- sr-users - kamailio.org

Setup timeout for in function jsonrpc_request
by Marco Capetta 06 Mar '18

06 Mar '18

Hi All, I'm currently in the process of upgrading a kamailio instance from 4.4.6 to 5.1.2. Since"janssonrpc-c" has been renamed in "jsonrpcc" I have to adapt my configuration. In version 4.4.6, the function "janssonrpc_request" allowed me to setup a specific timeout for the request. Is it possible to do the same in version 5.1.2 with the function "jsonrpc_request"? Because the documentation doesn't mention it. Thank you Best regards Marco

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Re: [SR-Users] Not documented parameter server_id_filter inside usrloc module
by Konstantin Polyakov 05 Mar '18

05 Mar '18

Victor, We are going to use this parameter, but we found that cleanup process should use such filter too. Otherwise different instances of proxy delete records of each other and it leads to the problems when db_mode 1 or 2 (cached) is used. Soon I am going to propose a pull request with filter in cleanup procedure. Could you tell me please, if it is necessary to add a new parameter for cleanup process or I can use this server_id_filter? Is there some case when we need to use this filter on startup but we don't it during cleanup? Thank you. Best regards, Konstantin

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Not documented parameter server_id_filter inside usrloc module
by Konstantin Polyakov 05 Mar '18

05 Mar '18

Hello everybody, Could somebody clarify why server_id_filter parameter is not listed in the usrloc module article? It is not ready for usage? Or it was forgotten to be added into description? Thank you Best regrads Konstantin

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Error using newest Siremis and kamailio 5.0.3
by Olli Attila 05 Mar '18

05 Mar '18

Hello, I installed Siremis today from master branch alongside with Kamailio 5.0.3 and entering "XMLRPC Commands" menu gives me this error: [2018-03-01 12:28:34 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/html/https/siremis/openbiz/bin/Resource.php:283 function: errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/html/https/siremis/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/html/https/siremis/openbiz/bin/sysheader.inc 117 function: userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/html/https/siremis/openbiz/bin/Resource.php", 283, Array(11) ) @ function: trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/html/https/siremis/openbiz/bin/Resource.php 283 function: getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/html/https/siremis/openbiz/bin/BizSystem.php 560 function: getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/html/https/siremis/siremis/modules/service/userEmailService.php 108 function: SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/html/https/siremis/siremis/modules/common/form/ErrorForm.php 34 function: Report ( ) @ /var/www/html/https/siremis/openbiz/bin/BizController.php 310 function: invoke ( ) @ /var/www/html/https/siremis/openbiz/bin/BizController.php 110 function: dispatchRequest ( ) @ /var/www/html/https/siremis/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/html/https/siremis/openbiz/bin/BizController.php" ) @ /var/www/html/https/siremis/siremis/bin/controller.php 6 (I'm running PHP 7.0.27-0+deb9u1 and MariaDb Server version: 5.1.73 Source distribution) What is going wrong here? I found this online but I think this is already fixed on the Siremis master branch? https://lists.kamailio.org/pipermail/sr-users/2016-December/095373.html Cheers, Olli

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Invoking RPC commands over HTTP
by Ben Hood 05 Mar '18

05 Mar '18

Hi, What is the idiomatic way to call an RPC from a remote HTTP client in the 5.1.x series? I'd like to invoke mtree.reload from a remote HTTP client. My goal is to avoid a dependency on kamcmd on the remote client system - it would be nice to be able to issue a HTTP RPC which could be implemented in any environment. Or should I be looking to implement a BINRPC client on the remote system? TIA, Ben

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Kamailio iOS Push Notifications
by Amine Mensori 04 Mar '18

04 Mar '18

Hello, We need help implementing push notifications within Kamailio. I have found a module relating to this: https://github.com/tvntsr/push <https://github.com/tvntsr/push> However, the developer did not provide any examples of the cfg file, only parts of certain functions. We need some help implementing the functions in our config file or a full example of another method to get push notifications to work when the application is closed on the device. Please note we are not very proficient in C programming so a clear example would help a lot. Many Thanks, Amine

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Non-standard SIP-over-TLS port
by Steve 03 Mar '18

03 Mar '18

Hello sr-users, How does one assign a non-standard SIP-over-TLS port rather than the usual port 5061 for client registration? Steve --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

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dialplan module reload fail
by Paulo Ferreira 02 Mar '18

02 Mar '18

Hi list, I'm using several modules, including dialplan. I can reload modules data from database to kamailio with kamctl reload, except the dialplan module. kamctl dialplan reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" }, "id": 21705 } looking at the error, it seems that this method was not implemented. This isn't making sense, so any idea in what can be wrong? thanks,and regards, Paulo -- ------------------------------------------- Paulo Ferreira VoIP@RCTS - Área de Infraestruturas Aplicacionaisz FCCN http://www.fccn.pt/ Av. do Brasil, n.º 101 1700-066 Lisboa - Portugal Telefone|Phone: +351 218440100; Fax: +351 218472167 Aviso de Confidencialidade/Disclaimer Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha rececionado indevidamente esta mensagem,solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218440100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218440100 and delete it immediately.

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question
by Syrine Riahi 02 Mar '18

02 Mar '18

bonjour je suis une étudiante en mastère professionnel des systèmes des réseaux et télécommunication j'ai choisie comme un sujet de mémoire mise en oeuvre de solution des VoIP et après des grands recherches j'ai choisie comme solution kamailio alors comme je suis débutante dans le domaine j'ai besoin des étapes exacte de l'instalation et la configuration de kamailio v 5.1.2 et j'utilise la distribution ubuntu v 16.04 merci énormément de me voir aider

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5.1.2 rtpengine regressions - broke working setup?
by Anthony Alba 02 Mar '18

02 Mar '18

Hi list, upgrade to 5.1.2 broke my setup due to some different behaviour in rtpengine.so. I use: rtpengine_manage("DTLS=off ICE=remove"); 1. I don't send any rtpengine_manage flags (RTP UDP SAVP). UAC sends RTP/SAVP in SDP, rtpengine changes to UDP/TLS/RTP/SAVP. Reverted to 5.1.1, and see that rtpengine sends RTP/SAVP 2. To SDP body, rtpengine is adding: a=setup:actpass a=fingerprint:sha-1 D6:D7:DA:AA:2D:DB:AD:94:29:46:FA:C4:D8:AE:0F:03:96:6F:8A:4F For some reason this broke FreeSWITCH which does not like this. Anthony

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Re: [SR-Users] rtpengine transcoding benchmarks?
by Jeff Brower 01 Mar '18

01 Mar '18

Andreas- > We've seen around 400 concurrent calls for G711<->Opus on a rather > slow 12-core E5-2420(a)1.90GHz for some first preliminary benchmarks. Was that using narrowband (8 kHz) for Opus ? Otherwise sampling rate conversion is needed. Also I assume Opus DTX was disabled. Also at what Opus bitrate ? Thanks. -Jeff > On 2018-02-26 21:27, Brooks Bridges wrote: >> I realize the transcoding support is rather new, however has anyone run any real world tests on it yet?Â Trying to get a sense of what the performance will be. >> >> Â >> >> Iâm aware that itâs forced into userspace for transcoding so itâs not **nearly** as efficient as the iptables module, however even if I have to throw a rack full of whiteboxes at it to get anywhere close to decent numbers thatâs still going to be comparable to a lot of the commercial options out there right now. >> >> Â >> >> *Brooks Bridges | *Engineering Manager >> >> *O^1 Communications* >> >> 4359 Town Center Boulevard, Suite 217 >> >> El Dorado Hills, California 95762 >> >> *office:*916.235.2097 |*main:*888.444.1111, Option 2 >> >> *email:*bbridges@o1.com <mailto:bbridges@o1.com> |Â *web:*www.o1.com <http://www.o1.com/> >> >> Â

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Kamailio v5.1.2 Released
by Daniel-Constantin Mierla 01 Mar '18

01 Mar '18

Hello, Kamailio SIP Server v5.1.2 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.2. For more details about version 5.1.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/03/kamailio-v5-1-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Planing next Kamailio releases - v4.4.7 v5.0.6 and v5.1.2
by Daniel-Constantin Mierla 01 Mar '18

01 Mar '18

Hello, I am considering to do releases from last three stable branches during next week. v4.4.7 is going to be released out of latest code in branch 4.4 to mark the end of official maintenance for series v4.4.x. The version 4.4.0 was released in March 2016, so it had more or less two years of maintenance. Commits can still go in 4.4 branch if developers wants to push there, end of official maintenance means that there is no plan to do packaging and release another version from the branch in the future, so you will have to build it from sources or generate packages yourself. v5.0.6 and v5.1.2 are the usual minor releases we do from the last two stable branches. If there are any issues you are aware of and not reported on bug tracker, create the items as soon as possible to get a better chance to review and eventually fix. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Siremis and kamailio 5.x.x
by Olli Attila 01 Mar '18

01 Mar '18

Hello, Question about Kamailio web GUI (Siremis): I believe the Siremis updates are still a ongoing work or has there been any updates when the new Siremis update (compatible with Kamailio 5.x.x) will be released? https://www.kamailio.org/w/2017/07/siremis-ongoing-updates/ Cheers, Olli

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Kamailio ims with TCP
by Amar Tinawi 01 Mar '18

01 Mar '18

Hello list For some reason i couldn't register on kamailio ims when i use only TCP protocols between nodes, the error was solved when i enabled UDP between I-CSCF and S-CSCF is it mandatory for those two nodes to talk over UDP only ? or i'm missing something ? and in general for sip protocol which is better ? UDP or TCP ? Regards,

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Trunks with registration
by Володимир Іванець 28 Feb '18

28 Feb '18

Hello all! I was testing Registration capability of UAC module and have two questions that hopefully someone have answers to: - Is it possible to tell Kamailio which transport protocol to use other than by configuring auth_proxy column of uacreg table with domain name that has SRV record? - What are the priorities of different transport protocols in SRV record when it is used? Thanks a lot!

