I don't seem to be able to get a match when I send a call from my gw to ser registered user. Debug indicates that user Not found in usrloc even though the database shows the users registered. I am able to make out bound calls the gw just fine. If anyone can shed some light in it it would be great. Thanks ============================================================================ ================================= mysql> select * from location; +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ | username | domain | contact | received | expires | q | callid | cseq | last_modified | replicate | state | flags | user_agent | +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ | 9901 | | sip:9901@69.250.205.31 | NULL | 2005-02-26 03:31:09 | -1.00 | e5201b3819008718(a)192.168.199.102 | 107 | 20050226033016 | 0 | 0 | 0 | Grandstream BT100 1.0.5.11 | +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ 1 row in set (0.01 sec) ====debugt output====================================================================== ========================= 20(18672) parse_headers: flags=256 20(18672) find_first_route: No Route headers found 20(18672) loose_route: There is no Route HF 20(18672) lookup(): '9901' Not found in usrloc ============================================================================ ================================= U 69.19.88.16:5060 -> 69.19.88.19:5060 INVITE sip:9901@69.19.88.19 SIP/2.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:4103532264 @69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118dc212 ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS , BYE, REFER..Content-Type: application/sdp..Content-Length: 240....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s=session..c=IN IP4 66. 159.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-16..a=silenceSupp:off - - - -.. # U 69.19.88.19:5060 -> 69.19.88.16:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" < sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Call-ID: 745bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.0 (i386/linux))..Content-Length: 0..Warning: 392 69.19.88.19:5060 "Noisy feedback tells: p id=18682 req_src_ip=69.19.88.16 req_src_port=5060 in_uri=sip:9901@69.19.88.19 out_uri=sip:9901@69.19.88.16:5060 via_cnt==1".... # U 69.19.88.19:5060 -> 69.19.88.16:5060 INVITE sip:9901@69.19.88.16:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:69.19.88.19;ftag=as149fbe93;lr=on>..Via: SIP/2.0/UDP 66 .159.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:410353 2264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118d c212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPT IONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 240..P-hint: GATEWAY....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s= session..c=IN IP4 69.19.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone -event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -.. # U 69.19.88.16:5060 -> 69.19.88.19:5060 SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 69.19.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4b K2fb13d17..From: "Cell Phone MD" <sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>;tag=as149fbe93..Call-ID: 74 5bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER ..Contact: <sip:4103532264@69.19.88.16>..Content-Length: 0.... ==ser.cfg=================================================================== ======================================== # # $Id: serconf.sh,v 1.3 2003/11/03 13:03:58 janakj Exp $ # # autogenerated SER configuration # # user: uid=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wheel) # system: Linux baltodb.popdial.com 2.6.10-1.760_FC3 #1 Wed Feb 2 00:14:23 EST 2005 i686 i686 i386 GNU/Linux # date: Sat Feb 26 02:57:50 EST 2005 # # ----------- global configuration parameters ------------------------ /* debug=4 fork=yes port=5060 log_stderror=no memlog=5 */ mhomed=yes fifo="/tmp/ser_fifo" alias=baltodb.popdial.com # uncomment to override config values for test # /* debug=4 # debug level (cmd line: -ddd) fork=yes port=5060 log_stderror=yes # (cmd line: -E) fifo="/tmp/ser_fifox" # */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=16 # if changing fifo mode to a more restrictive value, put # decimal value in there, e.g. dec(rw|rw|rw)=dec(666)=438 #fifo_mode=438 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/mysql.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/uri.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/enum.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- # all DB urls here #modparam("usrloc|acc|auth_db|group|msilo|uri", "db_url","mysql://ser:heslo@localhost/ser") modparam("usrloc|acc|auth_db|group|msilo|uri_db", "db_url","mysql://ser:heslo@localhost/ser") # -- usrloc params -- /* 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "user_column", "user_id") modparam("auth_db", "password_column", "password") modparam("auth", "nonce_expire", 300) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # that is the flag for which we will account -- don't forget to modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 3 ) # -- tm params -- modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 90 ) modparam("tm", "wt_timer", 20 ) # -- msilo params modparam("msilo", "registrar", "sip:registrar@baltodb.popdial.com") # -- enum params -- # modparam("enum", "domain_suffix", "e164.arpa.") # ------------------------- 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","Alas Too Many Hops"); break; }; #if (len_gt( max_len )) { if ( msg:len > max_len ) { sl_send_reply("513", "Message too large sorry"); break; }; # Make sure that requests dont advertise addresses # from private IP space (RFC1918) in Contact HF # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)&quot;)) { # 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:")) && !( src_ip==192.168.0.0/16 || src_ip==10.0.0.0/8 || src_ip==172.16.0.0/12 )) { log("LOG: Someone trying to register from private IP again\n"); sl_send_reply("479", "We dont accept private IP contacts" ); break; }; }; # anti-spam -- if somene claims to belong to our domain in From, # challenge him (skip REGISTERs -- we will chalenge them later) if (search("(From|F):.*(baltodb\.popdial\.com|66\.159\.88\.19)")) { # invites forwarded to other domains, like FWD may cause subsequent # request to come from there but have iptel in From -> verify # only INVITEs (ignore FIFO/UAC's requests, i.e. src_ip==myself) if (method=="INVITE" & !(src_ip==69.19.88.19)) { if (!