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37468 discussions

[Kamailio-Users] x-lite 3.0 on linux
by Daniel-Constantin Mierla 14 May '09

14 May '09

Hello, during last weeks while testing the kamailio (openser) modules with sip-router core I got x-lite 3.0 working on linux (ubuntu) with wine 1.1.20. I posted guidelines at: http://appdb.winehq.org/objectManager.php?sClass=version&iId=16413 Voice, IM&Presence seem to work just fine. Video was not tested as I couldn't make any wine application see the video cam, although it is working fine on linux with skype or ekiga. For me it is a great tool to test the presence, hope it will be useful for some of you as well. Apart of x-lite, I got other sip softphone working with wine: - pangolin - works fine, but the presence status changes are not sent to server (no PUBLISH) -- might be a bug in the software, maybe someone running it on windows can confirm it or not - zoiper communicator rc1 free edition for windows - linux version seems to be available now, but requires to register with their site Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] kamailio and/or ser 3.0 with sip router core
by Daniel-Constantin Mierla 14 May '09

14 May '09

Hello, long awaited period of SIP Router integration phase is more and more approaching the end. Apart of 4 modules that are not integrated yet from Kamailio - seas, siptrace, uac_redirect and nat_traversal (work being undertaken) -- the rest of 88 modules are ready for http://sip-router.org core. I have created a page to collect guidelines for the default Kamailio configuration file used with sip-router. I invite all of you to test it, not only this one, but your private configs as well. http://sip-router.org/wiki/migration/kamailio-3.0-config Feel free to add to the wiki page or post questions about it to sr-users(a)lists.sip-router.org. This is a new list created for users of sip-router project, you can subscribe to it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users At this stage, sip router can be used with modules from both Kamailio (OpenSER) and SER projects, mixed in same instance at the same time. Therefore you can choose the best ones that fit your needs. For Kamailio (OpenSER) users there is a page that tries to collect new features they got from SER side (still a lot to add there, hope ser developers will contribute what they find missing). Another page collects links for ser and kamailio users to find what the other project brings: http://sip-router.org/wiki/migration/new-features-for-k-users http://sip-router.org/wiki/features/inherited-features It was not only integration work in the past months, sip router has quite a bunch of brand new feature. The wiki page to collect the summary is available at: http://sip-router.org/wiki/features/new-in-devel Jan just set online documentation portal, available for web browsing: http://sip-router.org/docbook/sip-router/branch/master/ Common modules are located in directory 'modules' while kamailio and ser specific modules can be found under module_k and modules_s. Project's wiki portal got also good content and it is under heavy work -- note that everyone can contribute to wiki: http://sip-router.org/wiki Couple of core and tm features are not yet integrated or not exactly as they were in Kamailio. Please report the ones you are missing in your config at: http://sip-router.org/wiki/devel/kamailio-integration#tm_and_core_extensions It is the place that should set the priority rank for integration. I hope that in 1-2 months time frame to get into beta stage and prepare a stable release. Many thanks for those that allocated resources to make things evolve to this stage and many thanks in advance for those that will start testing and help improving the sip router project. Beware that is still very early stage and not recommended for production. Cheers, Daniel PS. Codename 3.0 refers to Kamailio (OpenSER) or SER using sip router core, there is not a stable release yet. -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] dialplan subst issue
by Andreas Granig 14 May '09

14 May '09

Hi all, I was playing around with dialplan module (kam 1.4) for the first time and it seems that the repl_exp doesn't handle back references correctly: mysql root@kamailio> select * from dialplan\G *************************** 1. row *************************** id: 1 dpid: 1 pr: 1 match_op: 1 match_exp: @voicebox\.local$ match_len: 0 subst_exp: ^sip:(.*)$ repl_exp: sip:xx_\1_yy attrs: voicebox 1 row in set (0.00 sec) In my config, I do: dp_translate("1", "$ru/$var(dptest)"); xlog("L_INFO", "DP result='$var(dptest)', attr='$avp(s:dialplan_attr)\n"); and the result is: DP result='sip:xx_user@voicebox.local', attr='voicebox' So the string after the back reference ("_yy" in this case) is not inserted. Any ideas? Andreas

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[Kamailio-Users] presence pres-rules and database watchers table
by Dubravko Caric 13 May '09

