- sr-users - kamailio.org

[Kamailio-Users] Asterisk 403 Forbidden error with port translation
by Vikram Ragukumar 21 Jan '10

21 Jan '10

Hello, I have made some progress since my previous post, but not enough :). ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the outbound proxy for the softphone(Outbound proxy a.b.c.d:9090). Authentication credentials for the softphone match the user registered in asterisk's sip.conf. F.W runs Kamailio and rtpproxy, with Kamailio listening on port 5060. The asterisk server is setup to listen on port 5060. The Firewall(F.W), uses a libnetfilter_queue based program to : (a) Rewrite the destination port 9090 as 5060, and rewrite all other occurrences of 9090 as 5060 in the SIP message, for packets from the softphone to the asterisk server. (b) Rewrite the source port 5060 as 9090, and rewrite all other occurrences of 5060 as 9090 in the SIP message, for packets from the asterisk server to the softphone. The following exchange of SIP messages take place -Sip softphone sends a REGISTER message to asterisk -Asterisk responds with a 401 UNAUTHORIZED -Sip softphone replies with a REGISTER message containing auth. info. -Asterisk responds with a 403 FORBIDDEN : BAD AUTHORIZATION The above setup works when the softphone uses port 5060, so there problem here does not have anything to do with Authorization credentials. Is it possible i might be modifying parts of the packet that shouldn't be modified or i might not be modifying some relevant parts of the packet ? Thanks in advance, Vikram.

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[Serusers] Hello , Help, and Thanks
by Justin 21 Jan '10

21 Jan '10

Hi, I have just begun to set up my soft phone, and am currently reading through your How to's Ser text's ,,, but I keep getting error 401-403-404-408 all doing with I am entering the wrong DOMAIN and PROXY for the Iptel.org SIP service. If you could contact me with a solution. Much Thanks (dark4ajaguar(a)iptel.org)

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[Kamailio-Users] Script variables Question
by alex pappas 21 Jan '10

21 Jan '10

Dear friends, When a sip 4rU is entering my kamailio script I'm cutting down the prefix of this customer and added to a script variable. In the same same route after i do some checks then i add back the customer prefix which I had kept in the script variable. The problem: Some times the call goes to failure Route for rerouting and then i try to add the prefix again but the is empty. How can I transfer a variable from route to route and be a unique variable for each call? Thank you Alex

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[Kamailio-Users] concurrent dialogs
by alex pappas 21 Jan '10

21 Jan '10

Dear Friends, I have the following question: When I'm running the command *kamctl fifo dlg_list* I'm getting the concurrent dialogs in that moment. When I disable the sip_trace module(which I don't see how they are related) I get NO output when I run the same command. Does anyone knows on why this is happening ? Alex

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[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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[Serusers] SER 0.9.7-pre4 and RPID
by Steven C. Blair 18 Jan '10

18 Jan '10

I'm trying to append an RPID header on my SER 0.9.7-pre4 system. The config has not changed and this use to work in SER 0.9.7-pre3. The header does not get appended and I cannot see why. The use_rpid parameter or the auth_db module does not seem to exist in this release. Is there a different way to append the header? Thanks,Steve

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[Kamailio-Users] automatic packaging building of master
by Henning Westerholt 18 Jan '10

18 Jan '10

Hi, we've updated the automatic packaging building process on the devel machine. This builds now every night from the git master packages for debian lenny, squeeze and sid. Hints about your archive configuration in order to use the packages can be found on http://devel.kamailio.org/ They are build from the control files in pkg/debian, so at the moment some packages with special dependencies are missing, and they are named with 'ser-$name'. The only thing that is modified at the moment is the version number. This packages are of course only for tests and not for production use. Please let me know if there are any problems with the building service. Henning

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[Kamailio-Users] %23 remove from rU
by alex pappas 18 Jan '10

18 Jan '10

Dear Friends, Do you have any idea on how to remove from $rU the %23 when is send in to my network? The scenario is: The customer will send (prefix # number). This is translated in sip in (prefix %23 number). I need to remove this in order to check against PDT module. Thanks Alex

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[Kamailio-Users] Kamailio 3.0 Crash when load with module siptrace.
by Quang Tri 18 Jan '10

18 Jan '10

Hello, When i try with module siptrace.so This is loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dialog.so" #!ifdef WITH_CALLCONTROL loadmodule "call_control.so" #!endif #!ifdef WITH_MEDIAPROXY loadmodule "mediaproxy.so" #!endif #!ifdef WITH_SIPTRACE loadmodule "siptrace.so" #!endif .... and with # ----- sip trace ------- #!ifdef WITH_SIPTRACE modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 2) #!endif and this is error log 8(7163) DEBUG: tm [t_lookup.c:764]: DEBUG: t_lookupOriginalT: searching on hash entry 60285 8(7163) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(7163) DEBUG: tm [t_lookup.c:854]: DEBUG: t_lookupOriginalT: no CANCEL matching found! 8(7163) DEBUG: tm [t_lookup.c:856]: DEBUG: t_lookupOriginalT completed 8(7163) DEBUG: tm [tm.c:952]: lookup_original: t_lookupOriginalT returned: (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [receive.c:280]: receive_msg: cleaning up *17(7186) : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 14* 17(7186) DEBUG: <core> [tcp_main.c:3038]: DBG: handle_ser_child: dead child 5, pid 7160 (shutting down?) 17(7186) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x81ede80, 14, -1, 0x0) fd_no=22 called 0(7151) ALERT: <core> [main.c:722]: child process 7160 exited by a signal 11 0(7151) ALERT: <core> [main.c:725]: core was generated 0(7151) INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD 6(7161) INFO: <core> [main.c:788]: INFO: signal 15 received 7(7162) INFO: <core> [main.c:788]: INFO: signal 15 received 16(7185) INFO: <core> [main.c:788]: INFO: signal 15 received 15(7183) INFO: <core> [main.c:788]: INFO: signal 15 received 8(7163) INFO: <core> [main.c:788]: INFO: signal 15 received 14(7179) INFO: <core> [main.c:788]: INFO: signal 15 received 9(7164) INFO: <core> [main.c:788]: INFO: signal 15 received 10(7165) INFO: <core> [main.c:788]: INFO: signal 15 received 13(7176) INFO: <core> [main.c:788]: INFO: signal 15 received 1(7155) INFO: <core> [main.c:788]: INFO: signal 15 received 17(7186) INFO: <core> [main.c:788]: INFO: signal 15 received 2(7156) INFO: <core> [main.c:788]: INFO: signal 15 received 3(7158) INFO: <core> [main.c:788]: INFO: signal 15 received 4(7159) INFO: <core> [main.c:788]: INFO: signal 15 received 11(7171) INFO: <core> [main.c:788]: INFO: signal 15 received 12(7174) INFO: <core> [main.c:788]: INFO: signal 15 received 0(7151) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(7151) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(7151) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(7151) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(7151) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(7151) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy 0(7151) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory lock 0(7151) DEBUG: <core> [main.c:741]: terminating due to SIGCHLD How can fix it. Thank you.

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[Kamailio-Users] Start/Stop Kamailion
by juergen＠glowka.de 18 Jan '10

18 Jan '10

Hi Dinesh do as Daniel already told You, or use the start/stop-script kamailion.init in the folder debian/ after unpacking the sourcepacket and change it according to Your system. Copy/rename it to /etc/init.d/kamailion check for errors: /usr/loca/sbin/kamailio -c Juergen (who likes india:)

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add_contact_alias
by Klaus Darilion 17 Jan '10

17 Jan '10

Hi Juha! Does the alias functions also support IPv6 addresses? regards klaus

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[Kamailio-Users] [kamailio-users] Implement kamailio sever failover
by RAJNIKANT VANZA 16 Jan '10

16 Jan '10

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

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[Kamailio-Users] How to enable all sip_trace messages?
by Dinesh gautam 16 Jan '10

16 Jan '10

Dear all, I have enabled the sip_trace module and able to log the messages ACK, INVITE, REGISTER , SUBSCRIBE. I want to log other messages like RING, OK, BYE etc. Please help me out how can I enable theses messages. -- With Regards, Dinesh Kumar Gautam

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Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 15 Jan '10

15 Jan '10

Authentication in my upstream server is a MUST in my case and I cannot disable it, sorry. In fact I would be going against the standards for what I am configuring if I disabled authentication. I only wanted SER for the special case of internal calls inside a private LAN, but I am realising it is not as simple as I though it would be. In any case, thanks for the support. Regards, Rebeca Martínez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 15/01/2010 12:06 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Not that I know of... Anyway, we had the same problem....but solved it by disabling authentication on the upstream server. After all...you are now in a "trusted zone"....you have successfully "auth challenged" on the SER...so no need to do it twice....SER "IS" the Registrar, thats what it is designed to be....an architecture with 2 Registrars is not good. If you can turn off auth on the upstream server then this is best. When you t_replicate Registers SER just ignores the 200ok back from the upstream server....after all it is just sending the register as a courtesy gesture. Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 15 January 2010 10:54 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Rupert you were right, SER does not permit to save the location based on the reg of a different upstream server; which means that I have to make "save_contacts("location") when SER receives the REGISTER. However, as it then sends directly to my client a 200 OK, when the client receives the "401 Unauthorized" error from my server, it does not authenticate with it because it has received a previous 200 OK (from SER...). So, here is my question: Is it possible to make SER not send a 200 OK when saving the location of the user? (I am only asking because I do not know). Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: lun 11/01/2010 12:37 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Rebeca, i am not an expert, but if I had to guess...then i would guess it would be because there is no contacts in the 200 to a Register. In our ser.cfg we save("location"); after testing if (method=="REGISTER") { but I can see that you are making registration and saving of info into location dependent on successful reg on a different upstream server. I do not think that is possible because you need to make the decision to save into location based on the reg that SER has received...and not based on the response of an upstream element....but if I am wrong I would be interested in the answer... Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 11 January 2010 11:21 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

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Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 15 Jan '10

15 Jan '10

Hi all, Rupert you were right, SER does not permit to save the location based on the reg of a different upstream server; which means that I have to make "save_contacts("location") when SER receives the REGISTER. However, as it then sends directly to my client a 200 OK, when the client receives the "401 Unauthorized" error from my server, it does not authenticate with it because it has received a previous 200 OK (from SER...). So, here is my question: Is it possible to make SER not send a 200 OK when saving the location of the user? (I am only asking because I do not know). Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: lun 11/01/2010 12:37 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Rebeca, i am not an expert, but if I had to guess...then i would guess it would be because there is no contacts in the 200 to a Register. In our ser.cfg we save("location"); after testing if (method=="REGISTER") { but I can see that you are making registration and saving of info into location dependent on successful reg on a different upstream server. I do not think that is possible because you need to make the decision to save into location based on the reg that SER has received...and not based on the response of an upstream element....but if I am wrong I would be interested in the answer... Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 11 January 2010 11:21 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

1 0

[Serusers] Max-Forwards missing in SER CANCEL during forking
by Andres 15 Jan '10

15 Jan '10

Hi, We encountered an issue in 0.9.6 when SER forks requests because there are several endpoints registered with the same account (ie office and home phones). When one of the endpoints answers the call, SER sends a CANCEL to the other endpoints so they can stop further call processing. It turns our that this SER generated CANCEL does not contain the "Max-Forwards" header which apparently violates RFC3261. Here is an example: U SER IP:5060 -> 10.10.1.50:36679 CANCEL sip:3055551234@10.10.1.50:36679;rinstance=C926AD943D258534F5F62913216C50ADC9D895D2; transport=udp SIP/2.0..Via: SIP/2.0/UDP 10.10.1.21:5065;branch=z9hG4bK1c1e.32eeb702.1.. From: "9545551234" <sip:9545551234@10.10.1.20:5066>;tag=as7d1243c9.. Call-ID: 4385c66230a310e714f020926aa2028c(a)10.10.1.20.. To: <sip:3055551234@X.X.X.X:5065>..CSeq: 102 CANCEL.. User-Agent: Sip EXpress router(0.9.6 (i386/linux)).. Content-Length: 0.... Is there any fix or workaround for this? How can we get that "Max-Forwards" header in the Message if it is internally generated by SER and no something that flows through the config script. Thanks, Andres http://www.telesip.net

