- sr-users - kamailio.org

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

3 8

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

4 3

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

2 2

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.

2 2

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

3 2

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

2 6

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

2 2

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

1 0

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

1 3

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

2 1

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,

2 5

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!

2 2

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

2 1

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

2 1

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>

3 3

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.

4 4

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

2 1

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/>

2 2

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

1 0

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

3 3

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

1 0

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

2 3

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.

1 0

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?

3 2

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

2 4

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

1 0

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

2 1

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

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users