sr-users October 2007

sr-users@lists.kamailio.org

130 participants
286 discussions

[OpenSER-Users] DNS blacklisting problem
by Tim Madorma 09 Oct '07

09 Oct '07

Hi, I'm interested in using DNS blacklisting to stop transmitting SIP requests towards UAS endpoints that are non-responsive. I am on version 1.2, but I can't get it to work properly. Here is what I am trying to do: 1) I get a message from a UAC that needs to be sent towards 1 of 4 destination UAS endpoints. 2) I setup the OpenSER to perform an SRV lookup which returns a record that has four A records associated with it. It looks as follows: ;; QUESTION SECTION: ;_sip._udp.aimcidfilter.com. IN SRV ;; ANSWER SECTION: _sip._udp.aimcidfilter.com. 60 IN SRV 1 100 5061 spinner.eng.rr.com. _sip._udp.aimcidfilter.com. 60 IN SRV 2 100 5061 bart.eng.rr.com. _sip._udp.aimcidfilter.com. 60 IN SRV 3 50 5061 homer.eng.rr.com. _sip._udp.aimcidfilter.com. 60 IN SRV 3 50 5061 crunch.eng.rr.com. 3) The OpenSER does a gethostbyname which performs a DNS A record lookup and gets a response for spinner.eng.rr.com. 4) When the OpenSER sends towards spinner.eng.rr.com, it does not get a SIP response from it. *It is at this point I was hoping that OpenSER would blacklist this UAS endpoint*, but for each incoming request that is received by the OpenSER, it continues to resolve the domain in the same manner and sends towards spinner.eng.rr.com even though there is no SIP response. Is what I am trying to do in accordance with how OpenSER blacklisting is supposed to work? I am trying to have OpenSER send towards bart.eng.rr.com when spinner.eng.rr.com is not responding or is not reachable. My configuration is as below. Note that I have even tried to disable sending towards spinner.eng.rr.com by manually adding a dst_blacklist entry - that doesn't even seem to work for me.. Am I doing something wrong? I have also attached a log file. Here is my output from openserctl # openserctl fifo list_blacklists 200 OK List:: net_filter owner=13 flags=1 Rule:: flags=0 IP:: 65.185.233.55 Mask:: 255.255.255.255 Proto:: 0 Port:: 5061 List:: dns owner=17 flags=6 --------------- # # Openser.cfg # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) maxbuffer=1048576 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #reply_to_via=1 children=4 log_facility=LOG_LOCAL4 dns_use_search_list=no dns_servers_no=2 user="sipproxy" group="sipproxy" disable_dns_blacklist=no disable_dns_failover=no dns_try_ipv6=no dns_retr_time=2 dns_retr_no=2 dst_blacklist = net_filter:{ ( any , 65.185.233.55, 5061 , "" )} # block towards spinner listen=udp:65.185.232.62:5060 alias=65.185.233.104:5060 # LOAD OpenSER MODULES mpath="/sw/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc","report_cancels", 1) modparam("acc","failed_transaction_flag", 1) modparam("acc","log_extra", "req_uri=$rU") modparam("tm", "fr_timer", 5) modparam("tm","fr_inv_timer",5) modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages within a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { route(1); }; if (method=="OPTIONS") { sl_send_reply("200", "OK"); exit; } if (uri =~ "^sip:feature_fs@" || method=="NOTIFY") { xlog("L_DBG", "TWC: received incoming message:\n <$mb>\n"); seturi("sip:feature_fs@aimcidfilter.com"); setflag(1); route(1); } else { sl_send_reply("404", "Not Found"); }; } ##################################################### # Default Message Handler ##################################################### route[1] { # Send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

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[OpenSER-Users] Problem with 2 PC behind the same NAT
by Daniel Grotti 09 Oct '07

09 Oct '07

I have 2 SJphone behind the same NAT (I know that it's a restricted con NAT). UA1 ----> (NAT 1) --------> PROXY | | UA2 <-----(NAT 1)<------------- I've configured my openser.cfg to use Mediaproxy (installed in the same pc) if 1 or 2 pc are behind NAT (in this case both of my pc). But when I call from UA1 to the other and I accept the call with UA2, the RTP stream doesn't flow through the Mediaproxy but it's direct from UA1 to UA2. Any clue? thanks daniel

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Re: [Serusers] Get RealTime Online Users status
by Jiri Kuthan 09 Oct '07

09 Oct '07

If you mean location database, check the "location" table then. -jiri At 10:05 09/10/2007, roayu wrote: >Hi there! > >Can anyone tell me how to get the RealTime Online user status other than using command �serctl ul show�? Or how can I store the realtime online user status to the MySQL db? > > >Thanks. > >Cheers, >Roa Yu J >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] The role of "rinstance"
by Daniel Grotti 09 Oct '07

09 Oct '07

Hi everybody, Somebody Could tell me which role the Rinstance tag play in the SIP message, when I use X-lite 3 ? Thank you, Daniel

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[OpenSER-Users] Can't Register SJPhone with IMS Core
by A Chaturvedi 09 Oct '07

