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37455 discussions

[Users] User Registration Security on Openser
by Howard Tang 18 Jan '07

18 Jan '07

Hi, I realized some one is able to make call and registered to my sip proxy while he/she is not in the subscriber table. I couldn't find his/her username in the subscriber table, but i was able to see him/her in the location table. I am able to see he/she made 10 calls from my sip proxy. Anyone have an idea on what i have done wrong? I have included the authentication part of code here. if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("x.x.x.x", "subscriber")) { www_challenge("x.x.x.x", "0"); exit; }; consume_credentials(); save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("x.x.x.x","0"); exit; } consume_credentials(); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; -- Howard Tang

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[Serusers] Resource List Notification Problem
by raj gopal 18 Jan '07

18 Jan '07

Hi, I have configured apache and SER as described in presence hand book. I am facing some problem in group resource list subscription and notification. SUBSCRIBE sip:venkatfriends@10.187.83.218 SIP/2.0 Supported: eventlist Subject: Subscription Request Via: SIP/2.0/UDP 10.187.83.222:5061;rport;branch=z9hG4bK08652 Max-Forwards: 70 To: <sip:venkatfriends@10.187.83.218> From: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489 Call-ID: 382598915012(a)10.187.83.222 CSeq: 1 SUBSCRIBE Contact: <sip:venkat@10.187.83.222:5061> Expires: 7200 User-Agent: XXXXXXXXXXXXXXX Event: presence Content-Length: 0 Accept: application/pidf+xml,multipart/related,application/rlmi+xml SIP/2.0 200 OK Via: SIP/2.0/UDP 10.187.83.222:5061;rport=5061;branch=z9hG4bK08652 To: <sip:venkatfriends@10.187.83.218 >;tag=c20c251a135762f9e7a5a4d8e6d1fe64-93f1 From: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489 Call-ID: 382598915012(a)10.187.83.222 CSeq: 1 SUBSCRIBE Require: eventlist Contact: <sip:10.187.83.218:5060> Expires: 6200 Server: Sip EXpress router (0.10.99-dev64 (i386/linux)) Content-Length: 0 NOTIFY sip:venkat@10.187.83.222:5061 SIP/2.0 Via: SIP/2.0/UDP 10.187.83.218;branch=z9hG4bK7fa8.3ee87012.0 To: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489 From: <sip:venkatfriends@10.187.83.218 >;tag=c20c251a135762f9e7a5a4d8e6d1fe64-93f1 CSeq: 10 NOTIFY Call-ID: 382598915012(a)10.187.83.222 Content-Length: 340 User-Agent: Sip EXpress router(0.10.99-dev64 (i386/linux)) Subscription-State: active;expires=6200 Contact: <sip:10.187.83.218:5060> Max-Forwards: 70 Event: presence Require: eventlist Content-Type: multipart/related;type="application/rlmi+xml";start="qwW388551513pPdxX1615182410";boundary="RewXdpxR742478745xA757064664"; --RewXdpxR742478745xA757064664 Content-Transfer-Encoding: binary Content-ID: qwW388551513pPdxX1615182410 Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list xmlns="urn:ietf:params:xml:ns:rlmi" uri=" sip:venkatfriends@10.187.83.218" version="0" fullState="true"> </list> --RewXdpxR742478745xA757064664-- I have created all entries in resource-list.xml and service entry in index file. But, the server sends NOTIFY with no resource list entries and doesn't give any error messages. I am using full-no-failover.cfg configuration file. Whats wrong here ?????? Can anyone help me.... -Venkat

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[Users] SipSAK with OpenSER
by Bill Neely 18 Jan '07

