[Serusers] Forward to Quintum GW - One way audio
Pavel Siderov
pi at hostmates.com
Tue Nov 30 17:58:57 CET 2004
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
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