sr-users October 2006

sr-users@lists.kamailio.org

189 participants
345 discussions

[Serusers] Compiling SER on Solaris 9
by Cheez Mirza 15 Oct '06

15 Oct '06

To compile SER on Solaris I ran (g)make all but get the following messages: Makefile.rules:84: action.d: No such file or directory Makefile.rules:84: crc.d: No such file or directory Makefile.rules:84: daemonize.d: No such file or directory Makefile.rules:84: data_lump.d: No such file or directory Makefile.rules:84: data_lump_rpl.d: No such file or directory Makefile.rules:84: dprint.d: No such file or directory Makefile.rules:84: dset.d: No such file or directory Makefile.rules:84: error.d: No such file or directory Makefile.rules:84: fifo_server.d: No such file or directory .... flex cfg.lex flex: fatal internal error, exec failed make: *** [lex.yy.c] Broken Pipe make: *** Deleting file `lex.yy.c' What am I doing wrong? Thanks, -CM __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] key exchange failed with MIKEY
by Gregoire 14 Oct '06

14 Oct '06

Hi everybody! I set up my Openser with TLS for signaling and it works well. I use Minisip as softphone and it registers successufully over TLS. But I have the following problem when I try to make a call using SRTP: Openser denies the INVITE cause its size is more than 2048 and then it makes an error as "Message too big". Even if I try to bypass this error in the configuration, the INVITE is still denied. In fact, I saw that Minisip uses MIKEY for the key exchange, and put it in the SDP body, and then it makes quite big but I don't understand why the size must be under 2048. Does someone have any idea to help me? Thanks Greg

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[Users] store in DB some searched content present in messages
by Ricardo Carvalho 14 Oct '06

14 Oct '06

Is there a way to search and store in a regular expression basis, some content of some kind of messages in an avp and then store that content in some table of MySQL database? Thanks, Ricardo.

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Re: [Users] acc_radius question
by Cseke Tamas 14 Oct '06

14 Oct '06

Hi, i use both setfalg(x), and acc_rad_request i need accounting for INVITE, ACK (for INVITE), CANCEL, BYE it was working ubkess error, but now i try failed_transaction flag. Could you help me in some better solution, I need acc_rad_request becasue of i need invite accounting immediately - for set the call setuptime ack - for set the call connecttime bye - for disconnecttime So i done it this way: (config in question) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "failed_transaction_flag", 4) if (is_method("ACK") || is_method("BYE")) { setflag(1); setflag(2); setflag(4); }; # accounting requests if (is_method("INVITE")) { if (!radius_www_authorize("$rd")) { www_challenge("$rd", "0"); exit; }; acc_rad_request("200 Invite received..."); consume_credentials(); } else if (is_method("ACK")) { if (t_check_status("[4-6][0-9][0-9]")) { acc_rad_request("Failed"); } else { acc_rad_request("200 Ack received..."); }; } else if (is_method("BYE")) { } else if (is_method("CANCEL")) { acc_rad_request("200 CANCEL received..."); }; Thanks, Tamas raviprakash sunkara wrote: > Can U send the Config > > U didn't set the flags properly in ur Config file > and > Don't use setflag("x"); > > Use acc_rad_request("200"); > > On 10/11/06, Cseke Tamas <cseke.tamas(a)eworldcom.hu> wrote: > >> >> Hi, >> >> When sends openser accounting packets for INVITE, and BYE message? >> it seems to be, when received 200 OK. >> >> 1)for INVITE i need radius packet immediately, not just when it was >> answered. >> 2)for BYE i need accounting packet, when any error occured too. >> >> modparam("acc", "radius_flag", 2) >> >> modparam("acc", "radius_missed_flag", 3) >> >> this attribute mean that, should sent radius packet, when any error >> occured too, isn't it? However openser don't send accounting packet in >> this case. >> >> i run into the following error: >> >> ########## >> T called -> openser >> BYE sip:21.16.8.92:5060 >> ## >> T openser->called >> SIP/2.0 477 Unfortunately error on sending to next hop occurred >> (477/TM) >> ########## >> T caller->openser >> BYE sip:66610038641368254@21.16.8.90:5060; >> ##### >> # openser -> caller >> SIP/2.0 481 Call Leg/Transaction Does Not Exist.. >> >> So no 200 OK for BYE received, therefore no accounting packet to radius >> sent. >> How can i solve these problems? >> >> Btw, what could cause this 477 error? >> >> Thanks, >> Tamas >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > > >