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Changing contact header
by Cibin Paul 28 Feb '18

28 Feb '18

Hello, In my setup, Kamailio acts as a registration server and route the calls to different asterisk boxes. The clients using IPV6 private ip and IPV4 public ip address are rejected by the asterisk server. On printing the contact header, I am getting the private ip (ipv6). Is there a way to change the contact header with public ip in all scenarios. I tried updating the contact header as below and unfortunately not working. Is there a better way to handle the scenario. As of now we don/t have a plan to implement ipv6 network. $var(p1) = "<sip:"; $var(p2) = "@"; $var(p3) = ":"; $var(p4) = ";ob>"; $var(x) = "" + $var(p1) + $fU + $var(p2) + $si + $var(p3) + $sp + $var(p4); remove_hf("Contact"); append_hf("Contact: $var(x)\r\n”); Thanks & Regards Cibin

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Question on contribution of documentation improvements
by Floimair Florian 28 Feb '18

28 Feb '18

Hi guys! While reading through documentation I realized there are a lot of typos (typing errors) and some misspellings so I decided to try to correct them. I dit this in all of Kamailios module and lib documentation (the xml in the doc folders) and they turned out to be quite a lot. Right now this sits on my fork of kamailo on github, but I would like to get this back into the main repository. What's puzzling me though, is how to most effectively get this in via a pull request due to the following: For now I have made a single commit for each module I touched. Therefore quite a lot of commits have accumulated. Actually 152 commits in modules and 4 commits in lib. Since most of the commits in modules are rather small (ranging from 1 line changed to 4) with only a few having more, I am not sure if that's the way to go to have a single commit for each module. Therefore I am unsure if a squashed version would be more appropriate. Therefore I created a second branch on my forked repository. Unsquashed: https://github.com/ffloimair/kamailio/tree/wip-fix-typos Squashed: https://github.com/ffloimair/kamailio/tree/wip-fix-typos In total a diff --stat to master results in 205 files changed, 521 insertions(+), 521 deletions(-) So please let me know what would be the wanted (and most efficient) way to proceed. With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg

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cnxcc module - make auth for redis connection
by Donat Zenichev 28 Feb '18

28 Feb '18

Hi community. This message is specially dedicated to developers of the kamailio project. I need to find solution how to make authentication for connections of cnxcc with redis. For that moment, cnxcc module is able to connect to redis only without password (Carlos Ruiz Díaz will add this possibility later?). So that, this makes database vulnerable. Well, furthermore I don't want to use local redis database, cuz I think machine that contains kamailio daemon, must perform just this kind of actions - SIP proxying (kamailio server is not to be a database server or anything else). For that moment we use separate redis db for cases like that, but this is not best practice I think. I already tried to find solution (for e.g. connections via another module like db_redis: https://github.com/kamailio/kamailio/pull/1432) but I got advice - achieving my goal by sr-users mailing list. Please pay attention for my request. I think this will be very helpfull for others. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 (ext 1320) https://w-net.us/ <http://wnet.ua>

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kamailio 5.1 compilation error
by Aqs Younas 28 Feb '18

28 Feb '18

Greetings list, I am trying to install Kamailio 5.1 from source code on Debian 8.10. But It is failing with below error. Don't know what could be wrong. Any suggestion is much appreciated. Thanks. make --no-print-directory -C ./squirrel/squirrel SQMARCH64="-m64" CC_EXTRA_FLAGS="-fPIC" \ OUT="../../libsquirrel.a" sq64 gcc -O2 -m64 -D_SQ64 -fno-exceptions -fno-rtti -Wall -fno-strict-aliasing -c sqapi.cpp sqbaselib.cpp sqfuncstate.cpp sqdebug.cpp sqlexer.cpp sqobject.cpp sqcompiler.cpp sqstate.cpp sqtable.cpp sqmem.cpp sqvm.cpp sqclass.cpp -I../include -I. -Iinclude -fPIC gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory Makefile:51: recipe for target 'sq64' failed make[3]: *** [sq64] Error 1 Makefile:44: recipe for target 'libsquirrel.a' failed make[2]: *** [libsquirrel.a] Error 2 Makefile:511: recipe for target 'modules' failed make[1]: *** [modules] Error 1 make[1]: Leaving directory '/usr/src/kamailio/src' Makefile:27: recipe for target 'all' failed make: *** [all] Error 2 Br, Aqs.

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please downgrade warning message
by Juha Heinanen 28 Feb '18

28 Feb '18

I noticed this when I start a correctly configured kamailio: Feb 27 21:55:45 rautu /usr/bin/sip-proxy[29529]: WARNING: registrar [registrar.c:369]: mod_init(): empty reg_xavp_cfg or sock_flag not set -> resetting Neither of those two are mandatory. I'm not using registered() function. Please downgrade the message to info or debug level. -- Juha

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rtpengine transcoding benchmarks?
by Brooks Bridges 27 Feb '18

27 Feb '18

I realize the transcoding support is rather new, however has anyone run any real world tests on it yet? Trying to get a sense of what the performance will be. I'm aware that it's forced into userspace for transcoding so it's not *nearly* as efficient as the iptables module, however even if I have to throw a rack full of whiteboxes at it to get anywhere close to decent numbers that's still going to be comparable to a lot of the commercial options out there right now. Brooks Bridges | Engineering Manager O1 Communications 4359 Town Center Boulevard, Suite 217 El Dorado Hills, California 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 v5.0.6 Released
by Daniel-Constantin Mierla 27 Feb '18

27 Feb '18

Hello, Kamailio SIP Server v5.0.6 stable release is out. This is a maintenance release of the previous stable branch, 5.0, that includes fixes since the release of v5.0.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.6 or v5.1.x. For more details about version 5.0.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/02/kamailio-v5-0-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.1, at this moment with its latest release v5.1.1 (and v5.1.2 planned to be out in the next few days). See more details about it at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Kamailio and dialogs..
by Laura 27 Feb '18

27 Feb '18

Dear list.. I'm using kamailio on a live enviroment with dialogs support.. what I made is a script that permit me to see the concurrent call (live) with all parameters using the info saved on dialogs tables. The problem is that if the call is not going on the first termination of LCR but for examples in the second, the dialog is not updated with the paramenter of the second supplier.. Is this problem only on my system or maybe i forgot to make some configuration steps ? Regards Laura

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Kamailio v4.4.7 Released
by Daniel-Constantin Mierla 26 Feb '18

26 Feb '18

Hello, Kamailio SIP Server v4.4.7 stable release is out. This is the last planned maintenance release of the old stable branch, 4.4, that includes fixes since the release of v4.4.6. There is no change to database schema or configuration language structure that you have to do on upgrading previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.7, or even better to v5.0.x or 5.1.x. For more details about version 4.4.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/02/kamailio-v4-4-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: theb branch 4.4 is going out of maintenance life time, the latest stable branch is 5.1, at this moment with its latest release v5.1.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Kamailio json_rpc isn't reloading htable data from database into memory
by José Seabra 26 Feb '18

26 Feb '18

Hello there, I'm using xHTTP and json_rpc module to receive http instructions to reload htable's from database to memory. The way i'm calling the json_rpc in kamailio script is the following: $var(json_rpc)='{"jsonrpc": "2.0", "method": "htable.reload", "id":"htable"}'; jsonrpc_exec($var(json_rpc)); The execution of this command doesn't give any errors, but the changes made on htable database weren't transmitted to the memory. I have attached the kamailio log where you can see that there isn't any erros on json_rpc execution during reload instructions, the strange thing here is that the htable module only says: 2018-02-23 14:33:44.968 DEBUG: htable [ht_db.c:96]: ht_db_open_con(): database connection opened successfully doesn't report if the reload was success or not. Those logs attached were printed with kamailio debug set to level 4. The kamailio version is 4.4.2. Thank you -- Best Regards José Seabra

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Thanks, Juha. You are correct. That feature request is directly relevant to this thread. I will upvote the feature request, and add a link back to this thread. The proposed workaround of using async workers is the one I opted to use.

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

>A common design which avoids this is to use TCP at the client edge and >UDP inside the network core. This is one of the reasons why TCP is not >optimal for use inside the core. That makes sense, but is unfortunately not an option for me due to strict security requirements. I need to use TLS on the whole path. Another option I explored was to have the edge proxies not always use the same TCP connection for sending to the registrar. If I could find a way to load balance across a number of TCP connections, that would probably work for me. Perhaps there is a way the DISPATHCER module can be configured to accomplish this. Maybe the dispatcher configuration can list multiple copies of the same destination, but each having a different send socket address, and then can load balance across those. Does that make any sense?

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Load-balance on 2 proxies
by David Villasmil 23 Feb '18

23 Feb '18

Hello guys, I hava a setup where there are 2 kamailios in active-active connected to the same db (no registration yet). They both forward requests load balancing to several freeswitches. My question is, is the load balance algorithm independent on the 2 servers, or do they act as 1 load balancing in term of sending the next request to the next FS in the list independent of which kamailio sent the last request? I.e. Kam1 -> fs1 Kam2 -> fs2 Kam1 -> fs3 Kam1 -> fs1 Kam2 -> fs2 And so on? Thanks for your help David

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Thanks. Regarding UDP, that all makes sense. For my short-term needs, to minimize risk and infrastructure changes, it looks like the ASYNC module will work. Here's the basic outline of what I'm doing: -------- async_workers=8 route { route(CHECK_IS_EXPENSIVE_OPERATION); if ($var(isExpensiveOperation)) { async_task_route("COMMON"); } else { route(COMMON); } } route[COMMON] { # Do all my normal routing logic. } -------- I detect at the very top of the main route whether the operation will be expensive. If so, I process the message with an async worker. If not, I directly process the message on the TCP worker. The rest of the routing logic remains unchanged. The only difference is that I execute the expensive operations using the async workers. It's not the prettiest thing in the world, but it seems like an OK short-term fix to relieve pressure. I will revisit the bigger architectural ideas in a later phase. Thanks again for all your help and good ideas. Happy Friday. -Cody

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Thanks very much for the quick replies, Alex and Brandon. The main reason I'm hitting a bottleneck is because my architecture is not optimal. I have a number of edge proxies which communicate with all my clients. The clients are usually distributed pretty evenly across all the proxies. On those proxies, client TCP connections are distributed pretty evenly across the TCP workers, so that's all fine. The problem occurs when the edge proxies communicate with the central registrar. When that happens, the SIP messages from a very large number of clients are all multiplexed onto a single TCP socket connection between the proxy and the registrar. That narrowing results in the registrar worker processes not being utilized efficiently. For example, say I have 3 edge proxies, and my registrar has 8 cores and 8 worker processes. I want to spread the message processing across all 8 workers, but I'm only able to utilize 3, because all the messages from a given edge proxy are being processed by a single TCP worker on the registrar. The long-term solution is to change my architecture. I should not be doing all the expensive work on a single central registrar. I'm planning to move to a mesh architecture, where all servers are the same, and the registration processing is divided amongst all the servers. That design makes sense, but it's more complex and will require more information sharing amongst the servers. I'm looking for a short-term improvement that will give me some breathing room, which led me to looking at the ASYNC module. >...one alternate suggestion that could help spread load on actual Kamailio TCP workers is by firing up additional workers on alternate ports. That makes sense. I could have my central registrar listen on several different ports, and could perhaps use the dispatcher module on my edge proxies to try and evenly divide the traffic across those ports. I will look into that. Thanks again. -Cody