(proxy_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */ ))) { proxy_challenge("baltodb.popdial.com" /* realm */, "0" /* no-qop */); break; }; # to maintain outside credibility of our proxy, we enforce # username in From to equal digest username; user with # "john.doe" id could advertise "bill.gates" in From otherwise; if (!check_from()) { log("LOG: From Cheating attempt in INVITE\n"); sl_send_reply("403", "That is ugly -- use From=id next time (OB)"); break; }; # we better don't consume credentials -- some requests may be # spiraled through our server (sfo@iptel->7141@iptel) and the # subsequent iteration may challenge too, for example because of # iptel claim in From; UACs then give up because they # already submitted credentials for the given realm #consume_credentials(); }; # INVITEs claiming to come from our domain } else if (method=="INVITE" && !(uri=~"[@:\.](baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" # ... and we serve our gateway too if present | uri=~"@66\.159\.88\.16([;:].*)*" )) { #the INVITE neither claims to come from our domain nor is it targeted to it # -> junk it sl_send_reply("403", "No relaying"); break; }; /* ********* RR ********************************** */ # to be safe, record route everything; UAs may use different # transport protocols and need to have SER in path record_route(); # if route forces us to forward to some explicit destination, # do so; check however first that a cheater didn't preload # a gateway destination to bypass PSTN ACLs if (loose_route()) { if (uri=~"sip:[+0-9]+@66\.159\.88\.16") { # it is gateway -- proceed to ACLs route(3); break; }; # route HF determined next hop; forward there append_hf("P-hint: rr-enforced\r\n"); t_relay(); break; }; /* ********* check for requests targeted out of our domain... ******* */ # sign of our domain: there is '@' (username) or : (nothing) in # front of our domain name ; ('.' is not there -- we handle all # xxx.iptel.org as outbound hosts);if none of these cases matches, # proceed with processing of outbound requests in route[2] if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) { route(2); break; }; /* ************ requests for our domain ********** */ /* now, the request is for sure for our domain */ # registers always MUST be authenticated to # avoid stealing incoming calls if (method=="REGISTER") { # Make sure that user's dont register infinite loops # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(baltodb\.popdial\.com|66\.159\.88\.19)&quot;)) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; if (search("^(Contact|m): .*66\.159\.88\.16")) { log(1, "LOG: alert: protected contacts\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; if (!www_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */ )) { # challenge if none or invalid credentials www_challenge( "baltodb.popdial.com" /* realm */, "0" /* no qop -- some phones can't deal with it */); break; }; # prohibit attempts to grab someone else's To address # using valid credentials; if (!check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id in REGISTERs"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; m_dump(); break; }; # some UACs might be fooled by Contacts our UACs generate to make MSN # happy (web-im, e.g.) -- tell its urneachable if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon is gone"); break; }; # is this an ENUM destination (leading +?)? give it a try, if the lookup # doesn't change URI, just continue if (uri=~"sip:\+[0-9]+@") { if (!enum_query("voice")) { # if parameter empty, it defaults to "e2u+sip" enum_query(""); # E2U+sip }; } else { # aliases (take precedences over PSTN number; provisioning interface # is set up to assinge aliases beginning with 8) lookup("aliases"); }; # check again, if it is still for our domain after aliases are resolved if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) { route(5); break; }; # now check if it's about PSTN destinations through our gateway; # note that 8.... is exempted for numerical non-gw destinations if (uri=~"sip:\+?[0-79][0-9]*@.*") { route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found ... route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to UsrLoc to bypass our authorization logic if (uri=~"sip:[+0-9]+@66\.159\.88\.16") { # it is gateway -- proceed to ACLs route(3); break; }; /* ... and also report on missed calls ... */ setflag(3); # we now know we may, we know where, let it go out now! append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------------------- OUTBOUND ---------------------------------------- # routing logic for outbound requests targeted out of our domain # (keep in mind messages to our users can end up here too: for example, # an INVITE may be UsrLoc-ed, then the other party uses outbound # proxy with r-uri=the usr_loced addredd (typically IP)) route[2] { append_hf("P-hint: OUTBOUND\r\n"); t_relay(); } #------- ALIASED OUTBOUND -------------------------------------------- # routing logic for inbound requests aliased outbound; unlike # with real outbound requests we do not force authentication # as these calls are server by our server and we do not want # to disqualify unathenticated request originatiors from other # domains route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); t_relay(); } #----------------- PSTN ---------------------------------------------- # logic for calls to the PSTN route[3] { # turn accounting on setflag(1); /* require all who call PSTN to be members of the "int" group; apply ACLs only to INVITEs -- we don't need to protect other requests, as they don't imply charges; also it could cause troubles when a call comes in via PSTN and goes to a party that can't authenticate (voicemail, other domain) -- BYEs would fail then; exempt Cisco gateway from authentication by IP address -- it does not support digest */ if (method=="INVITE" && (!src_ip==69.19.88.16)) { if (!proxy_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "baltodb.popdial.com" /* realm */, "0" /* no qop */ ); break; }; # let's check from=id ... avoids accounting confusion if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; if(!is_user_in("credentials", "int")) { sl_send_reply("403", "NO PSTN Privileges..."); break; }; consume_credentials(); }; # INVITE to authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("69.19.88.16:5060"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("69.19.88.16","5060"); } /* *********** handling of unavailable user ******************* */ route[4] { /**/ # message store if (method=="MESSAGE") { t_newtran(); if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); } else { t_reply("503", "Service Unavailable"); }; break; }; /**/ # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # voicemail subscribers ... t_newtran(); t_reply("404", "Not Found"); # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted if (method=="INVITE") { acc_db_request("404 missed call", "missed_calls"); }; }