2 3

[Kamailio-Users] mk_proxy: could not resolve hostname
by kévin l'heveder 14 Jan '10

14 Jan '10

Hi, I'm using kamailio V3.0 In the kamailio.cfg when I use the "send" function with an IP inside (as send("127.0.01:51001"), I see the following error : mk_proxy: could not resolve hostname: "127.0.01:51001" At first, kamailio was sending request to the DNS server which define in resolv.conf to resolve "127.0.01:51001" The DNS could not resolve this name then the error occured So I removed this defined DNS and now no request are sending but still have the same error. I tried to set dns and use_dns_search_list parameters to no, but that's change nothing Anybody have any ideas? Regards, Koon

3 2

[Serusers] ser scalability
by Sabatella, Michael 14 Jan '10

14 Jan '10

I just wanted to run this by the group. I did a complete overhaul on one of my ser environments and upgraded the physical server and all software around my sip proxies. I am also maintaining a complete front end for provisioning and billing that I also moved(PHP/mysql) I am running this on a Virtual machine (vmware esxi). If I need more proxies I can clone the VM and build in new service/capacity pretty fast. I am running 0.9.6 on Redhat ES 5. I have 4 cores dedicated to this server running Xeon 2.93GHz processers and 4 GIG memory. I am running the 32 bit OS. I hardly see the processor tick up to 1%. The machine I am testing has about 300 registered endpoints in the proxy db. My questions are: 1. What is the scalability of ser processing calls? 2. Are there any limits on the number of users that can be registered at one time. Is the registration memory based and am I severely limited with 4GB of memory? 3. Is there anyone that has 1000s of registration to the proxy and if so I would like to hear how you are scaling it. What type of server are you using? 4. What are some good metrics to look at on the sip proxy to gauge its performance? 5. Is there a way to capture calls per minute? Is anyone doing this? I have call accounting set up. Thanks, Michael ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

2 1

[Kamailio-Users] 64kbit, quad/dual core support
by mbsip 14 Jan '10

14 Jan '10

Hi all, I am wondering if kamailio do support 64-bit OS systems and dual/quad core cpu platforms? The question is raised because i am about to prepare hardware specification for kamailio server. Thanks in advance, Maciej.

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Re: [Kamailio-Users] about kamailio with sctp
by Daniel-Constantin Mierla 14 Jan '10

14 Jan '10

Hi Bruce, On 1/14/10 12:54 PM, Bruce McAlister wrote: > Hi Daniel, > > >> you may ask this one on sr-dev(a)lists.sip-router.org since the devel of >> SCTP use to be more active there. >> >> > Thanks, I will ask there too. > > >> When I implemented in in old kamailio, it was no option for me than >> using kamailio-to-kamailio SCTP channel. >> >> > When you did the testing between the two kamailio servers, how did you > enforce SCTP as the transport? > > Does t_relay offer a way to specify UDP/TCP/SCTP as the transport to relay > the messages between servers? > > if you set dst uri with transport parameter then t_relay() will use it, e.g.: $du = "sip:10.0.0.1;transport=sctp"; t_relay(); Or you can use t_relay_to_sctp() or t_relay_to() with appropriate parameters: http://kamailio.org/docs/modules/3.0.x/modules/tm.html Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

Re: [Kamailio-Users] about kamailio with sctp
by Daniel-Constantin Mierla 14 Jan '10

14 Jan '10

Hello, On 1/14/10 2:25 AM, Bruce McAlister wrote: > Hi Klaus, > > >> I do not know. For kamailio 1.5 I would just grep for SCTP in the >> sources to get the answer. >> For kamailio 3.0 (based on sip-router with a new (I guess much better) >> SCTP implementation there is probably another method needed. A cite from >> another email from Andrei: >> >> >>> Yes, make cfg SCTP=1 and then make all will compile with SCTP support. >>> I'm currently pondering whether or not I should make the SCTP support >>> automatic (if the needed *.h files are installed compile it >>> > automatically). > > I am compiling Kamailio 3.0.0 on Solaris 10 which does have SCTP support > inherent in the OS. I did a 'grep -i sctp Makefile*' and saw that there were > checks for SCTP in the makefiles. > > I ended up building Kamailio 3.0.0 with the following command line: > > make prefix=/opt/kamailio \ > SCTP=1 \ > CC_EXTRA_OPTS=-I/usr/gnu/include \ > group_include="standard postgres presence" \ > include_modules="snmpstats perl tls" \ > all > > The build went through successfully and I can see that sctp_server.o has > been linked into the kamailio binary when the build is running. > > Does anyone know of any tool that I can use to test Kamailio using the SCTP > protocol? > you may ask this one on sr-dev(a)lists.sip-router.org since the devel of SCTP use to be more active there. When I implemented in in old kamailio, it was no option for me than using kamailio-to-kamailio SCTP channel. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

2 1

[Kamailio-Users] running Qjsimple with kamailio 1.5 + tls
by mustafa samara 14 Jan '10

14 Jan '10

Hello i am trying Qjsimple 0.6.3 with Kamailio 1.5 with tls support, i make the user register successfully but when i am trying to invite another user i got the following error: " transport_srtp Failed generating random key: unspecified failure pjsua_call.c Error initializing media channel: Unknown error 259800 [status=259800] Dialog destroyed QjSimple: Error calling buddy: Unknown error 259800 [status=259800] " the sip setting as the following: protocol: tcp srtp : mandatory srtp requirements: tls also i got the same error if the sip setting as the following: protocol: tls srtp : mandatory srtp requirements: tls or: protocol: tls srtp : mandatory srtp requirements: sips any body help me please? -- Eng.Mustafa Al-Samara

2 2

Re: [Kamailio-Users] about kamailio with sctp
by Klaus Darilion 14 Jan '10

14 Jan '10

Bruce McAlister schrieb: > Hi All, > > I am wanting to play with Kamailio 3.0.0 and SCTP as well. I just want to > verify that I am building it in to Kamailio properly. When I build Kamailio > do I need to have USE_SCTP=1 as an option on the make command line to enable > SCTP support? > > I can see there are updates in the ChangeLog for SCTP but I cannot find how > to enable support for it in the INSTALL/README-MODULES documentation. > > Can someone point me in the right direction please. I do not know. For kamailio 1.5 I would just grep for SCTP in the sources to get the answer. For kamailio 3.0 (based on sip-router with a new (I guess much better) SCTP implementation there is probably another method needed. A cite from another email from Andrei: > Yes, make cfg SCTP=1 and then make all will compile with SCTP support. > I'm currently pondering whether or not I should make the SCTP support > automatic (if the needed *.h files are installed compile it automatically). regards klaus > > Thanks > Bruce > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 11 January 2010 09:57 > To: mustafa samara > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] about kamailio with sctp > > Hello, > > On 1/11/10 1:37 AM, mustafa samara wrote: >> Hello >> i do not know how to try sctp, imake kmailio 1.5 with sctp=1 >> could you help me? > if you want to play with sctp then I recommend to use 3.0.0 (to be > released today). It has far more improvements and new features than > older versions to SCTP. You can take it from git now: > http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git > > Some highlights about sctp and 3.0.0: > http://by-miconda.blogspot.com/2010/01/best-of-new-in-kamailio-300-15-sctp.h > tml > > Cheers, > Daniel >

2 1

[Kamailio-Users] Rtpproxy not recording media stream
by Vikram Ragukumar 13 Jan '10

13 Jan '10

Hello, I have setup Kamailio(v1.5) and Rtpproxy(v1.2.1) on a Centos5.4 linux box. Kamailio's config file has been edited to include Nat and mysql support. With Kamailio running, i started rtpproxy using rtpproxy -l ip_addrs -s udp:127.0.0.1:7722 -a -r /var/log -F However rtpproxy does not seem to be recording any session. I read that the -a switch can be used to record all sessions. Am i missing a switch/doing something wrong while invoking rtpproxy ? Regards, Vikram.

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[Kamailio-Users] Logical operators
by Antonio Reale 13 Jan '10

13 Jan '10

Hi, I'm trying this: if(is_avp_set("$avp(s:avp)") && !isflagset(1) || is_avp_set("$avp(s:avp)") && !isflagset(2)) { ...; } The first condition before OR operator is never matched. Could you please suggest the correct syntax? Thank you very much. Regards. -- Antonio

2 4

[Kamailio-Users] SIP proxy is expecting "Route:" header to be filled information from the last "Contact:" header ?
by Julien Chavanton 13 Jan '10

13 Jan '10

Hi all, first thank you for maintaining this list and free/Open source SIP proxy software. I am working on one interco problem with a remote SIP proxy running on Kamailio. Cisco(U1) --> Opensips(P1) --> Internet --> Kamailio(P2) --> Asterisk(U2) The remote proxy(P2) is expecting us to fill the "Route:" header taking information from the "Contact:" header, according to my research the "Route:" header as to be filled using the information found in the last "Record-route:" header. Note that the "Contact:" header IP address is the one of U2, as it is the destination UA. I think our partner using Kalamailio only interconnected UA not Proxy, since they never faced this problem. Of course I do not suspect Kalamailio to be doing something wrong, I just want to validate what we should expect, and investigate further from there. I will have to contact the technicien that as configured the Kalamailio server to validate why face this behavior. -------------------------------------------------------------------------- " 16.12.1.3 Rewriting Record-Route Header Field Values " http://www.ietf.org/rfc/rfc3261.txt -------------------------------------------------------------------------- We found a temporary solution to make it worko, where we hardcoded the replacement of the IP address in the "Route:" header with the one of the U2 subst('/^Route: <sip:P2;(.*)$/Route: <sip:U2;\1/');

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[Kamailio-Users] Does uac_auth work for you?
by Andrey Utkin 13 Jan '10

13 Jan '10

Hi all. Please share your experience. If you ever tried authenticating outgoing calls with Kamailio, did it work for you? Which devices or servers you know do/don't accept such authentication? Do you use it in some tricky way to get it working?

4 3

[Kamailio-Users] about kamailio with sctp
by mustafa samara 13 Jan '10

13 Jan '10

Hello i do not know how to try sctp, imake kmailio 1.5 with sctp=1 could you help me? thnaks -- Eng.Mustafa Al-Samara

3 2

Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 12 Jan '10

12 Jan '10

Hi everybody, this ML was forgotten yesterday to be cc-ed ...for the sake of archiving, here it is the announcement. After ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

1 0

[Kamailio-Users] Checking the destination IP after dispatcher
by Geoffrey Mina 12 Jan '10

12 Jan '10

What pseudo variable would i check after running ds_next_dst() to check the IP we are about to forward the INVITE to? Basically I want to have a check for $si == [something] to ensure I am not about to send the INVITE to the same UI which just requested it. While I have your attention, I may as well see if anyone has a better solution for what I am trying to accomplish: I am doing this because asterisk can't handle a hairpin scenario and I need to generate two distinct calls in my network, so I figure i'll just forward the request to another server in the network to deal with this. INVITE from PSTN --> Kamailio --> Asterisk#1 [Asterisk#1 needs to create a new session, but stay on-net] --> INVITE from Asterisk#1 to Kamailio --> [Dispatcher] --> INVITE from Kamailio to Asterisk#2 This would basically create 3 legs of the call, which is what I want PSTN --> Asterisk#1 --> Asterisk#2 I would prefer if Asterisk#1 could just INVITE to LocalHost, but that doesn't seem to be allowed. Not sure if that's a limitation of SIP or Asterisk... Thanks!