18 Jan '07

Received the following from Klaus a few weeks ago: Does anybody have an example of how to use sipsak with openser. Does openser call sipsak, or is it the other way around? Xantek, Inc. wrote: I am sure this must have been covered somewhere, but I cannot find it. I am trying to get openser to register with username and password and forward a call to that server. This is for outbound sip termination to the PSTN. openser is a proxy thus basically openser can'T do that. Now the good news. For registration at the termination provider you can use sipsak. For sending authentication credentials to the termination provider openser has the "uac" module. But be aware that this module does not always works as it does not increase the cseq when sending the INVITE with credentials (search the archive for details). E.g. the uac module wont work if the termination providers uses Asterisk (as Asterisk checks the cseq) regards klaus -- 702-874-3833 1-866-553-3833

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[Users] CID insertion
by nick 18 Jan '07

18 Jan '07

I have my user authentication in a MYSQL database and use openser administrator to manage users. With every user I can add the Caller ID which ends up in the database (I've confirmed the existence in the DB). I'd like to add a Remote Party ID header into my calls directed to my PSTN termination provider (this is the header they want). I have had a look at the Auth module, and if I understood correctly, needed to add the lines modparam("auth", "rpid_avp", "i:13") to obtain the rpid field from the database and append_rpid_hf(); to actually add the RPID field to my outgoing INVITES, etc. however, I still can't see the header being added to the calls in outgoing SIP traces, and I have the feeling that I'm missing something fundamental.. I've tried also adding this line to load the credentials from the database: modparam("auth_db", "load_credentials", "rpid") Do I need to set the load credentials to be i:13=rpid to load the rpid field into the avp??? It's not 100% clear from the documentation where and how these avps are used (automatically if they're filled, or if they need to be somehow referenced in the function that uses them) and exactly which function is necessary (do I need the auth_db function or not?)..

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[Users] Module Permissions
by Bruno Machado 18 Jan '07

18 Jan '07

Hi I'd like to use the 'permissions' module but there is a something I need to change: the table openser.trusted has a column that matches the 'From URI' with it (from_pattern). Is there a way to change it to RURI? I mean, I don't want to analyze the header From: I want the function 'allow_trusted()' matching the RURI instead. Is it possible? Thanks in advance. Bruno Machado __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

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Re: [Users] mi_xmlrpc - versions of dependancies for - libxml, libxmlrpc-c3, libxmlrpc-c3-dev
by Bogdan-Andrei Iancu 17 Jan '07

