6"); # t_relay_to_udp("194.24.1.6", "5060"); t_relay(); rewritehostport("194.24.1.6:5060"); append_hf("P-hint: VoipSwitch GATEWAY"); break; } } else { if (uri=~"^sip:[0]*@193.2.6.17") { ### Other SER setflag(1); rewritehostport("bla.mydomain.com:5060"); append_branch("bla.mydomain.com"); t_relay(); break; }; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; # }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_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 (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_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(); }; } Thanks In Advance! Pavel Siderov