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[Kamailio-Users] job offer: VoIP developer, team Bucharest
by Henning Westerholt 12 Jan '10

12 Jan '10

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional VoIP developer position open in the team in Bucharest, Romania at the moment. For more informations (requirements, how to apply..) please refer to the opening on the job web page. Some facts about our VoIP system can be found at e.g. [1], please feel free to contact me per mail as well. http://cariere.1and1.ro/xml/JobsDetail?job_id=2142 Best regards, Henning Westerholt [1] http://www.kamailio.org/events/2009-linuxtag/presentation.pdf -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] This has probably been asked a thousand times before...
by Robert B 12 Jan '10

12 Jan '10

Regardless, despite my Googlage, I can't seem to find what I am looking for. I've looked at the Wiki and the documentation. While I do have a programming background, I still am having a hard time grasping some of these tutorials. Many of them seem very specific and not at all relevant to what I am trying to do. I feel like I'm left without a good starting point. My goal is to have Kamailio setup as an ITSP would. That is, I have three sipxecs PBX systems here, and rather than have each one configured with umpteen SIP trunks that I'll need to manage, I want to have (for the time being) just one -- that one would be my personal Kamailio server. When a user makes a call, it first goes to my Kamailio server which then distribute the calls to my ITSPs (like Bandwidth.com, voip.ms, etc). I want the ACLs to be setup based on IP addresses and not user registrations. I want nothing to do with the media or RTP streams. First step would be termination (outbound). The obvious next step would be setting it up for origination (inbound). I've downloaded and installed the SIREMIS web UI for OpenSER 1.4.x, but as I said my lack of understanding of the terminology's context prohibits me from really exploring anything. Some areas look familiar, others seem redundant, etc. The reason I started with the web UI was because it's obviously easier and it also produces output that is syntactically correct for my own examination and learning purposes. What would the appropriate terminology for this kind of setup be? Can anyone point out a Wiki article which I may have missed? To me, this seems like the most basic of basic configurations and a seemingly obvious launching point for anyone getting into the software... Thanks! -- Robert

3 2

[Kamailio-Users] Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hi everybody, after ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

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[Kamailio-Users] preparing Kamailio 3.0.0 release
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hello, as announced previously, today will package the kamailio 3.0.0 release. Please commit any change to kamailio_3.0 branch before 14:00UTC to avoid overlappings. If you are aware of new bugs, just drop an email to devel mailing list. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [Kamailio-Users] Users Digest, Vol 56, Issue 19
by mustafa samara 11 Jan '10

11 Jan '10

thank you very much i will see kamailio 3.0 but is there any voip soft phone which support sctp, please? thanks

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Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 11 Jan '10

11 Jan '10

Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

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[Kamailio-Users] oracle sqlops URL
by alex pappas 11 Jan '10

11 Jan '10

Hi all, happy 2010 for all of us! Dear friends I'm trying to setup a sqlops connection to an Oracle db. I loaded the db_oracle module with (loadmodule "db_oracle.so") without parameters. The connection string that I gave in sqlops module is : modparam("sqlops","sqlcon","ca2=>user/pass@ip_addres:1521:db_name") When i try to start kamailio I get the following errors and kamailio does NOT start: ERROR:core:db_check_api: module db_user/pass@ip does not export db_use_table function ERROR:core:init_mod_child: failed to initializing module sqlops, rank 2 ERROR:core:main_loop: init_child failed for UDP listener Thank in advance and merry Christmas Alex

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[Kamailio-Users] about kamailio and tls
by mustafa samara 11 Jan '10

11 Jan '10

i am mustafa samara master degree student. i try to test qjsimple with kamailio openser is it possible (to test the tls support) ? also i want to ask about ( in sip preferences) what is the deference when we you use tls as a protocol or when we use (tls or sips) as a SRTP requirements. clould you help me please? i wat to include the result in my thesis -- Eng.Mustafa Al-Samara

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[Kamailio-Users] PDD
by Anders 09 Jan '10

09 Jan '10

Hi, Anyone with a smooth solution for tracking PDD of every call? Thanks!

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[Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 08 Jan '10

08 Jan '10

Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

3 2

[Kamailio-Users] Unreadable script
by Bruno Machado 08 Jan '10

08 Jan '10

Hi all I'm using the version 1.5.2 and I'd like to know from you if it is possible to 'hide' the config file. Using other words, becoming the file unreadable for us, humans. There are people that have root access to the machine and I don't want to let them read the script. Does Kamailio have a function to do this kind of job? tks, Machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Problems with avp_printf in Kamailio-3.0.0
by Santiago Gimeno 07 Jan '10

07 Jan '10

Hi, I'm getting errors when trying to use avp_printf in K-3.0.0. The script line, for example, is: avp_printf("$avp(i:905)", "$rU"); The error is: kamailio: DEBUG: <core> [sr_module.c:516]: find_export_record: <avp_printf> not found I've tried pv_printf with no luck either. This was working correctly in K-1.5 Could this be a bug? Best regards, Santi

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[Kamailio-Users] openser/kamailio subscribe / MWI
by Joel Webster 07 Jan '10

07 Jan '10

I am looking to MWI messages openser/kamailio and send the MWI message back based on a database lookup or other means within a routing block, when the users register though, i get errors like: Jan 3 12:07:20 ser1 /usr/local/sbin/kamailio[20317]: ERROR:presence:get_database_info: No matching subscription dialog found in database Jan 3 12:07:20 ser1 /usr/local/sbin/kamailio[20317]: ERROR:presence:handle_subscribe: getting stored info and when I disable the DB setup I had, I get a similar error: Jan 4 16:59:48 ser1 /usr/local/sbin/kamailio[5211]: ERROR:presence:get_stored_info: record not found in hash_table Jan 4 16:59:48 ser1 /usr/local/sbin/kamailio[5211]: ERROR:presence:handle_subscribe: getting stored info

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Re: [Kamailio-Users] [kamailio-users] Kamailio - Polycom Sound Point
by Ramu 07 Jan '10

