- sr-users - kamailio.org

force_send_socket
by Nagorny, Dimitry 26 Apr '16

26 Apr '16

Hi all, I've configured Kamailio to use TLS and it has two NICs, 192.x.x.x extern to a PSTN and 10.250.5.74 intern to FreeSWITCH. Kamailio has "listen=tls:10.250.5.74:5071" and later on in the config I fire "force_send_socket(tls:10.250.5.74:5071)". For some reason on calls from FreeSWITCH via Kamailkio to PSTN the TLS works but the other way around if Kamailio should do TLS to FreeSWITCH it throws this warning: "/kamailio[3308]: WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:10.250.5.74:5071, to udp:10.250.5.72:5071)" As a result of this Kamailio tries to send the PSTN INVITE via UDP and also tries to use the external IP (192.x.x.x) and of course this can't work. Why is it throwing this warning and how to understand it? Best regards Dimitry Nagorny Trainee

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BLF: Turning a key on/off without a dialog
by Ali Pey 25 Apr '16

25 Apr '16

Hello, I use presence and dialog modules for my BLF keys; however, I have a key (e.g. park) that I need to turn on/off outside a dialog: Certain Invite message would need to cause the BLF key to become busy and another unrelated Invite message would need to turn the BLF key off. What's the best and easiest way to do this? Best regards, Ali Pey

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Are sip calls working OK?
by Aloysius Bolskey Clark Donson 25 Apr '16

25 Apr '16

Dear Friends, I started to use my sip account few monts ago sip:4095xx@iptel.org It was working OK, but at the end of March I realised that I could not get any calls in my account. I own also a sip account in callcentric and works OK. So my zoiper version seems nothing to do on it Have you changed the setup of the account (iptel)? I got a real US phone number in IPCall associated to my sip account. IPCall is giving up, so that is the reason why the calls made to my real phone numbers ar not being forwarded to my sip account. But I have no idea why my sip:4095xx@iptel.org is not working thanks in advance looking forward your kind answer jose

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Using flags in t_branch_timeout() section of failure_route{}
by Marrold 24 Apr '16

24 Apr '16

Hi all, I'm having issues using flags in a t_branch_timeout() of failure_route{} A brief summary of the code - onreply_route{ setflag(14); } failure_route { if (t_branch_timeout()) { xlog("L_ALERT","t_branch_timeout()\n$mb"); if (isflagset(14)) { xlog("L_ALERT","isflagset(14) - YES\n$mb"); } } } The call flow is below, t_branch_timeout() is being triggered when the 200 OK is not ACK'd -- INVITE -> <- 100 Trying -- <- 180 Ringing -- <- 200 OK -- <- 200 OK -- <- 200 OK -- <- 200 OK -- I can attempt to create a minimal .cfg if anyone wishes to replicate it, assuming I'm not missing something stupid. Thanks, Matthew

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NO RTP when using kamailio on public IP
by Ankitt Sharma 23 Apr '16

23 Apr '16

hello, i've been trying to setup my communication setup , everything works fine when i use my kamailio setup within the local network but as soon as i assign a public IP to kamailio , there is no RTP flow between the clients i searched over internet and found that rtpengine module should be used when using the stup on public network ,and tried configureation(with some tweaks) given by https://github.com/caruizdiaz/kamailio-ws i also observed that the call still flows when the two clients are in same network. please help what should i do! thanx in advance! :)

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Kamailio IMS+FHoSS: Error "Visited Network Identity not found, responding with 400"
by Трущенков Алексей 22 Apr '16

22 Apr '16

Good afternoon! I use Kamailio from BOX from Article https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ The client can not register (use Boghe IMS Client as IMS client) i-cscf log 10(1991) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to > [sip:alice@net1.test] > 10(1991) ERROR: <script>: REGISTER (sip:alice@net1.test (10.0.0.10:1173) > to sip:alice@net1.test, 9848450f-c6dd-5b5e-6516-addaf85a9f2f) > 10(1991) ERROR: ims_icscf [registration.c:149]: > I_perform_user_authorization_request(): ERR:I_UAR: Visited Network Identity > not found, responding with 400 > 10(1991) ERROR: tm [t_lookup.c:1497]: t_get_trans_ident(): ERROR: > t_get_trans_ident: no transaction found I would be glad of any help. Thank you. --------------------- Trushenkov Alexey

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Understanding the new interface (kemi) for embedded interpreters
by Daniel-Constantin Mierla 22 Apr '16

22 Apr '16

Hello, (cross-posting because the details are useful for both devels and users) Yesterday during the IRC meeting about Kamailio, one group of topics was about Kamailio 5.0, and the new feature of using embedded interpreters for external programming languages was highly debated. So I am trying to clarify some of its aspects. The new framework is referenced as kemi, chosen by me as an abbreviation for Kamaialio EMbedded Interface. We already had embedded interpreters for many years, respectively: - Lua via app_lua module (probably most developed and most used) - Perl via app_perl module (iirc, the oldest embedded interpreter) - Python via app_python (I discovered during the last days that it has quite a lot of features, just not documented -- never used it before) - Java via app_java (this one is a bit of a blackhole for me, never looked at it, don't know what it offers/how it is supposed to be used) - .NET group of languages via app_mono (e.g., C#, but also some variants of JavaScript, Python and maybe even Java) To clarify, an embedded interpreter means that Kamailio is linking to itself the interpreter of that language and kamailio becomes the interpreter of the language. It is not launching the standard interpreter for that language, therefore it is very fast at runtime. Moreover, kamailio is extending the language with new modules, giving access to Kamailio C functions. In other words, for example with Python, Kamailio becomes equivalent to the "python" application plus two extensions (modules) named Router (developed in the past inside app_python) and KSR (developed for kemi). At this moment, the kemi was implemented in app_lua and app_python modules. To understand what is all about, I am going to give an example with a function from maxfwd module. In default kamailio.cfg next statement is present: if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } mf_process_maxfwd_header("10") is the function exported by maxfwd: - https://www.kamailio.org/docs/modules/stable/modules/maxfwd.html#maxfwd.f.m… The parameter can be as a static number (like above) or a number provided via a variable, like next: $var(p) = 10; mf_process_maxfwd_header("$var(p)") The corresponding C code for this config function is: static int w_process_maxfwd_header(struct sip_msg* msg, char* str1, char* str2) { int mfval; if (get_int_fparam(&mfval, msg, (fparam_t*) str1) < 0) { LM_ERR("could not get param value\n"); return -1; } return process_maxfwd_header(msg, mfval); } The function get_int_fparam() is practically returning the int value of the parmeter, no matter it was a static value or a variable, then process_maxfwd_header(...) is executed with the int parameter. The msg is the SIP message structure present mostly everywhere in the C code related to the classic config interpreter. To be used inside the config file, the C function has to be exported via a cmd_export_t structure added to "struct module_exports exports", respectively: static cmd_export_t cmds[]={ ... {"mf_process_maxfwd_header", (cmd_function)w_process_maxfwd_header, 1, fixup_var_int_1, 0, REQUEST_ROUTE}, ... To get the same function in a kemi interpreter (Lua, Python at this moment), following C code was added: static sr_kemi_t sr_kemi_maxfwd_exports[] = { { str_init("maxfwd"), str_init("process_maxfwd"), SR_KEMIP_INT, process_maxfwd_header, { SR_KEMIP_INT, SR_KEMIP_NONE, SR_KEMIP_NONE, SR_KEMIP_NONE, SR_KEMIP_NONE, SR_KEMIP_NONE } }, { {0, 0}, {0, 0}, 0, NULL, { 0, 0, 0, 0, 0, 0 } } }; int mod_register(char *path, int *dlflags, void *p1, void *p2) { sr_kemi_modules_add(sr_kemi_maxfwd_exports); return 0; } The structure sr_kemi_t sr_kemi_maxfwd_exports exports directly the existing function process_maxfwd_header(), which is also executed by the function from kamailio.cfg after resolving the parameter to an int value. In the case of kemi, the Lua or Python will give directly an integer value, so no need for resolving that with a fixup wrapper. The mod_register() function is something already existing, providing a way to execute code when a module was loaded (e.g., is used for many years by modules such as tls, mysql, etc.). In this case it was added, because maxfwd didn't need anything at load time before. Practically, no C code extension was written to maxfwd module from point of view of SIP routing. Just an existing function was exported. Now, what's actually the role of this kemi framework ... The app_lua already exported the kamailio.cfg function mf_process_maxfwd_header() as sr.process_maxfwd() since Kamailio v3.1, see its usage in the tutorial: - https://kb.asipto.com/kamailio:usage:k32-lua-routing#lua_script But a Lua specific wrapper had to be written: - https://github.com/kamailio/kamailio/blob/master/modules/app_lua/app_lua_ex… All the other embedded interpreters would have needed to get a wrapper as well to export this function. That's obviously costing time and maintenance resources. So kemi is a common framework to export C function to any of the embedded interpreters, meaning that once a function is exported from a module, it becomes available to all embedded interpreters, no longer being necessary to write a specif wrapper per interpreter. The app_* module needs a bit of coding to use the kemi exported functions (already done for Lua and Python), but then nothing needs to be done for new exported functions by modules. As a matter of fact, lots of functions can be just exported, as you could see in the example with maxfwd. In some cases, maybe some functions need to be split, so the fixup handling is done separately and then calls a common C function. So, what we had so far: [native kamailio cfg script] <===> [fixup framework for variables] <===> [C implementation of function] The above stays in place, no change to it. What we got extra: [embedded language script] <===> [kemi framework] <===> [C implementation of function] In addition, kemi allows to write the routing blocks all in embedded language. Core parameters, loading modules and modules' parameters stay like so far, but there is no need to write request_route{}, failure_route, etc.. blocks. Some functions with a specific names have to be written in the embedded language script, see more at: - https://www.kamailio.org/wiki/devel/config-engines In several days I expect to be possible to have routing blocks of kamailio-basic.cfg written via kemi interpreters. That will allow testing the performance impact. Right now Lua and Python use different mechanism of exporting Kamailio C functions, the one from Python should be faster, but relies on defining some C wrappers inside the module. Anyhow Lua might be faster as interpreter itself -- and at the end it can just get the same mechanism as Python. app_lua already supports reloading the routing script without restarting kamailio, one of the features asked from time to time. Besides that, of course, a big benefit is the access to a very large set of extensions already available as Lua/Python libraries. Also important, many peoples are already familiar with these languages and there is plenty of good documentation about those languages. The classic/native kamailio.cfg format (and the interpreter for it) stays there -- same common C functions are shared with kemi, so any addition in the future will be visible to everywhere. The native interpreter will remain the option for extreme optimizations for those that deal with enormous amount of SIP traffic (e.g., heavy load balancers), but I expect that at least Lua will be in pair of performances with a registrar deployment. I hope I could shed some light on various aspects of the new framework for embedded interpreters. Expect very soon news about ability to build full routing logic in Lua/Python, hopefully many of you will join to test the results and compare with native alternatives. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

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irc devel meeting summary
by Fred Posner 22 Apr '16

