Bogdan-Andrei Iancu wrote: I had a SER setup using 2 instances one of them for voicemail and voice responses (using vm and SEMS). I upgraded things to OpenSER but am unable to have the voicemail working. P.S. I inherited the script for SER and it looks so ugly, what I need is a simple script which should forward to voicemail on user unavailable and play messages for user not found and other things. -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101

Bogdan-Andrei Iancu wrote: Attached the configuration I used, one (openser.cfg) used for the main SIP server and the second was used as a second instance which serve for voicemail. -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101 # # $Id: rules.m4,v 1.1 2004/01/13 19:42:56 janakj Exp $ # # (c) 2003 iptel.org # # Rules to process ser.cfg templates # ### m4 macros to make the configuration easier ### End of m4 macro section # # $Id: ser.cfg,v 1.2 2003/11/01 18:18:16 jiri Exp $ # # # # Set the following in your CISCO PSTN gateway: # sip-ua # nat symmetric role passive # nat symmetric check-media-src # fork=yes port=5060 #log_stderror=yes fifo="/tmp/openser_fifo" debug=3 memlog=4 # memlog set high (>debug) -- no final time-consuming memory reports on exit mhomed=yes listen=ip-addr-1 alias="domain.tld" check_via=yes dns=yes rev_dns=no 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=0666 loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/group.so" loadmodule "/usr/lib/openser/modules/msilo.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/enum.so" loadmodule "/usr/lib/openser/modules/domain.so" #loadmodule "/usr/lib/openser/modules/permissions.so" # --------------------- database fifo settings ----------------------- fifo_db_url="mysql://openser:openserrw@localhost/openser" modparam("usrloc|acc|auth_db|group|msilo", "db_url", "mysql://openser:openserrw@localhost/openser") # -- usrloc params -- /* 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("usrloc", "use_domain", 1) modparam("registrar", "use_domain", 1) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #modparam("auth_db", "use_rpid", 1) modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@ip-addr-3>;party=calling;id-type=subscriber;screen=yes;privacy=off") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # report ACKs too for sake of completeness -- as we account PSTN # destinations which are RR, ACKs should show up modparam("acc", "report_ack", 1) modparam("acc", "log_level", 1) # if BYE fails (telephone is dead, record-routing broken, etc.), generate # a report nevertheless -- otherwise we would have no STOP event; => 1 modparam("acc", "failed_transactions", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) # Usage of flags is as follows: # 1 == should account(all to gateway), # 3 == should report on missed calls (transactions to iptel.org's users), # 4 == destination user wishes to use voicemail # 6 == nathelper # modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_missed_flag", 3) # report to syslog: From, i-uri, status, digest id, method modparam("acc", "log_fmt", "fisumdpr") # -- tm params -- modparam("tm", "fr_timer", 15) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 30) # -- msilo params modparam("msilo", "registrar", "sip:registrar at domain.tld") # -- enum params -- modparam("enum", "domain_suffix", "e164.arpa.") # -- multi-domain modparam("domain", "db_mode", 1) # NAT features turned off -- smartnat available only in nat-capable release # We will you flag 6 to mark NATed contacts modparam("registrar", "nat_flag", 6) # Enable NAT pinging modparam("nathelper", "natping_interval", 15) # Ping only contacts that are known to be behind NAT modparam("nathelper", "ping_nated_only", 1) # --------------------- request routing logic ------------------- route { if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483", "Alas Too Many Hops"); break; }; if (msg:len >= max_len) { sl_send_reply("513", "Message too large"); break; }; #emad setflag(3); # special handling for natted clients; first, nat test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding used); also, # the received test should, if complete, should check all # vias for presence 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(1,"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 smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) 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 append_to_reply("P-NATed-Caller: Yes\r\n"); }; }; # 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):.