Dear sir: I am using a Cisco 3660 router as a SIP-PSTN gateway with my SER server.When I have configure all the thing has the guideline,I meet a problem.If I use a telephone call my MSN Messager or use MSN Messager call the telephone,the receiver could ring,and the MSN Messager could send and receive signal,but the telephone could only receive.In other words,the RTP packages could not send to UA at computers from the router.Does anyone can help me? This is the configure at SER server: # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/acc/acc.so" loadmodule "modules/rr/rr.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/mysql/mysql.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" loadmodule "modules/group/group.so" loadmodule "modules/uri/uri.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db", "db_url","sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) modparam("acc", "log_flag", 1 ) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; /* ********* RR ********************************** */ /* grant Route routing if route headers present */ if (loose_route()) { t_relay(); break; }; /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") { record_route(); }; # now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, drop it -- # we cannot terminate it in PSTN; relay non-INVITE requests -- it may # be for example BYEs sent by gateway to call originator if (!uri=~"sip:\+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else { forward(uri:host, uri:port); }; break; }; # account completed transactions via syslog setflag(1); # free call destinations ... no authentication needed if ( is_user_in("Request-URI", "free-pstn") /* free destinations */ | uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */ | uri=~"sip:98[0-9][0-9][0-9][0-9]") { log("free call"); } else if (src_ip==192.168.0.10) { # our gateway doesn't support digest authentication; # verify that a request is coming from it by source # address log("gateway-originated request"); } else { # in all other cases, we need to check the request against # access control lists; first of all, verify request # originator's identity if (!proxy_authorize( "gateway" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "gateway" /* realm */, "0" /* no qop */ ); break; }; # authorize only for INVITEs -- RR/Contact may result in weird # things showing up in d-uri that would break our logic; our # major concern is INVITE which causes PSTN costs if (method=="INVITE") { # does the authenticated user have a permission for local # calls (destinations beginning with a single zero)? # (i.e., is he in the "local" group?) if (uri=~"sip:0[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local")) { sl_send_reply("403", "No permission for local calls"); break; }; # the same for long-distance (destinations begin with two zeros") } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", " no permission for LD "); break; }; # the same for international calls (three zeros) } else if (uri=~"sip:000[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "International permissions needed"); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "Forbidden"); break; }; }; # INVITE to authorized PSTN }; # authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("192.168.0.10:5060"); # forward the request now if (!t_relay()) { sl_reply_error(); break; }; } And this is the configure at the Cisco 3660: ! version 12.2 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname SIP ! enable secret 5 $1$Ov6R$SoCYqYuzY7a7spMUSEz2e. enable password 123456 ! ip subnet-zero no ip routing ! ! ip name-server 162.105.170.66 ip name-server 162.105.129.27 ! ! voice rtp send-recv ! voice service pots ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g729br8 codec preference 13 g711ulaw codec preference 14 g711alaw ! voice class codec 13 ! ! ! ! ! ! fax interface-type fax-mail mta receive maximum-recipients 0 ! ! ! ! interface FastEthernet0/0 ip address 192.168.0.200 255.255.255.0 no ip route-cache no ip mroute-cache speed auto half-duplex no cdp enable ! interface FastEthernet0/1 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial3/0 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface Serial3/1 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface Serial3/2 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface Serial3/3 no ip address no ip route-cache no ip mroute-cache shutdown serial restart_delay 0 no cdp enable ! interface FastEthernet5/0 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial5/0 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface FastEthernet5/1 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial5/1 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface FastEthernet6/0 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial6/0 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface FastEthernet6/1 no ip address no ip route-cache no ip mroute-cache shutdown duplex auto speed auto no cdp enable ! interface Serial6/1 no ip address no ip route-cache no ip mroute-cache shutdown clockrate 2000000 no cdp enable ! interface Dialer1 no ip address ! ip default-gateway 192.168.0.1 ip classless no ip http server ip pim bidir-enable ! ! vc-group 1 ! snmp-server community public RO ! call rsvp-sync ! voice-port 1/0/0 cptone CN description 3078 supervisory disconnect dualtone mid-call ! voice-port 1/0/1 ! voice-port 1/1/0 cptone CN description 3078 supervisory disconnect dualtone mid-call ! voice-port 1/1/1 ! voice-port 2/0/0 ! voice-port 2/0/1 ! voice-port 2/1/0 ! voice-port 2/1/1 ! ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice 86100 pots application session destination-pattern 8610T port 1/0/0 ! dial-peer voice 1000 voip application session destination-pattern .T session protocol sipv2 session target sip-server session transport udp codec g711ulaw ! dial-peer voice 100 pots destination-pattern 010T port 1/0/0 ! dial-peer voice 86000 pots destination-pattern 8600T port 1/1/0 ! sip-ua timers disconnect 200 sip-server dns:sipproxy.icst.pku.edu.cn ! ! line con 0 line aux 0 line vty 0 4 password 123456 login ! ! end