22 Apr '16

Greetings, Minutes from the recent devel meeting have been posted to the wiki: https://www.kamailio.org/wiki/devel/irc-meetings/2016a-minutes In summary, this was a 3+ hour meeting held on IRC #kamailio -- one of the longer meetings on record. No current "critical" issues were open, and the meeting moved to a discussion of for a testing framework and being able to load as many modules as possible in a running Kamailio system. Olle Johansson suggested a competition for Kamailio World: "The one with a kamailio with most loaded modules running at kamailio-world gets a beer." A discussion then occurred regarding the appropriateness of returning a 200 OK when no data exists during a xmlrpc request. The consensus seemed to side with how we currently return data with the suggestion of moving further discussion to the mailing list. The Kamailio team will be looking to both upgrading the Kamailio servers to Debian Jessie as well as utilizing a "responsive" template for the main website; making the site more mobile/user friendly. Kamailio 5 was discussed with a suggestion for a developers meeting in Stockholm, perhaps in June, to hammer out the framework. Daniel was thanked for moving forward with the Lua routing aspects of Kamailio 5 while ensuring that you will still be able to utilize the config file (as current) without Lua. Additional languages, such as python, will be coming. Daniel will be working on a tutorial for exporting the kamailio.cfg to an embedded interpreter in the future. As part of Kamailio 5, the source tree structure will be improved, most likely utilizing subdirectories to better organize the different elements of the software. There will also be a change in the method of checking the database schema for compatibility with the version of software running. This summary represents just an appetizer of the incredible meeting that took place. The minutes are available for you to read at your leisure and your participation is always encouraged through the mailing lists, IRC, as well as github. The meeting never ends... it just continues on the mailing lists. =) Best regards, Fred Posner http://www.palner.com

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A: Open discussion about Kamailio in about 30min
by Daniel-Constantin Mierla 22 Apr '16

22 Apr '16

Hello, long time involved in VoIP and friend of Kamailio, Suzanne Bowen has organized an open discussion about Kamailio on a web platform blah.im. The link of the session is: - https://blab.im/didx-exchange-kamailio-world-talk-on-privacy-scalability-re… It is planned to start at 12:00GMT (13:00 London, 14:00 Berlin), I will join it and I expect that the platform is allowing many to watch and ask questions. Hopefully few other Kamailio folks will join the conversation. Rather short notice here, but if you have something to ask about SIP, VoIP, Kamailio, you should try to participate. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

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BYE packet looping and loose routing
by Yasin CANER 22 Apr '16

22 Apr '16

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The book of "SIP Routing with Kamailio"
by Jay Li 22 Apr '16

22 Apr '16

Dear all, I'm looking for a document/book that explains the kamailio.cfg in detail. I've gone over the suggested study material "SER-Getting Started" which is such a nice book that explains ser.cfg line by line. I know Asipto has published a book named "SIP Routing with Kamailio". Just wondering if "SIP Routing with Kamailio" is written in a similar format when it comes to explain the kamailio.cfg. Hopefully I can get some information about the book before I make the purchase. Thanks. Regards,Jay

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Next Kamailio IRC Devel Meeting, Apr 20, 2016
by Daniel-Constantin Mierla 21 Apr '16

21 Apr '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, April 20, 2016, with alternative for next day, April 21. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016a Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

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DMQ_USRLOC sync CSEQ error
by José Seabra 21 Apr '16

21 Apr '16

Hi There, I'm using dmq and dmq_usrloc between 2 kamailio(version 4.4.0) registrar servers. I'm registering 15000 subscribers with sipp and everything is ok, the registration address of record are shared between both registrar servers. The problem here is when I restart one of the registrar server and then the other registrar server starts synchronization to restarted server. The error that i see on this process is the following shown bellow: ERROR: dmq_usrloc [usrloc_sync.c:75]: add_contact(): Invalid cseq But all the contacts are synchronized, what can cause this error? and what is the impact on normal behavior of kamailio? Best regards -- José Seabra

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about xml rpc response behavior
by Yasin CANER 21 Apr '16

21 Apr '16

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kazoo database not bound error
by Jayesh Nambiar 21 Apr '16

21 Apr '16

Hello, Using kazoo only to connect to amqp gives an error if the db_url is not specified. If db_url is specified it tries to do presence related stuff even if not intended. The solution was to use: modparam("kazoo", "pua_mode", 0) which I found here: http://lists.sip-router.org/pipermail/sr-users/2015-September/090092.html This is not documented in the kazoo module. Would be great if this is added in the module documentation to simplify the use of module. Thanks, - Jayesh

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How to distinguish 200 OK that have byed
by 张顺通 20 Apr '16

20 Apr '16

A KA B INVITE -> 200 OK(INVITE) <- ACK is lost BYE <- 200 OK(BYE) -> 200 OK(INVITE) <- 200 OK(INVITE) <- 200 OK(INVITE) <- ACK is lost AND A sent BYE cmd,B do not recv ACK and is also Retransmission 200 OK, How can I distinguish this 200 OK and drop it? thaks

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A: Kamailio World 2016 - Four Weeks Before
by Daniel-Constantin Mierla 20 Apr '16

20 Apr '16

Time is passing and Kamailio World Conference 2016 is approaching at fast pace – only four weeks left till the start of the event! The schedule [1] is pretty much nailed down, with some adjustments still expected to happen. The event starts like the past edition with a half a day of technical workshops, followed by two full conference days. A larger group of speakers [2] is participating to this edition. There was a big number of speaking proposals and we wanted to highlight more of the people that had a relevant contribution to the evolution of the project. To accommodate properly, two more discussion panels were added, keeping also the classic VUC panel. The topics cover many of the interesting aspects of real time communications, from security and scalability to WebRTC and VoLTE, touching Kamailio and other open source projects like Asterisk or FreeSwitch. More details can be found on the website of the event: - https://www.kamailioworld.com Don’t forget that this year Kamailio celebrates 15 years of development, the party is at Kamailio World! We expect to fill the capacity of the conference room, if you haven’t registered yet and plan to attend, do it as soon as possible to secure your seat! [3] Many thanks to our sponsors [4] that made possible this event: FhG Fokus, Asipto, Sipwise, Matrix.org, Sipgate, Simwood, NG Voice, Digium, VoiceTel, Evariste Systems, Core Network Dynamics, Pascom, Didx.net. Thank you for flying Kamailio and looking forward to meeting many of you at Kamailio World 2016! Daniel [1] https://www.kamailioworld.com/k04/schedule/ [2] https://www.kamailioworld.com/k04/speakers/ [3] https://www.kamailioworld.com/k04/registration/ [4] https://www.kamailioworld.com/k04/sponsors/ / / -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

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I have kamailio with drouting modulo and I want to pass only the RTP trafic to a FREESWITCH server
by Humberto H 20 Apr '16

20 Apr '16

Hello: I have kamailio with drouting and I want to pass only the RTP trafic to a FREESWITCH o what ever you think is better to manage de RTP traffic. What opcions do I have? I am a little bit confused using WITH_FREESWITCH. I will appreciate your help, Thanks

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another bug with JSONRPC-S?
by Brooks Bridges 20 Apr '16

20 Apr '16

I may have stumbled across another bug with this module. I believe the htable.dump method isn't correctly recognizing that a table name is being provided: [root@server ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "htable.dump", "name": "callstats"}' http://1.2.3.4:5060/jsonrpc {"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}} Debug log: 2016-04-19T17:42:30.046892+00:00 server /usr/local/sbin/kamailio[21634]: DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf=#012HTTP/1.1 500 No htable name given#015#012Sia: SIP/2.0/TCP 1.2.3.4:54912#015#012Content-Type: application/json#015#012Server: kamailio (4.4.0-rc1 (x86_64/linux))#015#012Content-Length: 69#015#012#015#012{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}} Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

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$shtval(htable=>key)
by Brooks Bridges 19 Apr '16

19 Apr '16

What is this? It's referenced here: http://kamailio.org/docs/modules/stable/modules/htable.html#idp1912904 but I cannot find it anywhere else. The only 2 places I find any reference to it at all in the entire codebase is here: modules/htable/doc/htable_admin.xml: <emphasis>$shtval(htable=>key)</emphasis> modules/htable/README: * $shtval(htable=>key) Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

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Kamailio Blind Call Forwarding
by Nagorny, Dimitry 19 Apr '16

19 Apr '16

Hi all, I'm trying to make a blind call forward to "1002" if a UA calls "1" into Kamailio. At the moment I have the following config: if ( $rU=~"^[1]$" && src_ip==$sel(cfg_get.pstn.gw_ip) ) { sl_send_reply("181", "Redirecting"); $ru = "sip:1002@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port) + ";user=phone"; force_send_socket(udp:MY_EXTERN_IP:5060); route(NORMAL_RELAY); #Just t_on_branch/reply/failure if (!t_relay()) { sl_reply_error(); } exit; } The calling UA is from the PSTN as well as the called UA. Problem right now is that the PSTN's anti-Loop protection jumps in and rejects the INVITE to 1002. Question: Is it possible to tell the calling UA "Hey wait, 181..." with Kamailio, then build seperately an independent Call Leg (with new CallID?) from Kamailio to UA "1002" and finally to put them together? Any help is very much appreciated even if they are just guides/examples. I'm using Kamailio 4.3.5 on CentOS 7.2. Very Respectfully Dimitry Nagorny Trainee

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Cleaning merged branches
by Daniel-Constantin Mierla 18 Apr '16

18 Apr '16

Hello, it looks that many branches still exist after being merged, discovered by running git-sweep. The list is next. If anyone from devs or users spots a branch that was not merged and needs to be reviewed, let me know. In several days I plan to remove them to keep the git repo a bit more sane. Cheers, Daniel $ git-sweep preview Fetching from the remote These branches have been merged into master: andrei/blst_send_flags andrei/cancel_reason andrei/cdefs2doc andrei/counters andrei/fixups andrei/mod_f_params andrei/mod_register andrei/module_interface andrei/path andrei/pointer_alias_warnings andrei/raw_sock andrei/rpc_async andrei/rve_f_params andrei/script_vars andrei/send_flags andrei/shm_early_init andrei/switch andrei/tcp_tls_changes andrei/to_parser_fix andrei/type_conversion co/crypto_name_collision co/tcp_closed_event_enhancements curl cvs-head ez/newstuff henning/trie henning/trie_mods janakj/missing_imports kamailio_modules lazedo-patch-1 lazedo-patch-2 lazedo/presence_improvements lazedo/presence_xml_fix_dummy luismartingil/msrp_crash mariusbucur/dmq mariuszbihlei/dnssec misi/ua-profile sd_journal_send_xavp ser_core_cvs ser_modules tirpi/cfg_framework_multivalue tirpi/script_callbacks tmp/build_local tmp/build_request tmp/core_events tmp/hpw_curl_improvements tmp/k3.0_sr_backports tmp/ruri_branch tsearle-futex-try tteras/db_sqlite -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

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loose_route fails
by Igor Potjevlesch 18 Apr '16

18 Apr '16

Hello, I'm trying to understand why a RE-INVITE is still discard by Kamailio (4.2.6). The RE-INVITE has the last Contact URI as Request-URI which is not Kamailio himself. But there are the route headers set. What are the other checks done by the function "loose_route()"? Regards, Igor.