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Re: [Users] Please help me to fix openser.cfg (rewritehostport)
by Daniel-Constantin Mierla 14 Oct '06

14 Oct '06

Please keep the list in cc. use xlog("L_ERR", "$rm is next to pass for transfer to Asterisk"); maybe you got the log message for another request, the syntax looks ok. Cheers, Daniel On 10/13/06 14:38, Ronald Wiplinger wrote: > Daniel-Constantin Mierla wrote: >> I am not sure what do you want to achieve, but I assume it should be >> before lookup("aliases"); > > I want that if I dial a number, like 601, it should go to another > machine, where Asterisk is running and a phone with 601 is connected. > > I have moved the paragraph just before the lookup("aliases"); > save("location"); > exit; > }; > > xlog("L_ERR", "INVITE is next to pass > for transfer to Asterisk"); > if (method=="INVITE") { > xlog("L_ERR", "INVITE passed for > transfer to Asterisk"); > if (uri=~"^sip:601@") { > rewritehostport("59.124.210.35:5060"); > xlog("L_ERR", "transfer to Asterisk"); > route(1); > exit; > }; > }; > > lookup("aliases"); > > With these three xlog statements, I get only the first one (INVITE is > next to pass for transfer to Asterisk). The next if is obviously not > fulfilled! > Do I have a syntax error here? > > > bye > > Ronald Wiplinger > > >> >> Cheers, >> Daniel >> >> >> On 10/12/06 17:01, Ronald Wiplinger wrote: >>> I try to rewritehostport to my Asterisk box with all starting >>> numbers with 6. The xlog never shows anything. >>> >>> >>> >>> if (method=="INVITE") { >>> if (uri=~"^sip:6.+@") { >>> rewritehostport("xx.xx.xx.xx:5060"); >>> xlog("L_ERR", "transfer to Asterisk"); >>> route(1); >>> exit; >>> }; >>> }; >>> >>> I believe I did not put it at the right place. Below is my entire >>> openser.cfg. Can you please tell me, where above paragraph should >>> be? Other parts maybe also not in the right order. >>> >>> >>> >>> bye >>> >>> Ronald Wiplinger >>> >>> >>> >>> >>> >>> # >>> # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ >>> # >>> >>> debug=3 # debug level (cmd line: -dddddddddd) >>> fork=yes >>> log_stderror=no # (cmd line: -E) >>> log_facility=LOG_LOCAL7 >>> log_name="Taipei-openser" >>> >>> check_via=no # (cmd. line: -v) >>> dns=no # was no (cmd. line: -r) >>> rev_dns=no # (cmd. line: -R) >>> port=5060 >>> children=4 >>> fifo="/tmp/openser_fifo" >>> >>> >>> mpath="/usr/local/lib64/openser/modules/" >>> loadmodule "mysql.so" >>> loadmodule "sl.so" >>> loadmodule "tm.so" >>> loadmodule "rr.so" >>> loadmodule "maxfwd.so" >>> loadmodule "usrloc.so" >>> loadmodule "registrar.so" >>> loadmodule "textops.so" >>> loadmodule "nathelper.so" >>> loadmodule "xlog.so" >>> loadmodule "enum.so" >>> loadmodule "acc.so" >>> loadmodule "auth.so" >>> loadmodule "auth_db.so" >>> modparam("usrloc|auth_db","db_url","mysql://shshshsh:shhhhrw@localhost/wodeopenser") >>> modparam("usrloc", "db_mode", 2) >>> modparam("registrar", "nat_flag", 6) >>> modparam("registrar", "sip_natping_flag", 7) modparam("auth_db", >>> "calculate_ha1", yes) >>> modparam("auth_db", "password_column", "password") >>> modparam("rr", "enable_full_lr", 1) >>> modparam("nathelper", "rtpproxy_sock", "udp:xx.xx.xx.xx:22222") # >>> default port 22222 >>> #modparam("nathelper", "rtpproxy_sock", >>> "unix:/var/run/rtpproxy.sock") modparam("nathelper", >>> "natping_interval", 30) >>> modparam("nathelper", "ping_nated_only", 1) >>> modparam("nathelper", "sipping_from", "sip:xx.xx.xx.xx") >>> modparam("enum", "domain_suffix", "e164.org.") >>> modparam("enum", "tel_uri_params", ";npdi") >>> >>> >>> # ------------------------- request