07 Jan '10

Hi, Your configuration looks fine. Can you try with 2 X-Lite phones on different machines just to narrow down the problem? On Wed, Jan 6, 2010 at 4:29 PM, Chandrakant Solanki < solanki.chandrakant(a)gmail.com> wrote: > Hi > > Thanks for reply > > Here is kamailio.cfg file > > ================================================================================================ > listen = 192.168.94.30 > port = 5060 > mpath = "/lib/kamailio/modules" > debug = 9 > fork = yes > disable_tcp = no > log_facility = LOG_LOCAL0 > log_stderror = no > dns = no > rev_dns = no > > loadmodule "usrloc.so" > modparam("usrloc", "user_column", "username") > modparam("usrloc", "domain_column", "domain") > modparam("usrloc", "contact_column", "contact") > modparam("usrloc", "expires_column", "expires") > modparam("usrloc", "q_column", "q") > modparam("usrloc", "callid_column", "callid") > modparam("usrloc", "cseq_column", "cseq") > modparam("usrloc", "methods_column", "methods") > modparam("usrloc", "flags_column", "flags") > modparam("usrloc", "user_agent_column", "user_agent") > modparam("usrloc", "received_column", "received") > modparam("usrloc", "socket_column", "socket") > modparam("usrloc", "use_domain", 0) > modparam("usrloc", "desc_time_order", 0) > modparam("usrloc", "timer_interval", 60) > modparam("usrloc", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "matching_mode", 0) > modparam("usrloc", "cseq_delay", 20) > modparam("usrloc", "nat_bflag", 6) > > loadmodule "pv.so" > > loadmodule "registrar.so" > modparam("registrar", "default_expires", 3600) > modparam("registrar", "min_expires", 60) > modparam("registrar", "max_expires", 0) > modparam("registrar", "default_q", 0) > modparam("registrar", "append_branches", 1) > modparam("registrar", "case_sensitive", 0) > modparam("registrar", "received_param", "received") > modparam("registrar", "max_contacts", 100) > modparam("registrar", "retry_after", 0) > modparam("registrar", "method_filtering", 0) > modparam("registrar", "path_mode", 2) > modparam("registrar", "path_use_received", 0) > modparam("registrar", "received_avp", "$avp(i:801)") > > loadmodule "rr.so" > modparam("rr", "enable_full_lr", 0) > modparam("rr", "append_fromtag", 1) > modparam("rr", "enable_double_rr", 1) > modparam("rr", "add_username", 0) > > loadmodule "tm.so" > modparam("tm", "fr_timer", 3) > modparam("tm", "fr_inv_timer", 35) > modparam("tm", "wt_timer", 5) > modparam("tm", "delete_timer", 2) > modparam("tm", "ruri_matching", 1) > modparam("tm", "via1_matching", 1) > modparam("tm", "unix_tx_timeout", 2) > modparam("tm", "restart_fr_on_each_reply", 1) > modparam("tm", "pass_provisional_replies", 0) > modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)") > > loadmodule "xlog.so" > modparam("xlog", "buf_size", 4096) > modparam("xlog", "force_color", 0) > > loadmodule "mi_fifo.so" > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > loadmodule "domain.so" > modparam("domain", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > > loadmodule "nathelper.so" > modparam("nathelper", "natping_interval", 60) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_method", "OPTIONS") > modparam("nathelper", "received_avp", "$avp(i:801)") > > loadmodule "textops.so" > > loadmodule "sl.so" > modparam("sl", "enable_stats", 1) > > loadmodule "maxfwd.so" > modparam("maxfwd", "max_limit", 256) > > loadmodule "db_mysql.so" > modparam("db_mysql", "ping_interval", 300) > modparam("db_mysql", "auto_reconnect", 1) > > loadmodule "auth.so" > modparam("auth", "nonce_expire", 300) > modparam("auth", "rpid_suffix", > ";party=calling;id-type=subscriber;screen=yes") > modparam("auth", "rpid_avp", "$avp(s:rpid)") > > loadmodule "auth_db.so" > modparam("auth_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("auth_db", "user_column", "username") > modparam("auth_db", "domain_column", "domain") > modparam("auth_db", "password_column", "password") > modparam("auth_db", "password_column_2", "ha1b") > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "use_domain", 1) > modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=uuid") > > loadmodule "uri_db.so" > modparam("uri_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > > loadmodule "avpops.so" > modparam("avpops", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("avpops", "avp_table", "usr_preferences") > modparam("avpops", "use_domain", 1) > modparam("avpops", "uuid_column", "uuid") > modparam("avpops", "username_column", "username") > modparam("avpops", "domain_column", "domain") > modparam("avpops", "attribute_column", "attribute") > modparam("avpops", "value_column", "value") > modparam("avpops", "type_column", "type") > > loadmodule "lcr.so" > modparam("lcr", "db_url", "mysql://database:database@192.168.90.12/openser > ") > modparam("lcr", "gw_table", "gw") > modparam("lcr", "gw_name_column", "gw_name") > modparam("lcr", "ip_addr_column", "ip_addr") > modparam("lcr", "port_column", "port") > modparam("lcr", "uri_scheme_column", "uri_scheme") > modparam("lcr", "transport_column", "transport") > modparam("lcr", "grp_id_column", "grp_id") > modparam("lcr", "lcr_table", "lcr") > modparam("lcr", "strip_column", "strip") > modparam("lcr", "prefix_column", "prefix") > modparam("lcr", "from_uri_column", "from_uri") > modparam("lcr", "priority_column", "priority") > modparam("lcr", "flags_avp", "$avp(i:712)") > modparam("lcr", "gw_uri_avp", "$avp(i:1400)") > modparam("lcr", "ruri_user_avp", "$avp(i:1402)") > modparam("lcr", "fr_inv_timer_avp", "$avp(s:fr_inv_timer_avp)") > modparam("lcr", "fr_inv_timer", 90) > modparam("lcr", "fr_inv_timer_next", 30) > modparam("lcr", "rpid_avp", "$avp(s:rpid)") > > loadmodule "alias_db.so" > modparam("alias_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("alias_db", "user_column", "username") > modparam("alias_db", "domain_column", "domain") > modparam("alias_db", "alias_user_column", "alias_username") > modparam("alias_db", "alias_domain_column", "alias_domain") > modparam("alias_db", "use_domain", 0) > > loadmodule "uac.so" > modparam("uac", "rr_store_param", "vsf") > modparam("uac", "from_restore_mode", "auto") > modparam("uac", "from_passwd", "s1p:Wiz4rd!") > > loadmodule "acc.so" > modparam("acc", "db_url", "mysql://database:database@192.168.90.12/openser > ") > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") > > loadmodule "mi_datagram.so" > modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8080" ) > modparam("mi_datagram", "children_count", 3 ) > modparam("mi_datagram", "unix_socket_mode", 0600 ) > modparam("mi_datagram", "unix_socket_group", "root" ) > modparam("mi_datagram", "unix_socket_user", "root" ) > modparam("mi_datagram", "socket_timeout", 2000 ) > > loadmodule "mediaproxy.so" > modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy/dispatcher.sock") > modparam("mediaproxy", "mediaproxy_timeout", 500) > modparam("mediaproxy", "signaling_ip_avp", "$avp(s:nat_ip)") > modparam("mediaproxy", "media_relay_avp", "$avp(s:media_relay)") > > loadmodule "nat_traversal.so" > modparam("nat_traversal", "keepalive_interval", 90) > modparam("nat_traversal", "keepalive_method", "OPTIONS") > modparam("nat_traversal", "keepalive_from", "sip:keepalive@mydomain.com<sip%3Akeepalive(a)mydomain.com> > ") > > route[0] > { > xlog("L_DBG", "New Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > force_rport(); > > if(msg:len > max_len) > { > xlog("L_DBG", "Message too big - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("513", "Message Too Big"); > exit; > } > if (!mf_process_maxfwd_header("10")) > { > xlog("L_DBG", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("483", "Too Many Hops"); > exit; > } > if(!is_method("REGISTER")) > { > if(nat_uac_test("19")) > { > record_route(";nat=yes"); > } > else > { > record_route(); > } > } > if (is_method("SUBSCRIBE")) { > # challenge/response > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > exit; > } > exit; > } > > if(is_method("CANCEL") || is_method("BYE")) > { > end_media_session(); > } > if(loose_route()) > { > if(nat_uac_test("19") || search("^Route:.*;nat=yes")) > { > fix_nated_contact(); > setbflag(6); > } > route(8); > } > if(is_method("REGISTER")) > { > route(7); > } > if(is_method("INVITE")) > { > use_media_proxy(); > route(9); > } > if(is_method("CANCEL") || is_method("ACK")) > { > route(13); > } > route(14); > } > > ######################################################################## > # Request route 'clear-usr-preferences-caller' > ######################################################################## > route[1] > { > xlog("L_DBG", "Clear caller preferences - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:caller_cli)/g"); > avp_delete("$avp(s:clir)/g"); > } > > ######################################################################## > # Request route 'clear-usr-preferences-callee' > ######################################################################## > route[2] > { > xlog("L_DBG", "Clear callee preferences - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:callee_fr_inv_timer)/g"); > avp_delete("$avp(s:cfu)/g"); > avp_delete("$avp(s:cfc)/g"); > } > > ######################################################################## > # Request route 'usr-preferences-caller' > ######################################################################## > route[3] > { > route(1); > xlog("L_DBG", "Load caller preferences for uuid > '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > # load caller avps > avp_db_load("$avp(s:caller_uuid)", "*"); > avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); > if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", > "eq/i:1")) > { > # mark for anonymization > setflag(28); > } > } > > ######################################################################## > # Request route 'usr-preferences-callee' > ######################################################################## > route[4] > { > xlog("L_DBG", "Load callee preferences for uuid > '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > # load callee avps > avp_db_load("$avp(s:callee_uuid)", "*"); > if(is_avp_set("$avp(s:cfu)/s")) > { > xlog("L_DBG", "Call-forward-unconditional to '$avp(s:cfu)' > found - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > avp_delete("$avp(s:caller_uuid)/g"); > avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); > avp_pushto("$ru", "$avp(s:cfu)"); > > route(3); > route(10); > exit; > } > if(is_avp_set("$avp(s:ringtimeout)/n")) > { > xlog("L_DBG", "Setting ring timeout to $avp(s:ringtimeout) > secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > avp_copy("$avp(s:ringtimeout)", > "$avp(s:callee_fr_inv_timer)/d"); > } > } > > ######################################################################## > # Request route 'clir' > ######################################################################## > route[5] > { > if(isflagset(28) && !isflagset(27)) > { > setflag(27); > > xlog("L_DBG", "Anonymize caller - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > > uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); > if(is_present_hf("Privacy")) > { > remove_hf("Privacy"); > } > append_hf("Privacy: id\r\n"); > } > } > > ######################################################################## > # Request route 'stop-rtp-proxy' > ######################################################################## > route[6] > { > if(isflagset(22)) > { > end_media_session(); > } > } > > ######################################################################## > # Request route 'base-route-register' > ######################################################################## > route[7] > { > sl_send_reply("100", "Trying"); > if(!www_authorize("", "subscriber")) > { > xlog("L_DBG", "Register authentication failed - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > www_challenge("", "0"); > exit; > } > if(!check_to()) > { > > xlog("L_DBG", "Spoofed To-URI detected - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > sl_send_reply("403", "Spoofed To-URI Detected"); > exit; > } > consume_credentials(); > if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) > { > fix_nated_register(); > setbflag(6); > } > > if(!save("location")) > { > xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > sl_reply_error(); > exit; > } > xlog("L_DBG", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > exit; > } > > ######################################################################## > # Request route 'base-outbound' > ######################################################################## > route[8] > { > if(is_present_hf("P-Asserted-Identity")) > { > remove_hf("P-Asserted-Identity"); > } > if(is_present_hf("Remote-Party-ID")) > { > remove_hf("Remote-Party-ID"); > } > if(is_avp_set("$avp(s:caller_cli)/s")) > { > if(!isflagset(28)) > { > xlog("L_DBG", "Set caller CLI '$avp(s:caller_cli)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > append_hf("P-Asserted-Identity: > <$avp(s:caller_cli)>\r\n"); > } > } > > route(5); > if(isbflagset(6)) > { > if(!isflagset(22) && !search("^Content-Length:[ ]*0")) > { > setflag(22); > # use_media_proxy(); > } > t_on_reply("2"); > } > else > { > t_on_reply("1"); > } > if(!isflagset(21)) > { > t_on_failure("2"); > } > if(isflagset(29)) > { > append_branch(); > } > if(is_present_hf("Proxy-Authorization")) > { > # consume_credentials(); > } > > xlog("L_DBG", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > # no 100 (we already sent it) and no DNS blacklisting > if(!t_relay("0x05")) > { > sl_reply_error(); > if(is_method("INVITE") && isbflagset(6)) > { > end_media_session(); > } > } > exit; > } > > ######################################################################## > # Request route 'base-route-invite' > ######################################################################## > route[9] > { > sl_send_reply("100", "Trying"); > if(from_gw()) > { > $avp(s:caller_uuid)="0"; > xlog("L_DBG", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > setflag(23); > $avp(s:ringtimeout)=1000; > > route(3); > if (nat_uac_test("3")) > { > xlog("L_DBG", "Inside NATed INVITE Processing\n"); > fix_nated_contact(); > > fix_nated_sdp("3"); > > t_on_reply("1"); > } > else > { > t_on_reply("1"); > } > > if(nat_uac_test("19")) > { > setbflag(6); > } > > route(51); > } > else > { > if(!proxy_authorize("", "subscriber")) > { > xlog("L_DBG", "Proxy authentication failed - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > proxy_challenge("", "0"); > exit; > } > if(!check_from()) > { > xlog("L_DBG", "Spoofed From-URI detected - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > sl_send_reply("403", "Spoofed From-URI Detected"); > exit; > } > > route(3); > if(nat_uac_test("19")) > { > fix_nated_contact(); > setbflag(6); > } > route(10); > } > } > > ######################################################################## > # Request route 'invite-find-callee' > ######################################################################## > route[10] > { > if(alias_db_lookup("dbaliases")) > { > xlog("L_DBG", "Callee was aliased - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > } > > route(2); > if(!is_domain_local("$rd")) > { > setflag(20); > $avp(s:callee_uuid) = "0"; > route(12); > } > avp_delete("$avp(s:callee_uuid)"); > avp_db_query("select uuid from subscriber where username = '$rU'", > "$avp(s:callee_uuid)"); > if(is_avp_set("$avp(s:callee_uuid)/s")) > { > xlog("L_DBG", "Callee is local, uuid='$avp(s:callee_uuid)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(11); > } > else > { > $avp(s:callee_uuid) = "0"; > xlog("L_DBG", "Callee is not local - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > # > # *86 - sending user direct to access voicemail from sip > client > # > if($rU=="*86") > { > route(12); > } > else > { > if($(rU{s.len})<=7) > { > route(11); > } > else > { > route(12); > } > } > } > exit; > } > > ######################################################################## > # Request route 'invite-to-internal' > ######################################################################## > > route[11] > { > route(4); > > > revert_uri(); > seturi("sip:*89@openser.org"); > > rewritehostport("voicemailserver.databasepbx.com:5060"); > route(8); > exit; > } > > ######################################################################## > # TEST ROUTE 51 > ######################################################################## > route[51] > { > if(alias_db_lookup("dbaliases")) > { > xlog("L_DBG", "Callee was aliased - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > } > > route(2); > if(!is_domain_local("$rd")) > { > setflag(20); > $avp(s:callee_uuid) = "0"; > route(12); > } > avp_delete("$avp(s:callee_uuid)"); > avp_db_query("select uuid from subscriber where username = '$rU'", > "$avp(s:callee_uuid)"); > if(is_avp_set("$avp(s:callee_uuid)/s")) > { > xlog("L_DBG", "Callee is local, uuid='$avp(s:callee_uuid)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(52); > } > else > { > $avp(s:callee_uuid) = "0"; > xlog("L_DBG", "Callee is not local - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > route(12); > } > exit; > } > > ######################################################################## > #TEST 52 > ######################################################################## > route[52] > { > route(4); > if(!lookup("location")) > { > xlog("L_DBG", "Local user offline - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > sl_send_reply("404", "User Offline"); > } > else > { > xlog("L_DBG", "Local user online - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:callee_srcip)"); > avp_db_query("select received from location where username > = '$rU'", "$avp(s:callee_srcip)"); > avp_printf("$avp(i:13)", "$(du{s.select,1,:})" ); > avp_printf("$avp(i:14)", > "$(avp(s:callee_srcip){s.select,1,:})" ); > xlog("L_DBG", "source IP [$avp(i:13)] destination IP > [$avp(i:14)]\n"); > if (avp_check("$avp(i:13)","eq/$avp(i:14)/i")) > { > xlog("L_INFO", "Detected Two Clients Behind the > Same NAT - Disabling Mediaproxy"); > # Do not use mediaproxy as the clients seem to be > behind the same NAT > xlog("L_DBG", "source IP is the same as > destination IP \n"); > resetflag(2); > resetflag(3); > } > avp_delete("$avp(i:13)/g"); > avp_delete("$avp(i:14)/g"); > route(8); > } > exit; > } > > ######################################################################## > # Request route 'invite-to-external' > ######################################################################## > route[12] > { > if(isflagset(20)) > { > xlog("L_DBG", "Call to foreign domain - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > route(8); > exit; > } > if(!isflagset(23)) > { > if(uri =~ "^sip:\*86@.*") > { > xlog("L_DBG", "User accessing Voicemail - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > rewritehostport(" > voicemailserver.databasepbx.com:5060"); > route(8); > } > # don't allow calls relaying from PSTN to PSTN, if not > explicitely forwarded > else if(uri =~ "^sip:[0-9]+@") > { > # only route numeric users to PSTN > if(!load_gws()) > { > xlog("L_ERR", "Error loading PSTN gateways > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > sl_send_reply("503", "PSTN Termination > Currently Unavailable"); > exit; > } > if(!next_gw()) > { > xlog("L_ERR", "No PSTN gateways available - > M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > sl_send_reply("503", "PSTN Termination > Currently Unavailable"); > exit; > } > setflag(21); > t_on_failure("1"); > route(8); > } > } > > xlog("L_DBG", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("404", "User Not Found"); > exit; > } > > ######################################################################## > # Request route 'base-route-local' > ######################################################################## > route[13] > { > t_on_reply("1"); > if(t_check_trans()) > { > xlog("L_DBG", "Request leaving server - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > } > else > { > xlog("L_DBG", "Dropping mis-routed request - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > exit; > } > if(!t_relay()) > { > sl_reply_error(); > } > exit; > } > > > ######################################################################## > # Request route 'base-route-generic' > ######################################################################## > route[14] > { > xlog("L_DBG", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("501", "Method Not Supported Here"); > exit; > } > > ######################################################################## > # Request route 'base-filter-failover' > ######################################################################## > route[15] > { > if(!t_check_status("408|500|503")) > { > xlog("L_DBG", "No failover routing needed for this response > code - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(6); > exit; > } > } > > > ######################################################################## > #Send the call out > ######################################################################## > > route[16] > { > #send the call outward > if(method=="INVITE" && !isflagset(10)) > { > t_on_failure("2"); > } > > if (!t_relay()) > { > xlog("L_WARN", "ERROR: t_relay failed"); > sl_reply_error(); > } > } > > ######################################################################## > # Request route 'Voicemail and Asterisk for further processing' > ######################################################################## > failure_route[3] > { > if(!t_was_cancelled()) > { > revert_uri(); > rewritehostport("voicemailserver.databasepbx.com:5060"); > append_branch(); > #PREVENT SOME CRAZY VOICEMAIL LOOP > xlog("L_DBG", "No Answer Sending the call to Voicemail - > M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > setflag(10); > route(8); > } > } > > ######################################################################## > # Reply route 'base-standard-reply' > ######################################################################## > onreply_route[1] > { > xlog("L_DBG", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); > if (nat_uac_test("1")) > { > fix_nated_contact(); > } > exit; > } > > ######################################################################## > # Reply route 'base-nat-reply' > ######################################################################## > onreply_route[2] > { > xlog("L_DBG", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si > ID=$ci\n"); > if(nat_uac_test("1")) > { > fix_nated_contact(); > xlog("L_DBG", "NAT-Reply==[Fix_nated_contact] - S=$rs D=$rr > F=$fu T=$tu IP=$si ID=$ci\n"); > } > > if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") > { > if(!search("^Content-Length:[ ]*0")) > { > use_media_proxy(); > xlog("L_DBG", "NAT-Reply==[Force Rtp proxy] - S=$rs > D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); > } > } > exit; > } > > ######################################################################## > # Failure route 'pstn-failover' > ######################################################################## > failure_route[1] > { > xlog("L_DBG", "Failure route for PSTN entered - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > route(15); > if(!next_gw()) > { > xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > route(6); > exit; > } > > t_on_failure("3"); > route(8); > } > > ######################################################################## > # Failure route 'base-standard-failure' > ######################################################################## > failure_route[2] > { > if(!t_was_cancelled()) > { > revert_uri(); > route(10); > } > if(t_check_status("422|481|487")) > { > xlog("L_DBG", "Final reply - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > route(6); > exit; > } > > t_on_branch("1"); > xlog("L_DBG", "Redirect from UAC intercepted - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > route(10); > if($avp(s:cfc) != NULL) > { > avp_pushto("$ru", "$avp(s:cfc)"); > setflag(29); > append_branch(); > > t_on_branch("1"); > xlog("L_DBG", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > route(10); > } > > route(15); > route(6); > } > > ######################################################################## > # Branch route 'cfc-drop-local' > ######################################################################## > branch_route[1] > { > if(is_domain_local("$rd")) > { > xlog("L_DBG", "Dropping local branch - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > drop(); > } > } > > =============================================================================================== > > MediaProxy's config.ini > > ; > ; Configuration file for MediaProxy > ; > > [Relay] > relay_ip = 192.168.94.30 > dispatchers = 192.168.94.30 > port_range = 8000:20000 > log_level = DEBUG > on_hold_timeout = 1800 > stream_timeout = 90 > > [Dispatcher] > socket = /var/run/mediaproxy/dispatcher.sock > listen = 192.168.94.30 > listen_management = 192.168.94.30 > log_level = DEBUG > management_use_tls = no > > [TLS] > certs_path = tls > verify_interval = 300 > > [OpenSIPS] > max_connections = 100 > > Is there any problem in config file then please suggest me. This > configuration was running with other phone but not with * > PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. > > > Thanks in advance > -- > Regards, > > Chandrakant Solanki > > On Wed, Jan 6, 2010 at 3:38 PM, Ramu <contactramu(a)gmail.com> wrote: > >> Hi, >> >> You have to do NATing if the UAs are behind NAT when REGISTER and INVITE >> packet comes to Kamailio. >> >> You have to call use_media_proxy() function for INVITE and 200 OK >> messages. >> >> Please send your kamailio.cfg file to resolve the issues. >> >> >> On Wed, Jan 6, 2010 at 11:12 AM, Chandrakant Solanki < >> solanki.chandrakant(a)gmail.com> wrote: >> >>> Hello >>> >>> I have installed kamailio - 1.5.2 with MediaProxy. >>> >>> I registered 101(a)domain.com and 102(a)domain.com from * >>> PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. >>> >>> But when I look for location table into database, I show received field >>> is NULL, while from other phone like X-Lite it shows some value in same >>> field like sip:X.X.X.X:5061. >>> >>> When I call from 101 to 102 using *PolycomSoundPointIP *phone, The phone >>> rings on the other extension but when I pick up no voice is heard, the >>> caller phone keeps hearing the wait-for-other party. >>> >>> Is there any changes required into kamailio.cfg or not. ?? >>> >>> Thanks in advance >>> >>> -- >>> Regards, >>> >>> Chandrakant Solanki >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >> >> >> Thanks, >> Ramu >> > > -- Ramu