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DISPATCHER module No destinations availables even if my Asterisk Boxes are up and running
by Sébastien Brice 18 Apr '16

18 Apr '16

Hello everyone. I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 503 error (please look at my attached kamailio.cfg) I use "ds_probing_mode", 1 and "ds_ping_reply_codes", "class=2;class=3;class=4" to OPTIONS ping the two asterisk instances but they do appear down both all the time. That is very annoying none of the INVITE go through Asterisk, they all are stopped because of my failure_route that sets no destination available ERROR: <script>: Destination down: INVITE sip:9980048587420715@XXX.XXX.XXX.XX (sip:ipofAsterisk1:5060) ERROR: <script>: Destination down: INVITE sip:9980048587420715@YY.YY.YYY.YY (sip:ipofAsterisk1:5060) ERROR: <script>: Destination down: INVITE sip:000442030044700@XXX.XX.X.XXX (sip:ipofAsterisk2:5060) ERROR: <script>: Destination down: INVITE sip:000442030044700@YY.YYY.YY.YYY (sip:ipofAsterisk2:5060) NOTICE: <script>: Media server sip:172.16.0.202:5060 failed to answer, selecting other one! NOTICE: <script>: No destination, sent 404 I don't know what's wrong with my setup, could someone give an eye to my kamailio.cfg and head me on the right track ? Thanks a lot. Seb

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mod_com_amd only sees silence
by Michael Jepson 18 Apr '16

18 Apr '16

When placing a call and checking for an answering machine using mod_com_amd, it always sees *silent_state*. Strangely, when I stream audio TO the session, mod_com_amd registers this audio and responds accordingly. So, it seems that somehow mod_com_amd is listening to the wrong end of the session. Any help would be greatly appreciated. Best regards, Michael Jepson

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Check if xavp is set
by Marrold 17 Apr '16

17 Apr '16

Hi all, What's the most efficient way to check if an xavp is set? I have tried- is_avp_set() from avpops module which resulted in- ERROR: avpops [avpops.c:915]: fixup_is_avp_set(): bad attribute name <$xavp(tm_contacts)> and if ($xavp(tm_contacts)) {} which resulted in - WARNING: <core> [rvalue.c:1007]: rval_get_int(): automatic string to int conversion for "<<xavp:0x7feac74cfcb8>>" failed To give a little background, I am using the UAC_redirect module and t_next_contacts() in conjunction with 302 redirects .However if there is only a single contact, $xavp(tm_contacts) is not created and t_next_contacts() fails, hence why I'd like to check if it exists first. Thanks

2 2

Kamailio not replay response
by Shiv Patidar 17 Apr '16

17 Apr '16

In kamailio server register is done with 200ok response and now i send invite but kamailio not give replay 200ok What i need to do configure of kamaulio file or not And kamailio.log file show some error like Parse_first_line():parse_first_line: bad message And Parse_msg:message=<INVITE ....to, from,via all filed.......> And Receive _msg(): CORE PARSING OF SIP MESSAGE FAILED

2 2

Random number function?
by Brooks Bridges 15 Apr '16

15 Apr '16

I feel like I have to be missing something obvious here, but I can't for the life of me find anything. Is there not a basic random number/string generation function in core or any of the modules anywhere? I find that incredibly unlikely that such a simple function that could have so many uses wouldn't exist, but I just can't find anything... Any pointers? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 1

Siremis install error
by Serge S. Yuriev 15 Apr '16

15 Apr '16

Hello, I'm trying to setup latest Siremis from GIT and stuck on DB stage. ERROR: SQLSTATE[08006] [7] FATAL: password authentication failed for user "www-data" FATAL: password authentication failed for user "www-data" Why installer tries to use web server user instead of supplied? DB is PGSQL at localhost. Kam DB created successfully via kamdbctl. -- Serge S. Yuriev Lead VoIP engineer

3 6

sipcapture module
by Ali Taher 15 Apr '16

15 Apr '16

Hello, I'm using sipcapture module in Kamailio to capture sip packets and save them into mysql database. Everything is working as expected and I got the packets registered in the database , yet I'm wondering if there is a way to open these packets directly in Wireshark , or do I need an intermediate step to convert these packets to pcap format (if possible) in order to be read by Wireshark. Any help or hint would be appreciated. Ali Taher Technical Support Engineer T&F M W A +961-5-457437 Ext 215 +961-70-010600 <http://www.vanrise.com/> http://www.vanrise.com <https://maps.google.com/maps/ms?msa=0&msid=208243362929850217065.0004cbc8d6 12c5f9b4237&ie=UTF8&t=h&vpsrc=0&z=19&iwloc=0004cbc8e4652b4edf71cC:\Users\Mar ie\Documents\My%20Received%20Files> Vanrise Building, Facing Galerie Semaan Hazmiyeh Highway - Beirut, Lebanon Join us at <http://www.facebook.com/pages/Vanrise-Solutions-Offshore/128309743915533> <http://www.linkedin.com/company/vanrise-solutions> <http://twitter.com/VanriseSolution>

2 1

sipcapture module with wireshark
by Ali Taher 15 Apr '16

15 Apr '16

Hello, I'm using sipcapture module in Kamailio to capture sip packets and save them into mysql database. Everything is working as expected and I got the packets registered in the database , yet I'm wondering if there is a way to open these packets directly in Wireshark , or do I need an intermediate step to convert these packets to pcap format (if possible) in order to be read by Wireshark. Any help or hint would be appreciated. Regards, Ali

2 1

Help with Kamailio app_java module
by Clarence Sandjon 15 Apr '16

15 Apr '16

Hi! I am trying to use the app_java module with kamailio-4.4.0 SIP server. Every time I load the app_java module, my users can't register and the sever replies with destintion unreachable (port unreachable). How can I fix this issue? Also, how can I use the Kamailio.java file since it doesn't have a main method? Thanks

2 2

should the AVPs be available in CANCEL?
by Andrew Pogrebennyk 14 Apr '16

14 Apr '16

Hi all, it seems that the AVPs not available when when processing CANCEL message, even though they have been set for this transaction initially. Is this the expected behavior? P.S. kamailio 4.3.4 Andrew

2 2

uac "restore_mode" is auto: effective when using core variables?
by Andrew Pogrebennyk 14 Apr '16

14 Apr '16

Per my understanding the uac module stores the "vsf" parameter in Record-Route and should be able to update the From/To URIs automatically in all in-dialog requests that carry this parameter. http://kamailio.org/docs/modules/stable/modules/uac.html#uac.p.restore_mode “auto” - all sequential requests and replies will be automatically updated based on stored original URI. For this option you have to set “modparam("rr", "append_fromtag", 1)”. What makes me wonder is: does that only work in From/To was changed by uac_replace_from()/uac_replace_to() or also when assigning directly to the $fu and $tu variables? I'm changing those variables and I am using restore_mode auto but that does not change anything in in-dialog ACK. I presume this is expected behavior because I'm not using uac when assigning to variables, isn't it? Thanks. Andrew

2 1

A real mystery - 1.5.x vintage edition!
by Alex Balashov 14 Apr '16

14 Apr '16

Hi, I've got an installation that is still running Kamailio 1.5.x. Yes, I know: "They're running WHAT?" As far as I'm concerned, it was EOL'd in early 2010. Nevertheless, that doesn't stop the customer from running it. They're probably the most change-resistant customer in the history of IP telecom, and, knowing this industry, that's a high bar. There's not much I can do about it. Anyway, I have a bit of a mystery: GW Proxy PBX ================================================== ------- INVITE ------> <---- 100 Trying ---- -------- INVITE --------> <------ 100 Trying ------ <- 503 Service Unavailable ---------- ACK ---------> <- 500 Service Una... -------- ACK --------> There is no basis in the configuration for the translation of the "503 Service Unavailable" reply to "500 Service Unavailable", so I'm thinking it has to be rooted in some default module behaviour. Any insight would be appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 5

RFC6442: geolocation
by Michael Jepson 14 Apr '16

14 Apr '16

Hello everybody, Can anybody help me out with sending a geolocation in the SIP messages that Kamailio sends? In order for the emergency telephone number to work properly, we need to send the location of our SIP trunk customers along with their INVITE when calling the emergency phone numbers. Is there any module that will facilitate this, or is there another way to add this information in the INVITE message? Thanks in advance! Best regards, Michael Jepson

1 0

Kamailio setting the reqeust URI
by NITESH BANSAL 13 Apr '16

13 Apr '16

Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in SIP INVITE coming from Asterisk looks like this<sip:kamailio@x.x.x.x>But my objective is to use Kamailio to forward the call to a remote endpoint. What header should I put in the SIP INVITE from Asterisk to Kamailio to conveythat Kamailio should use this 'SIP URI' to route the call onwards.I tried 'Route' header, but it doesn't seem very clean, as kamailio doesn't updatethe Request-URI in the forwarded INVITE if I use the Route header. Thanks,Nitesh

3 4

Kamailio TLS and SIP-Package building
by Nagorny, Dimitry 13 Apr '16

13 Apr '16

Good afternoon all, I have two questions regarding my configs: kamailio.cfg<http://pastebin.com/32PCh8n0> tls.cfg<http://pastebin.com/gnWZeD9e> . I am using Kamailio 4.3.5 on Cent OS 7.2 and all relevant pakage up-to-date. 1.) When I try to start Kamailio I get the following warning: WARNING: tls [tls_mod.c:287]: mod_init(): tls support is disabled (set enable_tls=1 in the config to enable it) Q: As you can see (line 36) TLS is enabled in config. So what could be the problem? 2.) Lines 324-330 indicate that I want to redirect to 2031 if I'm called at 1. For some reason the SIP-Trace tells me its still building the INVITE with the To-Header to sip:1@PSTN:5060 and the PSTN can't find 1, because no 1 is registered. Q: Why? (Outbound is set the same way and working perfectly). Info: I tried already with uac_replace_to and uac_replace_from w/o success. I would very much appreciate any hints on solving these issues. Very Respectfully Dimitry Nagorny Trainee

2 2

Ldap authentication
by Volkan Oransoy 13 Apr '16

13 Apr '16

Hi all, I am trying to use an ldap directory for authentication and found this tutorial. https://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth What I want is to use ldap bind with user's existing ldap password, not a seperate sip password. I would like to use same password for simplicity. There are two ways in my mind to achieve this and I need suggestions. One way is to add a new field to the ldap tree and populate it with user password's ha1 values but this needs an additional work and doesn't seems nice to me. Second way is to get clear-text password from auth request and bind to ldap server with this password but I don't know if I can access to the password or it this is a secure way. Does anybody have any suggestions or experience to share? Regards, /Volkan

1 0

Misrouting ACK packet for 200 Ok messages.
by Yasin CANER 13 Apr '16

13 Apr '16

4 3

Stable DEB
by Serge S. Yuriev 13 Apr '16

13 Apr '16

Hi, Shouldn't http://deb.kamailio.org/kamailio point to 4.4 version? And question about recent Ubuntu version support: Trusty is way to far behind latest Willy -- Serge S. Yuriev Lead VoIP engineer

2 2

Detecting calls with missing ACK (Lazy SIP scanners)
by Marrold 13 Apr '16

13 Apr '16

Hi, I have been running a couple of Asterisk honey pots to get a better understanding of the tools and methods potential hackers are using to exploit SIP servers. I have observed many attacks from the 'sipcli' user agent that don't send ACKs. At this stage I'm not sure what they're trying to achieve, whether it's a successful call to one of their test numbers, or maybe they will brute force anything that returns a 401 later, or maybe they're waiting for a 18X response. Below are three typical scenarios- ------ INVITE ------ > <--- 100 Trying --- <----- 200 OK ----- <----- 200 OK ----- <----- 200 OK ----- ( No ACK) ------ INVITE ------ > <-------- 503 -------- <-------- 503 -------- <-------- 503 -------- ( No ACK) ------ INVITE ------ > <-------- 401 -------- <-------- 401 -------- <-------- 401 -------- ( No ACK) Please could anyone point me in the right direction to detect these non completed calls with a missing ACK in Kamailio? I am unsure on the terminology I should be using to search the online documentation. Thanks

5 11

Kamailio 5.0 - Routing blocks in embedded scripting languages - Lua
by Daniel-Constantin Mierla 12 Apr '16

12 Apr '16

Hello, based on some of the plans for Kamailio 5.0 (https://www.kamailio.org/wiki/devel/kamailio-5.0-design), I started the implementation for the support to allow writing the active routing logic at runtime (respectively what are now the routing blocks) in other embedded languages than the native kamailio.cfg interpreter. Of course, besides a lot more extensive set of functions and expressions coming with the well established scripting languages, this way will allow reloading the logic without kamailio restart. I already committed the foundation for it, currently the request_route{} and reply_route{} can be implemented in Lua, relying on app_lua module. Lua was selected for first experiments given that I have some past experience with it, but other languages such as Perl, Python will hopefully follow soon, once the framework gets to a stable design. I wrote some initial docuentation on the wiki: - https://www.kamailio.org/wiki/devel/config-engines Would be helpful if people with experience on the other supported embedded languages will join the effort to design the framework to suit those languages, as well as other developers and community members to start exporting C functions. Also, there are some other things to decide on, listed in the to-do section, where suggestions/assistance are appreciated: - https://www.kamailio.org/wiki/devel/config-engines#to-do Many of them probably will be approached during the planned IRC devel meeting next week: - https://www.kamailio.org/wiki/devel/irc-meetings/2016a Any feedback is welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Strange behaviour in working with CNXCC
by Dirk Teurlings 12 Apr '16