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How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Hello. I'm curious as to how SIP messages received over TLS are divided amongst the available TCP workers. Based on some searching, I was hoping and expecting that the messages would be load balanced amongst the TCP workers at the level of individual SIP messages. However, with my configuration, it seems that the work is load balanced at the TCP connection level. I will now elaborate on my basic architecture. I have several edge proxies to which the clients connect. Those proxies do not handle registration, or do much of anything, they just forward traffic to a central registrar server. On the central registrar server, it seems that a particular TCP worker is assigned to handle all the traffic arriving from a particular edge proxy. As long as the TCP connection between the servers remains open, it seems that the particular worker is locked into handling all traffic on that connection. Does that sound like expected behavior? That behavior is not ideal for me, because I would like to evenly divide the SIP message load across all the TCP workers on my central registrar. Perhaps I'm doing something bad in my config file that is preventing the load balancing from working at the SIP message level. If that behavior is expected, is there anything I can do to achieve more even load balancing? Perhaps the ASYNC module might work for my needs. It seems like I could use async_task_route() to divide certain messages evenly amongst the async workers. Does that sound reasonable? In my use case, certain messages require a lot of CPU time to perform crypto calculations, and I want those calculations to be distributed evenly across all CPU cores. Thanks. -Cody

3 2

rtpengine codecs transcoding: performance
by Abdoul Osséni 23 Feb '18

23 Feb '18

Hello, Is there a study comparing the performance (quality and cpu/ram consumption) of codecs transcoding between rtpengine and asterisk/freeswith? Is rtpengine more efficient? Regards Abdoul

2 1

RFC: db_redisusrloc module - review and how to deal with
by Daniel-Constantin Mierla 23 Feb '18

23 Feb '18

Hello, (cross-posting being something I want to get developers and users express their opinions). We were recently in a (lucky) situation to have two new modules submissions targeting more or less the same purpose: allowing to use Kamailio with a Redis backend via database API. One was submitted by Andreas Granig (Sipwise) and already merged with the name db_redis, because it was designed from the beginning as a generic DB connector, so the module can be used with auth_db, acc, usrloc, ... * https://www.kamailio.org/docs/modules/devel/modules/db_redis.html The second one was submitted by Surendra Tiwari (Plivo), initially having a naming conflict with db_redis, but renamed to db_redisusrloc, see the pull request at: * https://github.com/kamailio/kamailio/pull/1446 Now, this email is about deciding the way to go forward with the second module. It was designed to be used only for usrloc in the first phase, with many specific usrloc attributes hard coded inside db_redisusrloc. Surendra said (in a private chat) that the long term plan is to make it work for other modules. Anyhow, at this moment is very tied to usrloc, therefore the name of the module. Given that the backend (Redis) is not an SQL engine, mapping over Kamailio's DB API needs some schema definition (see the readme of db_redis) in order to make it generic and work for all our modules that use a DB backend. On the other hand, to squeeze the best of the backend, specially in no-SQL cases, having a dedicated DB connector module optimized for a specific module might help to get more performances/high-availability/scalability from the backend itself. In this case, for example the expires value for a contact record can be set inside redis, so kamailio module doesn't have to run a timer routine to clean up (it doesn't mean db_redisusrloc does it right now, I didn't have time for a proper review, just giving an example). Surendra said they use it in production for couple of months now and it is several times faster than using usrloc with db_postgres (iirc, not db_mysql) for db_mode=3 (database only mode). But of course, the reverse of the medal with a dedicated db connector for a module: it adds overhead to code maintenance (besides generic updates due to external library changes, I expect changing something relevant in usrloc, like adding new columns, would require updates in this module as well). So, there are few things I want feedback on: 1) how do you fill about splitting from a generic-only DB connectors to have also some dedicated ones? This is more from confusion point of view, as a general rule so far, we do not deny contributions if there are other options for same kind of feature (e.g., many lcr or nat traversal options). As long as the contributor is willing to maintain the code, we were fine. 2) I guess usrloc, presence and dialog modules would be the main suspects that would benefit for such dedicated connectors, in other cases might not worth adding dedicated connectors. Missing any other module one would like to squeeze more performances with a dedicated connector? 3) should we set a different naming policy for such modules, for example: use *dbs_* prefix instead *db_*, to suggest better it is a DataBase Specific connector? 4) Andreas said he plans to do some performance testing of usrloc module with the two modules and see the differences. Anyone else that wants to do it? It can be a good metric to see if it worth going one way or another? 5) Helping to review the pull request, specially if you use Redis, is appreciated. Personally I am very short in available time these days, next week I plan do to new Kamailio stable releases, so the schedule is not getting lighter in my side. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

6 6

Checking dispatcher gw state
by Daniel Greenwald 23 Feb '18

23 Feb '18

Is there a way thru kamctl or kamcmd to check which dispatcher gateways are currently inactive? "kamctl dispatcher show" doesn't seem to have that info and I can't find anything in the docs. Thanks

2 2

How to Re-transmit the current Request when a negative reply arrives?
by Arun NV 21 Feb '18

21 Feb '18

I am using Kamailio 4.4. And I am forwarding an INVITE to my client. but sometimes this client will send a negative response.And I am able to drop that negative response from Kamailio. But I need to re-transmit the same request to the client again. How do I do that? Thank you, Arunbalan <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

2 1

Return codes within a route
by Jeff Bilyk 20 Feb '18

20 Feb '18

Hello, Running into odd behaviour in Kamailio 5.0.2 and I'm hoping to clarify. I have a simple route that uses the textops functions subst and remove_hf. Other than that, there is nothing else done within the route. In some cases, the remove_hf function returns false as there are no matching headers. Documentation in https://www.kamailio.org/wiki/cookbooks/5.0.x/core#return seems to show that if there are no explicit return statements, the route should return 1. However, this route appears to be returning -1. My question is: if a function within a route returns a negative value, and there are no explicit return statements, will the entire route return -1 (as it appears is happening), or should the route return 1 as the core cookbook indicates? Thanks in advance, Jeff

2 2

Kamailio 4.3 decrypting SIP-TLS
by Remsik,Robert 20 Feb '18

20 Feb '18

Hello! I'm working on setting up our kamailio instance to use TLS for inter-phone communication and so far I've been able to configure the GUI and get it to work which is great. However, I'm trying to grab a packet capture and decode the TLS session to get the SIP ladder (so I can troubleshoot issues) and so far I've been unable to. I've been following the article at: http://packetpushers.net/using-wireshark-to-decode-ssltls-packets/ and using the kamailio self-signed cert as I think that's the cert being used with no success. I'm getting the Cipher Suite: TLS_RSA_WITH_AES_256_CBC_SHA (0x0035) which doesn't use perfect forward secrecy so it should be able to be decoded. Is there a way to decode the kamailio TLS encoded communications so I can troubleshoot issues that may come up? Thank you in advance, Robert Robert Remsik Telecom Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

3 3

Inserting a message into a transaction?
by Daniel Tryba 20 Feb '18

20 Feb '18

For a retarded endpoint (an old ISDN PBX behind a Smartnode) I need to insert a 180 reply before sending a 183 (with SDP), otherwise the PBX fails to playback the inband audio to the caller behind the PBX. I can't find hints on how to do this. uac_req_send() can be used to start a new transaction, but I only need to generate a 180 based on the 183 in the on_reply route. Current Q&D hack is to send a 180 on the initial INVITE, works but not ideal.

2 3

Re: [SR-Users] lost mysql connections
by Jan Gaida 20 Feb '18

20 Feb '18

Hello juha, Sorry for getting back on this old topic. Can you confirm that with Kamailio v4.3 your problem was solved? Regards, Jan

1 0

Create ISUP Body
by Duarte Rocha 20 Feb '18

20 Feb '18

Hey Everyone I'm setting up a load testing scenario for my Kamailio. In order to do that i've created another Kamailio to answer "200 OK" to an Invite in order to simulate a call being answered. One of the scenarios I need to test is reading values from the ISUP IAM or ACM body. Here's the code on the Kamailio that replies with 200 : $var(acm) = "06 02 04 01 2c 01 fb 0c 08 83 10 19 7c 82 12 15 00 36 01 19 40 01 00 39 06 2c c0 36 c0 40 c0 00"; set_reply_body("$var(acm)","application/isup;base=itu-t92+;version=etsi356"); if( is_method("INVITE") && !has_totag() ) { · xnotice("First Invite"); send_reply("200","OK"); } However, when i use this and i do a pcap i get a "Message Type: Overload", which doesn't support optional parameters. What am i doing wrong ? Are there other methods to setup a simple ISUP part without hex ? Thanks in advance

1 0

Kamailio mod ALIAS_DB: multiple aliases
by Stefan Rüetschli 20 Feb '18

20 Feb '18

Hello Kamailio-Users I'm using kamailio 5.1 with Alias_DB module. My subscribers register with a username like 0000001, 0000002 etc... Every user has a list of multiple aliases which they can use on their SIP-Trunk. I want to avoid that the user can use any alias for outgoing calls, only the aliases which are boud to his username in the aliases-db. So, if the user uses an alias which does not belong to him, i want to overwrite it to the first alias for his username in the aliases-db. At the moment, the function alias_db_find returns me the last alias found in the db. Is there any way to do it? Or is there a way to extend the dbaliases with a new column marking the "primary" alias for an account?|||| This would be the perfect solution. Regards Stefan

2 2

Info: Kamailio World 2018 - Preview With 15 Sessions
by Daniel-Constantin Mierla 20 Feb '18