09 Oct '07

Hi, I tried to register SJPhone with IMS Core but it failed (even after using sip2ims gatway). A part from this when I used XLite sip phone it gets registered... Please guide me if pcscf need some modification to accept the Register request from it... Also find ate ethereal dump for it.. regards: Ashish Chaturvedi Mailto: a.chaturvedi(a)tcs.com =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[OpenSER-Users] MI_XMLPRC supports new versions of libxmlrpc-c3
by Bogdan-Andrei Iancu 09 Oct '07

09 Oct '07

Hi, I know this was from the beginning a hot issue, but now, thanks to Dragos Oancea <dragos(a)voice-system.ro>, the mi_xmlrpc can: 1) be compiled against the newest versions of libxmlrpc-c3. The Makefile will detect the library version an trigger the appropriate compile options for the module; So the module supports both the old version (0.9.10) and the newest versions (1.03.10 - 1.06.18). If none of these version is detected, a warning will be issued and the module will be compiled as for the new versions. If you detect the older versions (than 1.03.10 are supported, please let us know). 2) properly auto-detects the path of the header files and the libraries that the modules needs to compile against, so this should not be a problem anymore ;) any feedback will be useful :) regards, bogdan

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[OpenSER-Users] Catching 200 OK as part of INVITE CSeq.
by Alex Balashov 09 Oct '07

09 Oct '07

Greetings, I am proxying INVITEs and need to be able to know when the 200 OK that is part of that particular transaction dialogue (same CSeq as the INVITE) appears so that I stamp voice-bearing media start in my logs. However, I cannot seem to detect it. I see from the packet capture that the 200 OK from the far end of the call does go back through my OpenSER proxy (I do record_route() on all methods but REGISTER & OPTIONS as per the spec), but I can't seem to programmatically access it in any way. Are there special steps required to get at it? This is to say, my main route contains: if(is_method("INVITE")) { ... } But I never seem to catch anything with it but actual INVITE messages. What gives? I tried doing if(t_check_status("200")) within that block, but without anything that I would describe as success. I never even get another message whose overall transaction method is INVITE even when I dump them all for debugging purposes in the main route. Insights appreciated! Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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[OpenSER-Users] Direct (Postgres) SQL queries.
by Alex Balashov 09 Oct '07

09 Oct '07

Is there a better way to talk to a database directly than to use avp_db_query()? I need to use my own schemas; I have a platform that absolutely does not lend itself to storing data in the avpops' schema. Luckily, avp_db_query() saved the day. Is there anything even easier, or is this the definitive and methodologically correct way to go? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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[OpenSER-Users] Urgent!! Facing Problem while Registering SJPhone
by A Chaturvedi 09 Oct '07

09 Oct '07

Hi I was trying to register SJPhone with IMS Core and at Client end it shows Registered but at HSS it shows AUTH-PENDING. The Ethereal dump shows proper call flow with 200 OK. Have a look on HSS console debug statements... at de.fhg.fokus.diameter.DiameterPeer.transport.Communicator.run(Communicator.java:187) 2007-10-09 12:43:26,340 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:26,389 DEBUG de.fhg.fokus.hss.main.Task - execute Processing MAR! 2007-10-09 12:43:26,410 DEBUG de.fhg.fokus.hss.cx.op.MAR - generateAuthVector Auth-Scheme is Digest-MD5 2007-10-09 12:43:26,418 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:26,462 DEBUG de.fhg.fokus.hss.main.Task - execute Processing SAR! 2007-10-09 12:43:36,460 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:36,508 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:36,555 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:36,620 DEBUG de.fhg.fokus.hss.main.Task - execute Processing MAR! 2007-10-09 12:43:36,658 DEBUG de.fhg.fokus.hss.cx.op.MAR - generateAuthVector Auth-Scheme is Digest-MD5 2007-10-09 12:43:36,703 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! de.fhg.fokus.hss.cx.CxExperimentalResultException: IMS_Diameter_Error_User_Unknown at de.fhg.fokus.hss.cx.op.UAR.processRequest(UAR.java:112) at de.fhg.fokus.hss.main.Task.execute(Task.java:169) at de.fhg.fokus.hss.main.Worker.run(Worker.java:66) Please help me out in this.... If any body has made a successful call with SJPhone and IMS , plzz send me the configuration setting for that... Thanks in adv.... Ashish =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[OpenSER-Users] Snom number-guessing and OpenSER
by Edoardo Serra 09 Oct '07