12 Apr '16

Hi, Working with CNXCC for quite some time now with success. Though running into the following problem. Running Kamailio 4.4 (Debian Wheezy) from the Kamailio repository. Setting up CNXCC with a dialog start event route, a dialog end event route and a cnxcc:shutdown event route. In the start event I set some variables in $var_dlg, because I will need them later on in the dialog, could be the end event, or the shutdown event. Now the strange thing is, that in the shutdown event, when CNXCC runs out of credit for instance, I no longer have access to the $var_dlg pool. All of the variables have been lost. Another strange thing is, if I cut off the call with CNXCC after just 1 second, the variables all are still available. But after 5 seconds cutoff, all variables are lost. On another system, running a 4.2 Kamailio with a CNXCC ported from 4.1, this still works fine. I've also tested this with Kamailio 4.2 and 4.3, with the same CNXCC code from 4.4. Any insight would be greatly appriciated! Cheers, Dirk

1 0

4.4 crash at start
by Juha Heinanen 12 Apr '16

12 Apr '16

I build kamailio from latest 4.4 git source and got crash at start: (gdb) where #0 0x0000000000449af8 in add_interfaces ( if_name=0x7f0e770106e8 "192.98.102.10", family=0, port=13880, proto=1978, ai_l=0x1277600) at socket_info.c:1210 #1 0x000000000044ad77 in fix_socket_list (list=0x5, type_flags=0xffffffff) at socket_info.c:1406 #2 0x000000000044de70 in fix_all_socket_lists () at socket_info.c:1807 #3 0x000000000041b535 in main (argc=0, argv=0x7fff8cfaf1e8) at main.c:2390 -- Juha

3 9

Extracting P-Asserted-Identity value
by Grant Bagdasarian 11 Apr '16

11 Apr '16

Hello, How do I extract the value of "tel" from multiple P-Asserted-Identity headers? Example: P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@cisco.com> P-Asserted-Identity: tel:+14085264000 What I got so far is: $var(p_asserted_identity) = $(hdr(P-Asserted-Identity)[*]); $var(caller_id) = $var(p_asserted_identity){param.value,'tel',':'}; But this does not work. I keep getting parser errors when doing the transformation (2nd line of code example): I'm not sure how to format the line of code. I tried multiple ways, but none work. Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 60-64: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 65: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 66-68: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: Regards, Grant

3 6

Help with Kamailio java module
by Bonjour Madame 11 Apr '16

11 Apr '16

Hi! I am a newbie and I need help understanding how to use the java module with Kamailio. I am trying to create a basic java application that will print the SIP message type being processed by Kamailio. I read the app_java module but can't still figure it out. Any help or reference will be greatly appreciated. Thanks Sent from Mail for Windows 10

2 3

Error in USERLOCDB
by Alberto Sagredo 11 Apr '16

11 Apr '16

Im getting this error using USERLOCDB Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Duplicate entry '0--1' for key 'connection_idx' (1062) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [ucontact.c:684]: db_insert_ucontact(): inserting contact in db failed xxxx (uloc-5706bebd-7415-3) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [urecord.c:591]: insert_ucontact(): failed to insert in database Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: registrar [save.c:522]: insert_contacts(): failed to insert contact My config # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_ops_ruid", 1) modparam("usrloc", "db_check_update", 1) #!endif I have been using it without any problems in other installations but in a new one with 4. KAMAILIO im getting this error. I can see registrations on kamctl ul show but seems that option 1 is not working . Option 2 does not update registrations from that device. Thanks

2 3

truncating sip message
by Yasin CANER 11 Apr '16

11 Apr '16

4 10

Multiple SIP-servers with SRV-records and authentication secrets
by Alfred E. Heggestad 09 Apr '16

09 Apr '16

Dear SIP-experts and DNS-SRV gurus; I have some questions to the deployers of SER/Kamailio and best current practice for multiple SIP-servers with SRV-records and authentication. This is not a question about Kamailio itself but rather experience with deployment of it in the field. The current usecase is: 1. Multiple SIP-servers are deployed for the same domain 2. The DNS is configured with SRV-records for load balancing, example: (lets call the domain "example.com") $ host -t SRV _sip._udp.example.com _sip._udp.example.com has SRV record 20 0 5080 alpha1.example.com. _sip._udp.example.com has SRV record 20 0 5080 alpha2.example.com. 3. when a SIP client registers, it resolves the domain using RFC3263 [1] and the first REGISTER request is sent to SIP-Server #1 4. SIP-server #1 replies with 401 containing the authentication challenge 5. The SIP Client adds the authentication header to the REGISTER request and re-sends it, but this time also using RFC 3263, and due to DNS rotation the request is sent to SIP-Server #2 6. Now, because the SIP-Servers are configured with _different_ secrets in the "auth" module [2], the REGISTER request fails with authentication error. Now, I know that it is common for SIP user-agents to send both requests to the same SIP-server instance. Baresip [3] is not doing that, it does a new RFC 3263 lookup for all requests (except e2e ACK/CANCEL). so here are my questions: - What is common practice in the field, to configure auth module with the same "secret" or different "secret" values? - Do you know if there is any reference to IETF documents about how this should be handled? RFC 3263 says that every request should be resolved, except: "The procedures here MUST be done exactly once per transaction, where transaction is as defined in [1]. That is, once a SIP server has successfully been contacted (success is defined below), all retransmissions of the SIP request and the ACK for non-2xx SIP responses to INVITE MUST be sent to the same host. Furthermore, a CANCEL for a particular SIP request MUST be sent to the same SIP server that the SIP request was delivered to." - What is common practice for SIP user-agents to do in this case? /alfred [1] https://tools.ietf.org/html/rfc3263#section-4.4 [2] http://www.kamailio.org/docs/modules/3.4.x/modules/auth.html#auth.secret [3] https://github.com/alfredh/baresip/issues/39

4 4

Kamailio Updates on VUC 588 - this week on Friday
by Daniel-Constantin Mierla 08 Apr '16

08 Apr '16

Hello, The weekly VoIP Users Conference (VUC) #588 moderated by Randy Resnick will host a session for Kamailio Updates. Everyone from community is invited to join and actively participate to discussions. There are many ways to connect, from SIP to classic PSTN dial in as well as listening or watching live streams on the web. The live session will start on Friday, April 8, at 18:00 Berlin timezone (17:00 London, 12:00EDT). More details about the session are available at: - https://www.kamailio.org/w/2016/04/vuc-588-kamailio-updates/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Question about debugging using `kamcmd pkg.stats`
by Lucian Balaceanu 08 Apr '16

08 Apr '16

Hi Daniel, While debugging a 'no private memory left' issue by using `kamcmd pkg.stats` we stumbled upon data we don't understand. Could you help with some information? `kamcmd pkg.stats`returns free values of about 8M per process with an exception: kamailio with no apparent problems: `timer` process: free value: 8M used value: 100M real_used: 308M. kamailio with no private memory left: `timer` process: free value: 8/9M used value: 98M real_used: 662M (both mentioned kamailios have roughly the same load) Are these timer values normal? What is this real_used value and how can it be bigger than the allocated pkg mem? Additionally, how come we don't see the fifo process pkg stats in the output of `kamcmd pkg.stats`? We get the error message when doing reloading backlists via kamctl fifo. Thank you, Lucian Balaceanu

2 6

Compatibility between Siremis 4.2 and Kamailio 4.4
by Jay Li 08 Apr '16

08 Apr '16

Dear all, I got into the Kamailio world recently, and just wanted to get some info here about the compatibility between Siremis and Kamailio. From what I read, Siremis is wonderful web management tool for Kamailio. However the latest Siremis to date is 4.2.0 which was released more than a year ago. From the release note, it was intended to work with Kamailio 4.2, but Kamailio 4.4 has already released. I'm wondering if there's known compatibility issues between Siremis 4.2 and later released of Kamailio, like 4.3 and 4.4. Any information is appreciated. Regards,Jay

2 1

Insert callerid in Kamailio
by Nelson Migliaro 07 Apr '16

07 Apr '16

Hello, I am planning to insert the caller id in my Kamailio before sending the call to sip vendor. I am planning to send a prefix before the number and use that prefix in the SQL query to select the CLID. I was thinking in the sqlops to select the CLID and then append the result in the Remote-Party-ID: modparam("sqlops","sqlcon","cb=>mysql://kamailio:abc@10.10.1.1/testdb") modparam("sqlops","sqlcon","ca=>dbdriver://username:password@dbhost/dbname") ... ... sql_query("ca", "select clid from mitable where prefix = $var(prefix)", "ra"); xlog("number of rows in table domain: $dbr(ra=>rows)\n"); sql_result_free("ra"); Question..... ¿Is there an specific module to do this or a there a better way? Thank you Nelson.-

2 1

SIP Message Body
by Vladimer Gabunia 07 Apr '16

07 Apr '16

How can I Catch Text Message in SIP Method Message ? ________________________________ [gh.ge] ვლადიმერ გაბუნია IT სამსახურის უფროსი ტელ: (+995) 32 2505222 +8183 მობ: (995) 577 095333 შპს "ჯეო ჰოსპიტალს" სათავო ოფისი თბილისი 0160, ვაჟა-ფშაველას გამზ. № 16; http://www.gh.ge <http://gh.ge>

5 4

rtp proxy truncates sdp body
by Yasin CANER 06 Apr '16

06 Apr '16

1 0

SQLOPS select database in realtime
by Jon Bonilla (Manwe) 06 Apr '16

06 Apr '16

Hi all I have multiple auth credentials split among multiple databases (same server). I want to extract the password from the correct database for each request and pass it as arameter to the auth functions. The problem is, how can I use sqlops and select the correct database in realtime? the db connection already contains the database to connect to. Would it if I use a default connection but I put the db I want i the query? Something like: select * from database.table ... select * from $avp(s:database).table where i=1... cheers, Jon

3 2

pseudo variables in strings
by Juha Heinanen 05 Apr '16

05 Apr '16

If I have a string that contains pseudo variables, e.g,, $var(uri_pattern) = "sip:$fU@tutpro.com"; is it somehow possible to replace them with their values, something like: $var(uri) = eval("$var(uri_pattern)"); -- Juha

3 8

about rtpengine versions
by Juha Heinanen 05 Apr '16

05 Apr '16

I noticed that 15 days ago there was rtpengine commit https://github.com/sipwise/rtpengine/commit/aac8899b612dc8103b89f3f9c921f88… titled "Release new version 4.4.0.0+0~mr4.4.0.0", but I could not find the corresponding branch or tag. If someone wants to use that version, is 'git archive aac8899b612dc8103b89f3f9c921f88af3501303' the only way to do so? -- Juha

2 2

Re: [SR-Users] Ka--כרmailio asאעא ער מאיכקטח- a presence server
by Shlomi Gutman 05 Apr '16