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[Kamailio-Users] [kamailio-users] Kamailio - Polycom Sound Point
by Chandrakant Solanki 06 Jan '10

06 Jan '10

Hello I have installed kamailio - 1.5.2 with MediaProxy. I registered 101(a)domain.com and 102(a)domain.com from * PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. But when I look for location table into database, I show received field is NULL, while from other phone like X-Lite it shows some value in same field like sip:X.X.X.X:5061. When I call from 101 to 102 using *PolycomSoundPointIP *phone, The phone rings on the other extension but when I pick up no voice is heard, the caller phone keeps hearing the wait-for-other party. Is there any changes required into kamailio.cfg or not. ?? Thanks in advance -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] memlog and mem_dbg : kamailio 3.0.0
by Ricardo Martinez 05 Jan '10

05 Jan '10

Hello. I have a doubt about the memlog and memdbg log levels. I have in my configuration file the next lines : debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=4 port=5060 memdbg=0 memlog=0 I was expecting not to see any memory debugs, instead I have this lines in the log file : Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:406]: qm_malloc(0x826db60, 8) returns address 0x82f9408 frag. 0x82f93f0 (size=8) on 1 -th hit Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:428]: qm_free(0x826db60, 0x82f9408), called from nathelper: nathelper.c: nh_timer(3252) Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x82f93f0 alloc\'ed from nathelper: nathelper.c: nh_timer(3143) Is this ok? How can I remove such debug? Regards, Ricardo Martinez

2 1

[Kamailio-Users] t_replicate in Kamailio-3.0.0 ?
by Santiago Gimeno 05 Jan '10

05 Jan '10

Hi, I'm trying to use t_replicate in K-3.0.0 with no luck. t_replicate("sip:192.168.1.1:5060"); The logs report "unknown command" and by looking at the documentation of the TM module I've seen that t_replicate is no longer in the TM module. Is this correct? Any workaround? Thanks in advance. Best regards, Santi

2 3

[Kamailio-Users] odd query
by Anders 05 Jan '10

05 Jan '10

Hi, Using SIREMIS, there is a column called dst_domain in both the acc and cdrs tables. This is supposedly the ID of the gw used for the call. However, this seems to be messed up when there is more than one gw that has the same IP. I have gws with the same IP but different prefixes, but then it just seems to put into dst_domain whatever the ID is of the first gw with that IP. Not so good, since I need the info of which gw used to be able to find out how much the call cost me. Anyone knows about how to handle this? Is there a fix, hatch or work-around? Thanks!

1 0

[Kamailio-Users] multiple kamctl configurations
by Klaus Darilion 04 Jan '10

04 Jan '10

Hi! I have several Kamailio servers on a single server running. Is there a way to tell kamctl to use a different configuration (e.g. for fifo)? regards klaus

2 1

[Kamailio-Users] re-invite mid call?
by Antonio Goméz Soto 04 Jan '10

04 Jan '10

Hi, maybe I should go to the -dev list for this, but thought I'd ask here first. Is it possible to let openSER send a SIP reinvite in the middle of a call, to redirect the sound stream to some rtpproxy? Maybe using the MI? Or should OpenSER be modified to do this? Thanks, Antonio

5 25

[Kamailio-Users] Kamailio 3.0.0 - Jan 11, 2010
by Daniel-Constantin Mierla 04 Jan '10

04 Jan '10

Hello, it was a bit longer way than usual with the testing, but the achievements pay the bill - you can run kamailio and ser modules at the same time, as well as most of the core features from both applications. I propose to release Kamailio (OpenSER) 3.0.0 on 11th of January, by that time we should fix the small issues discussed on the devel list lately. If you are aware of other issues, report them to: sr-dev [at] lists.sip-router.org I encourage everyone to contribute to wiki pages for what is new and migration from 1.5.x: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 To get it from GIT and test, see: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git I started to detail some of the new features in blog posts, see ToC at: http://bit.ly/4qy9cn A great 2010 for kamailians and sip router users! Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Serusers] problems when starting ser
by "Martínez García, Rebeca" 04 Jan '10

04 Jan '10

Hello all, I have installed ser in Ubuntu Karmic by means of Synaptic without any problems or errors during the installation. But before starting to configure the ser.cfg, I decided firstly to try the configuration by default and I got this error (I only uncommented the "log_stderror=yes"): cng@cng:/etc/ser$ ser D E Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.1.3.207 [10.1.3.207]:5060 udp: 192.168.50.10 [192.168.50.10]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.1.3.207 [10.1.3.207]:5060 tcp: 192.168.50.10 [192.168.50.10]:5060 Aliases: tcp: cng.local:5060 tcp: cng:5060 tcp: localhost:5060 udp: cng.local:5060 udp: cng:5060 udp: localhost:5060 0(2956) init_tcp: using epoll_lt as the io watch method (auto detected) cng@cng:/etc/ser$ 0(2958) Maxfwd module- initializing 0(2958) ERROR: fifo_server.c:857: Can't create FIFO: No such file or directory (mode=384) 0(2958) ERROR: ctl: mod_init: init ctrl. sockets failed 0(2958) init_mod(): Error while initializing module ctl ERROR: error while initializing modules It seems that it starts ok but suddenly finds an error and stops. However if I run: cng@cng:/etc/ser$ sudo /etc/init.d/ser start Starting ser: serListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.1.3.207 [10.1.3.207]:5060 udp: 192.168.50.10 [192.168.50.10]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.1.3.207 [10.1.3.207]:5060 tcp: 192.168.50.10 [192.168.50.10]:5060 Aliases: tcp: cng.local:5060 tcp: cng:5060 tcp: localhost:5060 udp: cng.local:5060 udp: cng:5060 udp: localhost:5060 0(3016) init_tcp: using epoll_lt as the io watch method (auto detected) . 0(3018) unable to create pid file /var/run/ser/ser.pid: No such file or directory It only complains about the ser.pid file. I am a little bit confused. Where is the problem? Thanks in advance, Rebeca Martinez