*@((ip-addr-1)|(domain.tld))")) { # 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==fox) if ((method == "INVITE" || method == "SUBSCRIBE") && !((src_ip == ip-addr-1) || ((src_ip == ip-addr-2) || (src_ip == ip-addr-3)))) { if (!(proxy_authorize("domain.tld", "subscriber"))) { proxy_challenge("domain.tld", "0"); 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 at iptel->7141 at 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(); }; # non-REGISTER from other domain } else if ((method == "INVITE" || method == "SUBSCRIBE" || method=="REGISTER" ) && !(uri == myself || uri =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)")) { # and we serve our gateway too (we RR requests to it, so that # its address may show up in subsequent requests after loose_route sl_send_reply("403", "No relaying"); break; }; # By default we record route everything except REGISTERs if (!(method=="REGISTER")) record_route(); # if route forces us to forward to some explicit destination, do so # # loose_route returns true in case that a request included # route header fields instructing SER where to relay a request; # if that is the case, stop script processing and just forward there; # one could alternatively ignore the return value and treat the # request as if it was an outbound one; that would not work however # with broken UAs which strip RR parameters from Route. (What happens # is that with two RR /tcp2udp, spirals, etc./ and stripped parameters, # SER a) rewrites r-uri with RR1 b) matches uri==myself against RR1 # c) applies mistakenly user-lookup to RR1 in r-uri if (loose_route()) { # check if someone has not introduced a pre-loaded INVITE -- if so, # verify caller's privileges before accepting rr-ing if ((method=="INVITE" || method=="ACK" || method=="CANCEL") && uri =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)") { route(3); # Forward to PSTN gateway } else { append_hf("P-hint: rr-enforced\r\n"); # account all BYEs if (method=="BYE") {log(1,"loose setflag(1)\n"); setflag(1); } route(1); # Generic forward }; break; }; # ------- check for requests targeted out of our domain... ------- if (!(uri == myself || uri =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)")) { # ... and we serve our gateway too (we RR requests to it, so that # its address may show up in subsequent requests after # rewriteFromRoute append_hf("P-hint: OUTBOUND\r\n"); route(1); break; }; # ------- now, the request is for sure for our domain ----------- # registers always MUST be authenticated to # avoid stealing incoming calls if (method == "REGISTER") { /* if (!allow_register("register.allow", "register.deny")) { log(1, "LOG: alert: Forbidden IP in Contact\n"); sl_send_reply("403", "Forbidden"); break; }; */ # prohibit attempts to grab someone else's To address # using valid credentials; if (!www_authorize("domain.tld", "subscriber")) { # challenge if none or invalid credentials www_challenge("domain.tld", "0"); break; }; 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; }; # aliases # note: through a temporary error in provisioning interface, there # are now aliases 905xx ... they take precedence overy any PSTN numbers # as they are resolved first lookup("aliases"); # check again, if it is still for our domain after aliases if (!(uri == myself || uri =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)")) { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); route(1); break; }; # Remove leading + if it is a number begining with + if (uri =~ "^[a-zA-Z]+:\+[0-9]+@") { strip(1); prefix("00"); }; if (!does_uri_exist()) { # Try numeric destinations through the gateway if (uri =~ "^[a-zA-Z]+:[0-9]+@") { route(3); } else { sl_send_reply("604", "Does Not Exist Anywhere"); }; 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")) { setflag(4); }; # 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 =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)") { log(1, "LOG: Weird! Gateway address in UsrLoc!