05 Apr '16

אני אה כן בתאריך 4 באפר׳ 2016 8:45 לפנה״צ,‏ "Daniel-Constantin Mierla" < miconda(a)gmail.com> כתב: > Hello, > > can you run with debug=3 in config file and see if you get some hints from > syslog messages about what happens ? > > Cheers, > Daniel > > On 01/04/16 17:17, Igor Olhovskiy wrote: > > Hi! > I want to create custom Presence server. Main idea - phones subscribes to > it (for ex, extension 100@domain). This subscription actually lit BLF > button on a phone. > But main idea - change status of BLF can arrive from different locations > via PUBLISH, I assume. > Problem - phones has different type of SUBSCRIBE, like > dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some > of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not > domain, but IP, but in From or To - domains are used) > Phones are registered elsewhere, not on Kamailio. After reading RFC > (especially event:dialog part) I’ve found about SIP-If-Match part, but > problem about sync this data across multiple PUBLISH locations. > Can be Kamailio be a presence server in this scenario? > > For ex, SUBSCRIBE > > SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport > From: <sip:608@master.rufan.at>;tag=521811605 > To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 > Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA > CSeq: 20202 SUBSCRIBE > Contact: <sip:608@192.168.88.60:5060> > X-Grandstream-PBX: true > Max-Forwards: 70 > User-Agent: Grandstream GXP2160 1.0.5.33 > Expires: 180 > Supported: replaces, path, timer, eventlist > Event: dialog > Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, > UPDATE, MESSAGE > Content-Length: 0 > > > I’ve tried send to Kamailio this PUBLISH > > PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 > Max-Forwards: 70 > To: <sip:*55@master.rufan.at> > From: <sip:*55@master.rufan.at:5060>;tag=1 > Call-ID: 1-6930(a)127.0.1.1 > CSeq: 1 PUBLISH > Contact: <sip:mod_sofia@127.0.1.1:5060> > Event: presence > Expires: 3600 > Content-Type: application/pidf+xml > Content-Length: 486 > > <?xml version="1.0" encoding="ISO-8859-1"?> > <presence xmlns='urn:ietf:params:xml:ns:pidf' > xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' > xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' > xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:* > 55(a)master.rufan.at'> > <tuple id='z9hG4bK-6930-1-0'> > <status> > <basic>open</basic> > </status> > </tuple> > <dm:person id='1'> > <rpid:activities> > <rpid:on-the-phone/> > </rpid:activities> > <dm:note>Night</dm:note> > </dm:person> > </presence> > > Kamailio answers with 200OK, but after this - no NOTIFY to subscribed > clients. > > -- > Best regards, > Igor > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Kamailio as a presence server
by Igor Olhovskiy 05 Apr '16

05 Apr '16

Hi! I want to create custom Presence server. Main idea - phones subscribes to it (for ex, extension 100@domain). This subscription actually lit BLF button on a phone. But main idea - change status of BLF can arrive from different locations via PUBLISH, I assume. Problem - phones has different type of SUBSCRIBE, like dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not domain, but IP, but in From or To - domains are used) Phones are registered elsewhere, not on Kamailio. After reading RFC (especially event:dialog part) I’ve found about SIP-If-Match part, but problem about sync this data across multiple PUBLISH locations. Can be Kamailio be a presence server in this scenario? For ex, SUBSCRIBE SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport From: <sip:608@master.rufan.at>;tag=521811605 To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA CSeq: 20202 SUBSCRIBE Contact: <sip:608@192.168.88.60:5060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.5.33 Expires: 180 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 I’ve tried send to Kamailio this PUBLISH PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 Max-Forwards: 70 To: <sip:*55@master.rufan.at> From: <sip:*55@master.rufan.at:5060>;tag=1 Call-ID: 1-6930(a)127.0.1.1 CSeq: 1 PUBLISH Contact: <sip:mod_sofia@127.0.1.1:5060> Event: presence Expires: 3600 Content-Type: application/pidf+xml Content-Length: 486 <?xml version="1.0" encoding="ISO-8859-1"?> <presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:*55@master.rufan.at '> <tuple id='z9hG4bK-6930-1-0'> <status> <basic>open</basic> </status> </tuple> <dm:person id='1'> <rpid:activities> <rpid:on-the-phone/> </rpid:activities> <dm:note>Night</dm:note> </dm:person> </presence> Kamailio answers with 200OK, but after this - no NOTIFY to subscribed clients. -- Best regards, Igor

3 4

dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)
by TEG AMJG 04 Apr '16

04 Apr '16

Hi I am new to Kamailio and i am just trying to setup a kamailio (v4.2.3) dispatcher to distribute calls to 2 asterisk using the call load distribution algorithm (alg 10). I started to test and see how all it is working and in the logs appears that the call load record it cannot be found. - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (call id generated by asterisk) I am getting this error trying a 2 call leg conversation when asterisk is creating the BYE message destinated to the callee after receiving a BYE message from the caller, and the BYE message by Asterisk reach Kamailio to then apply the ds_load_update function >From what i understand about ds_load_update it is that it sets the load to an specific destination if Kamailio receives a 2xx reply of an INVITE message, and removes it if Kamailio receives a BYE message based from the Call-id received in that message, where it is actually failing in my implementation at the moment Here are the blocks of code of the dispatcher parameters and where the function ds_load_update() is being used: #!ifdef WITH_LOADBALANCE #Parameters using Dispatcher from DB # ---------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") # ---------- #Enabling failover mechanism # ----------- modparam("dispatcher", "flags", 2) # ----------- #Setting dispatcher parameters in AVP # ----------- modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # ----------- #Setting dispatcher result in PV variables # ----------- modparam("dispatcher", "attrs_pvname", "$var(attrs)") # ----------- #Enabling dispatcher algorithm according to call loads # ----------- modparam("dispatcher", "ds_hash_size",8) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) # ----------- # Setting parameters for probing dispatcher destinations # ----------- modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@<Kamailio IP>") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404") # ----------- ... route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_LOADBALANCE if(is_method("BYE|CANCEL")){ xlog("L_ERROR","UBICAR BYE ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n"); ds_load_update(); } #!endif ... } ... onreply_route[MANAGE_REPLY] { #!ifdef WITH_LOADBALANCE if(is_method("INVITE")) { if(status=~"2[0-9][0-9]") { xlog("L_ERROR","UBICAR setting ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs"); ds_load_update(); }else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]"){ route(NATMANAGE); } } This is what i am getting from logs after the 2 call legs conversation completes: kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR: FROMASTERISK 70175143146f92975a5bff8876435911(a)192.168.10.194 kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194 method=INVITE callid=28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to= sip:104@192.168.10.194 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4083]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194:5060 method=INVITE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.10.194:5060 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4084]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.10.190:5060 from=sip:103@192.168.30.1:5060 method=BYE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.30.1:5060 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.30.2:22523;transport=UDP from= sip:103@192.168.10.194 method=BYE callid= 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to=sip:104@192.168.10.194 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for ( 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194) Kamailio IP: 192.168.10.194 Asterisk IP: +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | 1 | 0 | sip:192.168.10.190:5060 | 2 | 0 | duid=PBX1;my=pbx1;maxload=200 | pbx1 | | 2 | 0 | sip:192.168.10.191:5060 | 2 | 0 | duid=PBX2;my=pbx2;maxload=200 | pbx2 | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+

3 4

get_redirects with t_next_contacts
by Marrold 04 Apr '16

04 Apr '16

Hi all, I am having some issues using get_redirects() from the UAC_Redirect module in conjunction with t_next_contacts from the TM Module. I've searched Google and the list but the only references I can find are several years old. I am attempting to configure Kamailio so it if receives a 3XX redirect message, it will store the Contacts it contains, and attempt them in 'Q' parameter order. So for the Contact below- Contact: <sip:10.0.97.21:5060>;q=0.2,<sip:5.29.13.139:5060>;q=0.5. I want Kamailio to try 5.29.13.139, and if that times out, or is rejected in someway, try 10.0.97.21. If I use t_relay rather than t_next_contacts, kamailio branches the calls and relays them in parallel regardless of Q value - this is not what I want. If I use t_next_contacts, it attempts 5.29.13.139 which is rejected with a 401, but it doesn't attempt 5.29.13.139 and responds "408", "Request Timeout MH 2" Relevant script section is below, has anyone been able to get this working? failure_route[302] { if (t_check_status("302")){ xlog("L_ALERT","Failure Route: 302\n$mb"); get_redirects("*"); if (!t_load_contacts()) { send_reply("500", "Server Internal Error - Cannot load contacts MH"); exit; }; if (!t_next_contacts()) { t_reply("408", "Request Timeout MH"); } else { t_on_failure("302"); t_relay(); exit; }; } else if (t_check_status("[456][0-9][0-9]")){ if (!t_next_contacts()) { t_reply("408", "Request Timeout MH 2"); } else { t_on_failure("302"); t_relay(); exit; } } } Thanks, Matthew

1 2

Using 302 redirects with Dispatcher Module
by Marrold 04 Apr '16

04 Apr '16

Hi All, I'd like to use the Dispatcher Module along with 30X redirects, to monitor gateway availability with SIP options and to provide redundancy / fail over. Some sources online suggest it's as easy as - if ( method=="INVITE" ) { dst_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } However when the 30X reply is sent, it contains no contact header. Please could someone advise if it's possible to use the Dispatcher module with 302 redirects, and what they syntax is? Thanks, Matthew

3 6

kamailio number translation and route
by liwei＠neulinx.com 04 Apr '16

04 Apr '16

hi all; I want to use kamailio achieve: First phase: 1) inbound caller number translation; 2) outbound caller number translation; 3) inbound callee number translation; 4) outbound callee number tanslation; Example : Call Flow backend server ------------------> kamailio --------------------------> GateWay ------------------->PSTN caller -->callee caller ---> callee 1000 --> 2000 111000--->222000 inbound outbound caller --->callee caller ---> callee 1000 ---> 2000 11000 ---> 22000 | translation | | translation | 11000 ---> 22000 111000---> 222000 Second phase: / ---->gateway1 ----->pstn / backend server ---> kamailio ----->gateway2 -----> pstn \ \ ----> gateway3 ------->pstn Could you give me some suggest or example configuration？ liwei(a)neulinx.com

2 1

Load balancing traffic based on SIP URI
by NITESH BANSAL 04 Apr '16

04 Apr '16

Hello, I want to use Kamailio to load balance traffic across multiple asterisks.The problem is that my SIP traffic is related to conference, so if a SIP callfor a particular URI is sent to an Asterisk instance, any subsequent calls for thatSIP URI should be sent to the same Asterisk instance.Is there anyway to achieve this in Kamailio? Thanks,Nitesh

3 5

SIP Authentication
by Girish 04 Apr '16

04 Apr '16

Hello Sir, I am a newbie to Kamailio and I am trying my first setup. I would like to use Kamailio to take in traffic using a SIP Trunk and Route the calls to another SIP trunk based on prefix. I have added the incoming SIP trunk IP address to the address table on the MYSQL but I am unable to get a call routed out. Is there any sample configuration for Authentication via IP address and route the incoming SIP INVITE. Regards, Phoenix

2 1

RPID and Auth DB
by Alberto Sagredo 04 Apr '16

04 Apr '16

Hi! I plan to use rpid field but when trying to get from database, subscriber table has rpid field filled, i get always NULL. Is there any more needed than: modparam("auth_db", "load_credentials", "rpid") ? Just a check of variable xlog(" Valor de RPID es : $avp(s:rpid)"); Shows me NULL all times Thanks