20 Feb '18

Hello, less than 3 months till the start of the 6th edition of Kamailio World Conference, time if flying fast! About one week ago we published the details for a group of accepted speakers, today we made a selection of sessions at the Kamailio World 2018. We had more proposals than we could accommodate, we are trying hard to fit in as many as possible, taking also in consideration the feedback from the participants at the past editions. For now you can head to the Schedule page and see the details of 15 sessions, from both workshops and conference days: https://www.kamailioworld.com/k06/schedule/ A very divers range of topics, from using Kamailio for emergency services (112/911), scaling with Redis backend, deploying in a containerized environment with Docker and Kubernetes, how to migrate the SIP routing logic to rich KEMI languages such as Lua, Python or Javascript, unit testing for Kamailio and test driven deployments, to blockchains in telephony, using Kamailio and FreeSwitch together, or latest updates from Asterisk PBX. The IMS/VoLTE workshop is going to show what you can do with latest Kamailio in mobile networks. And, of course, we have the very popular two sessions that never missed a Kamailio World edition: Dangerous Demos with James Body and VUC Visions with Randy Resnick. The details for other speakers and sessions will be published in the near future, stay tuned! Do not miss Kamailio World Conference 2018, it is going to be another great edition! You can register now at: https://www.kamailioworld.com/k06/registration/ Looking forward to meeting many of you at the next Kamailio World Conference, during May 14-16, 2018, in Berlin, Germany! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

IMS: maybe a bug in S-CSCF: should we create an issue?
by Valentin Christoph 20 Feb '18

20 Feb '18

Hi all, We are using kamailio 5.1.0 as S-CSCF as follows: We have two machines and two kamailio instances, one on each machine, both working on the same database (together they build one highly available S-CSCF). After registering a user via the first kamailio instance, then the first instance is stopped (simulating an outage), and the HSS sends an RTR to the second instance. However, the second instance does not find the registered user in its cache and it does not access the database --> the deregistration fails. Configuration: kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "enable_debug_file", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "matching_mode", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", DB_URL) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_mode", 2) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_interval", 10) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_procs", 4) Does anyone immediately "see" a mis-configuration, or should we create an issue? Details in the mail below. Thanks, Christoph From: Friedrich Peter Sent: Friday, February 16, 2018 6:38 PM To: Valentin Christoph <Christoph.Valentin(a)kapsch.net> Cc: Onic Roman <Roman.Onic(a)kapsch.net>; Habashizadeh Mahmoud <Mahmoud.Habashizadeh(a)kapsch.net> Subject: AW: Kein RTR Message nachdem SCSCF-1 outage Hello, following scenario at REGISTER of catt subscriber: start with empty db and empty usrloc cache node1(.23): MariaDB [scscf]> select * from impu; Empty set (0.00 sec) [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims ~]# [root@vict-ims2 etc]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims2 etc]# After catt REGISTER: catt_subscriber visible in DB and usrloc cache of .23 but missing in usrloc cache of .24 MariaDB [scscf]> select * from impu; +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | id | impu | barring | reg_state | ccf1 | ccf2 | ecf1 | ecf2 | ims_subscription_data | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | 196 | sip:catt178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 197 | sip:+4350255872178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 198 | tel:+4350255872178 | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ 3 rows in set (0.00 sec) MariaDB [scscf]> [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 3542 client: n/a } } } [root@vict-ims ~]# [root@vict-ims2 etc]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims2 etc]# After kamctl stop + kamctl start of .24 catt_subscriber is visible in usrloc cache of (.24): preload_data loads data from db at module init [root@vict-ims2 ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 2949 client: n/a } } } [root@vict-ims2 ~]# Now deregister catt_subscriber from HSS1: catt_subscriber disappears from db and usrloc cache of (.23) but persists on (.24) as registered in usrloc cache HSS >hss.deregister_subscriber catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> 0 [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims ~]# MariaDB [scscf]> select * from impu; Empty set (0.00 sec) MariaDB [scscf]> [root@vict-ims2 ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 2571 client: n/a } } } [root@vict-ims2 ~]# Scenario of RTR from HSS after REGISTER of catt_subscriber at (.23) and outage of (.23): subscr. visible in db but not visible in usrloc cache of .24 HSS sends diameter RTR message to (.24) and deregister callback in (.24) searches in usrloc cache and does not find catt_subscriber and fails. MariaDB [scscf]> select * from impu; +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | id | impu | barring | reg_state | ccf1 | ccf2 | ecf1 | ecf2 | ims_subscription_data | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | 196 | sip:catt178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 197 | sip:+4350255872178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 198 | tel:+4350255872178 | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ 3 rows in set (0.00 sec) Cfg file of (.24): kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "enable_debug_file", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "matching_mode", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", "cluster://cluster1") kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", DB_URL) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_mode", 2) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_interval", 10) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_procs", 4) lg P The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

2 4

issue with running kamailio
by justin barnes 20 Feb '18

20 Feb '18

I am installing from githum kamailio 5.0 on my ubuntu-16.04-x86_64 VPS. when I start the kamailio server I get this error : /# /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceJob for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. failed! Ilooking up the status, I get this kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled; vendor preset: enabled) Active: failed (Result: exit-code) since Mon 2018-02-19 15:23:41 EST; 4h 10min ago Process: 18969 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GR Feb 19 15:23:41 khynepassion systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Feb 19 15:23:41 khynepassion systemd[18969]: kamailio.service: Failed at step EXEC spawning /usr/sbin/kamailio: No such file or directory Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Control process exited, code=exited status=203 Feb 19 15:23:41 khynepassion systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Unit entered failed state. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Failed with result 'exit-code'.

1 0

LCR options as of 2018
by Francisco Valentin Vinagrero 19 Feb '18

19 Feb '18

Hello, I am configuring a Kamailio node that will interface with 3 carriers and will implement Least Cost Routing for around 900 destinations. I have already implemented a quick prototype with the LCR module and I have been reading the CarrierRoute doc and the CGrates site. I don't really need advanced functionality, just: - Re-routing to a backup carrier when the main one is not available. - Ping with OPTIONS to all carriers so I can detect if any of them is not available (à la Dispatcher) - Translate prices from a central DB to routing priorities. - Get some accounting post-call info about the routing decision (acc module) So far, while testing the LCR module I have had some issues with the rule IDs being integers that auto-increment, so the LCR rules need to be referenced from the rule_target table using this numeric ID. I can see some issues when scaling this to 900 lcr_rules and 2700 lcr_rule_targets with 3 priorities. In addition to that, I think the OPTIONS mechanism is just used by the lcr module when a gateway is marked as defunct, but the gateways are not actively monitored in the normal status. Is this correct? I have been considering CarrierRoute, which has a slightly simpler table structure, but I can't find any other advantage for my use case. Regarding CGrates, it looks great but it looks like it exceeds my current needs. For those who have implemented LCR in your kamailios, am I missing something regarding performance, scalability, etc? Are there any newer modules that can implement this, or maybe just a dispatcher-on-steroids setup will do the trick? Cheers, Francisco.

3 6

User state
by Magnus Löfqvist 19 Feb '18

19 Feb '18

Hi, We are running for a customer a kamailio server for user registration and routing of calls. Also using Asterisk for queue management and IVR. Now we are in the progress of writing a web portal where the customers users should be able to see if there collueges are busy in phone or availible. In asterisk I would use hints and listing in events in AMI. But what way are the best in Kamailio? Best Regards Magnus Sent from Nine<http://www.9folders.com/>

2 1

PR1447: first attempt at app_python3
by Anthony Alba 18 Feb '18

18 Feb '18

Hi Daniel and kamailio contributors, I have submitted a PR for app_python3 here: https://github.com/kamailio/kamailio/pull/1447. It is very rough, but would appreciate your feedback. Notes: 1. Almost a verbatim port of app_python; just changing obvious semantics. String->Unicode is practically the only large change. No change to symbol names and export functions so the two modules cannot coexist. (Don't loadmodule "app_python.so" and loadmodule "app_python3.so) 2. Module initialization semantics in Py3 have changed: add the top level modules to the inittab before calling Py_Initialize(). I have done this for KSR and Router. I.e. PyImport_AppendInittab() is called for Router and KSR *before* Py_Initialize() 3. Makefile looks for a python3 binary (PYTHON3 in the Makefile) to get include paths and link libraries. Here is an example compile line for IUS Python 3.6 on CentOS 7 (no native Python3 from RHEL) ## default: needs to find python3 and python3-config ## needs to find both python3.6 and python3.6-config make -C src/ PYTHON3=python3.6 SKIP_MODULES="${SKIP_MODULES}" quiet=verbose module_group_py="app_python3" group_include=py every-module 4. The semantics for GIL and thread management are too different for one-to-one replacements. I have therefore decided to comment out all threading calls. Fortunately, kamailio does not used threads at the moment. This is more complicated than it seems: even with Py2+thread management: there would be deadlocks if kamailio->python2 script->Cython extension with threads(e.g. grpcio). For an MVP it would be best to get it in without thread management. 5. It should backport to 5.1. 6. Tested locally, with a routing scripting making many python_exec() calls Thank you. Anthony Alba

1 1

PUBLISH and ETag
by Cyrille Demaret 16 Feb '18

16 Feb '18

Hi, I'm using Kamailio with presence enabled and Asterisk PJSIP and outbound-publish. My problem is happening when I place 2 consecutive calls from Asterisk : When I make a first call Asterisk sent the following: PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a610 82 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Event: dialog Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early.. Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e6972 3a61082;received=192.168.100.37 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-ff39 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19863.16.0 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 When the call is done, Asterisk sent another PUBLISH telling that the call if terminated : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c7 52 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19863.16.0 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 230 <?xml version="1.0" encoding="UTF-8"?> terminated.. And Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97 ab8c752;received=192.168.100.37 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-48b4 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19873.18.1 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 Here, the SIP ETag is a.1518775074.19873.18.1. The problem is if I make a new call before the expiration of the previous SUBSCRIBE, Asterisk reuse this SIP ETag according to the RFC : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj9d13bb82-31d9-48db-9672-bd4b6b4f22 f0 From: sip:201@mydomain.com;tag=33e6b028-0444-4b3a-8bc2-4a987a291528 To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10699 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19873.18.1 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early. Kamailio refuse it with this error : "Trying to update an already terminated state. Skipping update." because the call is considered as terminated. The RFC is stating : When updating previously published event state, PUBLISH requests MUST contain a single SIP-If-Match header field identifying the specific event state that the request is refreshing, modifying or removing. This header field MUST contain a single entity-tag that was returned by the ESC in the SIP-ETag header field of the response to a previous publication. Why Kamailio is acting like that? Best regards, Cyrille

2 2

Problem with "transport=tcp" and rtpproxy
by Pau Giralt 16 Feb '18

16 Feb '18

Hi! I'm working with kamailio and rtpproxy. When I configure the clients to send sip messages through TCP adding the ";transport=tcp" in the sip message the rtpproxy is not working anymore. The code is never passing the point pasted below, inside route[NATMANAGE] in kamailio.cfg if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; Therefore, rtpproxy_manage("co"); is never reached. Could it be a parsing problem? ( check_route_param("nat=yes") ) The config file is exactly the same as the one provided in the kamailio repository (the default one) Thanks!

