Hi,
I am trying to set up openser with mediaproxy (at xx.xx.xx.133) to
route calls from UA behind NAT to asterisk as voicemail (at
xx.xx.xx.134) and PSTN gateways (at xx.xx.xx.32)
I can see in the SDP payload that the RTP is being sent from asterisk
to mediaproxy, but in sessions.py it shows the private ip not the
public one and I have one way audio.
Can anybody help please.
opnser.cfg below.
Marnus
--
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
log_facility=LOG_LOCAL6
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/openser_fifo"
# ------------------ module loading ----------------------------------
loadmodule "/usr/local/lib/openser/modules/mysql.so"
loadmodule "/usr/local/lib/openser/modules/sl.so"
loadmodule "/usr/local/lib/openser/modules/tm.so"
loadmodule "/usr/local/lib/openser/modules/rr.so"
loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
loadmodule "/usr/local/lib/openser/modules/usrloc.so"
loadmodule "/usr/local/lib/openser/modules/registrar.so"
loadmodule "/usr/local/lib/openser/modules/textops.so"
loadmodule "/usr/local/lib/openser/modules/uri_db.so"
loadmodule "/usr/local/lib/openser/modules/domain.so"
loadmodule "/usr/local/lib/openser/modules/mediaproxy.so"
loadmodule "/usr/local/lib/openser/modules/nathelper.so"
# Logging
loadmodule "/usr/local/lib/openser/modules/xlog.so"
loadmodule "/usr/local/lib/openser/modules/auth.so"
loadmodule "/usr/local/lib/openser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
#tm timeout for voicemail params
modparam("tm", "fr_timer", 3)
modparam("tm", "fr_inv_timer", 35)
modparam("tm", "noisy_ctimer", 1)
# parms for NAT/mediaproxy
modparam("nathelper", "rtpproxy_disable", 1)
modparam("nathelper", "natping_interval", 0)
modparam("mediaproxy", "natping_interval", 30)
modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock")
modparam("mediaproxy", "sip_asymmetrics",
"/usr/local/etc/openser/sip-clients")
modparam("mediaproxy", "rtp_asymmetrics",
"/usr/local/etc/openser/rtp-clients")
modparam("registrar", "nat_flag", 6)
# ------------------------- request routing logic -------------------
# main routing logic
route {
#check for old messages: could mean a problem with the DNS
entries or some other loop-causer...
if (!mf_process_maxfwd_header("10"))
{
xlog("L_WARN", "WARNING: Too many hops\n");
sl_send_reply("483", "Too many hops, forward count
exceeded limit\n");
return;
};
#check for extremely large messages; we don't need a sip dos
attack
if (msg:len >= 2048)
{
xlog("L_WARN", "WARNING: Message too large, &>= 2048
bytes\n");
sl_send_reply("513", "Message too large, exceeded
limit\n");
return;
};
# Track what is happening
xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n");
#record everything besides registers and acks
if(method!="REGISTER" && method!="ACK")
{
setflag(1);
};
# Record Route Section
if (method=="INVITE" && client_nat_test("3"))
{
record_route_preset("xx.xx.xx.133:5060;nat=yes");
}
else if (method!="REGISTER")
{
record_route();
}
# Call tear down section
if (method=="BYE" || method=="CANCEL")
{
end_media_session();
}
#do not send to voicemail if BYE or CANCEL
#is used to end call before user pickup or timeout
if(method=="CANCEL" || method=="BYE")
{
setflag(10);
};
#grant route if route headers already present
if (loose_route())
{
# May need client_nat_test & use_media_proxy here...
route(1);
return;
};
#Always require authentication, which could result in a PSTN
if (method=="REGISTER")
{
if (!search("^Contact:[ ]*\*") && client_nat_test("7"))
{
setflag(6);
fix_nated_register();
force_rport();
};
if(!www_authorize("domain.tld", "subscriber"))
{
www_challenge("domain.tld", "0");
return;
}
else
{
if (!check_to())
{
sl_send_reply("401", "Unauthorized");
return;
};
#Save into user database, used below when
checking if user is available
xlog("L_INFO", "REGISTER: User $fu
Authenticated Correctly\n");
save("location");
return;
};
};
if (method=="INVITE")
{
if (client_nat_test("3"))
{
setflag(7);
force_rport();
fix_nated_contact();
};
if(uri=~"sip:[EMAIL PROTECTED]")
{
#authorize if a call is going to VM
if(!proxy_authorize("domain.tld", "subscriber"))
{
proxy_challenge("domain.tld", "0");
return;
};
xlog("L_INFO", "CALL: Call from $fu to check
voicemail\n");
rewritehostport("vm.domain.tld:5060");
}
else
{
if (does_uri_exist())
{
#Call is to sip client, so do nothing
but route
xlog("L_INFO", "CALL: Sip client\n");
if (!lookup("location"))
{
sl_send_reply("404", "Not
Found");
xlog("L_ERROR", "ERROR: User
$tu Not Found\n");
return;
};
}
else
{
#authorize if a call is going to PSTN
if(!proxy_authorize("domain.tld",
"subscriber"))
{
proxy_challenge("domain.tld",
"0");
return;
};
#Call destination is PSTN, so send it
to the gateway
xlog("L_INFO", "CALL: PSTN $tu from $fu
\n");
rewritehostport("ast1.domain.tld:5060");
};
};
#Make sure that all subsequent requests go through us;
#done at the top already
#record_route();
}
else
{
if (does_uri_exist())
{
#Call is to sip client, so do nothing but route
xlog("L_INFO", "CALL: Sip client\n");
if (!lookup("location"))
{
sl_send_reply("404", "Not Found");
xlog("L_ERROR", "ERROR: User $tu Not
Found\n");
return;
};
}
else
{
#Call destination is PSTN, so send it to the
gateway
xlog("L_INFO", "CALL: PSTN $tu from $fu \n");
rewritehostport("ast1.domain.tld:5060");
};
#record_route();
};
#ALL PROCESSING IS DONE, SO ROUTE
route(4);
route(1);
}
route[1]
{
#send the call outward
if(method=="INVITE" && !isflagset(10))
{
t_on_failure("2"); # voicemail if failure
};
if (!t_relay())
{
xlog("L_WARN", "ERROR: t_relay failed");
sl_reply_error();
};
}
# -----------------------------------------------------------------
# NAT Traversal Section
# -----------------------------------------------------------------
route[4]
{
if (isflagset(6) || isflagset(7))
{
if (!isflagset(8))
{
setflag(8);
use_media_proxy();
};
};
}
failure_route[2]
{
if(!t_was_cancelled() && !t_check_status("407"))
{
revert_uri();
rewritehostport("vm.domain.tld:5060");
append_branch();
#PREVENT SOME CRAZY VOICEMAIL LOOP
xlog("L_INFO", "INFO: CALL TO VOICEMAIL");
setflag(10);
route(1);
}
}
onreply_route[1]
{
if ((isflagset(6) || isflagset(7)) &&
(status=~"(180)|(183)|2[0-9][0-9]"))
{
if (!search("^Content-Length:[ ]*0"))
{
use_media_proxy();
};
};
if (client_nat_test("1"))
{
fix_nated_contact();
};
}