17 Jan '07

Hi Steven, yes, in the official distribution there is an older version of xmlrpc-c3. Not sure why this, but we decided to build the module against the version you currently find in all distro to avoid forcing the users to compile/install by hand additional software. regards, bogdan Steven Halerman wrote: >Thank you for the quick response. I didn't see >binaries for Fedora, so i downloaded the latest - >xmlrpc-c-1.06.09 (earlier versions would not cleanly >compile with GCC 3.4.4) from >http://sourceforge.net/projects/xmlrpc-c but it does >not have the xmlrpc_abyss.h file. Upon further >digging, i noticed in server_abyss.h, a note that >mentions > >Abyss contents handlers >xmlrpc_server_abyss_rpc2_handler() and >xmlrpc_server_abyss_default_handler() were available >in older Xmlrpc-c, but starting with Release 1.01, >they are not. Instead, call >xmlrpc_server_abyss_set_handlers() to install them. > >Steven > >--- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> >wrote: > > > >>Hi Steven, >> >>I have libxmlrpc-c3-dev version 0.9.10-4 on a >>unstable debian and >>compiles with no problem. The "xmlrpc_abyss.h" file >>belong to the >>libxmlrpc-c3-dev package: >>/usr/include/xmlrpc_abyss.h could you check >>if you have this hdr file installed on your machine? >> >>regards, >>bogdan >> >>Steven Halerman wrote: >> >> >> >>>I have been trying to make mi_xmlrpc (from cvs >>> >>> >>head). >> >> >>>I am running Fedora Core 3 - gcc version 3.4.4 >>> >>>I get the following error: >>> >>>xr_server.c:37:26: xmlrpc_abyss.h: No such file or >>>directory >>>mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or >>>directory >>>In file included from >>>/usr/local/include/xmlrpc-c/base.h:9, >>> from >>> >>> >>/usr/local/include/xmlrpc.h:1, >> >> >>> from mi_xmlrpc.h:35, >>> from mi_xmlrpc.c:38: >>>/usr/local/include/xmlrpc-c/util.h:20:56: >>>xmlrpc-c/c_util.h: No such file or dir >>>ectory >>>In file included from >>>/usr/local/include/xmlrpc-c/base.h:9, >>> from >>> >>> >>/usr/local/include/xmlrpc.h:1, >> >> >>> from mi_xmlrpc.h:35, >>> from mi_xmlrpc.c:38: >>>/usr/local/include/xmlrpc-c/util.h:150: error: >>> >>> >>syntax >> >> >>>error before "GNU_PRINTF_A >>>TTR" >>>/usr/local/include/xmlrpc-c/util.h:158: error: >>> >>> >>syntax >> >> >>>error before "GNU_PRINTF_A >>>TTR" >>>mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or >>>directory >>>mi_xmlrpc.c: In function `mod_child_init': >>>mi_xmlrpc.c:156: warning: implicit declaration of >>>function `xmlrpc_server_abyss_ >>>init_registry' >>>mi_xmlrpc.c:166: error: >>>`xmlrpc_server_abyss_rpc2_handler' undeclared >>> >>> >>(first >> >> >>>use >>>in this function) >>>mi_xmlrpc.c:166: error: (Each undeclared identifier >>> >>> >>is >> >> >>>reported only once >>>mi_xmlrpc.c:166: error: for each function it >>> >>> >>appears >> >> >>>in.) >>>mi_xmlrpc.c:172: error: >>>`xmlrpc_server_abyss_default_handler' undeclared >>>(first >>>use in this function) >>>make[1]: *** [mi_xmlrpc.o] Error 1 >>> >>>what versions of >>> libxml >>> libxmlrpc-c3 >>> libxmlrpc-c3-dev >>>did you use to successfully compile? >>> >>>Thanks in advance >>>Steven >>> >>>

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[Users] interrupts after 30 secconds
by Ion Minzu 17 Jan '07

17 Jan '07

Hello, I have a problem: When someone begins a call from PSTN to my voip network, the session interrupts after 30 secconds. The UA don't receive the ACK message(6)[the ACK doesn't arrive].The scheme below indicate this situation. P.S. This situation happens just with one type of equipment: atcom AG468; with other equipment (atcom AT320ED, audiocodes, atcom AG168) all works fine. Very strange. Thunks. |cisco-gateway| | openser | |UA's| X.X.X.X - public ip adress eth0-X.X.X.Y - public IP 172.17.6.0/24 | eth1-172.17.6.2 | | Invite | Invite | 1 |------------------------------->|----------------------------->| | Trying | Trying | 2 |<-------------------------------|<-----------------------------| | Ringing | Ringing | 3 |<-------------------------------|<-----------------------------| | PRACK | | 4 |------------------------------->| | | 200 ok | 200 ok | 5 |<-------------------------------|<-----------------------------| | ACK | | 6 |------------------------------->| | |................................|..............................|30 s | Bye | Bye | 7 |<-------------------------------|<-----------------------------| | 200 OK | | |------------------------------->| | | | | -- Best regards, Ion Minzu, Specialist Tehnologii Informationale, Administrator de sistem al Centrului de certificare, Administrator VoIP, I.S."Centrul de Telecomunicatii Speciale", tel:250-517 (office), 069501208 (mob), 382869185 (ICQ) mailto:ion.minzu@cts.md

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[Users] Fwd: [kphone-users] New KPhone in the CVS
by Cesc 17 Jan '07