3 4

[Kamailio-Users] New install woes - 1.5.3 w/ mysql
by Graham Wooden 02 Jan '10

02 Jan '10

Hi all, Apparently I am missing something fundamentally here; trying to get mysql functions working with the kamdbctl command, but it seems that while it locates the kamdbctl.mysql file, it can¹t locate/use the the locate_tool() function from the kamctl.base¹ (which is called at the top of kamdbctl.mysql). I have been going back through trunk/INSTALL file and not seeing anything that is standing out. Neither when I ³strace² the kamdbctl create¹ command. But seems that my environment is missing something. I don¹t recall such issues with my production 1.4.2 machine. It¹s a fresh install CentOS 5.4 with most items (e.g. mysql server/client) installed via the standard yum repo¹s. I tried 1.5.3 from both building it from source as as well as the linux/i386 binary set. Still the same results. Thoughts? [root@misfits kamailio]# kamctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded INFO: Starting Kamailio : INFO: started (pid: 9006) [root@misfits kamailio]# kamdbctl create /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 25: locate_tool: command not found error: 'mysql' tool not found: set MYSQL variable to correct tool path

2 5

[Kamailio-Users] A great 2010 for everyone!
by Daniel-Constantin Mierla 31 Dec '09

31 Dec '09

Hello, before starting the party, I want to wish everyone all the best for 2010. Many thanks for those helping during 2009 with Kamailio and SIP Router projects, with code, documentation, advocating and support on mailing list. It was a lot of work done and I am confident you will enjoy Kamailio 3.0.0 and its great improvements and new features. I started a series of posts trying to highlight the best of it from my point of view, link to first is here: http://by-miconda.blogspot.com/2009/12/best-of-new-in-kamailio-300-1-includ… Not finished, expect more before Kamailio 3.0.0 is out, planned to happen before mid of January 2010. The draft of new features is available on wiki: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x A fantastic night for all! Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] avpops OR sqlops
by alex pappas 31 Dec '09

31 Dec '09

Hi all, I'm trying to make an sql query through my kamailio script in order to check for number portability. i have a table with 2 col (phonenumber,prefix). I'm trying both modules avpops and sqlops but I'm not able to retreive any data. modparam("avpops", "db_url","mysql://openser:openserrw@localhost/openser") avp_db_query("SELECT prefix FROM ported WHERE phonenumber='$rU'","$avp(s:lnp_prefix)") ;The query run perfectly in command line. The return code is always 1 but when I print $avp(s:lnp_prefix) I get no result. I have 2 questions : 1. Which is the best way to check portability in my case 2. What I'm doing wrong and I'm not getting any result from the query. Thank you ALL Alex

5 46

[Kamailio-Users] 1.5.3 dialog module: dialogs stay in database when in "shutdown only" mode
by Nathan Angelacos 30 Dec '09

30 Dec '09

Hi, It appears that in "shutdown only" mode (modparm db_mode = 3 ) dialogs are not cleared from the database on startup: 1. kamctl start initiate call 2. kamctl stop 3. kamctl start terminate call (dialog is terminated correctly) 4. kamctl db show dialog shows the dialog record 5. kamctl stop 6. kamctl db show dialog shows the (now very dead) dialog record Perhaps the dialog table should be cleared right after step 3 or at the beginning of step 5 when the dialog module is in "shutdown only" mode.

2 1

[Kamailio-Users] Problems with Sip Redirect Client
by Daniel Cardoso Alves 30 Dec '09

30 Dec '09

Hello... I have a SIP Redirect Serve with a kamailio and it works fine with others system(NexTone, Asterisk, etc), but I have a problem with a SIP redirect client with Kamailio. I am using http://kamailio.org/docs/modules/1.5.x/uac_redirect.html to try manipulate the 3xx message. I would like to send one destiny at a time, but the get_redirects("10:2","redirect") send only 2 contacts and at teh same time. I get from my SIP Redirect Server this message: SIP/2.0 301 Moved Permanently..Via: SIP/2.0/UDP 2XX.XX.XXX.XXX:5080;branch=z9hG4bK633e.a3e32ff2.0;rport=5080;received=2XX.XX.XXX.XXX..Via : SIP/2.0/UDP 192.168.0.133:7191;received=201.XX.X.XXX;branch=z9hG4bK-d87543-254f7f3a1058e232-1--d87543-;rport=7191..To: <sip:05433330000 @2XX.XX.XXX.XXX:5080>;tag=609b19d62bd6886602a6c902875525b0.c8b2..From: "1938"<sip:1938@2XX.XX.XXX.XXX:5080>;tag=187fec3c..Call-ID: 4959bc 623e556571@ZGVzZW4tcGMtMDguZ292b2lwLmNvbS5icg....CSeq: 1 INVITE..Contact: sip:55341#5433330000@192.168.0.2, q=1, sip:123#5433330000@1 92.168.0.3, q=2, sip:55341#5433330000@192.168.0.4, q=3, sip:55341#5433330000@192.168.0.5, sip:55341#5433330000@192.168.0.6, sip:553 41#5433330000(a)192.168.0.7, sip:55341#5433330000@192.168.0.8, sip:55341#5433330000@192.168.0.9, sip:55341#5433330000@192.168.0.10. .Server: Test Redirect..Content-Length: 0.... How I can configure to send fist to sip:55341#5433330000@192.168.0.10 and after if I not sucess it sends to sip:55341#5433330000@192.168.0.9, etc...? Thanks... ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

3 6

[Kamailio-Users] [kamailio-users] Kamailio 1.5.0 + MediaProxy 2.3.8
by Chandrakant Solanki 30 Dec '09

30 Dec '09

Hi Kamailio - 1.5.0 MediaProxy - 2.3.8 CentOS 5.2 - 2.6.18-92.el5PAE My kamailio is running on *192.168.X.Y* ** *Here is my /etc/mediaproxy/config.ini* ------------------------------------------------- ; Configuration file for MediaProxy ; [Relay] dispatchers = 192.168.X.Y relay_ip = <public ip> port_range = 8000:10000 log_level = DEBUG stream_timeout = 90 [Dispatcher] socket = /var/run/mediaproxy/dispatcher.sock listen = 192.168.X.Y listen_management = 192.168.X.Y log_level = DEBUG [TLS] certs_path = tls verify_interval = 300 [OpenSIPS] max_connections = 100 =========================================================== Start media-dispatcher / media-relay # media-dispatcher --no-fork # media-relay --no-fork When I see log it show dispatcher to relay and relay to dispatcher connected.. But when I call 101 to 102 it gives me error, as show below *Media Dispatcher error* error: No suitable relay found *Media Relay error* warning: Cannot use port pair 8684/8685 warning: Cannot use port pair 8686/8687 -- Regards, Chandrakant Solanki

3 2

Re: [Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

On Tuesday 29 December 2009 01:55:27 juergen(a)glowka.de wrote: > Hello, > > I'm new here. So I say "hello" to everybody. > > Maybe, somebody can help me or push me in the right direction. > > If there are Caller and Callee behind the same NAT, so it should > be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), > instead of using the rtp-proxy (this causes a jitter noise) . > > > I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. > > Has anybody a code snip or a sample kamailio.cfg for me? > I would be thankful for it. Better than giving you the solution, try to guest it ... Think a little ... what have in common a REQUEST from UAC A to UAC B if they are behing THE SAME NAT ROUTER ? -- Raúl Alexis Betancor Santana Dimensión Virtual ____________________________________________ Dear Mr. Santana Thanks for trying to help. oh, I guess, they've got the same IP-Address ;-) I found this code-snip in : dokuwiki/doku.php/examples:caller-callee-behind-same-nat ______________________________________ else if ( isflagset(2) and isflagset(3) ) { log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("i:450", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(2); resetflag(3); } } _______________________________________ But this Code-snip contains many mistakes so I changed it to : _______________________________________ if ( isflagset(5) and isbflagset(6) ) { lookup("location"); log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("$avp(i:450)", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(5); resetbflag(6); } } _______________________________________ This seems to work, but I still have got this looping, jittering feeping noise witch is getting louder and louder ... Two or three words I can talk, then it sounds, as if a group of Navajo-Indians gone attack me .... :-) Could it be the client? I use 2 PDAs, Windows Mobile, Portsip Ver. 2.6 Best Wishes, also for me Juergen Glowka

3 2

[Serusers] using ser to involve nat
by WangZhenzhong 29 Dec '09

29 Dec '09

Hi every body I am using ser to solve the nat problem, In my environment: invite A-->SER-->B 1. SER receive the invite message REQA from A(invite B) 2. SER judge the use B, if the B is in public network, will fix nat in the REQA message if judge the B is in private network , will not fix anything and I want to know, how to judge the B's network, when the SER receive the INVITE from A? Is there any examle code used in ser.cfg thank you very much 2009-12-29 姓名：王振中公司：艾诺通信系统(苏州)有限公司地址：苏州工业园区苏虹西路81号苏虹工业坊A8单元二楼

1 0

[Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

Hello, I'm new here. So I say "hello" to everybody. Maybe, somebody can help me or push me in the right direction. If there are Caller and Callee behind the same NAT, so it should be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), instead of using the rtp-proxy (this causes a jitter noise) . I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. Has anybody a code snip or a sample kamailio.cfg for me? I would be thankful for it. Best wishes Juergen Glowka _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

2 1

Re: [Kamailio-Users] adding inbound and outbound routes seperately
by Klaus Darilion 28 Dec '09

28 Dec '09

please always Cc: the list.... On 23.12.2009 16:24, Rodney wrote: > Well could you explain how to actually get peering functioning > between Kamailio on inbound and outbound? > There is no "peering functioning" in kamailio. You self have to define how you want to peer. For example if you want to peer with somebody sending you inbound requests, you have to define at least: - number format - authentication (IP-based, TLS, digest) One you have defined those parameters, you can implement it, e.g. digest authentication or IP based authentication. The same for outbound route - you have to configure Kamailio so that outbound requests are sent in the proper format. And Kamailio can not register to any other SIP service. If you need such a feature, you better use Asterisk. regards klaus > > -----Original Message----- From: Klaus Darilion > [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 23, > 2009 8:49 AM To: Rodney Cc: users(a)lists.kamailio.org Subject: Re: > [Kamailio-Users] adding inbound and outbound routes seperately > > > On 22.12.2009 22:35, Rodney wrote: >> Ok, I have kamailio and siremis, and I need to know how to set up >> a separate in bound and outbound route. >> >> Peering with inbound and registering with outbound >> > > Kamailio is a sip proxy, thus it can not register to other SIP > proxy. > > regards klaus > >> Any ideas guys? >> >> R >> >> >> >> _______________________________________________ Kamailio (OpenSER) >> - Users mailing list Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

3 6

[Kamailio-Users] Newbie questions
by Antonio Goméz Soto 28 Dec '09

28 Dec '09

Hi, I have used asterisk before, but now am looking at kamailio for scaling up. I've read 'Building Telephony Systems with OpenSER', but I still have lots of questions.... so, here it goes: - I could not find a way to reboot a phone remotely, can it send SIP NOTIFY to all phones? - does it have something like the Asterisk Manager Interface, so I can connect/create an operator panel for it? Does that already exist somewhere? Something like FOP where operators can transfer calls, and check availability? - according to the docs it is not a BTBUA. What does this mean? I know that asterisk is a BTBUA, but what can a BTBUA do that a SIP proxy cannot? Is it only the moh and conferences? - lastly, is there a good tutorial on load-balancing somewhere? Thanks, Antonio.

5 12

[Serusers] About iptel 400 missing cookie
by Dennis.Cheng 28 Dec '09

28 Dec '09

Hi folks, I used a Draytek Vigor 2820PBX to register iptel, and I found 400 missing cookie issue. When my pbx send the reinvite to hold another side, iptel always responded pbx 400 missing cookie. But I checked invite Via field, its branch included magic cookie "z9hG4bK". I attached wireshark packets. BR Dennis

1 0

[Kamailio-Users] Merry Christmas and a Happy New Year!
by Elena-Ramona Modroiu 25 Dec '09

25 Dec '09

Hi everybody, before you leave the computers and go on holidays, I want to thank you all for the good work and support of the Kamailio and SIP Router projects. I wish you a Merry Christmas and a Happy New Year! As the project name is a Hawaiian word - the "Kamailian" greeting is: "Mele Kalikimaka me ka Hau'oli Makahiki Hou!" How would it be in your language? May the new year bring you joy, peace, good health and prosperity! Enjoy the winter holidays! Ramona

9 9

[Kamailio-Users] Transparent bridge mode
by Vikram Ragukumar 24 Dec '09

24 Dec '09

Hello All, I am trying to setup a test scenario, where i have Kamailio and rtpproxy running on one CentOS box (Server1) and i have Asterisk running on another CentOS box (Server2). Server1 has 2 NIC's eth0 and eth1 that are both assigned Public IP's. There is a transparent bridge br0 connecting eth0 and eth1 which also has its own Public IP. Finally eth0 on Server2 also has a Public IP. Server2 must be assigned a Public IP. My goal is to modify rtpproxy so that i can intercept packets traveling to Server2, process them and let them resume along their original path. I would like to know if there is another way of setting this up so that i dont use as many Public IP's ? Do any of you see a problem with this setup, things that may not work eventually, or any other concerns ? Thanks, Vikram.