routing logic ------------------- >>> >>> # main routing logic >>> >>> route{ >>> xlog("L_ERR", "request method: [$rm], from URI: [$fu], To URI: >>> [$tu]"); >>> if (!mf_process_maxfwd_header("10")) { >>> sl_send_reply("483","Too Many Hops"); >>> exit; >>> }; >>> >>> if (msg:len >= 4096 ) { >>> sl_send_reply("513", "Message too big"); >>> exit; >>> }; >>> >>> # >>> ----------------------------------------------------------------- >>> # NAT detection >>> # >>> ----------------------------------------------------------------- >>> route(2); >>> >>> >>> # >>> ----------------------------------------------------------------- >>> # if you want to make ENUM work with numbers starting with "00", >>> # use the following to convert "00" it into a "+" >>> # >>> ----------------------------------------------------------------- >>> >>> >>> if (uri=~"^sip:00[1-9][0-9]*@") { strip(2); >>> prefix("+"); >>> }; >>> if (uri=~"sip:\+[0-9]+@") { >>> if(!enum_query("e164.arpa.") && !enum_query("e164.org.") && >>> !enum_query("e164.info.") && !enum_query("enum.org.")) >>> { >>> xlog("L_ERR", "Invalid enum? $uri >>> Route to PSTN"); >>> }; >>> xlog("L_ERR","we matched just number $tu"); >>> strip(1); # remove + >>> t_relay(); >>> exit; >>> }; >>> # >>> ----------------------------------------------------------------- >>> if (!method=="REGISTER") >>> record_route(); >>> >>> if (loose_route()) { >>> # mark routing logic in request >>> append_hf("P-hint: rr-enforced\r\n"); route(1); >>> }; >>> >>> # ----------------------------------------------------------------- >>> # Call Type Processing Section >>> # ----------------------------------------------------------------- >>> if (!uri==myself) { >>> route(1); >>> }; >>> >>> # ----------------------------------------------------------------- >>> if (uri==myself) { >>> >>> if (method=="REGISTER") { >>> if (!www_authorize("taipei.example.net", "subscriber")) { >>> www_challenge("taipei.example.net", "0"); >>> exit; >>> }; >>> >>> if (isflagset(5)) { >>> setflag(6); >>> }; >>> >>> save("location"); >>> exit; >>> }; >>> >>> lookup("aliases"); >>> if (!uri==myself) { >>> append_hf("P-hint: outbound alias\r\n"); >>> route(1); >>> }; >>> >>> if (!lookup("location")) { >>> sl_send_reply("404", "Not Found"); >>> exit; >>> }; >>> append_hf("P-hint: usrloc applied\r\n"); >>> if (method=="INVITE") { >>> if (uri=~"^sip:6.+@") { >>> rewritehostport("xx.xx.xx.xx:5060"); >>> xlog("L_ERR", "transfer to Asterisk"); >>> route(1); >>> exit; >>> }; >>> }; }; >>> >>> route(1); >>> } >>> >>> >>> route[1] { >>> if (subst_uri('/(sip:.*);nat=yes/\1/')){ >>> setflag(6); >>> }; >>> >>> if (isflagset(5)||isflagset(6)) { >>> route(3); >>> } >>> if (!t_relay()) { >>> sl_reply_error(); >>> }; >>> exit; >>> } >>> >>> >>> route[2]{ >>> force_rport(); >>> if (nat_uac_test("19")) { >>> if (method=="REGISTER") { >>> fix_nated_register(); >>> } else { >>> fix_nated_contact(); >>> }; >>> setflag(5); >>> }; >>> } >>> >>> >>> route[3] { >>> if (is_method("BYE|CANCEL")) { >>> unforce_rtp_proxy(); >>> } else if (is_method("INVITE")){ >>> force_rtp_proxy(); >>> t_on_failure("1"); >>> }; >>> if (isflagset(5)) >>> search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); >>> t_on_reply("1"); >>> } >>> >>> failure_route[1] { >>> if (isflagset(6) || isflagset(5)) { >>> unforce_rtp_proxy(); >>> } >>> } >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> --- >> avast! Antivirus: Inbound message clean. >> Virus Database (VPS): 0641-3, 2006/10/12 >> Tested on: 2006/10/13 ¤U¤È 05:15:28 >> avast! - copyright (c) 1988-2006 ALWIL Software. >> http://www.avast.com >> >> >> >> > >