17 Jan '07

FYI ---------- Forwarded message ---------- From: Klaus Fleischmann <kgfleischmann(a)t-online.de> Date: Jan 16, 2007 7:01 PM Subject: [kphone-users] New KPhone in the CVS To: kphone-devel <kphone-devel(a)lists.sourceforge.net>, kphone-users <kphone-users(a)lists.sourceforge.net > Hi, last night I uploaded new code for kphoneSI, the most important features are: - SIMPLE presence: presence server support (PUBLISH, subscription to watcherlist) - SIMPLE presence end to end: After answered subsription, the buddy will be shown in the contact window (in red) even if he not subscribe us too. - new Call Widget design - help file and dtmf panels with own window - the last error message will be shown in the call widget instead of alarm windows - some more ALSA improvements - document and man page update If you are not used to CVS-downloads, see http://sourceforge.net/docman/?group_id=47742 or http://sourceforge.net/docman/display_doc.php?docid=34779&group_id=47742 for details. Yours Klaus ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV _______________________________________________ kphone-users mailing list kphone-users(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/kphone-users

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[Users] REMOVED: old fifo implementation
by Bogdan-Andrei Iancu 17 Jan '07

17 Jan '07

Hi everybody, the old fifo implementation (from core) was removed and replaced by the MI FIFO module. See http://www.openser.org/docs/modules/1.2.x/mi_fifo.html why? See: http://www.openser.org/pipermail/users/2006-December/008154.html To maintain the previous FIFO behaviour, you just have to: 1) remove "fifo=/tmp/openser_fifo" from your config 2) load the mi_fifo module 3) configure modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") that's all :) for an working example, see the default openser.cfg which was already updated: http://openser.cvs.sourceforge.net/openser/sip-server/etc/openser.cfg?revis… regards, bogdan

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[Users] Rtpproxy and re-invites
by Joshua Laroff 17 Jan '07

17 Jan '07

Hello, I am new to openser and rtpproxy. I have tried to configure openser so that reinvites are also handled by rtpproxy. It seems however that they are not. I have included my config below: debug=9 fork=yes log_stderror=no check_via=no dns=no rev_dns=no port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ Module Loading Section ------------------------- mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "permissions.so" loadmodule "options.so" #------------------- Module Parameters ----------------------------- modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") #--- nathelper params --- modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:8899") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") #--- usrloc params --- modparam("usrloc", "db_mode", 2) #--- registrar params --- modparam("registrar", "nat_flag", 6) #--- rr params --- modparam("rr", "enable_full_lr", 1) #--- tm params --- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "i:45") #--- permissions params --- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #------------------------ Main Routing Block ------------------------------ route { # ----------------------------------------- # Sanity check Section # ----------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ------------------------------------------- # Record Route Section # ------------------------------------------- if (method!="REGISTER") { record_route(); }; # ------------------------------------------- # Call Tear Down Section # ------------------------------------------- if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; # ------------------------------------------- # - Loose Route Section # ------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); }; if (nat_uac_test("19")) { setflag(6); }; }; route(4); route(1); exit; }; # --------------------------------------------- # - Call Type Processing Section # --------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; exit; }; if (method=="ACK") { route(1); exit; } if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (uri==myself) { if (method==OPTIONS) { options_reply(); } } lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # -------------------------------------------- # - Default Message Handler # -------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # -------------------------------------------- # - REGISTER Message Handler # -------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ------------------------------------------------ # - INVITE Message Handler # ------------------------------------------------ if (!allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); }; if (nat_uac_test("19")) { setflag(6); }; if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(5); exit; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); route(5); exit; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); route(5); exit; }; sl_send_reply("404", "User Not Found"); exit; }; route(4); route(5); } route[4] { # --------------------------------------- # - NAT Traversal Section # --------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # ---------------------------------------- # - PSTN GATEWAY Handler # ---------------------------------------- rewritehost("204.x.x.x"); # avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { # --------------------------------------- # - REPLY ROUTE # --------------------------------------- if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } if (nat_uac_test("1")) { fix_nated_contact(); }; }

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