\n"); route(3); break; }; # if user is on-line and is in voicemail group, enable redirection # no voicemail currently activated if (method == "INVITE" && isflagset(4)) { t_on_failure("1"); }; # ... and also report on missed calls ... note that reporting # on missed calls is mutually exclusive with silent C timer setflag(3); log(1,"Set flag to 3\n"); # we now know we may, we know where, let it go out now! append_hf("P-hint: USRLOC\r\n"); route(1); } # # Forcing media relay if necesarry # route[1] { # if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")) { # sl_send_reply("479", "We don't forward to private IP addresses"); # break; # }; if (isflagset(6)) { if (!is_present_hf("P-RTP-Proxy")) { force_rtp_proxy(); append_hf("P-RTP-Proxy: YES\r\n"); }; append_hf("P-NATed-Calee: Yes\r\n"); }; # nat processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # natted at the moment of request processing); look at replies t_on_reply("1"); if (method=="BYE"||method=="INVITE") { log(1,"route[1] setflag(1)\n"); setflag(1); }; if (!t_relay()) { sl_reply_error(); break; }; } onreply_route[1] { # natted 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(); }; # keep Cisco gatweay sending keep-alives if (isflagset(7) && status=~"2[0-9][0-9]") { remove_hf("Session-Expires"); append_hf("Session-Expires: 60;refresher=UAC\r\n"); fix_nated_sdp("1"); }; # if (method=="BYE") { if (method=="BYE"||method=="INVITE") { log(1,"On Reply[1] setflag\n"); setflag(1); }; } # # logic for calls to the PSTN # route[3] { #log(1,"In route [3]\n"); # discard non-PSTN methods if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "OPTIONS" || method == "BYE")) { sl_send_reply("500", "only VoIP methods accepted for GW"); break; }; # turn accounting on # if (method=="BYE") { setflag(1); # }; # continue with requests to PSTN gateway ... # no authentication needed if the destination is on our free-pstn # list or if the caller is the digest-less gateway # # 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 if (method == "INVITE") { if (!(is_user_in("Request-URI", "free-pstn") || ((src_ip == ip-addr-2) || (src_ip == ip-addr-3)))) { if (!proxy_authorize("domain.tld", "subscriber")) { proxy_challenge("domain.tld", "0"); break; }; # let's check from=id ... avoids accounting confusion if (!check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; if (uri=~"sip:02[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local")) { rewriteuri("sip:9999@ip-addr-1:5090"); t_relay_to_udp("ip-addr-1", "5090"); # sl_send_reply("403", "No permission for local calls"); break; }; rewritehostport("ip-addr-3:5060"); #t_on_failure("7"); # the same for long-distance (destinations begin with two zeros") } else if (uri=~"sip:0[1-9][0-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { rewriteuri("sip:9999@ip-addr-1:5090"); t_relay_to_udp("ip-addr-1", "5090"); # sl_send_reply("403", " no permission for LD "); break; }; rewritehostport("ip-addr-3:5060"); # t_on_failure("7"); # the same for international calls (three zeros) } else if (uri=~"sip:001[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { rewriteuri("sip:9999@ip-addr-1:5090"); t_relay_to_udp("ip-addr-1", "5090"); # sl_send_reply("403", "International permissions needed"); break; }; strip(2); rewritehostport("ip-addr-5:5060"); # everything else (e.g., interplanetary calls) is denied } else if (uri=~"sip:0044[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { rewriteuri("sip:9999@ip-addr-1:5090"); t_relay_to_udp("ip-addr-1", "5090"); # sl_send_reply("403", "International permissions needed"); break; }; strip(2); rewritehostport("ip-addr-6:5060"); # everything else (e.g., interplanetary calls) is denied }else { rewriteuri("sip:9998@ip-addr-1:5090"); t_relay_to_udp("ip-addr-1", "5090"); # sl_send_reply("403", "Forbidden"); break; }; consume_credentials(); append_hf("P-Hint: GATEWAY\r\n"); log(1,"TRELAY IN ROUTE[3]\n"); t_relay(); }; # authorized PSTN break; } failure_route[7] { rewritehostport("ip-addr-4:5060"); append_branch(); t_relay(); } # ------------- handling of unavailable user ------------------ route[4] { # message store if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; 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; }; if (method == "INVITE") { acc_log_request("404 missed