2 7

Raspberry Pi deb packages -> built
by Mititelu Stefan 15 Feb '18

15 Feb '18

Hi people! After a bit of research and lib installations, I've managed to build the full kamailio deb packages for a raspberry pi 3 (version B) running raspbian stretch (lite) that can be found at [1] (with a really few module exceptions that might be used in really specific scenarios -> I think) Just for fun, I'm planning to do some CPS stats tomorrow (if I got the time), with a basic config and also share here, but, if someone else tried it before and is able to share some stats/benchmarks, would be greatly appreciated (and interesting to compare, given my "decent" rpi overclock), because I'm also curious about VoIP stuff on raspberry pi! :D [1] https://github.com/smititelu/rpi/tree/master/kamailio/5.1.1 Thank you, Stefan

2 3

KEMI
by Aleksandar Sosic 15 Feb '18

15 Feb '18

Hi everyone, we're valuating on rewriting our kamailio routes in python. We're doing a lot of string manipulation and GraphQL API queries so the kamailio scripting language is a little bit limited for this use case. Can someone tell us if the KEMI framework is stable enough for production? Kind regards, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

5 11

Re: [SR-Users] RTPEngine Transcoding
by Social Boh 15 Feb '18

15 Feb '18

Hello, I'm trying to use this configuration: Softphone A audio codec g722 and Opus Softphone B audio codec PCMA and PCMU Kamailio with rtpengine_manage with codec-transcode-PCMU codec-transcode-PCMA in the request Softphone A call Softphone B The RTPEngine DEBUG show many Decoder Error: https://pastebin.com/k1Zhw8HS Thank you Regards --- I'm SoCIaL, MayBe El 13/02/2018 a las 20:05, Richard Fuchs escribió: > On 2018-02-13 07:22 PM, Social Boh wrote: >> Thank you. >> >> which is the best way to implement ir >> >> codec-strip-all and the add codecs? >> >> Is there a document about how work the transcoding? > It's really up to you and depends on what you want to achieve. Doing a > strip-all and then adding codecs you want to offer is definitely one > way. Another is to just add codecs that you know or think the peer may > want to see. > > Cheers >

2 2

Invalid payload location
by Soltanici Ilie 15 Feb '18

15 Feb '18

Hi All, Can someone explain why i'm receiving such errors in syslog: [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:228]: set_sdp_payload_fmtp(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location This is how the SIP Packet is looking: 2018/02/15 10:51:40.916040 46.166.139.7:30189 -> X.X.X.X:5060 INVITE sip:01031203697179@X.X.X.X:5060 SIP/2.0 Via: SIP/2.0/UDP 46.166.139.7:30189;branch=z9hG4bK-5342365299746297160962993236828333618949;rport=30189 From: <sip:88893855391@46.166.139.7:30189>;tag=0vkn019bur To: <sip:01031203697179@X.X.X.X:5060> Contact: <sip:88893855391@46.166.139.7:30189> Call-ID: 0889633580294714035892119863939070195463 CSeq: 101 INVITE User-Agent: Hello Annonymous Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 356 v=0 o=SIP-Session 950029666 1307322426 IN IP4 46.166.139.7 s=SIP c=IN IP4 46.166.139.7 t=0 0 m=audio 5075 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=fmtp:9 bitrate=64000 a=rtpmap:104 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Thank You.

1 0

python_exec return 0 causes script hangs; return -1 is treated like 0
by Anthony Alba 15 Feb '18

15 Feb '18

5.1.1: if python_exec method returns 0, then the script hangs If the python_exec method return -1, then expressions treat it as if it returned zero. Somewhere there seems to be an offset by 1 issue. if (!python_exec("method")) { # if you return -1, the script does not hang but treated like zero } ################## Example 1: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return 0 Calling: python_exec("hello") will cause script hang ################### Example 2: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return -1 Calling python_exec("hello") if (!python_exec("hello")) { ## method returns -1 ## why did we get here? xlog("L_INFO","method return zero!\n"); } shows INFO: <script>: method return zero!

2 1

[IGNORE] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Apologies for the noise; it is working now. Not sure how I got the extra " 0" appended to the method arguments for python_exec(). AAlba

1 0

[BUG?] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Using 5.1, the python_exc args string is getting an extra [_0' appended When I call python_exec("method", "string"); In python: def method(self, msg, args): KSR.info("DBG: {} {} {}".format(msg, args)) This will show 'string 0'; instead of string. AAlba

1 0

RTPEngine Transcoding
by Social Boh 14 Feb '18

14 Feb '18

Hi List, I just read on the RTPEngine page the Proxy RTP now support transcoding: https://github.com/sipwise/rtpengine#transcoding is there a plan to support it? Regards -- --- I'm SoCIaL, MayBe

5 10

RTPEngine QoS statistics and leg labelling
by Alex Balashov 14 Feb '18

14 Feb '18

Hello, I am excited about the new media statistics from RTPEngine, exposed conveniently into PVs, e.g. jitter, packet loss, MOS, time offset of lowest and highest MOS scores, etc. Great work! I am, however, having a bit of trouble understanding how to properly apply this new functionality, as it is not clearly documented (IMHO): 1. What is the meaning of "global" statistics which do not take into account an "A" and a "B" side? That is to say, if I have a packet loss or a jitter or MOS score statistic using just the generic toplevel values, e.g. https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… What does that mean? A functional call ordinarily consists of an inbound and outbound RTP stream direction relative to the endpoints on both sides of RTPEngine. Inbound A --> server --> Outbound A Outbound B <-- server <--- Inbound B Where do the "global" stats situate themselves in relation to all this? 2. I am given to understand that the statistics for the "A" and "B" sides of a bidirectional RTP stream can be separated based on an arbitrary label, using these PVs: https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… How do I use it properly? My intuition was to try something like this: rtpengine_offer("replace-origin replace-session-connection ICE=remove label=outbound"); And in the answer/manage handler on the SDP answer: rtpengine_answer("replace-origin replace-session-connection ICE=remove label=inbound"); Later, when processing a BYE and obtaining subsequent media statistics, I tried this: if(check_route_param("proxy_media=yes")) { $avp(rtpengine_a_label) = 'outbound'; $avp(rtpengine_b_label) = 'inbound'; rtpengine_delete(); route(LOG_RTPENGINE_STATS); } But this only gets me stats for the "A" leg in the JSON object I emit: { "server_addr": "udp:127.0.0.1:5050", "a_label": "outbound", "b_label": "inbound", "a_leg": { "min": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "max": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "avg": { "mos": "4.4", "packetloss": "0", "jitter": "0", "roundtrip": 0, "samples": 2 } }, "b_leg": { "min": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "max": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "avg": { "mos": "<null>", "packetloss": "<null>", "jitter": "<null>" } } } I also tried rtpengine_delete("label=inbound") (and outbound) individually. Some clarity on how this all fits together would be appreciated. :-) Many thanks to Sipwise for great work on it! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 14

How to add a new module to existing Kamailio Server?
by Arun NV 14 Feb '18

14 Feb '18

I am using a *Kamailio 4.4* server. And I would like to install *http_async_client* module to this server. I have looked in the module directory(*/usr/local/lib64/kamailio/modules*) to check whether this *http_async _client* module already exists there, but it was not there. And I searched for a solution and only found <http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-modular-install> the method to include new module at the compile time. I could find the modules source code on GitHub <https://github.com/sipwise/kamailio/tree/master/modules/http_async_client>. But I don't know the steps to install a new module to the existing Kamailio Thank you, Arunbalan

2 1

Use RTPProxy in 302 scenario
by Fabian Borot 13 Feb '18

13 Feb '18

I have this scenario Customer --> Kamailio --> Provider1 The route where l route the call to Provider1 is called route["INCOMING"] and I "arm" the replies and failures to : t_on_reply("INCOMING") and t_on_failure("INCOMING"). Provider1 always replies with a 302 Redirect and in the failure_route["INCOMING'] I have this code: if (t_check_status("302")) { get_redirects("*"); t_on_branch("REDIRECT_BRANCH"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL_FORKING"); t_relay(); exit; } In the t_on_failure("SERIAL_FORKING") block I do t_relay() as long as there are more contacts to be processed. Lets say I need to try Provider_2 and Provider_3 failure_route["SERIAL_FORKING"] { xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Reply [$rs]\n"); if (!t_next_contacts()) { exit; } xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Trying next Contact.\n"); t_on_failure("SERIAL_FORKING"); t_relay(); } I want to use RTPProxy in this scenario and I am using the function rtpproxy_manage() to let kamailio handle it automatically. RTPProxy ↑ | | Customer --> Kamailio --> Provider1 | | ↓ Provider N After the 302 response is processed I would like to have the signaling flow "Customer -> Kamailio --> ProviderN" and the RTP flow "Customer --> RTPProxy --> ProviderN" I have tried inserting the function in several places but can't make it work and what I mean is that sometimes the rtpproxy creates the 1st session and then deletes it as soon as I call it the 2nd time with a message similar to this one (log of rtpproxy) : "INFO:416f20b74a58fb75168175e61b2edece@192.168.1.61:5060:rtpp_cmd_delete_ematch: forcefully deleting session 1 on ports 63502/0". Sometimes the kamailio log shows this message: "ERROR: rtpproxy [rtpproxy.c:2536]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" I think I must call rtpproxy_manage() before I send the INVITE to each of the Providers in the Contact list so they receive a SDP offer with an IP:Port combination provided by the RTPProxy (instead of the ones coming from the Customer). That would be in my route["INCOMING'] Then once Provider_N replies with a 183 with SDP or a 200 OK I would call it again to get a new session to be passed on the reply to my customer. But is the part I am struggling with. Should I add a new reply route inside the failure_route["INCOMING'] only if I am dealing with 302 and do it there ? I am out of ideas guys and any idea is welcome

2 2

Can rtpengine change network interface of estabished call?
by Yufei Tao 13 Feb '18

13 Feb '18

Hi, Is it possible to define two pairs of network interfaces for rtpengine, e.g. int/ext and up/down? If so during a call established between interfaces 'int' and 'ext', is it possible to send a re-INVITE from the calling party on interface 'ext', to update the RTP destination to an IP on a different interface 'up', for example? Or does the updated RTP destination have to be on the same interface as when the call was established? I have some success updating the RTP destination if it is on the same interface (different IP:port) however haven't managed to divert it to a different interface. I wonder if it is possible at all or me missing something? Cheers, Yufei