5 12

[Kamailio-Users] Server Internal Error with sample config file : Kamailio 3.0 rc3
by Ricardo Martinez 24 Dec '09

24 Dec '09

Hello list. I'm using the sample configuration that comes with Kamailio RC3. (version: kamailio 3.0.0-rc3 (i386/linux)). I have two IP phones registered in the proxy. If I call from one to another and the call is cancelled a "500 Server Internal Error" message is send back to the caller (please see the debug attached) by the proxy. This is not happening with Kamailio 1.4 or Opensips 1.6 (using the sample configurations). Could be some problem? 10.0.0.220 - Caller 10.0.0.211 - Proxy 10.0.0.152 - Callee U 10.0.0.220:5060 -> 10.0.0.211:5060 INVITE sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==1". . U 10.0.0.211:5060 -> 10.0.0.152:5062 INVITE sip:5501122334@10.0.0.152:5062 SIP/2.0. Record-Route: <sip:10.0.0.211;lr=on>. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.220:5060 -> 10.0.0.211:5060 CANCEL sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Supported: gruu, path. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 CANCEL sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9831 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.211:5060 via_cnt==1". . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 ACK sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM). Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==2". . U 10.0.0.220:5060 -> 10.0.0.211:5060 ACK sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . Ricardo.-

3 9

[Kamailio-Users] kamailio/ sip-router at 26c3
by Henning Westerholt 24 Dec '09

24 Dec '09

Hi, I'll (like in the last years) participate in the annual congress of the CCC in Berlin next week - the 26c3. More informations can be found on their site: http://events.ccc.de/congress/2009/wiki/Welcome So if you are in Berlin as well in this week (not necessarly on this event) and are interested in meeting for some discussions, please drop me a mail. Best regards, Henning

1 0

[Serusers] How to remove npdi & rn parameters from INVITE messages?
by Frank Durda IV 24 Dec '09

24 Dec '09

I have another large carrier that I am trying to send calls to, and they have a requirement that you not send them the npdi and rn=nnnnnnnn parameters for all calls. If you do, the call handling will be botched and the call will fail, even if the LRN information is exactly correct. Apparently this carrier (who once owned the Death Star) is incapable of filtering or ignoring this information on their side, and the sending switch is incapable of not sending this information because they thought RFC 4694 said they MUST ALWAYS send npdi/rn on every SIP call. Of course, the RFC does not say this. Don't get me started on which party is dumber. Anyway, I was hoping that SER could act as diplomat or interpreter and with some rewrite-something() calls, SER could remove the npdi/rn, or alter them to the point that the receiving switch no longer recognized the parameters as rn/npdi and so happily does it's own thing, and the calls don't die. However, based on what documentation exists, none of the various rewrite functions appear to operate on this part of the INVITE message and replace doesn't appear to work on the first line of a message. So there seems to be a spot whose contents can't be reached or manipulated by the existing functions. Am I correct in thinking this? So, is there a way to do something with rn=NNNNNNNNNN and npdi/npdi=yes parameters without having to write a custom function to do it? Thanks in advance!

2 1

[Serusers] How to drop a redundant/broken SIP reply message?
by Frank Durda IV 24 Dec '09

24 Dec '09

I have a large carrier that I am trying to send calls to, and their major brand SIP/ISUP switch (whose name begins with a "S") has a quirk where it emits a 183 Session Progress message with no SDP payload immediately after 100 Trying, and before ACM has actually been received. When ACM is received, it emits either a 180 Ringing with a SDP payload, or a 183 Session Progress with a SDP payload. So INVITE 100 Trying 183 Session Progress (no SDP payload) 180 Ringing (or 183 Session Progress) (with SDP payload) 200 OK (with SDP payload) is what you normally get, with the two 18x messages coming one or two seconds apart. However, if the called number is busy, you get 100 Trying 183 Session Progress (no SDP payload) 486 Busy Here What the calling party experiences for that is a second or two of ring-back commencing with the 183 Session progress, then a busy signal. That's very non-compliant. So far, the carrier has been unable to get this switch to stop emitting the first 183 Session Progress message. What I would like to do is have SER detect the redundant 183 Session Progress message (which I can already detect), and then have SER discard it, eg not send that one message back to whomever the INVITE came from. So far, I have been unable to get SER to do this. While "drop" is allowed in onreply_route sections, it is silently ignored, and you get a reply anyway. I tried putting a route(666) in the onreply_route section and putting the drop in that route["666"] section, but that had no obvious effect either, although some other functions can now be used in route["666"] that so far aren't solving the problem either. It occurred to me that I could change the 183 Session Progress of the offending message to something else in the "so-what" department, such as a "181 Your Call Is Being Forwarded", but "replace()" doesn't seem to work on the first line of replies, but can certainly alter other things in the reply message. There also doesn't seem to be any way (that actually works) to change the numeric SIP Response Code value to something else, even though clearly SER has it in the variable "status". I even thought of changing the port number or IP address via a rewrite command and send the message off to nowhere, but those don't work on replies or don't appear to. So, any suggestions on how to simply drop a chosen reply message, or misdirect it or malform it to the point that the call originator doesn't recognize it anymore? Yeah, kludgy but I only need it to work for two months. Thanks in advance!

2 2

[Serusers] Ser2.1.0.
by Aleš Podskalský 23 Dec '09

23 Dec '09

Hi all, I tried to install the newest snapshot of the Ser (version 2.1.0+cvs20091126). Server has been instaled correctly, but I can't add new user to the database. I noticed that serctl was changed with new tool called sercmd. With this tool I should be able to create new user using appropriate command, but I can't find some help for it. So my question is, if this version is unable for normal use by then, or if I can fill user data in some another way and how. Thank a lot. Ales P.

2 1

[Kamailio-Users] adding inbound and outbound routes seperately
by Rodney 23 Dec '09

23 Dec '09

Ok, I have kamailio and siremis, and I need to know how to set up a separate in bound and outbound route. Peering with inbound and registering with outbound Any ideas guys? R

3 2

Re: [Kamailio-Users] PCRE Regex / Transformations Bug? (Daniel-Constantin Mierla)
by Brandon Armstead 23 Dec '09

23 Dec '09

Inaki: That is what I was looking for and it does appear you are correct! Daniel, I could see that being as a useful addition, do you imagine there being much involved to implement such a method? Thank you guys for all of your help! Sincerely, Brandon Armstead

1 0

[Kamailio-Users] Question about Contact in NAT
by Ricardo Martinez 22 Dec '09

22 Dec '09

Hello list. I know that probably this questions is more related to the SIP protocol, but is causing me problems and this is a very supportive list. I'm using a HT-286 from Grandstream behind NAT. This is the REGISTER sequence : U 200.120.46.99:29786 -> 200.100.154.35:5060 REGISTER sip:myproxy.mydomain.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>. Contact: *. Supported: replaces, timer. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. Expires: 0. User-Agent: Grandstream HT287 1.1.0.42 DevId 000b822197fe. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE. Content-Length: 0. . U 200.100.154.35:5060 -> 200.120.46.99:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb;received=200.120.46.99. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 200.100.154.35:5060 "Noisy feedback tells: pid=29777 req_src_ip=200.120.46.99 req_src_port=29786 in_uri=sip:myproxy.mydomain.net out_uri=sip:myproxy.mydomain.net via_cnt==1". . U 200.100.154.35:5060 -> 200.120.46.99:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb;received=200.120.46.99. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=6eb14299de2cc731d17 3c135064daec5.1fe9. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. WWW-Authenticate: Digest realm="myproxy.mydomain.net", nonce="4b30e9281c52f640a2713c3011bb835d06005946", qop="auth". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 200.100.154.35:5060 "Noisy feedback tells: pid=29777 req_src_ip=200.120.46.99 req_src_port=29786 in_uri=sip:myproxy.mydomain.net out_uri=sip:myproxy.mydomain.net via_cnt==1". As you can see the initial REGISTER has a Contact header like this : Contact: *. Is this ok?. Is this ok by RFC? What does this means? In my cfg file I have : route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; So I think is not matching the regexp "(!search("^Contact:\ +\*")" therefore is not fixing the contact. Any ideas? Ricardo.-

2 3

[Kamailio-Users] append_hf
by Corey Edwards 22 Dec '09

22 Dec '09

I'm upgrading from OpenSER 1.2.2 to Kamailio 1.5.3 and I've run into a problem with append_hf. It seems that my header only gets appended once in a while. Here's the shortened version of what I'm doing: failure_route[1]{ if (t_check_status("(486)|(600)")){ # busy handler route(15); } } route[15]{ # look up forwarding number .... if (does_uri_exist()){ append_hf("Diversion: <sip:$var(orig_user)@$Ri>;user-id=\"$var(orig_user)@$Ri\";reason=busy\r\n"); t_relay(); } } Unfortunately, the Diversion header doesn't get set every time. I can't see any difference in the logs from when it does get set and when it doesn't. I added an xlog right above append_hf and confirmed that append_hf runs each time. On a whim I added a branch_route and called it using t_on_branch. I then added duplicate code to call append_hf. I now seem to be getting more instances of Diversion being set (twice!), but still not every time. Is there something obvious I'm missing here? Any way I can ensure those headers get set every time? Corey

1 1

[Kamailio-Users] listing users that were added
by Rafi Gordon 22 Dec '09

22 Dec '09

Hello, I had added successfully several users by : kamctl add xxx yyy How can I list all users that exist in the DB? Rgs, Rafi

3 5

[Kamailio-Users] Question about AVP in route()
by Ricardo Martinez 21 Dec '09

21 Dec '09

Hello. I have the next situation. In my config file I have something like this.. route { ... If INVITE route(3); } ... route[1] { Route(39); t_relay() } route[3] { ... different check for the INVITE radius_load_caller_avps("$fU@$fd") (I load the next AVPS from the radius server : Attributes: SIP-AVP = "tra:sip:0217001234567@10.0.0.211:5060" SIP-AVP = "channels:1") is_avp_set("$avp(s:channels)/n") <-- this AVP seem to be load and is set OK. route(10); } route[10] { rewritehost("myproxy.mydomain.com"); route(1); } route[39] { is_avp_set("$avp(s:channels)/n") } The last check in the route(39) is not working!. Why? Ricardo Martinez Ingeniero de Desarrollo VOISSNET S.A.