2 1

KAMAILIO tcp_send query, connection problem
by Ed Todd 04 Apr '16

04 Apr '16

KAMAILIO tcp_send query, connection problem I have a KAMAILIO SIP server on a static IP from Comcast, using a modem/router Cisco DPC39412B. I have a PC application server which is successfully signing on to the SIP server, via a LAN cable on the same network as the SIP server (Cisco). In my second test, the PC application server is connected to the internet via a ZTE Falcon Z-917. In a third test, the PC application server is connected to the internet via a Talk-Talk fiber broadband. The connection does not occur. >From the log files, the successful connection says: Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (7817 (14)), reusing fd Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f6270b8a4a0 n=565 fd=1 On the failed connection: Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.0.146 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 55133, type 3 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 1776:802:0, 803 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSc<default A while later, I get: Apr 1 09:41:37 SIP-Server kamailio[7804]: DEBUG: usrloc [urecord.c:357]: wb_timer(): Binding 'hvs_us_iphone','sip:hvs_us_iphone@172.56.8.182:44341;transport=TLS;line=75dc89bd64369c7' has expired Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7f6270b8a4a0 (ticks=862851529, timeout=862909176 (3602 s), wr_timeout=862851529 (0 s)), write queue: 197 bytes Apr 1 09:41:38 SIP-Server kamailio[7820]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.0.146:55133 failed (timeout) Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7f6270b8a4a Any clues or tips would be greatly appreciated! Ed

2 1

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 03 Apr '16

03 Apr '16

Sure it is! This is exactly what I was looking for! Actually I wasn't able to see that there was a way to perform this action! Thanks for your help, Olle! I will try to create this, and when I have it done, I will post my script here, so if someone else need the same problem I can help. Now, I have just one more question about my scenario: when I register a user, is there a way to create something like a "custom field" on the location? To explain better: If possible, I can add something like a custom header that says if the user is registered from a phone device or the web interface, then I can perform the loop and replace only the oldest contact using that device... *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > This is my problem... I want to have 2 contacts for AOR, and not only one. > > When the 3rd arrives, it must remove the oldest and continue with 2... > > *If I am using the 0x04 parameter, it will save just one contact, and the > user will not be able to get the Invite both, in the mobile device and web > interface.* > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Hello, *Olle!* >> >> *Thanks for your help in this case, and don't worry about the time >> (actually I wasn't even expecting to get an answer today).* >> >> >> *I know the max_contacts parameter. But the problem is that if I set the >> max_contacts to 2, when the user tries to register again, he gets an error >> message 503. The point is that I want to get the user registered, and >> replace the oldest location entry for the newest one.* >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello all. >>> >>> I have problem here and I tried to find a solution and search over >>> internet, but without success. >>> >>> My scenario is the following: I have an application that must be >>> registered in Kamailio when a user logs in the web interface, so he can get >>> calls (something like a web softphone using webrtc). At this point, we are >>> OK, and everything is working fine. >>> >>> To get these register functions I am using the parameter >>> "save("location", "0x04")" as described in the REGISTRAR module >>> documentation, so if a user logs in another web browser or computer, only >>> the last one will continue registered and all calls will be forwarded to >>> him. >>> >>> The point is that now we are creating a phone app that will do almost >>> the same thing as the web interface, allowing users to receive calls using >>> the mobile device, and here is my problem: I want to allow my users to be >>> registered on two devices at the same time, but if a user logs into another >>> device, I don't want to reply with a 503. I want to allow the user to >>> register again, deregistering the oldest contact. >>> >>> I saw that there is a module named "ims_usrloc_scscf" and on its >>> description it says "implemented overwrite oldest contact behaviour", >>> but I couldn't find any documentation about it. >>> >>> So, is there a way to get this working today? >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >> >> >

3 2

kamailio proxy
by liwei＠neulinx.com 03 Apr '16

03 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

2 1

SQlite DB Lock with Presence
by Kristian Kielhofner 02 Apr '16

02 Apr '16

Hello everyone, I'm trying to use BLF with the config here (except with SQlite): http://kb.asipto.com/kamailio:presence:k43-blf BLF seems to work perfectly except after the call is ended I get "sqlite commit failed: database is locked" error messages as seen here: http://pastebin.com/xVijj98H BLF is then broken at this point. kamailio -v version: kamailio 4.3.5 (x86_64/linux) 950657 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, 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, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 950657 compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 Has anyone ever seen this before? Thanks! -- Kristian Kielhofner

3 9

MSRP question
by Hussein Said 02 Apr '16

02 Apr '16

2 4

kamailio proxy
by liwei＠neulinx.com 02 Apr '16

02 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

1 0

Trouble initializing TLS on Fedora 23
by Anthony Messina 02 Apr '16

02 Apr '16

Happy New Year! I've just upgraded my Kamailio (build from master@c7e411e) instance to Fedora 23 from Fedora 22. I've built the packages specifically for Fedora 23 with the following current sources: openssl-1.0.2e-3.fc23.x86_64 openssl-libs-1.0.2e-3.fc23.x86_64 openssl version reports... OpenSSL 1.0.2e-fips 3 Dec 2015 Even so, the following error occurs. It seems like Kamailio having trouble detecting that I'm using running with the same version that I have installed, and the same version that I have compiled against. tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.0-fips 29 Mar 2010" (0x10000003), compiled "OpenSSL 1.0.2d-fips 9 Jul 2015" (0x1000204f). Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) CRITICAL: <core> [main.c:2558]: main(): could not initialize tls, exiting... Any pointers? -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello, *Olle!* *Thanks for your help in this case, and don't worry about the time (actually I wasn't even expecting to get an answer today).* *I know the max_contacts parameter. But the problem is that if I set the max_contacts to 2, when the user tries to register again, he gets an error message 503. The point is that I want to get the user registered, and replace the oldest location entry for the newest one.* *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all. > > I have problem here and I tried to find a solution and search over > internet, but without success. > > My scenario is the following: I have an application that must be > registered in Kamailio when a user logs in the web interface, so he can get > calls (something like a web softphone using webrtc). At this point, we are > OK, and everything is working fine. > > To get these register functions I am using the parameter "save("location", > "0x04")" as described in the REGISTRAR module documentation, so if a user > logs in another web browser or computer, only the last one will continue > registered and all calls will be forwarded to him. > > The point is that now we are creating a phone app that will do almost the > same thing as the web interface, allowing users to receive calls using the > mobile device, and here is my problem: I want to allow my users to be > registered on two devices at the same time, but if a user logs into another > device, I don't want to reply with a 503. I want to allow the user to > register again, deregistering the oldest contact. > > I saw that there is a module named "ims_usrloc_scscf" and on its > description it says "implemented overwrite oldest contact behaviour", but > I couldn't find any documentation about it. > > So, is there a way to get this working today? > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com >

2 3

Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello all. I have problem here and I tried to find a solution and search over internet, but without success. My scenario is the following: I have an application that must be registered in Kamailio when a user logs in the web interface, so he can get calls (something like a web softphone using webrtc). At this point, we are OK, and everything is working fine. To get these register functions I am using the parameter "save("location", "0x04")" as described in the REGISTRAR module documentation, so if a user logs in another web browser or computer, only the last one will continue registered and all calls will be forwarded to him. The point is that now we are creating a phone app that will do almost the same thing as the web interface, allowing users to receive calls using the mobile device, and here is my problem: I want to allow my users to be registered on two devices at the same time, but if a user logs into another device, I don't want to reply with a 503. I want to allow the user to register again, deregistering the oldest contact. I saw that there is a module named "ims_usrloc_scscf" and on its description it says "implemented overwrite oldest contact behaviour", but I couldn't find any documentation about it. So, is there a way to get this working today? *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com

2 2

Make Kamailio Great Again!
by Alex Balashov 01 Apr '16

01 Apr '16

For immediate release: ATLANTA, GA (1 April 2016)--Alex J. Balashov, a self-styled businessman based in Atlanta, Georgia, USA, has a plan to "Make Kamailio Great Again". "Evariste Systems is huge. My name is on the building," said Balashov of his iconic VoIP consulting brand. "And you know what, I have been very successful. Everybody loves me." Balashov has capitalised on a contentious election cycle marked by deep political polarisation, growing income inequality and geopolitical challenges such as global terrorism. And his sharp message of alarm about the declining influence of the Kamailio SIP server project has resonated with increasing numbers in the CxO suite, vaulting him to the lead in the race for the IETF SIP Working Group nomination, according to recent polls of primary voters. He has been quick to tout his competitive credentials in a tough global open-source ecosystem. At a recent colloqium on unified communications, he asked: "When was the last time anybody saw us beating, let's say, OpenSIPS in Git commits? They kill us. I beat OpenSIPS all the time. All the time." As Balashov sees it, a major cause of the beleaguered Kamailio project's woes lies in its liberal patch acceptance policy and lax scrutiny of third-party contributions: "When GitHub sends its people, they're not sending their best. They're not sending you. They're not sending you. They're sending people that have lots of problems, and they're bringing those problems. They're bringing drugs. They're bringing crime. They're rapists. And some, I assume, are good people." He has proposed a controversial solution that has drawn ire from liberal ranks in the open-source community, but has also attracted applause and standing ovations at his speaking engagements: "We have to have a firewall around the Kamailio source code. We have to have an access control list. And in that firewall, we're going to have a big fat door where commits and pull requests can come into the master branch, but they have to come in legally. The firewall will go up, and GitHub will start behaving." Balashov's firewall proposal has been met with scorn from critics who deride it as impractical and quixotic. In particular, commentators have raised questions about funding and resources as well as GitHub's willingness to entertain a boundary around a project in its vicinity. Balashov isn't concerned, however: "I will build a great firewall--and nobody builds firewalls better than me, believe me--and I'll build them very inexpensively. I will build a great, great stateful packet inspection wall on our border with GitHub, and I will make GitHub pay for that wall. Mark my words." He has also been rebuked by rival IETF leadership candidates for his often acerbic Twitter remarks directed at Lennart Poettering and the developers of "firewalld". As he sees it, however, the network effects of social media are a strength: "My Twitter has become so powerful that I can actually make my enemies tell the truth." He scoffed at the suggestion that his characterisations of industry actors behind the RedHat-led "systemd" movement are misleading: "RedHat was the worst Steward of Linux in the history of the kernel. There has never been a Steward so bad as RedHat. The source code blew up around us. We lost everything, including all synergies. There wasn't one good thing that came out of that administration or them being Stewards of Linux." Balashov's idiosyncratic campaign is not standing still. He has proven to be a capable populist, adapting rapidly to an evolving sense of the kinds of pronouncements that activate his swelling crowds of devotees. Along the way, he has deftly deflected calls to subject his policy proposals to expert review. "I know what I'm doing, and I listen to a lot of people, I talk to a lot of people, and at the appropriate time I'll tell you who the people are. But I speak to a lot of people, but my primary consultant is myself, and I have a good instinct for this stuff." At a recent gathering of SIP stack interoperability specialists, Balashov the latest pillar of his platform to "Make Kamailio Great Again", in view of growing security vulnerabilities in the latest Kamailio modules: "Alex J. Balashov is calling for a total and complete shutdown of commits entering the master branch from the territory of the European Union until our project's representatives can figure out what's going on. According to Netcraft, among others, there are a lot of buffer overflows in Kamailio by large segments of the EU population."