13 May '09

Hi all, is there any way to automatically change "reason" column from "rejected" back to "NULL" or "allow" in watchers table after UAC#1 moves the UAC#2 from its blacklist to whitelist? i see that kamailio (1.5.0) correctly reads index.xml file from database, but "rejected" reason is still in the watcher table until I delete it manually and restart UAC. ...and is this even suppose to work like that or should I use different approach? Thanks Dubravko

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[Kamailio-Users] kamailio
by Abou NDIAYE 13 May '09

13 May '09

Hello, In fact I started with kamailio. The calls go smoothly on my router (with NAT) kamailio that the proxy is started or not, the calls pass. I put what in the conf file for the proxy intercepts all calls sip and then forward. Thank you in advance. _________________________________________________________________ Découvrez Windows Live Spaces et créez votre site Web perso en quelques clics ! http://spaces.live.com/signup.aspx

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[Kamailio-Users] Get send_socket if I have no new branch
by catalina oancea 13 May '09

13 May '09

Hello all In my kamailio.cfg I don't use append_branch at all and I listen on multiple addresses. Before running t_relay I need to know on which socket will the request be sent. I don't want to change the socket, but I want to know which socket is chosen. I tried $(branch(send_socket)[0]) and $(branch(send_socket)) but I got error "ERROR:pv:pv_get_branchx: error accessing branch [0]" Does anybody know how I can get this info? Thanks Catalina

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Re: [Kamailio-Users] two UACs behind same NAT
by Dubravko Caric 13 May '09

13 May '09

Hi, ----- Original Message ---- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Iñaki Baz Castillo <ibc(a)aliax.net> Cc: users(a)lists.kamailio.org Sent: Monday, May 11, 2009 10:04:49 AM Subject: Re: [Kamailio-Users] two UACs behind same NAT >> >> Bad luck then... :( >> SIP ALG is the worst enemy for SIP. >> > workarounds could be: > - run sip server also on a different port than 5060 (say 5070) - > kamailio is just fine doing it. Point the users behind such ALGs to this > port. Still alg can detect it, but most of them do the detection by port > 5060 > - use TCP if the phone supports it - most of algs do not touch TCP > connections, but ... > - use TLS if the phone supports it - safest - the alg cannot touch it yes, we are planning to use TLS and I hope we won't have that kind of problems then > - recommended - send back the router and ask for money return well, I could do that but the problem is not just with this router. I saw a list of this kind of routers (small home routers) that are not recommended to use with SIP, and I think that TLS is only universal solution. Best regards Dubravko Cheers, Daniel

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[Serusers] SER with RADIUS
by Leon Li 13 May '09

13 May '09

Hi, I am trying to make SER working with FreeRADIUS. However, I got an error using www_challenge, " 0(3184) parse error (119,25-26): unknown command, missing loadmodule?" The 119 line is www_challenge("", "0"); I did load auth module. Any idea? Here is my config: # # $Id: ser-basic.cfg,v 1.1.2.2 2008/06/12 13:15:06 alfredh Exp $ # # This a very basic config file w aliases and anamed route but # w/o authentication, accounting, database, multi-domain support etc. # Please refer to ser.cfg for a more complete example # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) #memdbg=10 # memory debug message level #memlog=10 # memory statistics log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* 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) listen=202.158.197.134 port=5060 children=4 #user=ser #group=ser #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/ctl.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.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) # ctl params # by default ctl listens on unixs:/tmp/ser_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") # listen on tcp, localhost #modparam("ctl", "binrpc", "tcp:localhost:2046") # -- auth_radius params -- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_reply("483","Too Many Hops"); break; } if (msg:len >= max_len ) { sl_reply("513", "Message too big"); break; } # 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 withing 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(FORWARD); break; } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(FORWARD); break; } # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save_contacts("location"); break; } # native SIP destinations are handled using our USRLOC DB if (!lookup_contacts("location")) { sl_reply("404", "Not Found"); break; } append_hf("P-hint: usrloc applied\r\n"); } route(FORWARD); } route[FORWARD] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } } Thanks, Leon

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[Kamailio-Users] BAN a scanner
by BERGANZ François 12 May '09

12 May '09

Hello, Is it possible to ban an IP which scan my servers ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] WARNING:core:fm_free: free(0) called in kamailio
by rachid 12 May '09

12 May '09

Hello, when i start a sip client (x-lite) i have in kamailio debug message this error: WARNING:core:fm_free: free(0) called someone can tell me where is the problem?? thank you ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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