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[Serusers] change authorization packet format
by Sagar 14 Oct '06

14 Oct '06

Hi Guys , I am using ser-0.8.14 with free radius . I have a small issue with my authorization packet presently i am getting authorization packect on radius as bellow NAS-IP-Address = '192.168.3.78' User-Name = '215412145(a)192.168.3.78' Service-Type = '15' NAS-Port = '5060' call-id = '44e5770-1745420b(a)192.168.218.9' but i need this packe in the format given bellow NAS-IP-Address = '192.168.3.78' User-Name = '215412145' Service-Type = '15' NAS-Port = '5060' call-id = '46cf5bdb-13c4-16-5869-2204' Can anyone please help me from where i can change this packe format . Thanks in advance. Vidhya sagar dixit --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

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[Serusers] documentation for SEMS (cc from sems)
by Stefan Sayer 14 Oct '06

14 Oct '06

Hello, I am happy to announce that there will be a whole new set of documentation available for SEMS, and the first two documents are already online: A design overview describes the design of SEMS and important structures, while the application developers tutorial explains step by step how to write applications within the SEMS framework, both in C++ and Python, from the empty template up to a calling card application (with sample config). All documentation is available through the SEMS homepage http://www.iptel.org/sems/ . The documentation effort has been made possible thanks to iptego GmbH. iptego stands behind SEMS, we think that it is and will be a good platform to create voice services. We want to get a larger user, and hopefully also developer base. Thus the first step is to improve documentation; this will be followed by more stability and monitoring improvements. Yes, we like SEMS. Stefan -- Stefan Sayer Software Developer iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de

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[Serusers] OT: job offers for SIP and open source gurus
by michel Carette 13 Oct '06

13 Oct '06

Sorry for being out of topic... My client, a leading international VoIP carrier, is looking to hire two VoIP gurus for a permanent role based in Brussels (Belgium). One role would be for technical customer support, and the other role would be for developments. If any interest, please respond to my email mcarrette8(a)yahoo.fr and I will forward more details. Thanks, Michel ___________________________________________________________________________ Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Demandez à ceux qui savent sur Yahoo! Questions/Réponses http://fr.answers.yahoo.com

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[Serusers] SEMS is definitely back in business!
by Greger V. Teigre 13 Oct '06

13 Oct '06

This is a teaser on SEMS 0.10.0 (http://iptel.org/sems), not an official announcement... New documentation is in the process of being developed, but I find the ongoing things so exciting that I just wanted to share this with you. g-) ----------------------------- Some of you have looked at SEMS before, others probably don't know what it is. The answer is that if you cannot do it in SER, you can probably do it in SEMS :-) Here is a list (from README): * voicemail: records voice messages and mail them to the callee. * conferencing: enables many people to talk together at the same time. * announcement: plays an announcement. * echo: test module echoing your voice. * mailbox: saves voicemails into an IMAP server. Users can dial in to check their messages * ann_b2b: plays ann announcement before connecting the callee in b2bua mode * conf_auth: collect a PIN number, verify it against an XMLRPC authentication server and connect in b2bua mode * early_announce: announcement for early media (183) To those of you who looked at SEMS, but found it lacking, it is time to have another look... A lot of new functionality has not been added, but the basic tools for easily creating your own applications are now there. This is from a recent update Stephan did to the WHATSNEW file: -------------------- What is new in SEMS version 0.10.0 Between 0.9.0 (CVS) versions and 0.10.0, quite a lot has changed. Almost 50% of the code has been rewritten: the design has been simplified a lot, and to make a slim, clean core a lot of functionality has been dropped. Instead, for the core we just focus on the essentials: basic signalling, session and media handling, and loading plugins. An inter-plugin API ("DI-API") has been introduced, such that functionality can be added using plugins, everybody can implement their favorite functionality as a reusable plug-in, and applications can be built in a modular manner. A new kind of modules, session component plugins, can even modify the basic signaling behaviour, the session timer plugin is the first one to use this. Major additional changes: * Interface to Ser has been rewritten. * Application plug-in interface has been partially rewritten. Applications are now exclusively event driven and asynchronous. * Media is processed by one thread for all sessions, improving the performance extremely due to less task-switching * Back-to-back User Agent (B2BUA) functionality has been added. * IVR python code has been completely rewritten: Applications are now developed in the IVR like their C++ counterparts * Session-Timer has been added (as module), replacing the ICMP watcher * Adaptive playout buffer has been added * Audio processing simplified

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[Users] WARNING: run_actions: null action list (rec_level=1) ????
by Ronald Wiplinger 13 Oct '06

13 Oct '06

I find in my log file: WARNING: run_actions: null action list (rec_level=1) What does it mean? Where does it come from? bye Ronald Wiplinger

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sr-users October 2006