2 1

Acc INVITE dialog start timing difference
by Dirk Teurlings - Signet B.V. 01 Apr '16

01 Apr '16

Hi, I'm seeing a difference in timestamp for acc modules recorderd INVITES and the dialog start event. This is the log entry: Apr 1 12:55:59 host /usr/sbin/kamailio[15642]: INFO: <script>: 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:5060 - dialog:start And this has been added by ACC in the database: INVITE as2880319e 56FE53B9-49E3774-3E86ED1B 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:50... 200 OK 2016-04-01 12:56:01 Why are these two seconds apart? Shouldn't the acc module record the same dialog start event time as the start of a dialog? When I look at the value of DLG_lifetime this confirms my findings, as it results in: dialog_time = TS - dlg_start When looking at the database BYE-time - INVITE time, the call is two seconds shorter. Any insight would be appreciated! Cheers, Dirk

1 0

Kamailio start up issue
by NITESH BANSAL 01 Apr '16

01 Apr '16

Hello, I installed Kamailio 4.3 on debian wheezy machine using debian package from https://www.kamailio.org/wiki/packages/debsThe installation went fine, but I'm unable to start kamailio. It seems that Kamailio is having issue locating the issues.Kamailio installed all the modules in location '/usr/lib/x86_64-linux-gnu/kamailio/modules', when building from source, it usedto copy modules in '/usr/lib/kamailio/modules'.Any suggestions on how can I fix it? Thanks,Nitesh

1 1

Framework/Lib to create SIP packets
by Igor Olhovskiy 01 Apr '16

01 Apr '16

Hi, all Not a Kamailio-related question, but. I want to form SIP-packets with info I need and receive answers. At first iteration in would be PUBLISH (with XML info) messages (PUBLISH - 200 OK) sent to Kamailio. I want to use Python (more preferable) or Lua (less preferable). Is there any lib I can create messages with, or forget and just use plain text + UDP socket? -- Best regards, Igor

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HOW send 4XX to the caller when the callee response to bye is 481 Call leg/transaction does not exist
by Humberto H 01 Apr '16

01 Apr '16

Hello; [image: Inline image 1] HOW can I send 4XX to the caller when the callee response to a BYE message is: *481 Call leg/transaction does not exist. BUT the caller is waiting.* Before this message. I drop the bye sent by the callee, my purpose here is "*Do not send the BYE TO THE CALLER"* JUST send a message in order to the caller do failover, If I send *BYE* to the caller it can't do failover Thanks in advance. 10.929596| | BYE | | |SIP Request BYE CSeq:104 | | |(5060) ------------------> (5060) | | |10.929981| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) <------------------ (5060) | | |10.929983| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930445| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930451| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist

2 1

How can I send 404 to the caller instead of BYE?
by Humberto H 01 Apr '16

01 Apr '16

Hello I am trying to send 404 to the caller instead of BYE. Using the configuration showned bellow. when the callee send BYE Kamailio sends -- *404* to the callee (to the source of the BYE signaling) --- and *BYE* to the caller But I need to send 404 response to the caller only Thanks in advance event_route[dialog:end] { if(is_method("BYE") && ((src_ip==XXXXXXX) )){ send_reply("404", "Not found"); xlog("L_INFO","ENVIO 404 ON BYE \n"); } exit; }

2 1

stateful messages in a dialog
by Al S 01 Apr '16

01 Apr '16

Hi, I would like to know how I can make subsequent messages in a dialog (ACK and BYE) to be stateful.Your help will be greatly appreciated. // reroute INVITE after timeout.failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} Thanks,AS

2 1

Record-Route header
by jignesh gandhi 01 Apr '16

01 Apr '16

Hello All, I have a need to save the Record-Route header from the INVITE and use it as a route header in requests like BYE. PROXY --> Kamailio(as PROXY) --> Asterisk I am using rr module to add a record_route when passing the INVITE to Asterisk. Thanks, Jignesh Gandhi

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Re: [SR-Users] Proxy configuration
by Jignesh Gandhi 31 Mar '16

31 Mar '16

Thanks Alex for you reply. I understand the dispatcher module can be used to load balance inbound requests. But my questions is more related with outbound requests thru Kamailio. Are you saying that in-dialog request can be routed automatically by Kamailio to the correct Asterisk instance on the back end ? And this is included in the stock config ? I was just looking for an example. Thanks, --Jignesh On 3/30/16, 6:00 AM, "sr-users on behalf of sr-users-request(a)lists.sip-router.org" <sr-users-bounces(a)lists.sip-router.org on behalf of sr-users-request(a)lists.sip-router.org> wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Destination Timeout (ACK message) (Al S) > 2. Proxy configuration (Jignesh Gandhi) > 3. Re: Proxy configuration (Alex Balashov) > 4. Local Kamailio share a hosted Kamailio/Asterisk sip server > (S?bastien Brice) > 5. SQlite DB Lock with Presence (Kristian Kielhofner) > 6. Kamailio local and hosted (S?bastien Brice) > 7. Re: Releasing Kamailio v4.4.0 (Daniel-Constantin Mierla) > 8. Re: SQlite DB Lock with Presence (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Tue, 29 Mar 2016 09:27:56 -0400 >From: Al S <alirsm(a)outlook.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Destination Timeout (ACK message) >Message-ID: <BAY177-W1513432E7A3AFDAAB1DE8DD4870(a)phx.gbl> >Content-Type: text/plain; charset="iso-8859-1" > >Hi, >The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). > >failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} >However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. >Thank, >AS >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/b91e2df…> > >------------------------------ > >Message: 2 >Date: Tue, 29 Mar 2016 14:24:01 +0000 >From: Jignesh Gandhi <Jignesh.Gandhi(a)MoviusCorp.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Proxy configuration >Message-ID: <011222F3-5820-41FB-A0D8-8D727EA121FA(a)moviuscorp.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Is it possible to use kamailio to be a stateful outbound proxy ? > >The use case is kamailio is an outbound proxy for multiple instances of Asterisk. >Thus all responses need to go back to the same instance of Asterisk that generated them. > >After the call is established. >How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? > >Any help is appreciated. > >Thanks, >--Jignesh >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/5081718…> > >------------------------------ > >Message: 3 >Date: Tue, 29 Mar 2016 10:29:19 -0400 >From: Alex Balashov <abalashov(a)evaristesys.com> >To: sr-users(a)lists.sip-router.org >Subject: Re: [SR-Users] Proxy configuration >Message-ID: <56FA913F.6080903(a)evaristesys.com> >Content-Type: text/plain; charset=windows-1252; format=flowed > >Jignesh, > >Everything you're asking for is part of Kamailio's core functionality: > >1. Most useful applications of Kamailio involve stateful relay, and the >stock included configurations are stateful. > >2. The loose-routing section of the configuration, and, more broadly, >RFC 3261 in general, take care of routing in-dialog requests (ACKs, >reinvites, BYEs, etc.) to the appropriate dialog party. > >3. Load-balancing/distribution among Asterisk instances can be done with >the 'dispatcher' module; there are many examples floating around, as >it's one of the most common applications for Kamailio. > >-- Alex > >-- >Alex Balashov | Principal | Evariste Systems LLC >1447 Peachtree Street NE, Suite 700 >Atlanta, GA 30309 >United States > >Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) >Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > > >------------------------------ > >Message: 4 >Date: Tue, 29 Mar 2016 16:32:36 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Local Kamailio share a hosted Kamailio/Asterisk > sip server >Message-ID: <2011796741.12246629.1459261956126.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? > >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 5 >Date: Tue, 29 Mar 2016 20:50:14 -0400 >From: Kristian Kielhofner <kris(a)kriskinc.com> >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] SQlite DB Lock with Presence >Message-ID: > <CAKDfjge4kLhdLjV5V7gsOOsLsdByeLgtwnqE-EEXHhxbVeV8Ag(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >Hello everyone, > > I'm trying to use BLF with the config here (except with SQlite): > >http://kb.asipto.com/kamailio:presence:k43-blf > > BLF seems to work perfectly except after the call is ended I get >"sqlite commit failed: database is locked" error messages as seen >here: > >http://pastebin.com/xVijj98H > >BLF is then broken at this point. > >kamailio -v >version: kamailio 4.3.5 (x86_64/linux) 950657 >flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, >USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, >SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, >FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, 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, DEFAULT PKG_SIZE 8MB >poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >id: 950657 >compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 > >Has anyone ever seen this before? > >Thanks! > >-- >Kristian Kielhofner > > > >------------------------------ > >Message: 6 >Date: Tue, 29 Mar 2016 14:38:40 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Kamailio local and hosted >Message-ID: <1194659979.12170808.1459255120868.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? >, >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 7 >Date: Wed, 30 Mar 2016 10:02:02 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, sr-dev <sr-dev(a)lists.sip-router.org>, > "business(a)lists.kamailio.org" <business(a)lists.kamailio.org> >Subject: Re: [SR-Users] Releasing Kamailio v4.4.0 >Message-ID: <56FB87FA.8080304(a)gmail.com> >Content-Type: text/plain; charset=utf-8 > >Hello, > >the final packaging for 4.4.0 is expected to start after 14:00UTC today. >Backports to branch 4.4 can be done up to 12:00UTC, after that let's >sync on sr-dev mailing list, because I will start testing how it builds >on different platforms and it is good to have the a consistent source code. > >Cheers, >Daniel > >On 29/03/16 09:04, Daniel-Constantin Mierla wrote: >> Hello, >> >> refreshing about the major release 4.4.0 planned for tomorrow -- if any >> dev has patches to backport to branch 4.4 should do it as soon as possible. >> >> Cheers, >> Daniel >> >> On 18/03/16 12:16, Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> with no major issues reported lately, I think we are on a good track for >>> releasing stable v4.4.0. Next week a lot of people prepare for Easter, >>> therefore I propose to do the release on Wednesday, March 30, 2016. If >>> new things pop up, the exact date can be adjusted a bit before or after >>> the current proposal. >>> >>> Any new discovered issue should be reported to bug trucker: >>> >>> * https://github.com/kamailio/kamailio/issues >>> >>> Help to complete the upgrade guide would be very appreciated: >>> >>> * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel >>> >>> A draft list of what is new in 4.4 is already available at: >>> >>> * https://www.kamailio.org/wiki/features/new-in-4.4.x >>> >>> Cheers, >>> Daniel >>> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Message: 8 >Date: Wed, 30 Mar 2016 11:37:39 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> >Subject: Re: [SR-Users] SQlite DB Lock with Presence >Message-ID: <56FB9E63.7040104(a)gmail.com> >Content-Type: text/plain; charset=windows-1252 > >Hello, > >not using sqlite here, but could be related to presence modules doing db >transactions and not releasing the locks. Do you get the issue after the >first call, or it runs for a while and at some point it breaks? > >Cheers, >Daniel > >On 30/03/16 02:50, Kristian Kielhofner wrote: >> Hello everyone, >> >> I'm trying to use BLF with the config here (except with SQlite): >> >> http://kb.asipto.com/kamailio:presence:k43-blf >> >> BLF seems to work perfectly except after the call is ended I get >> "sqlite commit failed: database is locked" error messages as seen >> here: >> >> http://pastebin.com/xVijj98H >> >> BLF is then broken at this point. >> >> kamailio -v >> version: kamailio 4.3.5 (x86_64/linux) 950657 >> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, >> USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, >> SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, >> FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, 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, DEFAULT PKG_SIZE 8MB >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> id: 950657 >> compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 >> >> Has anyone ever seen this before? >> >> Thanks! >> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >------------------------------ > >End of sr-users Digest, Vol 130, Issue 29 >*****************************************

2 1

timeout function
by Al S 31 Mar '16

31 Mar '16

Hi, I am routing call traffic thru proxy to destination A. If destination A is down, we need to reroute to destination B. What function we can use to process timeout so we can reroute the calls to different node? Thanks,ARM

2 1

TCP connection select problem
by 张顺通 31 Mar '16

31 Mar '16

my program structure is LinPhone------TCP------KA_EDGE------UDP------KA_REGISTER------UDP------FS KA_EDGE:all clients TCP connection is in this server KS_REGSITER:all register info is process here when User A want to use program,Linphone send SIP REGISTER to KA_EDGE by TCP(and TCP is always present), and KA_EDGE send to KA_REGISTER by UDP. when FS want to Call A, FS generate SIP: INVITE A@KA_REGISTER_IP KA_REGISTER recive INVITE, lookup the register list, send to KA_EDGE INVITE TCP_IP:TCP_PORT Route:KA_EDGE_IP:KA_EDGE_PORT and LinPhone recive that INVITE. *my question is:* LinPhone A and LinPhone B both register in KA_REGISTER. FS send INVITE: INVITE B: KA_REGISTER_IP and KA_REGISTER lookup correct，send to KA_EDGE INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT but KA_EDGE is not correct.sip is correct,* but KA_EDGE send the INVITE to A' TCP connnection* INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT Someone had this problem?Is this a bug?

