sr-users January 2010

sr-users@lists.kamailio.org

73 participants
128 discussions

[Kamailio-Users] kamailio and sip-router at FOSDEM 2010 - social event
by Henning Westerholt 21 Jan '10

21 Jan '10

Hello, in about two and a half week the annual open source developer conference FOSDEM (http://fosdem.org/2010/) takes place in Brussels. As in the last year we would like to meet here for a social event, probably a dinner on Saturday evening, 6th February. This developers already confirmed their participance: - Daniel-Constantin Mierla - he is also giving a talk about Kamailio 3.0.0 on Saturday: http://fosdem.org/2010/schedule/events/kamailio_sip_server - Elena-Ramona Modroiu - Marius Zbihlei - 1&1 kamailio and sip-router development Bucharest - Timo Reimann - 1&1 kamailio and sip-router development Karlsruhe - Henning Westerholt Participation at the event is free of charge, everybody pays for its own expenses. In order to organize a restaurant and do a reservation please send me a short mail if you would like to attend. Suggestions for a nice place with good food and beer are also welcome. Looking forward to see you in Brussels. Best regards, Henning P.S.: Photos of the last year event can be found on our web site: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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[Kamailio-Users] Question about Timer, i need milliseconds
by Marco Bungalski 20 Jan '10

20 Jan '10

Hello all :-) I have a question about how to get a time-stamps with milliseconds. To create my own CDR's, i write to my databases timestamps, with a command like this: sql_query("ca", "UPDATE active_calls SET state = 'BILLING', end_time=$Ts WHERE call_id='$ci'", "ra"); I use the "$Ts"-variable, but with this i can get only "full" seconds. But i need milliseconds for my billing-system. Otherwise i have differences between my CDR's and my carrier's one. Example: A call has a duration of 0.2 seconds. In my CDR start- and endtime is the same (=0 seconds), to my carrier a have 1 second to pay ;-) Is there a possibility to get a time-stamp or time with milliseconds? Cheers, Marco Marco Bungalski GmbH Traversale 5 27283 Verden Telefon: +49 4231 - 776 9999 Fax: +49 4231 - 776 9998 Mobil: +49 172 4204774 e-mail: Marco(a)Bungalski.de web: www.bungalski.de Geschäftsführender Gesellschafter: Marco Bungalski Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586

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[Kamailio-Users] 3G+ and LTE forum - meeting in Berlin
by Daniel-Constantin Mierla 20 Jan '10

20 Jan '10

Hello, I will be in Berlin for few days (Jan 20-22), attending 3G+ and LTE Forum. If anyone is around and want to meet, feel free to drop me an email. Next event I will talk about Kamailio and SIP Router is FOSDEM in Brussels -- more details will follow soon. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] BYE packet is not received in siptrace module !!
by Dinesh Gautam 20 Jan '10

20 Jan '10

Dear all, I am using siptrace module and getting all the packets except the BYE. when the call disconnected BYE packet is not received. I am also getting the packets OK, REGISTER, INVITE two times in sip_trace table. Can anyone help me to sort out this problem?? -- Dinesh Gautam !!

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[Serusers] deactivate the Via field
by "Martínez García, Rebeca" 20 Jan '10

20 Jan '10

Hello all, I know it might sound strange but, is there any way to deactivate that SER puts its Via on the SIP message when forwarding it to other entity? Thanks in advance, Rebeca Martinez

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Re: [Kamailio-Users] [kamailio-users] Implement kamailio sever failover
by Klaus Darilion 20 Jan '10

20 Jan '10

ram schrieb: > in that case we are going to loose registration right ? not if you do database replication of "location" table > and ongoing calls will be dropped right ? No. only ongoing transactions may be affected. regards klaus PS: please always Cc the mailing list > > Ram > > > > On Sat, Jan 16, 2010 at 2:58 AM, Klaus Darilion > <klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> wrote: > > Hi! > > For failover there are several possibilities. The simplest on, I > guess the one you are referring to, is IP takeover. > > This is: 2 servers. On both is Kamailio and the database (e.g. > mysql) installed. The SIP service uses a virtual IP address which is > either configured at server 1 or server 2 (never on both). Kamailio > process is only running on the server which has the virtual IP > address provisioned. > > Then you need some other software to monitor the SIP service and > trigger the failover from one server to the other server (e.g. > either write some shell scripts or use existing software like > hearbeat, ultramonkey, ...) > > Further - depending on your scenario - you might need some database > replication (e.g. mysql cluster or mysql with master-master > replication). > > regards > klaus > > RAJNIKANT VANZA wrote: > > Hi, > > I want to implement kamailio server failover. > so, anybody know about how to configure kamailio server failover? > > e.g. 2 kamailio severs run and when any one server fail then > other server take over thats resources. > > > -- > Best Regards, > > Rajnikant Vanza > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > >

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[Kamailio-Users] no answer to invite
by Anders 19 Jan '10

19 Jan '10

Hi, I have a problem that I don't understand at all (yet another one ;-)). One client has Asterisk (like many others) from where he sends calls to our Kamailio. It worked fine, until it suddenly stopped for no apparent reason on Friday. No changes in configuration on his or our side. We can ping his server just fine, and vice versa. On any INVITE he sends, he gets no, zero, zip response from Kamailio on our server. I wanted to figure out why, so I put an 'ngrep' on port 5060 for his IP on our Kamailio server, and I got...nothing. No communication at all on this port containing this IP. Why? I have no clue. The ping should show that we can communicate, and a 'ngrep' on his server shows that he sends plenty of INVITEs, but gets absolute zero response from our Kamailio box. Any ideas where to look? Thanks a lot in advance!

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[Kamailio-Users] siremis on sourceforge.net
by Elena-Ramona Modroiu 19 Jan '10

19 Jan '10

Hi, the sources of siremis are in sourceforge.net svn. If any of you has developed (plan to develop) extensions, contact me to grant devel access. Also, patches should be sent against latest svn trunk (no branches so far): http://sourceforge.net/projects/siremis/ The svn repo does not store the openbiz core framework -- you have to get it from siremis-1.0.0.tgz (the openbiz dir) or download it from openbiz project and apply the patch from misc directory (note that you also have to download zend and others according to openbiz guides). Once you fetch siremis svn and add openbiz, run make to create the symlinks required for web dirs and you are set. Regards, Ramona

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[Kamailio-Users] SIREMIS v1.0.0 Released
by Elena-Ramona Modroiu 18 Jan '10

18 Jan '10

Hi, I have just released SIREMIS 1.0.0 - web admin interface for Kamailio (OpenSER) 3.0.0 and SIP Router. Among new features: * presence service management – watchers, active watchers and presentity * sip trace view – access sip_trace module records, highlight headers and group by call-id * user location statistics charts – online user agents, supported SIP methods, contacts and nat related charts * update of the core components to latest stable versions * update to work with latest Kamailio (OpenSER) v3.0.0 * patch to work with older Kamailio (OpenSER) v1.5.x * time-based X-axis for charts print labels as HH:MM * charts types supported: line, line dot and area Some screenshots specific for this version: http://www.asipto.com/gallery/v/siremis/siremis_22.png.html http://www.asipto.com/gallery/v/siremis/siremis_23.png.html http://www.asipto.com/gallery/v/siremis/siremis_24.png.html Download and installation steps: http://siremis.asipto.com/install/ More screenshots: http://www.asipto.com/gallery/v/siremis/ Demo site (it works on a database with random data, username: admin, password: admin): http://siremis.asipto.com/demo/ Regards, Ramona

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[Kamailio-Users] setbflag not working?
by Ricardo Martinez 18 Jan '10

18 Jan '10

Hello. I have the next situation with kamailio 3.0.0 In my config file I have something like this : route If INVITE then route(AUTH_REQUEST) . . . route[AUTH_REQUEST] . . . if (client_nat_test("3")) { append_hf("P-hint: route(AUTH_REQUEST)|setflag7,forcerport,fix_contact\r\n"); if (setbflag("7")) { xlog("L_INFO","[$ci]:[AUTH_REQUEST] : setbflag(7)"); } force_rport(); fix_contact(); xlog("L_INFO","[$ci]:[AUTH_REQUEST] : CLIENT_NAT_TEST-3"); }; . . . route(IN_TO_IN) route[IN_TO_IN] . . . route(LOCAL_CALL) . . . route[LOCAL_CALL] . . . if (isbflagset("7") ) { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET begin"); } else { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NO SET begin"); } if ( !lookup("location") ) { if ( avp_check("$avp(s:4)", "eq/s:1/g") ) { if( !avp_pushto("$br", "$avp(s:5)/g") ) { sl_send_reply("403","Forbidden - Forward Number NULL"); exit; }; prefix("222"); route(TO_VOICEMAIL_NODISP); exit; }; if ( avp_check("$avp(s:3)", "eq/s:1/g") ) { xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm - From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL ON\n"); route(TO_VOICEMAIL_NODISP); exit; } else { sl_send_reply("480", "Usuario fuera de Servicio"); xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm - From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL OFF\n"); exit; }; }; if (isbflagset("7") ) { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET end"); } else { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NOT SET end"); } . . . For some reason in the route[LOCAL_CALL] the flagb "7" is changed from "SET" to "NOT SET". According to my debugs this unset is made in the "lookup(location)" part. But I don't understand why. Can someone help me here? Thanks in advance, Ricardo Martinez.-

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sr-users January 2010