[Serusers] problem getting calls to my gateway

Hi everyone, I have been having lots of problems delivering calls to my companies gateway. I have tried several different configs found online however nothing seems to work. I am including the ser.cfg and a dump from ngrep in hopes a kind person will see what the problem is. I do not know very much about sip or ser so any help is greatly needed. Currently our company uses a cisco voip solution and I am setting up Ser as a test. Unfortunately our admin does not seem very helpful, I'm not sure if he has things setup correctly for my calls on the gateway or if it's my ser.cfg file. I was hoping from the information I'm sending someone can tell me where the problem looks like it's coming from, wether it's his gateway or my config file. I'm thinking it's me as I don't see any attempt of passing the call to the gateway in the ngrep output. Also, I have had to alter my ip's listed in this email. I have been warned under penalty of pain not to broadcast their ip addresses... :p I hope this does not cause a problem. ATA 64.189.165.206 Ser Box 64.189.165.205 Cisco GW 65.189.155.101 Thank you, # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line -dddddddddd) fork=yes log_stderror=no # (cmd line -E) #/* Uncomment these lines to enter debugging mode #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" # # $Id pstn.cfg,v 1.2 2003/06/03 031812 jiri Exp $ # # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/group.so" modparam("auth_db", "db_url","sql//secret@localhost/ser") modparam("usrloc", "db_url", "sql//secret@localhost/ser") # ----------------- setting module-specific parameters --------------- 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 (msglen >= max_len ) { sl_send_reply("513", "Message too big"); 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(urihost, uriport); }; 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[7][0-9][0-9][0-9]@.*" /* local PBX */ | uri=~"sip98[0-9][0-9][0-9][0-9]") { log("free call"); } else if (src_ip==65.189.155.101) { # 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=~"sip0[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=~"sip00[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=~"sip000[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("65.189.155.1015060"); # forward the request now if (!t_relay()) { sl_reply_error(); break; }; } ################ ngrep output####################### # U 64.189.165.2065060 -> 64.189.165.2055060 INVITE sip776044445556(a)64.189.165.205;user=phone SIP/2.0..Via SIP/2.0/UDP 64.189.165.2065060..From <sip6 044848235@64.189.165.205;user=phone>;tag=409936633..To <sip776044445556@64.189.165.205;user=phone>..Call-ID 2945885252@64.189.165.206..CSeq 1 INVITE..Contact &lt;sip6044445555(a)64.189.165.2065060;user=phone;transpor t=udp>..User-Agent Cisco ATA 186 v2.16.2 ata18x (030909a)..Expires 300..Content-Length 257..Content-Typ e application/sdp....v=0..o=6044445555 62848 62848 IN IP4 64.189.165.206..s=ATA186 Call..c=IN IP4 64.189.165.206..t=0 0..m=audio 16384 RTP/AVP 18 8 0 101..a=rtpmap18 G729/8000/1..a=rtpmap8 PCMA/8000/1..a=rtpmap0PCMU/8000/1..a=rtpmap101 telephone-event/8000..a=fmtp101 0-15.. # U 64.189.165.2055060 -> 64.189.165.2065060 SIP/2.0 407 Proxy Authentication Required..Via SIP/2.0/UDP 64.189.165.2065060..From <sip6044445555@64.189.165.205;user=phone>;tag=409936633..To <sip776044445556@64.189.165.205;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0ed0..Call-ID 2945885252@64.189.165.206..CSeq 1 INVITE..Proxy-Authenticate Digest realm="gateway", nonce="3fcf790810cb0daaf030be719aa79e574b96b535"..Server Sip EXpress router (0.8.12 (i386/linux)). .Content-Length 0..Warning 392 64.189.165.2055060 "Noisy feedback tells pid=32407 req_src_ip=64.189.165.206 req_src_port=5060 in_uri=sip776044445556(a)64.189.165.205;user=phone out_uri=sip776044445556(a)64.189.165.205;user=phone via_cnt==1".... # U 64.189.165.2065060 -> 64.189.165.2055060 ACK sip776044445556(a)64.189.165.205;user=phone SIP/2.0..Via SIP/2.0/UDP 64.189.165.2065060..From <sip6044 445555@64.189.165.205;user=phone>;tag=409936633..To &lt;sip776044445556(a)64.189.165.205;user=phone&gt;;tag=b27e1a1 d33761e85846fc98f5f3a7e58.0ed0..Call-ID 2945885252@64.189.165.206..CSeq 1 ACK..User-Agent Cisco ATA 186 v2.16.2 ata18x (030909a)..Content-Length 0....