4 19

db_cassandra status and plans
by Andreas Granig 31 Mar '16

31 Mar '16

Hi all, I was checking db_cassandra with Kamailio 4.3 recently to try making it work as a usrloc back-end, and my observation is that the module has quite some issues. In general, the module states that it's only tested with cassandra 1.0.6, which is really out-dated. I tried with the latest version of the 1.0 series (1.0.12) and even with lots of fiddling I didn't manage to make it work, could be my fault though. There has not been any work for 4 years other than cosmetical documentation fixes two years ago. Documentation is missing important information how to set up the version table, that could be fixed though. The schema definition in the documentation is missing new columns which have been introduced, that could be fixed as well. The module uses the legacy thrift interface, whereas nowadays (since cassandra 1.2+) a native cql driver is available as open-source. The module can not be used together with nathelper for nat pings, since the queries don't provide a key, which is required for cassandra. Our plan is to revamp the whole module and make it work with recent cassandra versions and remove the nat ping limitations. Main question is whether to go for a new module or replacing the old one, since it doesn't seem to work anymore anyways. Feedback whether you got it working with Kamailio 4.3+ and your views on future plans is much appreciated! Andreas

3 5

acc_cdrs empty
by Nelson Migliaro 31 Mar '16

31 Mar '16

Hello, I am not able to set up cdr. I see acc table is updated with 2 new entries after each call (INVITE and BYE) but acc_cdrs is empty. There are errors in the log. I think that I set up everything but is not working, - load the module, set module params and set the flag? Am I missing something, should I use a store procedure or something? ------------------------------------------------------------------------------------------------------------ loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route[WITHINDLG] { if (has_totag()) { if (loose_route()) { route(DLGURI); if (is_method("BYE")) { CREATE TABLE `acc_cdrs` ( `id` int(10) unsigned NOT NULL AUTO_INCREMENT, `start_time` varchar(32) NOT NULL DEFAULT '', `end_time` varchar(32) NOT NULL DEFAULT '', `duration` varchar(32) NOT NULL DEFAULT '', PRIMARY KEY (`id`), KEY `start_time_idx` (`start_time`) ) ENGINE=MyISAM AUTO_INCREMENT=2 DEFAULT CHARSET=latin1$$ Thank you.

1 1

UAC Register Asterisk
by Alberto Sagredo 31 Mar '16

31 Mar '16

Hi Im trying to modify example for Asterisk/Kamailio integration that miconda did, and trying to handle Forwarded register to Asterisk in an authenticated way instead removing secret in asterisk. I have made this changes route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $var(astip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $var(astport); $uac_req(furi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(turi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; xlog("Dentro de REGFWD"); t_on_failure("REG_AST"); uac_req_send(); } failure_route[REG_AST] { if (t_is_canceled()) { exit; } xlog("Dentro de REG AST"); if(t_check_status("401|407")) { xlog("Dentro de 401"); $avp(auser) = "test"; $avp(apass) = "test"; uac_auth(); t_relay(); exit; } } What i see on traces: U x.x.x..240:5060 -> x.x.x..56:5060 REGISTER sip:x.x.x.56:5060 SIP/2.0. Via: SIP/2.0/UDP x.x.x.240;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0. To: <sip:200@x.x.x.56>. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. CSeq: 10 REGISTER. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. Max-Forwards: 70. Content-Length: 0. User-Agent: ANY. Contact: <sip:emp_001_200@x.x.x.240:5060>. Expires: 3600. . # U x.x.x.56:5060 -> x.x.x..240:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP x.x.x..240 ;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0;received=x.x.x..240 ;rport=5060. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. To: <sip:200@x.x.x.56>;tag=as62b47138. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. CSeq: 10 REGISTER. Server: FPBX-2.11.0(11.17.1). Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59e39692". Content-Length: 0. . But Kamailio is not sending authentication . Any idea what could be wrong? Or any tip to do what i try. The idea is that a sip endpoint register with authentication on kamailio and it sends register with authentication to asterisk. Thanks in advance

4 7

Kamailio v4.4.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Kamailio v4.4.0 is out – it comes with 9 new modules and a consistent set of improvements touching more than 60 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.4.0! The development will start to focus on 5.0.0 - join the discussion on sr-users mailing list to design and decide what should Kamailio 5.0.0 look like - some notes are already collected at: * https://www.kamailio.org/wiki/devel/kamailio-5.0-design Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Releasing Kamailio v4.4.0
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Hello, with no major issues reported lately, I think we are on a good track for releasing stable v4.4.0. Next week a lot of people prepare for Easter, therefore I propose to do the release on Wednesday, March 30, 2016. If new things pop up, the exact date can be adjusted a bit before or after the current proposal. Any new discovered issue should be reported to bug trucker: * https://github.com/kamailio/kamailio/issues Help to complete the upgrade guide would be very appreciated: * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel A draft list of what is new in 4.4 is already available at: * https://www.kamailio.org/wiki/features/new-in-4.4.x Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Kamailio local and hosted
by Sébastien Brice 29 Mar '16

29 Mar '16

Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario? In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918 address? , thx you PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration

1 1

Proxy configuration
by Jignesh Gandhi 29 Mar '16

29 Mar '16

Hello, Is it possible to use kamailio to be a stateful outbound proxy ? The use case is kamailio is an outbound proxy for multiple instances of Asterisk. Thus all responses need to go back to the same instance of Asterisk that generated them. After the call is established. How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? Any help is appreciated. Thanks, --Jignesh

2 1

Destination Timeout (ACK message)
by Al S 29 Mar '16

29 Mar '16

Hi, The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. Thank, AS

1 0

SIP / WebRTC / GSoC proposals
by Daniel Pocock 29 Mar '16

29 Mar '16

Hi all, We had a few GSoC applications[1] from students who would like to do work relating to SIP and WebRTC this summer. The students have a range of skills. Some are interested in the server side (C, C++) code. Some are interested in Android and Java projects (e.g. the Lumicall and CSipSimple dialers) and some are web developers, interested in making practical use of WebRTC. Would anybody else from the community be interested in helping to mentor one of these projects? Prospective mentors can review the proposals privately before making any firm commitment, please contact me for details. Mentors can have some input in the direction of the students' work, just as long as all code is released publicly. Regards, Daniel 1. http://danielpocock.com/gsoc-2016-voice-video-chat

2 2

XML-RPC reply doesn't contain any data
by Arsen 29 Mar '16

29 Mar '16

Hi guys! We have an issue with XML-RPC module. >From this doc: http://kamailio.org/docs/modules/4.3.x/modules/xmlrpc.html "Success replies always contain at least one return value. In our case the simplest success replies contain single boolean with value 1" We are trying to reload domain module after adding/removing new domain, so we send the following request: XX - web server; YY - kamailio T XX.XX.XX.XX:55028 -> YY.YY.YY.YY:8080 [AP] POST /RPC2 HTTP/1.1. Host: YY.YY.YY.YY:8080. Accept: */*. Accept-Encoding: deflate, gzip. User-Agent: cURL. Accept-Charset: UTF-8. Content-Length: 120. Content-Type: application/x-www-form-urlencoded. . <?xml version="1.0" encoding="iso-8859-1"?> <methodCall> <methodName>domain.reload</methodName> <params/> </methodCall> T YY.YY.YY.YY:8080 -> XX.XX.XX.XX:55028 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP XX.XX.XX.XX:55028. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 108. . <?xml version="1.0"?> <methodResponse> <params> <param> <value></value> </param> </params> </methodResponse> As you can see in the response from the kamailio there is no any data, so our script is waiting until timeout... Please advice. Thanks, -- Regards, Arsen.

4 6

Timeout Handling (update)
by Al S 28 Mar '16

28 Mar '16

Hi, I am trying to re-route to a different $ru after timeout and right before Kamailio reply with a 408.I made the changes right before sl_reply_error(), however it does not work as I expected. Any help will be appreciated. Also, $rm represents the request message received by Kamailio. Is there a variable represents the messages leaving Kamailio?. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { // re-route to a new $ru before sending 408 sl_reply_error(); } exit;} Thanks,AS

2 2

using DMQ , Kamailio shutsdown
by Sunil More 28 Mar '16

28 Mar '16

Hello All, I am running two instances of kamailio and using dmq. Kamailio starts good. However one of the node shutsdown unexpectedly. Unsure what is going wrong. Please let me know what could be wrong. below is a the conf detail. loadmodule "dmq.so" modparam("dmq", "server_address", "sip:sunilmore.in:5060") modparam("dmq", "notification_address", "sip:sunilmore.in:5064") loadmodule "usrloc.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", DBURL2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "nat_bflag", FLB_NATB) modparam("usrloc", "db_insert_null", 1) loadmodule "dmq_usrloc.so" modparam("dmq_usrloc", "enable", 0) Below is the backtrace of the logs. ar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d7-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:1723]: main_loop(): Expect maximum 2177 open fds Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: htable [htable.c:237]: child_init(): rank is (12) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tm Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tm [callid.c:134]: child_init_callid(): DEBUG: callid: ' 0e4b331663a1f7ae-14041(a)sunilmore.in' Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tmx Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tmx [tmx_mod.c:272]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: sl Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tls Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_mongodb Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [ulcx-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq_usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 40, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: htable [htable.c:237]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: dmq_usrloc [dmq_usrloc.c:110]: child_init(): child_init PROC_INIT Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: registrar Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [uloc-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nathelper Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nat_traversal Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_fifo Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: avpops Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: rtimer Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dialog Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: domain Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: uac Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: presence Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: htable [htable.c:237]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 38, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa32028 and 0xa76028 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 8, 1, 0x7fab28a78438), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 9, 1, 0x7fab28a786a8), fd_no=1 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 10, 4, 0x7fab082bb21c), fd_no=2 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 11, 4, 0x7fab082bb2a8), fd_no=3 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 12, 4, 0x7fab082bb334), fd_no=4 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 13, 4, 0x7fab082bb3c0), fd_no=5 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 14, 4, 0x7fab082bb44c), fd_no=6 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 15, 4, 0x7fab082bb4d8), fd_no=7 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 16, 4, 0x7fab082bb564), fd_no=8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 17, 4, 0x7fab082bb5f0), fd_no=9 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 18, 4, 0x7fab082bb67c), fd_no=10 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 19, 4, 0x7fab082bb708), fd_no=11 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 20, 4, 0x7fab082bb794), fd_no=12 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 21, 4, 0x7fab082bb938), fd_no=13 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 22, 4, 0x7fab082bb9c4), fd_no=14 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 23, 4, 0x7fab082bba50), fd_no=15 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 24, 4, 0x7fab082bbadc), fd_no=16 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 27, 4, 0x7fab082bbb68), fd_no=17 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 25, 4, 0x7fab082bbbf4), fd_no=18 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 26, 4, 0x7fab082bbc80), fd_no=19 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 29, 4, 0x7fab082bbd0c), fd_no=20 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 31, 4, 0x7fab082bbd98), fd_no=21 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 28, 3, 0x7fab28d647f0), fd_no=22 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 30, 3, 0x7fab28d64810), fd_no=23 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 36, 3, 0x7fab28d64830), fd_no=24 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 39, 3, 0x7fab28d64850), fd_no=25 Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14027]: DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fab0853a9d8 from timer.c:648 Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 21 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [tcp_main.c:3448]: handle_ser_child(): dead child 14, pid 14031 (shutting down?) Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa31e60, 21, -1, 0x0) fd_no=26 called Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:739]: handle_sigs(): child process 14031 exited by a signal 11 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:742]: handle_sigs(): core was generated Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14038]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14040]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14037]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14035]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received There were no UAs that tried to register while this error occured. Thanking You, Sunil More Ph : 9503338275

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