1 0

(no subject)
by Tihomir Culjaga 13 Feb '18

13 Feb '18

Hi, Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=) i have a setup like this: (Kamailio version 5.1) UA > Kamailio > Freeswitch Kamailio forwards registration to freeswitch without saving location. I have configured sip over websockets and im able to register to Freeswitcha via kamailio. Also, i use add_path_received() so kamailio adds a path header to register messages recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809: ------------------------------------------------------------------------ REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 Max-Forwards: 69 To: <sip:1001@192.168.50.60> From: <sip:1001@192.168.50.60>;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90", uri="sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef", qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001 Contact: <sip:1001@192.0.2.110 ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6> ------------------------------------------------------------------------ send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 From: <sip:1001@192.168.50.60>;tag=vq30modpgd To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600 Date: Fri, 09 Feb 2018 21:38:07 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Content-Length: 0 ------------------------------------------------------------------------ Once, registered, im trying to originate a call to the registered endpoint from freeswitch this results in an invite like this: send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549: ------------------------------------------------------------------------ INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0 Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Max-Forwards: 70 From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK To: <sip:1001@192.0.2.110;transport=ws> Call-ID: 8212b700-8885-1236-08b3-000c293ae94e CSeq: 118770326 INVITE Contact: <sip:mod_sofia@192.168.50.65:15060> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 891 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:1002@192.168.50.65 >;party=calling;screen=yes;privacy=off Freeswitch appends a route header to the invite specifying kamailio as route and a received with original uri. well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio my RELAY route is like this route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_ALERT", "Its a Branch, go manage that !!! <$ru>\n"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_ALERT", "its onreply_route, go manage !!! <$ru>\n"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_ALERT", "Its failure_route, go manage !!! <$ru>\n"); t_on_failure("MANAGE_FAILURE"); } } $fs = "udp:192.168.50.60:5060"; xlog("L_ALERT", "Trying to rely !!! <$ru>\n"); if (!t_relay()) { xlog("L_ALERT", "failed to rely !!! <$ru>\n"); sl_reply_error(); } exit; } here is a part of the log: Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:192.168.50.60:5060, to ws:192.0.2.110:5060) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core> [core/msg_translator.c:424]: clen_builder(): content-length: 916 (916) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws> is that because i set a $fu before t_relay() ? any idea how to approach the issue ?

2 4

onend_route() ?
by Wilkins, Steve 13 Feb '18

13 Feb '18

I meant to reply to Kamailio mailing list. From: Wilkins, Steve Sent: Friday, February 9, 2018 12:40 PM To: 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: [SR-Users] t_relay()? Hi Daniel, I am not sure what I am doing wrong but I cannot get the forwarded message to print. I am using onend_route() and trying to print the results using xdbg. I have searched for examples and have had no luck. Thank you From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Friday, February 9, 2018 7:48 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Wilkins, Steve <swwilkins(a)mitre.org<mailto:swwilkins@mitre.org>> Subject: Re: [SR-Users] t_relay()? Hello, before t_relay() you can't see what is going to the network, because some headers are added when knowing target address and local socket (e.g, Via, Record-Route). You can use onsend_route for seeing the request just before writing to the network. https://www.kamailio.org/wiki/cookbooks/devel/core#onsend_route $snd(buf) will give outgoing buffer -- for more see $sndfrom(...) and $sndto(...) vars. Cheers, Daniel On 09.02.18 11:24, Wilkins, Steve wrote: Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com<http://www.asipto.com> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com>

2 1

Request Timeouts
by Wilkins, Steve 13 Feb '18

13 Feb '18

Hello, This is actually a follow up to my last question about some providers calls being dropped. I noticed that when calls are dropped, I see a Kamailio log message "Request Timeout" about 5-10 seconds before the call is dropped. I am wondering what could cause this while a call is in progress. Does Kamailio need to respond to Asterisk with a '200 OK' for an INFO request sent to Kamailio from Asterisk? I am not sure what else could cause this call dropping behavior. Note: It does not appear to be caused from a missing 'ACK'. Thank you, -Steve

3 6

What is the mechanism used for sending push notification from Kamailio, For both IOS and Android?
by Arun NV 13 Feb '18

13 Feb '18

I am creating custom phone client, which uses *Kamailio *as a proxy and it's required to send *push notification* from *Kamailio *to the clients when a call towards the client reached on *Kamailio*. I found a module for APNS here* https://github.com/tvntsr/push <https://github.com/tvntsr/push>*. But I am developing both Android and IOS clients. So What mechanism I can use for the Android side? Is there any module available for *GCM*(Google Cloud Messaging) in Kamailio? Thank you, Arunbalan

2 1

is_audio_on_hold() doesn't support rfc3264
by Julia Boudniatsky 13 Feb '18

13 Feb '18

FYI, After some tests found, that function "*is_audio_on_hold()*" (textops) *supports only rfc 254*3, where "hold" is indicated by setting the "c" destination addresses for the media streams to zero (0.0.0.0). The newest *rfc3264*, where "hold" is indicated by a=sendonly/inactive, *is not supported*. Kamailio version 4.4.6. Best regards, Julia

4 8

Load testing: Taking CPU & MEM stats
by jeevan Chaitanya 12 Feb '18

12 Feb '18

Hi All, Found python library to take system stats during the load. >>> from getstats import GetStats >>> stat_obj = GetStats.getstats(dur=300,rate=10) >>> stat_obj.run_stats() Running.......... ***** Completed capturing the STATS ***** ######################################## # stats_2018-02-12_16-47-47_.csv # ######################################## Doc URL: http://getstats.readthedocs.io/en/latest/ If you require use and let me know. Thanks, K.Jeevan.

1 0

Very inconsistent calls
by Wilkins, Steve 12 Feb '18

12 Feb '18

Hello, I have found that calls are very inconsistent. I use Kamailio 5, Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls into a WebRTC client (tryit-jssip), sometime the calls stay up until I close them (10-15 minutes), others times those calls drop in 30 seconds. This is extremely confusing...does anyone else experience this type of behavior? -Steve

4 7

t_suspend usage in branch_route
by Aymeric Moizard 09 Feb '18

09 Feb '18

Hi, I'm trying to use t_suspend in my kamailio.cfg from a branch_route. I'm using kamailio 5.1.1 on debian jessie from http://deb.kamailio.org/kamailio51 >From the documentation: http://www.kamailio.org/docs/modules/5.1.x/modules/tmx.html#tmx.f.t_suspend i can see that "This function can be used from ANY_ROUTE ." However, the INVITE is still forwarded for this branch. So obviously the "outgoing transaction" is not suspended. May be "t_suspend" is only suspending "incoming transactions"? I tried to call "drop()". This prevent the INVITE from being sent. But in such case, i have an answer is sent immediatly to the caller (SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL)) I want the script to wait as expected after my "t_suspend". And in the block indicated by t_continue, i want to append a new branch. Any way to do this? Otherwise, do you advise me to control the list of branch before t_relay? I guess I can access the list. Not sure how.. so if you have any tips... Tks Aymeric -- Antisip - http://www.antisip.com

3 9

I: - w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it
by Angelo Colucci 09 Feb '18

09 Feb '18

Hello Everyone, I've been resolved the problem. It was the setflag(FLT_ACC) at the beginning of the block if (is_method("INVITE")) { setflag(FLT_ACC); <----- using instead following function now the dialog is correctly created. setflag(FLT_DIALOG); Best Regards Angelo Da: Angelo Colucci Inviato: giovedì 8 febbraio 2018 18:33 A: 'sr-users(a)lists.kamailio.org' <sr-users(a)lists.kamailio.org> Oggetto: [SR-Users] - w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it Hello Everyone, I'm new in Kamailio environment. We trying to setup a IMS Credit Control Client as per RFC 4006. We've trying to implement the suggesting configuration in ims_charging module doc. The Kamailio version is 5.1.1 Stable. The Server is configured with two local users (A and B) registered on local mysql database. The scenario is the following: 1. In the request_route object we trying to intercept the first INVITE and trigger a diameter credit control request using Ro_CCR function. 2. When A calls B without triggering mechanism the call is fine. 3. When A calls B with credit control triggering, the calls failed and the diamater request is not generated. In the debug we find the following error: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=540 a=25 n=xlog ERROR: <script>: Individuato primo INVITE: generata nuova transazione exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=63 n=assign exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=29 n=Ro_CCR DEBUG: ims_charging [ims_charging_mod.c:494]: w_ro_ccr(): Ro CCR initiated: direction:orig, reservation_units:30, route_name:CHARGING_CCR_REPLY, incoming_trunk_id:1 outgoing_trunk_id:1 DEBUG: ims_dialog [dlg_hash.c:926]: internal_get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_hash.c:956]: get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_handlers.c:1950]: dlg_get_msg_dialog(): dlg with callid 'yH5kdBGSrM6CowdkkRHTUw..' not found ERROR: ims_charging [ims_charging_mod.c:511]: w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it The error indicates that w_ro_ccr function doesn't find any dialog. Following you can find the relative configuration of kamailio.cfg. The request_route block is equal to default configuration except the following command: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_DEBUG loadmodule "cdp" loadmodule "cdp_avp" loadmodule "ims_charging" loadmodule "ims_dialog" modparam("ims_charging", "origin_host", "kcc-t-lu-03.mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "origin_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "destination_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging","interim_update_credits",30) modparam("ims_charging","timer_buffer",5) # -- ims_dialog params -- modparam("ims_dialog", "dlg_flag", FLT_DIALOG) request_route { ... if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting t_newtran(); $var(cc_ret) = Ro_CCR("CHARGING_CCR_REPLY", "orig", 30, "1", "1"); if ($var(cc_ret) < 0) { xlog("Chiamata a Ro_CCR() fallita\n"); sl_send_reply("402","Payment required"); exit; } } ... } route[CHARGING_CCR_REPLY]{ xlog("cca_return code is $avp(s:cca_return_code): $avp(s:cca_result_code)\n"); switch ($avp(s:cca_return_code)){ case 1: #success break; case -1: #failure switch ($avp(s:cca_result_code)){ case 4010: send_reply("402", "Payment required - Unsufficient funds"); break; case 5006: send_reply("486", "Line limit exceeded"); break; default: send_reply("402","Payment required ($avp(s:cca_result_code))"); break; } default: xlog("L_ERR", "Unknown return code from CCR: [$avp(s:cca_return_code)] \n"); send_reply("500", "Charging Error"); break; } } What's goes wrong? Thanks in advance Angelo Colucci

1 0

Regular exepression issue in permissions module
by Paulo Ferreira 09 Feb '18

09 Feb '18

Hi list, I'm testing permissions module with kamailio 5.1.1 and I was struggling with regular expressions in trusted requests part of module. I noticed that 9[0-9]{4} doesn't work as expected so I had to change it for 9[0-9][0-9][0-9][0-9] to work. Is it supposed to be like that? Is there other way in this module to simplify the expression since it is not accepting the POSIX way? Thanks, Paulo -- ------------------------------------------- Paulo Ferreira VoIP@RCTS - Área de Infraestruturas Aplicacionaisz FCCN http://www.fccn.pt/ Av. do Brasil, n.º 101 1700-066 Lisboa - Portugal Telefone|Phone: +351 218440100; Fax: +351 218472167 Aviso de Confidencialidade/Disclaimer Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha rececionado indevidamente esta mensagem,solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218440100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218440100 and delete it immediately.