2 2

[Kamailio-Users] Presence_Dialoginfo versioning
by kamailio.org＠spam.lublink.net 21 Dec '09

21 Dec '09

Hey, I have noticed that the XML body of the dialog messages contains a version attribute. The server is counting the versions using the latest subscription as a reference point, and the phone is counting the versions from the first subscription ( at reboot ). Which is the correct way to count these versions? Consider : 10:00 Subscription 10:05 Notification version 1 10:35 Notification version 2 11:01 Subscription 11:05 Notification version X Is X = 3 because it is the third notification or 1 because it is the first after the last subscription? If it's version 1, it could confuse the phone cause a notification that is sent at the same time as the notification would have a confusing version. On the other hand, each subscription, has it's own versioning, so would it not logically follow that the different subscriptions for the same device have seperate versioning? From my phone : BLF Notify received for line: 3 has older version: 3 last version:135 To confirm my theory, I redialed the number 133 times, and once the version number had run up to 136, the lights started flashing again. Is this a bug in Grandstream, Kamailio, or perhaps an RFC which was unclear? Thanks, David

7 23

[Kamailio-Users] Clients with late offer-answer
by Klaus Darilion 21 Dec '09

21 Dec '09

Hi! I want to test rtpproxy with late offer-answer (SDP in 200-ok and ACK). Thus, I'm looking for a SIP client which support that. Do you know any? thanks klaus

3 2

[Kamailio-Users] add Allow header to sl_send_reply
by Andreas Heise 19 Dec '09

19 Dec '09

Hello, if I create a response with sl_send_reply then the message contains not an Allow header. How could I add an Allow header to this responses? thanks in advance, Andreas

4 10

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 19 Dec '09

19 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

2 1

[Kamailio-Users] Transparent bridge mode
by Sungjoo Kim 17 Dec '09

17 Dec '09

Hello All, I'm trying to setup a test scenario, where i have Kamailio and rtpproxy running on one CentOS box(server1) and asterisk running on another CentOS box(server2). Server1 has 2 NIC's installed, eth0 connected to the WAN and eth1 connected to Server2 in a bridge mode. I will be modifying rtpproxy to intercept data flowing to Server2 and process them before i let them continue to flow along their original path. Server2 has to have a Public IP. For this i am having to use 4 Public IP's- eth0,eth1 and bridge br0 on Server1 and eth0 on Server2. Is there an alternate way of doing this ? Do you suspect there will be a problem with setting things up this way while trying to use rtpproxy to intercept data. Thanks and regards, Vikram.

1 0

[Kamailio-Users] Kamailio + CSQL
by Ramu 17 Dec '09

17 Dec '09

Hi All, Did anybody tried to setup Kamailio with CSQL database. Please post your experiences using CSQL and what are parameters we need to change in modules to connect CSQL DB Server. Thanks, Ramu

2 2

[Kamailio-Users] Kamailio and RTPProxy
by Bruce McAlister 16 Dec '09

16 Dec '09

Hi All, I am trying to setup kamailio 1.5.3-notls for use with rtpproxy 1.2.1. I have a partially working setup, however, when I have a softphone configured behind nat it appears that rtpproxy is using the internal ip (prior to nat) to setup the sdp ports. I have rtpproxy configured in bridge mode as follows: rtpproxy -f -F -l public_ip/internal_ip -s udp:*:11211 I have kamailio's nathelper module setup as follows: modparam("nathelper", "rtpproxy_sock", "udp:internal_ip:11211") I then use a softphone (x-lite) behind a nat firewall (ip = 192.168.11.2) to establish a call, and the log of rtpproxy is as follows: INFO:handle_command: new session ZmRkNjRlNjM0NzExZjQ5MWUxYjllOTFhZWFmZGZkMjY., tag 8330585e;1 requested, type strong INFO:handle_command: new session on a port 52572 created, tag 8330585e;1 INFO:handle_command: pre-filling caller's address with 192.168.11.2:35498 INFO:handle_command: lookup on ports 52572/40380, session timer restarted INFO:handle_command: pre-filling callee's address with internal_ip:11250 INFO:handle_delete: forcefully deleting session 1 on ports 52572/40380 INFO:remove_session: RTP stats: 906 in from callee, 0 in from caller, 906 relayed, 0 dropped INFO:remove_session: RTCP stats: 3 in from callee, 0 in from caller, 3 relayed, 0 dropped INFO:remove_session: session on ports 52572/40380 is cleaned up In the log above it looks like rtpproxy is using 192.168.11.2:35498 as the remote ip/port combination to contact for the sdp session of the x-lite softphone. I would have thought that it would use the IP address that initial request was received from to establish this connection (via the fix_nated_contact() function). My route logic in kamailio is a slightly modified default configuration that is installed with kamailio : route{ . . . # NAT detection route(4); . . . route(1); } route[1]{ if (check_route_param("nat=yes")) { setbflag(6); } # if (isflagset(5) || isbflagset(6)) { route(5); # } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } Would the "fix_nated_contact" function in the "nat detection" route not fix the contact before passing over the request to rtpproxy to establish the sdp session? route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ # force_rtp_proxy(); if(src_ip==internal_ip_of_media_server) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaie"); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); # if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); if(src_ip == internal_ip_of_media_server) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaie"); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } } Any pointers would be greatly appreciated. Thanks Bruce

2 5

[Kamailio-Users] Problem
by Ricardo Martinez 15 Dec '09

15 Dec '09

Hello. What this error means? Dec 14 15:20:19 kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/nat_traversal.so>: /usr/local/lib/kamailio/modules_k/nat_traversal.so: undefined symbol: register_module_stats Dec 14 15:20:19 kamailio: : <core> [cfg.y:3210]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 48, column 12-29: failed to load module I have : # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/module s/" # Uncomment this if you want to use SQL database loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "siputils.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "group.so" loadmodule "mi_fifo.so" loadmodule "avpops.so" loadmodule "xlog.so" loadmodule "dialog.so" loadmodule "nathelper.so" loadmodule "nat_traversal.so" loadmodule "mediaproxy.so" I'm using Kamailio version: kamailio 3.0.0-rc2 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id$ main.c compiled on 18:09:39 Dec 9 2009 with gcc 4.3.2 In the directory /usr/local/lib/kamailio/modules_K/ the file nat_traversal.so exists : [root@ nat_traversal]# ls /usr/local/lib/kamailio/modules_k/ acc.so cfgutils.so drouting.so mi_datagram.so permissions.so registrar.so speeddial.so uac.so alias_db.so db_text.so exec.so mi_fifo.so pike.so rr.so sqlops.so uri_db.so auth_db.so dialog.so group.so msilo.so presence_mwi.so rtimer.so sst.so userblacklist.so auth_diameter.so dispatcher.so htable.so nathelper.so pua_mi.so siptrace.so statistics.so usrloc.so auth.so diversion.so imc.so nat_traversal.so pv.so siputils.so textops.so xlog.so benchmark.so domainpolicy.so kex.so path.so qos.so sl.so tmx.so call_control.so domain.so maxfwd.so pdt.so ratelimit.so sms.so uac_redirect.so Can someone help me? Regards, Ricardo

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Re: [Kamailio-Users] kamailio load problem
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, cc-ed back the list as others may look for same issues in the future ... On 12/15/09 11:30 AM, alex pappas wrote: > Daniel hi, > > In my config I don't have any authentication. I also configured in > mysql the log-slow-query with long_query_time = 2 > In my kamailio.cfg is a lot of ACC loging: > > Log acc to a file (extra fields) > Log to mysql with axtra fields(I disabled this for the moment because > was causing high load) > Log to a Radius (the radius is in the same machine and as far as I > know the radius after i swriting his data to a mysql also) I diden't > know that we have radius installed localy > > From the log-slow-query now I see a message about sip_trace query wich > obiesly took time more than 2 sec. Now I know that I'm doing also > sip_trace inserts in mysql sip_trace is writing big amount of data, this can make mysql do a lot of work. Try without it and see the difference. Also, doing acc to syslog brings some penalty if syslog is not configured in asyncronous mode (see my previous email). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [Kamailio-Users] kamailio load problem
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, cc-ed back the list as others may look for same issues in the future ... On 12/15/09 11:30 AM, alex pappas wrote: > Daniel hi, > > In my config I don't have any authentication. I also configured in > mysql the log-slow-query with long_query_time = 2 > In my kamailio.cfg is a lot of ACC loging: > > Log acc to a file (extra fields) > Log to mysql with axtra fields(I disabled this for the moment because > was causing high load) > Log to a Radius (the radius is in the same machine and as far as I > know the radius after i swriting his data to a mysql also) I diden't > know that we have radius installed localy > > From the log-slow-query now I see a message about sip_trace query wich > obiesly took time more than 2 sec. Now I know that I'm doing also > sip_trace inserts in mysql sip_trace is writing big amount of data, this can make mysql do a lot of work. Try without it and see the difference. Also, doing acc to syslog brings some penalty if syslog is not configured in asyncronous mode (see my previous email). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] debian packages for kamailio 3.0.0-rc3
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, I have uploaded debian packages for stable version: http://www.kamailio.org/pub/kamailio/3.0.0-rc3/packages/debian-lenny/ If you want to create debs by yourself, get Kamailio 3.0 branch from GIT, see here: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git And then run: make deb Testing and feedback is very much appreciated. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] kamailio load problem
by alex pappas 15 Dec '09

15 Dec '09

Dear users, I'm using kamailio in the following scenario: 1. A call comes in and is checked against PDT module. From this module depend from the prefix I'm getting back the domain which is used as the dialplan group. 2. After that the call is checked against Dialplan module using the previous value from PDT as Dialplan dpid. From the Dialplan module i get back the the attributes values. 3. these attributes values are user in the dispathcher module where I continue with failover scenarios. Also the script is working with a radius for accounting purposes. This scenario works perfecttly and it is still in a testing mode. An the problem: I'm running SIPP against this scenario and until the CPS is 5 everything looks good. The scenario for SIPP is the simplest. uac and uas default. Uac is sending a call and uas is answering. When the CPS goes above 9 then I have call failures. All the statistics are going down. I'm cheking for CPU load or other system problems but I don't see anything strange. Does anyone understand why this is happening? I'm attachng the 2 routes that i'm using: test values in order to understand the script *PDT value = 1 (used for the dialplan group) Dialplan value = 100.101.102.# Dispatcher values: 100 = 10.0.0.5 101 = 10.0.0.6 102 = 10.0.0.7* Thank you in advance Alex route[7] { if(prefix2domain("2", "0")) { $var(dial_grp) = $(rd{s.int}); if(dp_translate("$var(dial_ grp)", "$rU/$rU")) { $var(i) = 0; while($(avp(s:dest){s.select,$var(i),.})!="#") { $avp(s:dstgrp) = $(avp(s:dest){s.select,$var(i),.}{s.int}); $var(i) = $var(i) + 1; } $avp(s:user) = $rU; if(ds_select_domain("$avp(s:dstgrp)", "4")) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $rU + "@" + $rd; } else { $ru = "sip:" + $rU + "@" + $rd; } $avp(s:dstgrp) = null; t_on_failure("2"); t_relay(); exit; } } sl_send_reply("404", "No route found"); exit; } } failure_route[2] { if(t_was_cancelled()) { exit; } if(t_check_status("4[0-9][0-9]|5[0-9][0-9]")) { if(ds_next_domain()) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; } else { $ru = "sip:" + $avp(s:user) + "@" + $rd; } t_on_failure("2"); append_branch(); t_relay(); exit; } else { if($avp(s:dstgrp)!=null) { if(ds_select_domain("$avp(s:dstgrp)", "4")) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; } else { $ru = "sip:" + $avp(s:user) + "@" + $rd; } $avp(s:dstgrp) = null; t_on_failure("2"); append_branch(); t_relay(); exit; } } else { t_reply("444", "No more tries for you!"); } } }

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[Kamailio-Users] Reboot phones
by David 14 Dec '09

14 Dec '09

Hey, I have noticed that when I try to use kamctl to reboot phones, it comes back with simply "400". I tried using the fifo interface and the phone asked the server to identify itself using the phone's username and password. How do people generally use the kamctl to reboot phones? Thanks, David

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[Kamailio-Users] Kamailio 3.0.0-RC3 Released
by Daniel-Constantin Mierla 14 Dec '09

14 Dec '09

Kamailio v3.0.0-RC3 is out - the tarball with sources is available at: http://www.kamailio.org/pub/kamailio/3.0.0-rc3/src/kamailio-3.0.0-rc3_src.t… Lot of fixes to packaging, code and documentation have been committed since RC1, Kamailio 3.0.0-RC3 becoming ready for pre-production phase. If you like to work with sources, then use the tutorial guiding the installation of Kamailio 3.0 branch from GIT repository: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Two wiki pages were created to collect what is new in Kamailio 3.0 and how to migrate to this version: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 Feel free to contribute to wiki pages, helping to build migration tutorials. This is the last RC before full 3.0.0 relase. Please report any issue you find to sr-dev(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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