call\n"); acc_db_request("404 missed call", "missed_calls"); }; # if (t_newtran()) { # if (method == "ACK") { # log(1, "CAUTION: strange thing: ACK passed t_newtran\n"); # break; # }; if (!isflagset(4)) { # route(6); rewriteuri("sip:9998@ip-addr-1:5090"); t_relay_to_udp("ip-addr-1", "5090"); # sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; # forward to voicemail now rewritehostport("ip-addr-1:5090"); t_relay_to_udp("ip-addr-1", "5090"); # t_reply("404", "Not Found"); # }; # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted } #route[5]{ # log(1,"route 5555555555555555555555555555\n"); # if (status=~"408") { # rewritehostport("ip-addr-1:5090"); # append_branch(); # t_relay_to_udp("ip-addr-1", "5090"); # }; #} # if forwarding downstream did not succeed, try voicemail running # at bat.iptel.org:5090 failure_route[1] { # if (method == "INVITE") { # acc_log_request("404 missed call\n"); # acc_db_request("404 missed call", "missed_calls"); # }; log(1,"FAILURE ROUTE 1\n"); if (t_check_status("408|486")) { log(1,"408 or 486\n"); revert_uri(); rewritehostport("ip-addr-1:5090"); append_branch(); t_relay_to_udp("ip-addr-1", "5090"); } } # # $Id: voicemail.cfg,v 1.5 2004/01/14 18:23:50 rco Exp $ # # this script is configured for use as voicemail UAS; it processes # INVITEs and BYEs and asks SEMS to record media via "vm"; in this # script, all record-routing and other constructs known from proxy # scripts are not present -- it is a simple UAS # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) #listen=212.103.160.198 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5090 children=4 fifo="/tmp/vm_ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- #modparam("vm", "db_url","mysql://ser:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==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; }; if (!uri==myself) { sl_send_reply("404", "not reponsible for host in r-uri"); break; }; # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ if (uri=~"sip:999[0-9]@.*") { if (!vm("/tmp/am_fifo", "announcement")) { log(1,"couldn't contact announcement server\n"); t_reply("500", "couldn not contact announcement server"); }; log("!!!!!!!!!!!!!!!!!!!!!ANNOUNCEMENT!!!!!!!!!!!!!!!!!!!!!!\n"); } else if(uri=~"sip:1234567890@.*") { if (!vm("/tmp/am_fifo", "number_reader")) { log("could not contact the number reader server\n"); t_reply("500","could not contact ivr server"); }; log ("!!!!!!!!!!!!!!!!!!!!!!!!!IVR!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!\n"); } else if (uri=~"sip:55[0-9][0-9][0-9]@.*") { if (!vm("/tmp/am_fifo", "conference")) { log("Could not contact conference server\n"); t_reply("500","Could not contact conference server"); }; log ("!!!!!!!!!!!!!!!!!!!!!CONFERENCE!!!!!!!!!!!!!!!!!!!!!!!!!\n"); } else { if(!vm("/tmp/am_fifo","voicemail")) #if(!vm("/tmp/am_fifo","ivr")) { log(1,"could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("!!!!!!!!!!!!!!!!!!!!!!VOICEMAIL!!!!!!!!!!!!!!!!!!!!!!!!!\n"); }; # # log("**************** vm start - begin ******************\n"); # if(!vm("/tmp/am_fifo","voicemail")){ # log("could not contact the answer machine\n"); # t_reply("500","could not contact the answer machine"); # }; # log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); }

Bogdan-Andrei Iancu wrote: I used the config u sent me now am getting this error in my log. am using SEMS 0.9.0 and I have no idea what these errors mean Sep 11 14:33:30 serry /usr/sbin/openser[20658]: ERROR: fifo_t_reply: lookup failed Sep 11 14:33:30 serry /usr/sbin/openser[20658]: ERROR: fifo_server: command (t_reply) processing failed Sep 11 14:33:30 serry Sems[19839]: Error: (AmRequest.cpp)(reply)(368): AmRequestUAS::reply: 481 fifo_t_reply: no such transaction Sep 11 14:33:30 serry Sems[19839]: Error: (AmSession.cpp)(run)(217): 500 could not send response. Sep 11 14:33:30 serry /usr/sbin/openser[20658]: ERROR: fifo_t_reply: lookup failed Sep 11 14:33:30 serry /usr/sbin/openser[20658]: ERROR: fifo_server: command (t_reply) processing failed Sep 11 14:33:30 serry Sems[19839]: Error: (AmRequest.cpp)(reply)(368): AmRequestUAS::reply: 481 fifo_t_reply: no such transaction Sep 11 14:33:33 serry /usr/sbin/openser[20666]: LOG: Someone trying to register from private IP, rewriting Sep 11 14:33:35 serry /usr/sbin/openser[20678]: LOG: Someone trying to register from private IP, rewriting Sep 11 14:33:35 serry /usr/sbin/openser[20624]: WARNING: t_reply: ACKs are not replied -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101