2 4

t_relay()?
by Wilkins, Steve 09 Feb '18

09 Feb '18

Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve

3 2

- w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it
by Angelo Colucci 08 Feb '18

08 Feb '18

Hello Everyone, I’m new in Kamailio environment. We trying to setup a IMS Credit Control Client as per RFC 4006. We’ve trying to implement the suggesting configuration in ims_charging module doc. The Kamailio version is 5.1.1 Stable. The Server is configured with two local users (A and B) registered on local mysql database. The scenario is the following: 1. In the request_route object we trying to intercept the first INVITE and trigger a diameter credit control request using Ro_CCR function. 2. When A calls B without triggering mechanism the call is fine. 3. When A calls B with credit control triggering, the calls failed and the diamater request is not generated. In the debug we find the following error: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=540 a=25 n=xlog ERROR: <script>: Individuato primo INVITE: generata nuova transazione exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=63 n=assign exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=29 n=Ro_CCR DEBUG: ims_charging [ims_charging_mod.c:494]: w_ro_ccr(): Ro CCR initiated: direction:orig, reservation_units:30, route_name:CHARGING_CCR_REPLY, incoming_trunk_id:1 outgoing_trunk_id:1 DEBUG: ims_dialog [dlg_hash.c:926]: internal_get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_hash.c:956]: get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_handlers.c:1950]: dlg_get_msg_dialog(): dlg with callid 'yH5kdBGSrM6CowdkkRHTUw..' not found ERROR: ims_charging [ims_charging_mod.c:511]: w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it The error indicates that w_ro_ccr function doesn’t find any dialog. Following you can find the relative configuration of kamailio.cfg. The request_route block is equal to default configuration except the following command: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_DEBUG loadmodule "cdp" loadmodule "cdp_avp" loadmodule "ims_charging" loadmodule "ims_dialog" modparam("ims_charging", "origin_host", "kcc-t-lu-03.mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "origin_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "destination_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging","interim_update_credits",30) modparam("ims_charging","timer_buffer",5) # -- ims_dialog params -- modparam("ims_dialog", "dlg_flag", FLT_DIALOG) request_route { … if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting t_newtran(); $var(cc_ret) = Ro_CCR("CHARGING_CCR_REPLY", "orig", 30, "1", "1"); if ($var(cc_ret) < 0) { xlog("Chiamata a Ro_CCR() fallita\n"); sl_send_reply("402","Payment required"); exit; } } … } route[CHARGING_CCR_REPLY]{ xlog("cca_return code is $avp(s:cca_return_code): $avp(s:cca_result_code)\n"); switch ($avp(s:cca_return_code)){ case 1: #success break; case -1: #failure switch ($avp(s:cca_result_code)){ case 4010: send_reply("402", "Payment required - Unsufficient funds"); break; case 5006: send_reply("486", "Line limit exceeded"); break; default: send_reply("402","Payment required ($avp(s:cca_result_code))"); break; } default: xlog("L_ERR", "Unknown return code from CCR: [$avp(s:cca_return_code)] \n"); send_reply("500", "Charging Error"); break; } } What's goes wrong? Thanks in advance Angelo Colucci

1 0

Feature/help android fcm and ios related push notification module
by Dipen Patel 08 Feb '18

08 Feb '18

hello.. Now most sip soft client in required push wakeup phone. So for that required android, ios push notifications module. I have seen some one have developed 'Tsilo' module based on GCM but it is old. So if possible so please upgrade for Android FCM so it iwll supported for Android and ios both. Thanks

3 2

Using t_relay_to_tcp()
by Wilkins, Steve 08 Feb '18

08 Feb '18

Hi All, Some providers only allow tcp so I switched from t_relay to t_relay_to_tcp(); however, nothing now gets relayed. Has anyone else had luck switching? Thank you, -Steve

3 2

Random 'BYE" sent?
by Wilkins, Steve 08 Feb '18

08 Feb '18

Hello All, I am running Kamailio 5 and using Asterisk 14 as PBX. I have a strange issue. When I receive calls from some providers to a WebRTC client, calls hang up after about 25 seconds. This does not happen with all providers and does not happen with WebRTC client to WebRTC client calls. It appears the "BYE" is coming from Kamailio. Any ideas? Thank you, -Steve

3 2

Re: [SR-Users] How to avoid automatic string to int conversion?
by Yuriy Gorlichenko 08 Feb '18

08 Feb '18

Hi. Yep! I found issue was in if. Fixed it. thx for response! On Feb 8, 2018 18:44, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: It looks like you are using it in some conditional expression (like if(...)) or some assignment with integer expression. Can you paste here the part of the config where you use it? Cheers, Daniel On 08.02.18 04:08, Yuriy Gorlichenko wrote: Hi im using kamailio 5.1 with native cfg file and using json_get_value( method) form jsonso module When i get json string that contains strings and i need to have these strings kamailio tries to convert string i find into int and affcource it can't then i see warning and can't get vaules of the string i have rval_expr_eval_int(): rval expression conversion to int failed (24,6-24,14) How to say kamailio to not convers strng to int? _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Info: Kamailio World 2018: First Group Of Speakers Announced
by Daniel-Constantin Mierla 08 Feb '18

08 Feb '18

Hello, the details for the first group of speakers at Kamailio World 2018 have been published: - https://www.kamailioworld.com/k06/speakers/ So far they come from three continents: Europe, North America and Asia, many presenting for the first time at our event. The two sessions present at all editions so far will be there also in 2018, at our 6th edition, respectively Dangerous Demos with James Body and VUC Visions with Randy Resnick. Besides covering various use cases for Kamailio, Asterisk or FreeSwitch, the sessions go into WebRTC, VoLTE/IMS, IoT, blockchains for telecommunications or scalability using NoSQL data storage systems. Definitely another edition with very interesting content – soon we will publish more details about the sessions as well. You can register now to benefit of the early registration price: - https://www.kamailioworld.com/k06/registration/ Looking forward to meeting many of you at Kamailio World Conference, May 14-16, 2018, in Berlin, Germany! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

How to avoid automatic string to int conversion?
by Yuriy Gorlichenko 08 Feb '18

08 Feb '18

Hi im using kamailio 5.1 with native cfg file and using json_get_value( method) form jsonso module When i get json string that contains strings and i need to have these strings kamailio tries to convert string i find into int and affcource it can't then i see warning and can't get vaules of the string i have rval_expr_eval_int(): rval expression conversion to int failed (24,6-24,14) How to say kamailio to not convers strng to int?

2 1

dmq_usrloc - which server?
by Alex Balashov 07 Feb '18

07 Feb '18

Hi, The Socket parameter of location entries is not replicated under dmq_usrloc. To make matters worse, the technical characteristics of the registrar module are such that I cannot straightforwardly modify the incoming Contact to add a rider indicating the server on which the register was received. The AoR domain is the same for all registrations received on any server in the DMQ mesh. To reach the registrants, I need to forward the registration to the correct server that corresponds to the true provenance of the registration. What is a reasonable way of finding out what that is? I don't see any method of injecting such values into the replicated contacts, as dmq_usrloc does not provide hooks for this. Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 19

any interest in segmentation fault at startup ?
by Jean Cérien 07 Feb '18

07 Feb '18

Hi I guess I'm a bit too creative when it comes to kamailio syntax.... just edited the cfg file and got the following error upon startup: kamctl start INFO: Starting Kamailio : /usr/sbin/kamctl: line 1915: 5294 Segmentation fault $OSERBIN -P $PID_FILE -f $ETCDIR/kamailio.cfg $STARTOPTIONS > /dev/null 2> /dev/null ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I assume it is my syntax in cfg that creates this, I just need to correct it, but if there is interest, I can provide the faulty cfg file J.

4 4

Re: [SR-Users] Blind Transfer WS Error
by Matteo Campana 07 Feb '18

07 Feb '18

Dear Kamailians, does anyone have any idea on the issue below? Alessio told me that is always reproducible, and he can also provide captures and more logs. Thanks in advance! Matteo *From: *sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Alessio Miranda <smirolandia(a)gmail.com> *Reply-To: *"Kamailio (SER) - Users Mailing List" < sr-users(a)lists.kamailio.org> *Date: *Tuesday, 16 January 2018 at 14:18 *To: *"sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> *Subject: *[SR-Users] Blind Transfer WS Error Hi, I have a problem with the blind transfer via WebSocket, debugging from the browser we see that in some cases when the REFER is done and the sip message "202 Accepted" arrives, the message "180 Ringing" arrives first and then the "100 Trying" and the transfer goes in error with "500 JsSIP Internal Error" I have an architecture with Kamailio in frontend and asterisk in backend on the same network, the sip dialogue between FE and BE is in SIP / UDP. The WebRTC client connects to WebSocket at Kamailio's public address. We have verified that the sip messages exchanged between kamailio and asterisk are always correct. Kamailio Ver. 4.4.4 Asterisk Ver. 13.18.3 PJSIP Ver. 2.6 JSSIP Ver. 3.2.2 Thanks Alessio

2 2

Contact and Record-Route are using Private IP Address- is this correct?
by Wilkins, Steve 07 Feb '18

07 Feb '18

Hello All, Contact and Record-Route are using Private IP Address - is this correct? I though the Public IP should be used. I do not know what setting is causing this to happen. Does anyone have any ideas? Thank you, -Steve

3 4

Info: Markus Monka replacing Marcus Hunger in management group
by Daniel-Constantin Mierla 07 Feb '18