09 Oct '07

Hi all, I'm trying to develop an extension to support SNOM's number guessing feature (I don't know if it's somehow a standard but description on how it works is here: http://kb.snom.com/kb/index.php?View=entry&CategoryID=14&EntryID=14) As I have to do some 'complicated' db queries I was thinking to use PERL module to handle number guessing request from the phone. I see there's no way (please correct me if I'm wrong) from OpenSER, neither from OpenSER::Message, to send a stateless reply with a customized body (SNOM says that possible number completion shoud be included in the body of a 200 OK reply). For this reason I have to use Net::SIP to generate the reply packet but I'm doing something wrong as OpenSER is not forwarding it back to the client. Test client is a Snom 360 and is behind a NAT, OpenSER has a public IP Here is my routing logic in openser.cfg ------------------- BEGIN openser.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (nat_uac_test("19")) { fix_nated_contact(); force_rport(); setbflag(6); } if (method=="PUBLISH") { if ($hdr(Event)=="number-guessing") { log("- Initial PUBLISH\n"); t_newtran(); t_on_reply("1"); perl_exec("numberguessing"); exit; } sl_send_reply("501", "Not implemented"); exit; } if (method=="REGISTER") { if (nat_uac_test("19")) { fix_nated_contact(); force_rport(); setbflag(6); } save("location"); exit; }; exit; } route[1] { log("--------------------- Stateful Reply\n"); if (!t_relay()) { sl_reply_error(); }; exit; } ------------------- END openser.cfg Here is my Simple Perl ------------------- BEGIN net-sip.pl use OpenSER qw ( log ); use OpenSER::Constants; use Net::SIP; # Predefined body, not querying db yet my $body = '"Franky Chang" sip:101@192.168.0.1;user=phone "Steven Jones" sip:102@192.168.0.1;user=phone "Marie Sun" sip:103@192.168.0.1;user=phone '; sub numberguessing { my $m = shift; my $pkt = Net::SIP::Request->new( $m->getMessage() ); my $resp = $pkt->create_response(200, "OK"); $resp->set_body($body); $resp->add_header('Content-Type','application/number-guessing'); my $leg = Net::SIP::Leg->new( addr => '111.222.333.444', port => '5061'); $leg->deliver( $resp, '111.222.333.444:5060' ); # 111.222.333.444 is the IP of OpenSER # I'm sending from port 5061 to port 5060 return 1; } ------------------- END net-sip.pl OpenSER in verbose debugging module says that it cannot find a matching transaction for the 200 OK reply (probably because is originated from port 5061 and it has never forwarded it to that port ? ) Here is OpenSER DEBUG log: 0(24828) SIP Reply (status): 0(24828) version: <SIP/2.0> 0(24828) status: <200> 0(24828) reason: <OK> 0(24828) parse_headers: flags=2 0(24828) get_hdr_field: cseq <cseq>: <1> <PUBLISH> 0(24828) DEBUG:parse_to:end of header reached, state=10 0(24828) DBUG:parse_to: display={"Edoardo"}, ruri={sip:eserra@111.222.333.444} 0(24828) DEBUG: get_hdr_field: <to> [43]; uri=[sip:eserra@111.222.333.444] 0(24828) DEBUG: to body ["Edoardo" <sip:eserra@111.222.333.444> ] 0(24828) Found param type 232, <branch> = <z9hG4bK-m3j054b6d8q6>; state=6 0(24828) Found param type 235, <rport> = <n/a>; state=17 0(24828) end of header reached, state=5 0(24828) parse_headers: Via found, flags=2 0(24828) parse_headers: this is the first via 0(24828) After parse_msg... 0(24828) DEBUG:forward_reply: found module tm, passing reply to it 0(24828) DEBUG: t_check: start=0xffffffff 0(24828) parse_headers: flags=22 0(24828) DEBUG: t_reply_matching: failure to match a transaction 0(24828) DEBUG: t_check: end=(nil) 0(24828) parse_headers: flags=4 0(24828) DEBUG: get_hdr_body : content_length=135 0(24828) found end of header 0(24828) ERROR:forward_reply: no 2nd via found in reply 0(24828) DEBUG:destroy_avp_list: destroying list (nil) 0(24828) receive_msg: cleaning up Here is also the tshark dump of the 2 packets (PUBLISH and its reply) Request-Line: PUBLISH sip:eserra@213.92.23.108 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.254.151:2051;branch=z9hG4bK-ma9r5n0t4jfq;rport From: "Edoardo" <sip:eserra@111.222.333.444>;tag=39wlni6ex1 To: "Edoardo" <sip:eserra@111.222.333.444> Call-ID: 3c27c039249b-jml9w7t9sywx CSeq: 1 PUBLISH Max-Forwards: 70 Event: number-guessing Content-Type: application/text Content-Length: 25 Message body Number: 10\r\n Max-Hits: 3\r\n Status-Line: SIP/2.0 200 OK Message Header call-id: 3c27c0daa600-emnlo1xcejtn cseq: 1 PUBLISH from: "Edoardo" <sip:eserra@111.222.333.444>;tag=6c1y8pp3fb to: "Edoardo" <sip:eserra@111.222.333.444> via: SIP/2.0/UDP 192.168.254.151:2051;branch=z9hG4bK-623oddiyzg56;rport content-type: application/number-guessing Content-Length: 135 Message body "Franky Chang" sip:101@192.168.0.1;user=phone\n "Steven Jones" sip:102@192.168.0.1;user=phone\n "Marie Sun" sip:103@192.168.0.1;user=phone\n Hope this info is enough for a debug :) Tnx in advance Regards Edoardo Serra EXSORSA LLC

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sr-users October 2007