Daniel-Constantin Mierla wrote: I did not get around it, It is definitely not a network problem since sems and openser are on the same machine -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101

Daniel-Constantin Mierla wrote: This is what I got from ngrep and I can't decrypt it :) U IP:5060 -> IP:5090 INVITE sip:4000@IP:5090;user=phone SIP/2.0..Record-Route: <sip :IP;ftag=c793260e386ee639;lr=on>..Via: SIP/2.0/UDP 212.103.160 .198;branch=z9hG4bKaf6b.be857c25.0..Via: SIP/2.0/UDP 10.99.4.2;rport=2474;r eceived=212.103.165.18;branch=z9hG4bKe9a96c235da61c20..From: "Taymour A. El Erian" <sip:1101@tedata.net;user=phone>;tag=c793260e386ee639..To: <sip:400 0@tedata.net;user=phone>..Contact: <sip:1101@212.103.165.18:2474;user=phone algorit hm=MD5, uri="sip:4000@tedata.net;user=phone", nonce="433b9e156151c6fff850be 0d40656ab14853786f", response="097488dc945c51fdea29ce3dde36df28"..Call-ID: 07f54cd444bc0a36@10.99.4.2..CSeq: 63492 INVITE..User-Agent: Grandstream BT1 00 1.0.5.10..Max-Forwards: 16..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OP TIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Content-Length: 305... .v=0..o=1101 8000 8000 IN IP4 10.99.4.2..s=SIP Call..c=IN IP4 10.99.4.2..t= 0 0..m=audio 5004 RTP/AVP 18 99 4 2 15 9..a=rtpmap:18 G729/8000..a=rtpmap:9 9 iLBC/8000..a=fmtp:99 mode=20..a=rtpmap:4 G723/8000..a=rtpmap:2 G726-32/80 00..a=rtpmap:15 G728/8000..a=rtpmap:9 G722/8000..a=ptime:20..a=direction:ac tive.. # U IP:5090 -> IP:5060 SIP/2.0 100 Trying -- just wait a minute !..Via: SIP/2.0/UDP 212.103.160.19 8;branch=z9hG4bKaf6b.be857c25.0..Via: SIP/2.0/UDP 10.99.4.2;rport=2474;rece ived=212.103.165.18;branch=z9hG4bKe9a96c235da61c20..From: "Taymour A. El Er ian" <sip:1101@tedata.net;user=phone>;tag=c793260e386ee639..To: <sip:4000@t edata.net;user=phone>..Call-ID: 07f54cd444bc0a36@10.99.4.2..CSeq: 63492 INV ITE..Server: OpenSer (0.9.5 (i386/linux))..Content-Length: 0..Warning: 392 IP:5090 "Noisy feedback tells: pid=821 req_src_ip=212.103.160 .198 req_src_port=5060 in_uri=sip:4000@IP:5090;user=phone out_ uri=sip:4000@IP:5090;user=phone via_cnt==2".... # U IP:5060 -> IP:5090 CANCEL sip:4000@IP:5090;user=phone SIP/2.0..Via: SIP/2.0/UDP 2 12.103.160.198;branch=z9hG4bKaf6b.be857c25.0..From: "Taymour A. El Erian" < sip:1101@tedata.net;user=phone>;tag=c793260e386ee639..Call-ID: 07f54cd444bc 0a36@10.99.4.2..To: <sip:4000@tedata.net;user=phone>..CSeq: 63492 CANCEL..U ser-Agent: OpenSer (0.9.5 (i386/linux))..Content-Length: 0.... # U IP:5090 -> IP:5060 SIP/2.0 200 cancels are junked here..Via: SIP/2.0/UDP IP;branc h=z9hG4bKaf6b.be857c25.0..From: "Taymour A. El Erian" <sip:1101@tedata.net; user=phone>;tag=c793260e386ee639..Call-ID: 07f54cd444bc0a36@10.99.4.2..To: <sip:4000@tedata.net;user=phone>;tag=295de06b31c02f814974318d00ff602d.3f4c. .CSeq: 63492 CANCEL..Server: OpenSer (0.9.5 (i386/linux))..Content-Length: 0..Warning: 392 IP:5090 "Noisy feedback tells: pid=823 req_sr c_ip=IP req_src_port=5060 in_uri=sip:4000@IP:5090 ;user=phone out_uri=sip:4000@IP:5090;user=phone via_cnt==1".. -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101

Bogdan-Andrei Iancu wrote: You did decrypt right :). I fixed the fifo issue but now this is what I get Sep 29 12:29:11 serry /usr/sbin/openser[16942]: ERROR: fifo_server: command DB is not available Sep 29 12:29:11 serry /usr/sbin/openser[16942]: ERROR: fifo_server: command must begin with :: email_address Sep 29 12:29:11 serry /usr/sbin/openser[16942]: ERROR: fifo_server: command must begin with :: subscriber Sep 29 12:29:11 serry /usr/sbin/openser[16942]: ERROR: fifo_server: command must have at least 3 chars Sep 29 12:29:11 serry Sems[16265]: Error: (AmSession.cpp)(startSession)(512): 404 voicemail: no email address for user <4000> User 4000 already has an email address in the database (am using sems-0.9.0) -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101

Bogdan-Andrei Iancu wrote: Thanks a million, now it is working -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101