Hello My SER implementation includes Asterisk voicemail for unavailable users, Radius Accounting, Digest Authentication and PSTN gateway forwarding, everything works very well but now I am trying NatHelper/rtpproxy for nated endpoints, nated clients are registering with public IP but I cannot hear incoming audio in nated X-lite clients even if I use Port Forwarding or enable DMZ in NAT device (LinkSys), I am very confused because I can hear audio from Asterisk... maybe I have some problem in ser, please take a look to my configuration file and send me some advice, thanks
rafael
PS: myser.cfg based on http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/nathelper.cfg?r...
/usr/local/etc/ser/Ser_VM_RadAcc_NatHelp-Test1.cfg
# Version: We are using: Sip EXpress router (0.8.99-dev1 (i386/linux)) (Agosto 2004) # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper
# ----------- global configuration parameters ------------------------
#/* Uncomment these lines to enter debugging mode debug=9 fork=yes log_stderror=yes #*/
listen=100.110.*.* listen=127.0.0.1 port=5060
# hostname matching an alias will satisfy the condition uri==myself". alias=my.domain.com.pe alias=100.110.*.*
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/ser_fifo"
# sip_warning - Should replies include extensive warnings? # By default yes, it is good for trouble-shooting. sip_warning=yes
# ------------------ module loading ---------------------------------- 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/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/textops.so"
# digest authentication loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
modparam("usrloc", "db_mode", 2)
# storing passwords in our database in plain text: # modparam("auth_db", "calculate_ha1", yes) # modparam("auth_db", "password_column", "password")
# For Rad Accounting modparam("acc", "radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc
modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 30 ) modparam("tm", "wt_timer", 20 )
# add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
modparam("group", "db_url", "mysql://ser:heslo@localhost/ser") # "mysql" in cvs head version # modparam("uri", "db_url", "sql://ser:heslo@localhost/ser") # "sql" in ser0814 modparam("uri_db", "db_url", "mysql://ser:heslo@localhost/ser") # "mysql" in cvs head version
# ------------- registration parameters modparam("registrar", "nat_flag", 6) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 86400) modparam("registrar", "default_expires", 3600) modparam("registrar", "desc_time_order", 1) modparam("registrar", "append_branches", 1)
# !! Nathelper # modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# -------------------------- request routing logic --------------------------
route {
log(1, "-------------------------------------------\n"); log(1, "entering main loop\n");
# 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"); break; }; if ( msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; };
# !! 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 (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 }; };
# 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 (!method=="REGISTER") record_route();
# 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"); # t_relay(); ### use If don't want to enforce RTP proxy route(1); ### Nathelper!! break; };
# set Flag for Radius Accounting:
if (method=="INVITE") { log(1, "INVITE MESSAGE RECEIVED - START ACC\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ };
if (method=="BYE") { log (1, "BYE - STOP ACCOUNTING\n"); setflag(1); };
if (method=="CANCEL") { log (1, "CANCEL - STOP ACCOUNTING\n"); setflag(1); };
setflag(3); # Set Radius Missed Flag (radius_missed_flag param...)
if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # t_relay(); route(1); break; };
if (uri==myself) {
if (method == "REGISTER") { log(1, "ANALYZING REGISTER REQUEST\n"); # to use digest authentication if (!www_authorize("my.domain.com.pe", "subscriber")) { www_challenge("my.domain.com.pe", "0"); break; }; if (!save("location")) { sl_reply_error(); }; break; };
/* ***************** very insecure Dial out to PSTN logic ****************** */ ### Pendiente agregar seguridad a esta etapa, usar Digest-Auth o "credentials" ### ver http://www.voip-info.org/wiki-SER+example+pstn
# forward n digit requests to gateway AS5350 if(uri=~"^sip:9"){ log(1,"n digit expression match - Celulares Lima"); rewritehostport("100.110.*.*:5060"); route(2); break; };
# forward international calls to Asterisk (using Oh323 module to connect with H323 GWs) if(uri=~"^sip:00"){ rewritehostport("100.110.*.*:5060"); log(1,"n digit expression match - LDI"); route(2); break; };
/* ******************************************************************** */
lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); # t_relay(); route(1); break; };
# does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc
if (is_user_in("Request-URI", "voicemail")) { log(1, "requested user is in voicemail group"); setflag(4); };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log(1,"unable to locate user"); # handle user which was not found route(4); break; };
}; # End of "if(uri==myself)"
append_hf("P-hint: usrloc applied\r\n"); route(1);
# if user is on-line and is in Voicemail group, enable redirection if (method == "INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); };
# t_relay(); }
route[1] { # !! Nathelper if (uri=~"[@:](192.168.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; };
# 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(); }; }
# ----------------- SIP-to-PSTN call routed -------------------
route[2]{ log(1,"route[2]:SIP-to-GW call routed"); if(!t_relay()){ sl_reply_error(); }; }
# --------------- Handling of Unavailable user ---------------- route[4] {
# non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE")) { sl_send_reply("404", "Not Found"); acc_rad_request("404 Not Found"); break; };
# not voicemail subscriber if (!isflagset(4)) { sl_send_reply("404", "Not Found and no voicemail turned on"); acc_rad_request("404 Not Found"); break; };
### Forward to * voicemail adding prefix "vm" to simplify * "extension.conf" to this: ### exten => _vmXXXXXXX,1,Voicemail(u${EXTEN:2}) ### exten => _vmXXXXXXX,2,Hangup
prefix("vm"); rewritehostport("100.110.**.**:5060"); t_relay_to_udp("100.110.**.**","5060"); }
# if forwarding downstream did not succeed, try voicemail running at Asterisk
failure_route[1]{ if (t_check_status("485")){ revert_uri (); prefix("vm"); rewritehostport ("100.110.**.**:5060"); append_branch(); t_relay(); break; } }