07 Feb '18

Hello, Markus Monka has just replaced Marcus Hunger in Kamailio project management group. Marcus (still at sipgate) has moved to work more on frontend applications than backend, no longer interacting with Kamailio project. Markus Monka has managed the VoIP operations at sipgate for more than 15 years, helping the project over the years with various resources, mainly in respect of organizing events and testing infrastructure, sipgate being one of the oldest VoIP services using Kamailio (since first releases of SER). The change is reflected now on the website: * https://www.kamailio.org/w/management/ Warm welcome to Markus and many thanks to both of them for what they did so far for Kamailio project! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Info: Kamailio project administration updates
by Daniel-Constantin Mierla 07 Feb '18

07 Feb '18

Hello, after several discussions at some of the past IRC devel meetings, finally we started to build a team to be involved more actively in the administration of Kamailio. The project has grown steadily, not only in terms of code, but also packaging, continuous integration, social networking interactions as well as participation to events world wide. For a better coordination and ability to handle related tasks, I invited the most active developers and community members to join so called Kamailio Administration Team, the initial details about it are published as part of management page on kamailio.org: * https://www.kamailio.org/w/management/ Some more details about its rules and purpose: * https://www.kamailio.org/w/kadmin/ It will still take some time to get it properly rolling, more or less now looking to see if the community has suggestions/improvements on what can be done in these aspects. I wrote to all mailing lists for the initial announcement, but you can just reply to sr-users, if you are subscribed there, otherwise it's fine to write back to any of the lists. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Re: [SR-Users] Contact and Record-Route are using Private IP Address- is this correct?
by Ammar Tinawi 07 Feb '18

07 Feb '18

Is it mandatory to have an interface with public ip in order to overcome NAT problems ?

1 0

Issue with Socket selection on Forwarding ACK message
by Thomas Carvello 06 Feb '18

06 Feb '18

Hello, i have an issue with my Kamailio 4.1.9 configuration. This configuration is multi-homed, we have*two network* interfaces, one on a private network and on the public Internet. Kamailio is configured to listen on port 5060 and 5066 on both interfaces. We register two users Alice and Bob on the public Internet using port 5066. Both users are behind a NAT and we capture the SIP exchange on the proxy server. We have set the parameter mhomed=1 When Alice calls Bob, we have Alice Proxy Bob src=5063 dst=5066 INVITE ------------------> src=5066 ------ INVITE ---------------> dst=5060 dst=5066 <------- 200 OK -------------- src=5060 dst=5063 <------- 200 OK --------- src=5066 src=5063 dst=5066 -------- ACK -----------> *src=5060 (blocked by NAT)* ------ ACK-----x dst=5060 The ACK packet gets relayed with the wrong source port. Then the NAT rejects the packet and the call cannot be established. For some reason, when Bob calls Alice, the call is correctly established. Could it be because Bob happend to use 5060 as local port? Also, if we set nhomed=0 it works BUT we are not sure that multi homed is handled correctly. I was wondering if you have encounter this issue before? I have investigated the code for selection socket and what is the logic of this selection ? /*How does kamailo knows that it should choose 5066 as src port if the user is registered using port 5066 instead of 5066?* / Thank you for your time. Thomas

4 8

rtpproxy - no media connection
by Steve 05 Feb '18

05 Feb '18

Dear sr-users, I am trying to set up a simple Kamailio server with rtpproxy on the same host computer, behind a NAT router. “Run your own Skype-like service” is my set-up guide, but I’m using Kamailio version 5.1 so I modified the config files that came with the download. My problem is that SIP clients can register over the internet via TLS encryption with the server and establish a call session connection, but the ZRTP media connection is not being made. Once the call session connection is made, both clients send out UDP packets to each other, but neither receives them—the media traffic is one way and being dropped somewhere. I’ve tried different rtpproxy start settings. If I start the rtpproxy service with “sudo /etc/init.d/rtpproxy start” then it uses the default “unix:/var/run/rtpproxy/rtpproxy.sock”. I’ve also tried starting with “sudo rtpproxy –l /my_public_ip/ –s udp:localhost:7722” which then asks for either –F (superuser) or –u (user). I’ve tried it both with options “–F” and “-u kamailio kamailio”. All the variations tried so far have the same result as noted above. The server behind the router is in a DMZ zone that bypasses the router firewall. Is there anything in the user-list archives that addresses this issue? Or some likely place to start troubleshooting the problem? Any suggestions would be greatly appreciated. Thanks in advance. Steve --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

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IMS (ng-voice vs github)
by Amar Tinawi 04 Feb '18

04 Feb '18

Hello Dears, can you please tell me what are the differences between the github version of ims modules and the ng-voice modules introduced by Mr Carsten repo ? Thanks in advance

2 1

Fosdem dinner
by Daniel-Constantin Mierla 03 Feb '18

03 Feb '18

Hello, in the past 10 years or so, many of us met at Fosdem conference in Brussels (the edition this year happens during February 3-4) and had a dinner on Saturday evening. I know already couple of people from community that will go there, so the question is if there is any interest to organize again such event. If yes, it will be again on Saturday, on the evening of the 3rd of February. At the past editions, it turned into an all-RTC dinner, but we can consider to have it alone again (eventually with few other invited friends), if there are too many or people want a smaller meeting. Reply this week if you want to join, so we can estimate how many seats we need and have enough time to find a place that can accommodate us. More about Fosdem can be found at: - https://fosdem.org There is a devroom on Sunday with topics covering real time communications. I will have a presentation there. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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SDP Rewrite issues
by Steven Meyer 02 Feb '18

02 Feb '18

Our topology is: Client <> Client Router <> Internet <> Firewall (NAT) <> Kamailio <> FS Unfortunately, when enabling SDP Rewrite: fix_nated_sdp, 0x08, it rewrites packets to the firewall external IP, not the client router external IP. Right now we have to do a local NATed IP on each phone.

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Routing based on A and B numbers
by Grant Bagdasarian 02 Feb '18

02 Feb '18

Hi, Which module would be best suited for routing calls based on A and B numbers? We're currently using the LCR module, with some additional htables to add this functionality, but I was wondering if there is a module which does exactly this? Or perhaps a combination of modules? Regards, Grant Bagdasarian Senior Developer +31765727054<tel:+31765727054> cm.com<https://cm.com> [cid:image001.png@01D39C36.945A76C0]

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Distributed Registrations.
by Володимир Іванець 02 Feb '18

02 Feb '18

Hello! Yesterday I read email conversation "Distributed Presence between Multiple Kamailios" where it was suggested to use Presence module with new dmq parameter. So was wondering if someone was thinking about approach to share registrations between multiple Kamailio servers without using database? Thanks!

2 1

Contact Header need update?
by Duarte Rocha 02 Feb '18

02 Feb '18

Greetings, My proxy does some number normalization in order to guarantee E164 numeration. It also verifies for valid Calling Numbers and replaces them if needed. In order to this, i'm updating the $fU pseudo-variable. Should i update the "Contact" Header too? I've made some testing and i can get calls to work with keeping the Contact intact and changing it to the new number. What is the most correct approach? Cheers, Duarte Rocha

1 0

Distributed Presence between Multiple Kamailios
by SamyGo 01 Feb '18

01 Feb '18

Hi, I'm working on multiple Kamailios that have users distributed evenly among them, I now need to enable presence in those Kamailio boxes. The scenario is a mesh where any user may have subscribed to BLF of other user registered & calling on any other proxy. What modules can help me achieve this? can this be done without creating a new Presence-Node for the whole environment? Hope to get some suggestions on how experts here have done this. Looking forward to positive response. Regards, Sammy

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Case insensitive regexp in textops::search() function
by Andrey Deykunov 01 Feb '18

01 Feb '18

Hi, I have an htable containing some black list words (i.e. user IDs or phone models). To find matches of the words in every message I use the following script: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { if ( search($shtitval(i1)) ) { // do something } } sht_iterator_end("i1"); How can I modify search($shtitval(i1)) to get case insensitive matching? The example below does not work: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { $var(re) = $shtitval(i1) + "/i"; if ( search($var(re)) ) { // do something } } sht_iterator_end("i1"); Other possible modifications of $var(re) aren't working too. Thanks, Andrey

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Partial/regex match on hash table key?
by Anthony Joseph Messina 31 Jan '18

31 Jan '18

Is is possible (without iterating through all entries in a hash table) to return the value based on a partial or regex match to the key? Something like... $var(one)=$sht(a=>*$var(two)) -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

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question about force_send_socket
by Fabian Borot 31 Jan '18

31 Jan '18

our kamailo has 2 IPs, a public (87.x.x.x) and a private (192.168.1.1). It is listening on all of them. For all calls it listens on the public IP but for some calls I need to forward the incoming INVITE to a server in the private network so I do a " force_send_socket(privateip:port)" before the route that forwards the INVITE to the server in the private network and that works. But then when the server in the private network replies with 302 I have to contact the servers inside the Contact header using the Public IP now so I do this hoping that it will switch back to the public IP: if (t_check_status("302")) { force_send_socket(PUBLIC_IP:5060); get_redirects("*"); t_relay(); } But it stays in the private, so the INVITEs to the servers inside the Contact header are sent from the Private IP and they are blocked. This would be the flow: customer INVITE -> kamailio(publicIP) kamailio (publicIP) 100 trying -> customer kamailio(privateIP) INVITE -> Server(privateIP) Server(privateIP) 302 redirect -> kamailio(privateIP) kamailio(privateIP) ACK -> Server(privateIP) after I would like this to happen kamailio(publicIP) INVITE -> Server1(publicIP) but instead this happens: kamailio(privateIP) INVITE -> Server1(publicIP) how can I switch back to the public IP to continue with this call flow? thank you

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[4.2] how to send out BYE successfully to close a TCP/TLS connection which misses the connection ID?
by Jack Wang 31 Jan '18

31 Jan '18

For some reasons, My settings will cause kamailio to be restarted even if there were connections connected to it. And when one SIP client in a connection wanted to hangup the call with a WebRTC Client (SIP over websocket) after the proxy-restarting, local client was closed, however , the other side was not. >From the log below it seems that the connection missed. Jan 30 18:19:31 ./kamailio[27857]: WARNING: <core> [msg_translator.c:2777]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message So, Is it equivalent to connection ID missing? And is there any way to make Websocket found the connection? Btw, after proxy-restarting for seconds , I typed `sh kamctl stats` : websocket:ws_current_connections = 1 It showed that the WebRTC Client will re-connect automatically, So I think this connection could be found by proxy, Any suggestion? Thanks :)

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